Ganesh
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Posts posted by Ganesh
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Hello,
I would like to know if there is a possibility of this feature being included in the new release.
Ganesh
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Hi,
One of our customer using snomONE is looking for having the recorded call information tab under "Call Logs". By clicking on the tab, they should be able to play/download the audio file. Currently they are accessing the recorded calls from c:\program files\snomone\recording but it is very difficult to search for a specific call in the available files.
Hope this feature would be avaialble in the next release.
Regards
Ganesh
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We will open a feature request for this item.
Hi,
Any further progress on this please? We hope to see this feature in the next release.
Regards
Ganesh
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Well, that means that there are too many calls within a short time. You can increase the value of max_udp_invite or use TCP for SIP.
Thanks. Let me know how we can increase this value (max_udp_invite). Is this value in the snomONE configuration settings?
Regards,
Ganesh
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Hi,
Our customer is using snomONE version 4.2.0.3958 (Linux)and everthing was working fine. Today they are unable to call between the extensions. The snom phone displays the message as "Temporarily unavailable". In the PBX logs we see "DoS protection: Not accepting more calls"
Let me know how we can fix this problem. We are using version 4.2.0.3958 (Linux.
Regards
Ganesh
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Hi Support,
Our customer using the snomONE PBX is receiving the daily CDR to the email ID configured. It is a screen shot of call logs and these CDRs are not in editable format and it is difficult to search for specific numbers in the CDR.
It would be nice if we could receive the CDR in MS-Excel/CSV file format. If you could include a tab in the "Call Logs" for downloading the CDR in CSV format, it would be a great option!
Hope this feature can be included in the next release
Cheers
Ganesh
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This is the startup file that we are using for CentOS and this is working with older versions and not working with version 4025. The error displayed while starting the service is "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR"
Let me know the changes needed in the file to run version 4025.
#!/bin/bash
#
# Init file for pbxnsip PBX
#
# Copyright © 2006 pbxnsip Inc., USA
#
# chkconfig: 2345 20 80
# description: SIP-based PBX
#
# processname: pbxctrl
# pidfile: /var/run/pbxctrl.pid
# source function library
. /etc/rc.d/init.d/functions
RETVAL=0
# Installation location
INSTALLDIR=/usr/local/pbxnsip
PBX=$INSTALLDIR/pbxctrl
start()
{
echo -n "Starting PBX:"
$PBX --dir $INSTALLDIR
echo
RETVAL=1
}
stop()
{
echo -n "Stopping PBX:"
killproc $PBX -TERM
echo
RETVAL=1
}
case "$1" in
start)
start
;;
stop)
stop
;;
restart)
stop
start
;;
status)
status $PBX
RETVAL=$?
;;
*)
echo $"Usage: $0 {start|stop|restart|status}"
RETVAL=1
esac
exit $RETVAL
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Hello,
We are using Cent OS and upgraded the pbxnsip to version 4.2.1.4025. After upgrade while I try to start the service I see the error as "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR"
The start up script file works fine with the older version and not with version 4025. Can you please tell me what changes we need to make in start up file? I have checked the installation directory. It is as per the path in the start up file.
Regards
Ganesh
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Hello,
We are using Cent OS and upgraded the pbxnsip to version 4.2.1.4025. After upgrade while I try to start the service I see the error as "[root@sip3 ~]# service pbxnsip start Starting PBX:/etc/init.d/pbxnsip: line 19: 4471 Segmentation fault $PBX --dir $INSTALLDIR"
The start up script file works fine with the older version and not with version 4025. Can you please tell me what changes we need to make in start up file? I have checked the installation directory. It is as per the path in the start up file.
Regards
Ganesh
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We have two SIP registration trunk setup to two different service providers and both trunks are registered to service provider. The failover behaviour on both trunks is configured to "on all error codes". The dial plan is configured with first preference to trunk A and then same dial plan number to second preference to trunk B.
Now if the Trunk A registration fails then snomONE trunk status shows "408 request timed out". At this state if any user makes a call, the call is not redirected to Trunk B as trunk A is in failed state. We have seen that the snomONE is still trying to route the call only to the failed trunk. It never tries to re-route the same call to the other trunk.
Why does the PBX try routing call to a trunk that is in failed state? How do we setup the server to try route call to trunk B when A is in failed state?
Is it possible to first route the call to trunk A and wait for a while (few seconds) and if no response is received from far end then re-route a new Invite to trunk B?
Regards
Ganesh
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Hi Ganesh,
If the preference in your dial plan is to always send calls to 10.10.11.9 first, then the call routing will only get to 10.10.11.10 when 10.10.11.9 already failed. After 10.10.11.10 also fails, why do you want it to try 10.10.11.9 again (since we already know it fails)?
Anyway, if you do want this, you can try to simply add one more dialplan rule for 10.10.11.9, after ther rule to 10.10.11.10.
Thanks for your reply.
When 10.10.11.9 fails, the call routing is happening with 10.10.11.10.
The problem observed is only when 10.10.11.10 fails while 10.10.11.9 is in working state. Ideally it should continue to work with 10.10.11.9 but it does not.
Let me try the option of adding one more dial plan.
Regards
Ganesh
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We have 2 snom one servers at a central location on IP address 10.10.11.9 and 10.10.11.10. The remote location snom one server has 2 SIP gateway trunks configured to the central location servers. In the trunk 1 we have specified the domain as 10.10.11.9 and outbound proxy as 10.10.11.10. Similarly on trunk 2 we have specified the domain as 10.10.11.10 and outbound proxy as 10.10.11.9. The failover on both the trunks is set to "Always except when busy". In the dial plan of the remote location server the first preference is set to 10.10.11.9 and second is 10.10.11.10. When the remote location users call the central site the call is well handled by the server 10.10.11.9.
During a failover of the server 10.10.11.9 at central site, the remote location snom one server is automatically routing calls to 10.10.11.10 and this is working fine. But during the failover of server 10.10.11.10, the remote location server stops routing calls even to 10.10.11.9. Why is it so? The server 10.10.11.9 works only if the other server is connected and working. Else this stops responding. Whereas the server 10.10.11.10 works fine even if the first server becomes unavailable. Has someone tried this?
Regards
Ganesh
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So why does the phone not include the rport parameter? Maybe it was turned off (http://wiki.snom.com/Settings/enable_rport_rfc3581). As a result, the PBX sends the response back to the port that was advertized by the phone (according to the RFC).
Generally, I would suggest to factory-reset the phones and then use plug and play for the phones (including the upgrade to 8.4.18). You can try this with one phone and then if it solves the problems then you can include the other phones as well.
Thanks. This is now working after the upgrade.
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Is this an environment where the PBX has multiple IP addresses? There is no REGISTER message in the snomONE scenario. What is the output of "route print"?
There is definitively something wrong in the network setup. Even the RTP statistics coming from the phone show that there is a dramatic packet loss.
Also what is the outbound proxy of the phone that sends the "use proxy"? Did you plug and play it with the PBX? Or did you at least factory reset it before registering it to the PBX?
This is is a pretty much classical setup and should work without problems pretty much!
The PBX has a single domain (localhost) and the server has single IP address (10.10.71.10). Attached is the SIP registration and calls traces of the snom phones with snomONE server.
There is no outbound proxy configured in the snom phone. We are using only the Registrar as 10.10.71.10. If we use the same registrar in the outbound proxy, still the same problem is observed. If we use the outbound proxy as 172.21.11.x (as seen in the SIP registration traces of snomONE) then the phones does not register.No plug and play is used, the phones are manually configured with snomONE. Factory reset was done before registering the phones.
The phones (ext 111 and 112) at location B can call to phones (203 and 204) at location A whereas the reverse is not working.
The phones (111 and 112) at location B cannot call each other.
The phones (203 and 204) at location A can call each other.
The "route print" of snomONE network is
C:\Users\Snom Technology>route print
===========================================================================
Interface List
16...1c 65 9d 25 ea c3 ......Microsoft Virtual WiFi Miniport Adapter
14...1c 65 9d 25 ea c3 ......Realtek RTL8191SE 802.11b/g/n WiFi Adapter
11...1c c1 de 9e 41 84 ......Realtek PCIe FE Family Controller
1...........................Software Loopback Interface 1
19...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter
37...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter #2
17...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter #3
15...00 00 00 00 00 00 00 e0 Teredo Tunneling Pseudo-Interface
===========================================================================
IPv4 Route Table
===========================================================================
Active Routes:
Network Destination Netmask Gateway Interface Metric
0.0.0.0 0.0.0.0 10.10.71.1 10.10.71.10 276
10.10.71.0 255.255.255.0 On-link 10.10.71.10 276
10.10.71.10 255.255.255.255 On-link 10.10.71.10 276
10.10.71.255 255.255.255.255 On-link 10.10.71.10 276
127.0.0.0 255.0.0.0 On-link 127.0.0.1 306
127.0.0.1 255.255.255.255 On-link 127.0.0.1 306
127.255.255.255 255.255.255.255 On-link 127.0.0.1 306
224.0.0.0 240.0.0.0 On-link 127.0.0.1 306
224.0.0.0 240.0.0.0 On-link 10.10.71.10 274
255.255.255.255 255.255.255.255 On-link 127.0.0.1 306
255.255.255.255 255.255.255.255 On-link 10.10.71.10 276
===========================================================================
Persistent Routes:
Network Address Netmask Gateway Address Metric
0.0.0.0 0.0.0.0 10.10.71.1 Default
===========================================================================
IPv6 Route Table
===========================================================================
Active Routes:
If Metric Network Destination Gateway
1 306 ::1/128 On-link
11 276 fe80::/64 On-link
11 276 fe80::d991:e426:3991:9937/128
On-link
1 306 ff00::/8 On-link
11 276 ff00::/8 On-link
===========================================================================
Persistent Routes:
None
C:\Users\Snom Technology>
The "route print" of phone at location B is
Microsoft Windows [Version 6.1.7600]
Copyright © 2009 Microsoft Corporation. All rights reserved.
C:\Users\Snom Technology>route print
===========================================================================
Interface List
16...1c 65 9d 25 ea c3 ......Microsoft Virtual WiFi Miniport Adapter
14...1c 65 9d 25 ea c3 ......Realtek RTL8191SE 802.11b/g/n WiFi Adapter
11...1c c1 de 9e 41 84 ......Realtek PCIe FE Family Controller
1...........................Software Loopback Interface 1
19...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter
37...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter #2
17...00 00 00 00 00 00 00 e0 Microsoft ISATAP Adapter #3
15...00 00 00 00 00 00 00 e0 Teredo Tunneling Pseudo-Interface
===========================================================================
IPv4 Route Table
===========================================================================
Active Routes:
Network Destination Netmask Gateway Interface Metric
0.0.0.0 0.0.0.0 10.11.51.1 10.11.51.10 276
10.11.51.0 255.255.255.0 On-link 10.11.51.10 276
10.11.51.10 255.255.255.255 On-link 10.11.51.10 276
10.11.51.255 255.255.255.255 On-link 10.11.51.10 276
127.0.0.0 255.0.0.0 On-link 127.0.0.1 306
127.0.0.1 255.255.255.255 On-link 127.0.0.1 306
127.255.255.255 255.255.255.255 On-link 127.0.0.1 306
224.0.0.0 240.0.0.0 On-link 127.0.0.1 306
224.0.0.0 240.0.0.0 On-link 10.11.51.10 274
255.255.255.255 255.255.255.255 On-link 127.0.0.1 306
255.255.255.255 255.255.255.255 On-link 10.11.51.10 276
===========================================================================
Persistent Routes:
Network Address Netmask Gateway Address Metric
0.0.0.0 0.0.0.0 10.11.51.1 Default
===========================================================================
IPv6 Route Table
===========================================================================
Active Routes:
If Metric Network Destination Gateway
1 306 ::1/128 On-link
11 276 fe80::/64 On-link
11 276 fe80::d991:e426:3991:9937/128
On-link
1 306 ff00::/8 On-link
11 276 ff00::/8 On-link
===========================================================================
Persistent Routes:
None
C:\Users\Snom Technology>
SIP registration traces of snomONE.txt
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The "Use Proxy" method comes from the phone it seems, and AFAIK there is a setting on the phone that restricts inbound traffic to come from the "proxy" (AKA PBX). You can try to change the setting on the phone if it then accepts the traffic from that address.
Basically the phones at location B are registered to the snomone server at location A but are unable to receive any incoming calls. The SIP trace of snom phone at location B says “403, Use Proxy” for the Invites received from the snomone server.
The issue seems to be with snomone unable to understand the NAT. If we replace the snomone server with other SIP server, the entire system is working fine. The snom phones can communicate from location B to A and vice versa using the other SIP server.
Call flow while snomone is used
1. Calls from location A to B is NOT working.
2. Calls from location B to A is working
3. Calls between phones at location B is NOT working
4. Calls between phones at location A is working
Attached is the wireshark traces where you see a successful call from location B to A using other SIP server.
wireshark trace of 4s for working call.zip
wireshark trace of snom one for non working call between extensions.zip
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Hello,
We have few snom phones registered with the snom ONE. The snom ONE and few snom phones are at location A. Few more snom phones are at location B and registered to the snom ONE at location A.
We are able to make calls between the snom phones connected within the location A. But we cannot call between the snom phones connected at location B.
We can make calls from location B to A but not from A to B.
The SIP trace of snom phones at location B says "403 use proxy" for any invites coming from the snom ONE.
Attached is the wireshark capture and SIP traces of snom ONE and snom phone. Can someone check this and help asap?
wireshark trace of snom one for non working call between extensions.zip
snom one SIP trace for non working call between extensions.txt
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We have a setup running in which there is one way speech. For inbound calls via the trunk the far end person can hear the agent but the agent cannot hear any voice. Below is log. The firewall is open for UDP ports. Can you see if we need to do any configuration changes in Pbxnsip?
[9] 2009/07/24 14:46:25: SIP Rx udp:192.168.200.204:5060:
OPTIONS sip:201@192.168.200.204 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.204;branch=z9hG4bKac608188397
Max-Forwards: 70
From: <sip:201@192.168.200.204>;tag=1c608184426
To: <sip:201@192.168.200.204>
Call-ID: 60818406411200020958@192.168.200.204
CSeq: 1 OPTIONS
Contact: <sip:201@192.168.200.204>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.016.003
Content-Length: 0
[9] 2009/07/24 14:46:25: Resolve 25: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:25: Resolve 25: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:25: Resolve 25: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:25: SIP Tx udp:192.168.200.204:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.200.204;branch=z9hG4bKac608188397
From: <sip:201@192.168.200.204>;tag=1c608184426
To: <sip:201@192.168.200.204>;tag=094c02e6ea
Call-ID: 60818406411200020958@192.168.200.204
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Content-Length: 0
[9] 2009/07/24 14:46:35: Resolve 26: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 26: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 26: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 27: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 27: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 27: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 28: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 28: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 28: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 29: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 29: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 29: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: SIP Rx udp:216.52.221.144:51560:
INVITE sip:222@122.166.31.203:5060 SIP/2.0
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>
Date: Fri, 24 Jul 2009 09:12:45 GMT
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
Supported: timer,replaces
Min-SE: 1800
Cisco-Guid: 4227719171-2003309022-3116105749-3332332504
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:2199612221@216.52.221.144>;party=calling;screen=no;privacy=off
Timestamp: 1248426765
Contact: <sip:2199612221@216.52.221.144:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 676
v=0
o=CiscoSystemsSIP-GW-UserAgent 3750 881 IN IP4 216.52.221.144
s=SIP Call
c=IN IP4 216.52.221.144
t=0 0
m=audio 16748 RTP/AVP 0 8 18 2 98 99 4 3 100 101
c=IN IP4 216.52.221.144
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-16/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=yes
a=rtpmap:3 GSM/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194,200-202
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38
[9] 2009/07/24 14:46:35: UDP: Opening socket on 0.0.0.0:36494
[9] 2009/07/24 14:46:35: UDP: Opening socket on 0.0.0.0:36495
[5] 2009/07/24 14:46:35: Identify trunk (IP address and DID match) 6
[9] 2009/07/24 14:46:35: Resolve 30: aaaa udp 216.52.221.144 5060
[9] 2009/07/24 14:46:35: Resolve 30: a udp 216.52.221.144 5060
[9] 2009/07/24 14:46:35: Resolve 30: udp 216.52.221.144 5060
[9] 2009/07/24 14:46:35: SIP Tx udp:216.52.221.144:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>;tag=5577fa45a8
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 101 INVITE
Content-Length: 0
[9] 2009/07/24 14:46:35: Resolve 31: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 31: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 31: udp 192.168.200.204 5060
[6] 2009/07/24 14:46:35: Sending RTP for FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144#5577fa45a8 to 216.52.221.144:16748
[5] 2009/07/24 14:46:35: Trunk 4 (not global) sends call to account 222 in domain localhost
[8] 2009/07/24 14:46:35: Play audio_moh/noise.wav
[7] 2009/07/24 14:46:35: Hunt Group 222: Moving to next stage
[7] 2009/07/24 14:46:35: Hunt group 222 called 1 registrations
[9] 2009/07/24 14:46:35: Resolve 32: aaaa udp 216.52.221.144 5060
[9] 2009/07/24 14:46:35: Resolve 32: a udp 216.52.221.144 5060
[9] 2009/07/24 14:46:35: Resolve 32: udp 216.52.221.144 5060
[9] 2009/07/24 14:46:35: SIP Tx udp:216.52.221.144:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>;tag=5577fa45a8
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 101 INVITE
Contact: <sip:222@192.168.200.203:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: SI-PBX/3.3.1.3177
Content-Length: 0
[9] 2009/07/24 14:46:35: UDP: Opening socket on 0.0.0.0:61334
[9] 2009/07/24 14:46:35: UDP: Opening socket on 0.0.0.0:61335
[9] 2009/07/24 14:46:35: Resolve 33: url sip:202@192.168.200.204
[9] 2009/07/24 14:46:35: Resolve 33: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: SIP Tx udp:192.168.200.204:5060:
INVITE sip:202@192.168.200.204 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-2e80dbcdbced790202a8fddcc7404e19;rport
From: <sip:2199612221@localhost;user=phone>;tag=63556
To: <sip:222@localhost>
Call-ID: 4cc7ab93@pbx
CSeq: 20197 INVITE
Max-Forwards: 70
Contact: <sip:202@192.168.200.203:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: SI-PBX/3.3.1.3177
Content-Type: application/sdp
Content-Length: 341
v=0
o=- 8812 8812 IN IP4 192.168.200.203
s=-
c=IN IP4 192.168.200.203
t=0 0
m=audio 61334 RTP/AVP 0 8 18 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2009/07/24 14:46:35: Resolve 34: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 34: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: Resolve 34: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: SIP Rx udp:192.168.200.204:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-2e80dbcdbced790202a8fddcc7404e19;rport
From: <sip:2199612221@localhost;user=phone>;tag=63556
To: <sip:222@localhost>;tag=1c621280194
Call-ID: 4cc7ab93@pbx
CSeq: 20197 INVITE
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.016.003
Content-Length: 0
[9] 2009/07/24 14:46:35: SIP Rx udp:192.168.200.204:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-2e80dbcdbced790202a8fddcc7404e19;rport
From: <sip:2199612221@localhost;user=phone>;tag=63556
To: <sip:222@localhost>;tag=1c621280194
Call-ID: 4cc7ab93@pbx
CSeq: 20197 INVITE
Contact: <sip:202@192.168.200.204>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Require: 100rel
RSeq: 1
Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.016.003
Content-Length: 0
[9] 2009/07/24 14:46:35: Resolve 35: url sip:202@192.168.200.204
[9] 2009/07/24 14:46:35: Resolve 35: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:35: SIP Tx udp:192.168.200.204:5060:
PRACK sip:202@192.168.200.204 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-9292f13a6cf04f6e0080f17e70b421c3;rport
From: <sip:2199612221@localhost;user=phone>;tag=63556
To: <sip:222@localhost>;tag=1c621280194
Call-ID: 4cc7ab93@pbx
CSeq: 20198 PRACK
Max-Forwards: 70
Contact: <sip:202@192.168.200.203:5060;transport=udp>
RAck: 1 20197 INVITE
Content-Length: 0
[8] 2009/07/24 14:46:35: Play audio_en/ringback.wav
[9] 2009/07/24 14:46:35: SIP Rx udp:192.168.200.204:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-9292f13a6cf04f6e0080f17e70b421c3;rport
From: <sip:2199612221@localhost;user=phone>;tag=63556
To: <sip:222@localhost>;tag=1c621280194
Call-ID: 4cc7ab93@pbx
CSeq: 20198 PRACK
Contact: <sip:202@192.168.200.204>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.016.003
Content-Length: 0
[7] 2009/07/24 14:46:35: Call 4cc7ab93@pbx#63556: Clear last request
[9] 2009/07/24 14:46:36: SIP Tr udp:216.52.221.144:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>;tag=5577fa45a8
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 101 INVITE
Contact: <sip:222@192.168.200.203:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: SI-PBX/3.3.1.3177
Content-Length: 0
[9] 2009/07/24 14:46:39: Last message repeated 3 times
[9] 2009/07/24 14:46:39: Resolve 36: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:39: Resolve 36: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:39: Resolve 36: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:42: Resolve 37: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:42: Resolve 37: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:42: Resolve 37: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:43: SIP Tr udp:216.52.221.144:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>;tag=5577fa45a8
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 101 INVITE
Contact: <sip:222@192.168.200.203:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: SI-PBX/3.3.1.3177
Content-Length: 0
[7] 2009/07/24 14:46:45: Hunt Group 222: Moving to next stage
[7] 2009/07/24 14:46:45: Hunt group 222 called 0 registrations
[9] 2009/07/24 14:46:46: Resolve 38: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:46: Resolve 38: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:46: Resolve 38: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:48: Resolve 39: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:46:48: Resolve 39: a udp 192.168.200.204 5060
[9] 2009/07/24 14:46:48: Resolve 39: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:51: SIP Tr udp:216.52.221.144:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>;tag=5577fa45a8
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 101 INVITE
Contact: <sip:222@192.168.200.203:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: SI-PBX/3.3.1.3177
Content-Length: 0
[7] 2009/07/24 14:46:55: Hunt Group 222: Moving to next stage
[7] 2009/07/24 14:46:55: Hunt group 222 called 0 registrations
[9] 2009/07/24 14:46:57: SIP Rx udp:192.168.200.204:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-2e80dbcdbced790202a8fddcc7404e19;rport
From: <sip:2199612221@localhost;user=phone>;tag=63556
To: <sip:222@localhost>;tag=1c621280194
Call-ID: 4cc7ab93@pbx
CSeq: 20197 INVITE
Contact: <sip:202@192.168.200.204>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.016.003
Content-Type: application/sdp
Content-Length: 237
v=0
o=AudiocodesGW 621288964 621288882 IN IP4 192.168.200.204
s=Phone-Call
c=IN IP4 192.168.200.204
t=0 0
m=audio 6010 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[7] 2009/07/24 14:46:57: Call 4cc7ab93@pbx#63556: Clear last INVITE
[7] 2009/07/24 14:46:57: Set packet length to 20
[6] 2009/07/24 14:46:57: Send codec=pcmu/8000 afrer answer
[6] 2009/07/24 14:46:57: Sending RTP for 4cc7ab93@pbx#63556 to 192.168.200.204:6010
[9] 2009/07/24 14:46:57: Resolve 40: url sip:202@192.168.200.204
[9] 2009/07/24 14:46:57: Resolve 40: udp 192.168.200.204 5060
[9] 2009/07/24 14:46:57: SIP Tx udp:192.168.200.204:5060:
ACK sip:202@192.168.200.204 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-7f9cab7579ac99c088a3ad09463ac6b3;rport
From: <sip:2199612221@localhost;user=phone>;tag=63556
To: <sip:222@localhost>;tag=1c621280194
Call-ID: 4cc7ab93@pbx
CSeq: 20197 ACK
Max-Forwards: 70
Contact: <sip:202@192.168.200.203:5060;transport=udp>
Content-Length: 0
[7] 2009/07/24 14:46:57: Determine pass-through mode after receiving response
[6] 2009/07/24 14:46:57: send codec=pcmu/8000
[9] 2009/07/24 14:46:57: Resolve 41: aaaa udp 216.52.221.144 5060
[9] 2009/07/24 14:46:57: Resolve 41: a udp 216.52.221.144 5060
[9] 2009/07/24 14:46:57: Resolve 41: udp 216.52.221.144 5060
[9] 2009/07/24 14:46:57: SIP Tx udp:216.52.221.144:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>;tag=5577fa45a8
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 101 INVITE
Contact: <sip:222@192.168.200.203:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: SI-PBX/3.3.1.3177
Content-Type: application/sdp
Content-Length: 269
v=0
o=- 37660 37660 IN IP4 192.168.200.203
s=-
c=IN IP4 192.168.200.203
t=0 0
m=audio 36494 RTP/AVP 0 8 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[7] 2009/07/24 14:46:57: 4cc7ab93@pbx#63556: RTP pass-through mode
[7] 2009/07/24 14:46:57: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144#5577fa45a8: RTP pass-through mode
[9] 2009/07/24 14:46:58: SIP Tr udp:216.52.221.144:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>;tag=5577fa45a8
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 101 INVITE
Contact: <sip:222@192.168.200.203:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: SI-PBX/3.3.1.3177
Content-Type: application/sdp
Content-Length: 269
v=0
o=- 37660 37660 IN IP4 192.168.200.203
s=-
c=IN IP4 192.168.200.203
t=0 0
m=audio 36494 RTP/AVP 0 8 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2009/07/24 14:47:05: Last message repeated 4 times
[9] 2009/07/24 14:47:05: Resolve 42: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 42: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 42: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 43: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 43: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 43: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 44: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 44: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 44: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 45: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 45: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 45: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 46: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 46: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 46: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 47: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 47: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:05: Resolve 47: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:08: Resolve 48: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:08: Resolve 48: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:08: Resolve 48: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:10: SIP Rx udp:192.168.200.204:5060:
OPTIONS sip:201@192.168.200.204 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.204;branch=z9hG4bKac664470479
Max-Forwards: 70
From: <sip:201@192.168.200.204>;tag=1c664466501
To: <sip:201@192.168.200.204>
Call-ID: 664466140112000201043@192.168.200.204
CSeq: 1 OPTIONS
Contact: <sip:201@192.168.200.204>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.016.003
Content-Length: 0
[9] 2009/07/24 14:47:10: Resolve 49: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:10: Resolve 49: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:10: Resolve 49: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:10: SIP Tx udp:192.168.200.204:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.200.204;branch=z9hG4bKac664470479
From: <sip:201@192.168.200.204>;tag=1c664466501
To: <sip:201@192.168.200.204>;tag=73dbbdce2f
Call-ID: 664466140112000201043@192.168.200.204
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Content-Length: 0
[9] 2009/07/24 14:47:12: Resolve 50: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:12: Resolve 50: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:12: Resolve 50: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:13: SIP Tr udp:216.52.221.144:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 216.52.221.144:5060;branch=z9hG4bK521724
From: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
To: <sip:222@122.166.31.203>;tag=5577fa45a8
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 101 INVITE
Contact: <sip:222@192.168.200.203:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: SI-PBX/3.3.1.3177
Content-Type: application/sdp
Content-Length: 269
v=0
o=- 37660 37660 IN IP4 192.168.200.203
s=-
c=IN IP4 192.168.200.203
t=0 0
m=audio 36494 RTP/AVP 0 8 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2009/07/24 14:47:16: Resolve 51: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:16: Resolve 51: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:16: Resolve 51: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:18: Resolve 52: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:18: Resolve 52: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:18: Resolve 52: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:19: SIP Rx udp:192.168.200.204:5060:
BYE sip:202@192.168.200.203:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.200.204;branch=z9hG4bKac675660303
Max-Forwards: 70
From: <sip:222@localhost>;tag=1c621280194
To: <sip:2199612221@localhost;user=phone>;tag=63556
Call-ID: 4cc7ab93@pbx
CSeq: 1 BYE
Contact: <sip:202@192.168.200.204>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.60A.016.003
Content-Length: 0
[9] 2009/07/24 14:47:19: Resolve 53: aaaa udp 192.168.200.204 5060
[9] 2009/07/24 14:47:19: Resolve 53: a udp 192.168.200.204 5060
[9] 2009/07/24 14:47:19: Resolve 53: udp 192.168.200.204 5060
[9] 2009/07/24 14:47:19: SIP Tx udp:192.168.200.204:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.200.204;branch=z9hG4bKac675660303
From: <sip:222@localhost>;tag=1c621280194
To: <sip:2199612221@localhost;user=phone>;tag=63556
Call-ID: 4cc7ab93@pbx
CSeq: 1 BYE
Contact: <sip:202@192.168.200.203:5060;transport=udp>
User-Agent: SI-PBX/3.3.1.3177
RTP-RxStat: Dur=44,Pkt=1076,Oct=185072,Underun=0
RTP-TxStat: Dur=22,Pkt=1,Oct=172
Content-Length: 0
[7] 2009/07/24 14:47:19: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144#5577fa45a8: Media-aware pass-through mode
[7] 2009/07/24 14:47:19: Other Ports: 1
[7] 2009/07/24 14:47:19: Call Port: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144#5577fa45a8
[9] 2009/07/24 14:47:19: Resolve 54: url sip:2199612221@216.52.221.144:5060
[9] 2009/07/24 14:47:19: Resolve 54: udp 216.52.221.144 5060
[9] 2009/07/24 14:47:19: SIP Tx udp:216.52.221.144:5060:
BYE sip:2199612221@216.52.221.144:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-4dd697b98164e15b1920a9b53fe2b4ea;rport
From: <sip:222@122.166.31.203>;tag=5577fa45a8
To: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 509 BYE
Max-Forwards: 70
Contact: <sip:222@192.168.200.203:5060;transport=udp>
RTP-RxStat: Dur=44,Pkt=0,Oct=0,Underun=0
RTP-TxStat: Dur=22,Pkt=1076,Oct=185072
Content-Length: 0
[9] 2009/07/24 14:47:19: SIP Tr udp:216.52.221.144:5060:
BYE sip:2199612221@216.52.221.144:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-4dd697b98164e15b1920a9b53fe2b4ea;rport
From: <sip:222@122.166.31.203>;tag=5577fa45a8
To: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
CSeq: 509 BYE
Max-Forwards: 70
Contact: <sip:222@192.168.200.203:5060;transport=udp>
RTP-RxStat: Dur=44,Pkt=0,Oct=0,Underun=0
RTP-TxStat: Dur=22,Pkt=1076,Oct=185072
Content-Length: 0
[9] 2009/07/24 14:47:20: SIP Rx udp:216.52.221.144:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.203:5060;branch=z9hG4bK-4dd697b98164e15b1920a9b53fe2b4ea;rport;received=122.166.31.203
From: <sip:222@122.166.31.203>;tag=5577fa45a8
To: <sip:2199612221@216.52.221.144>;tag=18D0EFC0-1B12
Date: Fri, 24 Jul 2009 09:13:30 GMT
Call-ID: FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 509 BYE
[7] 2009/07/24 14:47:20: Call FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144#5577fa45a8: Clear last request
[5] 2009/07/24 14:47:20: BYE Response: Terminate FC13D89C-776811DE-B2ADC16F-10A1CBE6@216.52.221.144
-
We upgraded the CS410 (white box) to the latest version of pbxnsip application. The upgrade was done as there was no CLI displayed for incoming calls on FXO port. After upgrade, the FXO ports are not working. There is no incoming and out going calls on FXO ports. While the PSTN line is connected to FXO port, the LED is not flashing when there is an incoming call. No logs are captured for an incoming call on FXO port.
Below is the logs captured while an out going call is made. In these logs it shows INVITE, TRYING, RINGING, 200 OK etc. But the call does not go thru. The dialed PSTN number does not ring either.
[9] 2008/10/27 05:27:19: SIP Rx udp:192.168.0.2:5060:
REGISTER sip:192.168.0.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK24636383504932026;rport
From: 15 <sip:15@192.168.0.17:5060>;tag=202514052
To: 15 <sip:15@192.168.0.17:5060>
Call-ID: 2426712224-419229253@192.168.0.2
CSeq: 16 REGISTER
Contact: <sip:15@192.168.0.2:5060>
Max-Forwards: 70
Expires: 60
User-Agent: Voip Phone 1.0
Content-Length: 0
[9] 2008/10/27 05:27:19: Resolve 46: aaaa udp 192.168.0.2 5060
[9] 2008/10/27 05:27:19: Resolve 46: a udp 192.168.0.2 5060
[9] 2008/10/27 05:27:19: Resolve 46: udp 192.168.0.2 5060
[9] 2008/10/27 05:27:19: SIP Tx udp:192.168.0.2:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK24636383504932026;rport=5060
From: 15 <sip:15@192.168.0.17:5060>;tag=202514052
To: 15 <sip:15@192.168.0.17:5060>;tag=5012bb89f2
Call-ID: 2426712224-419229253@192.168.0.2
CSeq: 16 REGISTER
Contact: <sip:15@192.168.0.2:5060>;expires=31
Content-Length: 0
[9] 2008/10/27 05:27:20: SIP Rx udp:192.168.0.2:5060:
INVITE sip:9980160006@192.168.0.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2158616889645429424;rport
From: 15 <sip:15@192.168.0.17:5060>;tag=202761955
To: 9980160006 <sip:9980160006@192.168.0.17:5060>
Call-ID: 4373886024080-20982265928144@192.168.0.2
CSeq: 1 INVITE
Contact: <sip:15@192.168.0.2:5060>
Max-Forwards: 70
Supported: replaces, 100rel
User-Agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 254
v=0
o=100 18910302 27129223 IN IP4 192.168.0.2
s=A conversation
c=IN IP4 192.168.0.2
t=0 0
m=audio 10018 RTP/AVP 0 4 18 8 9
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/16000
a=sendrecv
[9] 2008/10/27 05:27:20: UDP: Opening socket on port 54146
[9] 2008/10/27 05:27:20: UDP: Opening socket on port 54147
[8] 2008/10/27 05:27:20: Could not find a trunk (1 trunks)
[9] 2008/10/27 05:27:20: Resolve 47: aaaa udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: Resolve 47: a udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: Resolve 47: udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: SIP Tx udp:192.168.0.2:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2158616889645429424;rport=5060
From: 15 <sip:15@192.168.0.17:5060>;tag=202761955
To: 9980160006 <sip:9980160006@192.168.0.17:5060>;tag=42c51291c3
Call-ID: 4373886024080-20982265928144@192.168.0.2
CSeq: 1 INVITE
Content-Length: 0
[6] 2008/10/27 05:27:20: Sending RTP for 4373886024080-20982265928144@192.168.0.2#42c51291c3 to 192.168.0.2:10018
[9] 2008/10/27 05:27:20: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 9980160006@192.168.0.17
[5] 2008/10/27 05:27:20: Dialplan FIRE FRO: Match 9980160006@192.168.0.17 to <sip:9980160006@localhost;user=phone> on trunk PSTN
[8] 2008/10/27 05:27:20: Play audio_moh/noise.wav
[9] 2008/10/27 05:27:20: UDP: Opening socket on port 61642
[9] 2008/10/27 05:27:20: UDP: Opening socket on port 61643
[9] 2008/10/27 05:27:20: Resolve 48: url sip:127.0.0.1:5062
[9] 2008/10/27 05:27:20: Resolve 48: udp 127.0.0.1 5062
[9] 2008/10/27 05:27:20: SIP Tx udp:127.0.0.1:5062:
INVITE sip:9980160006@localhost;user=phone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-10cf26a12f8eba1a2b94748c9612676d;rport
From: "Kitchen" <sip:15@localhost;user=phone>;tag=1167038513
To: <sip:9980160006@localhost;user=phone>
Call-ID: 17e82b84@pbx
CSeq: 6056 INVITE
Max-Forwards: 70
Contact: <sip:15@127.0.0.1:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Type: application/sdp
Content-Length: 268
v=0
o=- 233681333 233681333 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 61642 RTP/AVP 0 8 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2008/10/27 05:27:20: Resolve 49: aaaa udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: Resolve 49: a udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: Resolve 49: udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: SIP Tx udp:192.168.0.2:5060:
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2158616889645429424;rport=5060
From: 15 <sip:15@192.168.0.17:5060>;tag=202761955
To: 9980160006 <sip:9980160006@192.168.0.17:5060>;tag=42c51291c3
Call-ID: 4373886024080-20982265928144@192.168.0.2
CSeq: 1 INVITE
Contact: <sip:15@192.168.0.17:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 170
v=0
o=- 2095781365 2095781365 IN IP4 192.168.0.17
s=-
c=IN IP4 192.168.0.17
t=0 0
m=audio 54146 RTP/AVP 0 8
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=sendrecv
[9] 2008/10/27 05:27:20: SIP Rx udp:192.168.0.2:5060:
PRACK sip:15@192.168.0.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK178518297184232442
From: 15 <sip:15@192.168.0.17:5060>;tag=202761955
To: 9980160006 <sip:9980160006@192.168.0.17:5060>;tag=42c51291c3
Call-ID: 4373886024080-20982265928144@192.168.0.2
CSeq: 2 PRACK
Max-Forwards: 70
User-Agent: Voip Phone 1.0
rack: 1 1 INVITE
Content-Length: 0
[9] 2008/10/27 05:27:20: Resolve 50: aaaa udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: Resolve 50: a udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: Resolve 50: udp 192.168.0.2 5060
[9] 2008/10/27 05:27:20: SIP Tx udp:192.168.0.2:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK178518297184232442
From: 15 <sip:15@192.168.0.17:5060>;tag=202761955
To: 9980160006 <sip:9980160006@192.168.0.17:5060>;tag=42c51291c3
Call-ID: 4373886024080-20982265928144@192.168.0.2
CSeq: 2 PRACK
Contact: <sip:15@192.168.0.17:5060>
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Length: 0
[9] 2008/10/27 05:27:21: SIP Tr udp:127.0.0.1:5062:
INVITE sip:9980160006@localhost;user=phone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-10cf26a12f8eba1a2b94748c9612676d;rport
From: "Kitchen" <sip:15@localhost;user=phone>;tag=1167038513
To: <sip:9980160006@localhost;user=phone>
Call-ID: 17e82b84@pbx
CSeq: 6056 INVITE
Max-Forwards: 70
Contact: <sip:15@127.0.0.1:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Type: application/sdp
Content-Length: 268
v=0
o=- 233681333 233681333 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 61642 RTP/AVP 0 8 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2008/10/27 05:27:24: Last message repeated 3 times
[9] 2008/10/27 05:27:24: SIP Rx udp:192.168.0.4:55308:
REGISTER sip:192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:55308;branch=z9hG4bK-d87543-e357671db26f8a7b-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:15@192.168.0.4:55308;rinstance=710ef2fda04e0fbc>
To: "15"<sip:15@192.168.0.17>
From: "15"<sip:15@192.168.0.17>;tag=677d8039
Call-ID: 0874fd67c34d7260M2FlNjIwMmY4MTA3ZGNiNDgyNzFmN2I4YTQ4NmIxNTc.
CSeq: 17 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1003l stamp 30942
Content-Length: 0
[9] 2008/10/27 05:27:24: Resolve 51: aaaa udp 192.168.0.4 55308
[9] 2008/10/27 05:27:24: Resolve 51: a udp 192.168.0.4 55308
[9] 2008/10/27 05:27:24: Resolve 51: udp 192.168.0.4 55308
[9] 2008/10/27 05:27:24: SIP Tx udp:192.168.0.4:55308:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.4:55308;branch=z9hG4bK-d87543-e357671db26f8a7b-1--d87543-;rport=55308
From: "15" <sip:15@192.168.0.17>;tag=677d8039
To: "15" <sip:15@192.168.0.17>;tag=69df673bd9
Call-ID: 0874fd67c34d7260M2FlNjIwMmY4MTA3ZGNiNDgyNzFmN2I4YTQ4NmIxNTc.
CSeq: 17 REGISTER
Contact: <sip:15@192.168.0.4:55308;rinstance=710ef2fda04e0fbc>;expires=32
Content-Length: 0
[9] 2008/10/27 05:27:28: SIP Tr udp:127.0.0.1:5062:
INVITE sip:9980160006@localhost;user=phone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-10cf26a12f8eba1a2b94748c9612676d;rport
From: "Kitchen" <sip:15@localhost;user=phone>;tag=1167038513
To: <sip:9980160006@localhost;user=phone>
Call-ID: 17e82b84@pbx
CSeq: 6056 INVITE
Max-Forwards: 70
Contact: <sip:15@127.0.0.1:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Type: application/sdp
Content-Length: 268
v=0
o=- 233681333 233681333 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 61642 RTP/AVP 0 8 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2008/10/27 05:27:48: Last message repeated 2 times
[9] 2008/10/27 05:27:48: SIP Rx udp:192.168.0.2:5060:
REGISTER sip:192.168.0.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1762136931286821264;rport
From: 15 <sip:15@192.168.0.17:5060>;tag=202514052
To: 15 <sip:15@192.168.0.17:5060>
Call-ID: 2426712224-419229253@192.168.0.2
CSeq: 17 REGISTER
Contact: <sip:15@192.168.0.2:5060>
Max-Forwards: 70
Expires: 60
User-Agent: Voip Phone 1.0
Content-Length: 0
[9] 2008/10/27 05:27:48: Resolve 52: aaaa udp 192.168.0.2 5060
[9] 2008/10/27 05:27:48: Resolve 52: a udp 192.168.0.2 5060
[9] 2008/10/27 05:27:48: Resolve 52: udp 192.168.0.2 5060
[9] 2008/10/27 05:27:48: SIP Tx udp:192.168.0.2:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1762136931286821264;rport=5060
From: 15 <sip:15@192.168.0.17:5060>;tag=202514052
To: 15 <sip:15@192.168.0.17:5060>;tag=5012bb89f2
Call-ID: 2426712224-419229253@192.168.0.2
CSeq: 17 REGISTER
Contact: <sip:15@192.168.0.2:5060>;expires=31
Content-Length: 0
[7] 2008/10/27 05:27:50: Call 17e82b84@pbx#1167038513: Clear last INVITE
[5] 2008/10/27 05:27:50: INVITE Response: Terminate 17e82b84@pbx
[7] 2008/10/27 05:27:50: Other Ports: 1
[7] 2008/10/27 05:27:50: Call Port: 4373886024080-20982265928144@192.168.0.2#42c51291c3
[9] 2008/10/27 05:27:50: Resolve 53: aaaa udp 192.168.0.2 5060
[9] 2008/10/27 05:27:50: Resolve 53: a udp 192.168.0.2 5060
[9] 2008/10/27 05:27:50: Resolve 53: udp 192.168.0.2 5060
[9] 2008/10/27 05:27:50: SIP Tx udp:192.168.0.2:5060:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2158616889645429424;rport=5060
From: 15 <sip:15@192.168.0.17:5060>;tag=202761955
To: 9980160006 <sip:9980160006@192.168.0.17:5060>;tag=42c51291c3
Call-ID: 4373886024080-20982265928144@192.168.0.2
CSeq: 1 INVITE
Contact: <sip:15@192.168.0.17:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/3.0.1.3023
Content-Length: 0
[9] 2008/10/27 05:27:50: SIP Rx udp:192.168.0.2:5060:
ACK sip:9980160006@192.168.0.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2158616889645429424;rport
From: 15 <sip:15@192.168.0.17:5060>;tag=202761955
To: 9980160006 <sip:9980160006@192.168.0.17:5060>;tag=42c51291c3
Call-ID: 4373886024080-20982265928144@192.168.0.2
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[9] 2008/10/27 05:27:52: SIP Rx udp:192.168.0.4:55308:
REGISTER sip:192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:55308;branch=z9hG4bK-d87543-7d107733872f1f33-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:15@192.168.0.4:55308;rinstance=710ef2fda04e0fbc>
To: "15"<sip:15@192.168.0.17>
From: "15"<sip:15@192.168.0.17>;tag=677d8039
Call-ID: 0874fd67c34d7260M2FlNjIwMmY4MTA3ZGNiNDgyNzFmN2I4YTQ4NmIxNTc.
CSeq: 18 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1003l stamp 30942
Content-Length: 0
[9] 2008/10/27 05:27:52: Resolve 54: aaaa udp 192.168.0.4 55308
[9] 2008/10/27 05:27:52: Resolve 54: a udp 192.168.0.4 55308
[9] 2008/10/27 05:27:52: Resolve 54: udp 192.168.0.4 55308
[9] 2008/10/27 05:27:52: SIP Tx udp:192.168.0.4:55308:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.4:55308;branch=z9hG4bK-d87543-7d107733872f1f33-1--d87543-;rport=55308
From: "15" <sip:15@192.168.0.17>;tag=677d8039
To: "15" <sip:15@192.168.0.17>;tag=69df673bd9
Call-ID: 0874fd67c34d7260M2FlNjIwMmY4MTA3ZGNiNDgyNzFmN2I4YTQ4NmIxNTc.
CSeq: 18 REGISTER
Contact: <sip:15@192.168.0.4:55308;rinstance=710ef2fda04e0fbc>;expires=32
Content-Length: 0
-
Does pbxnsip support H.281 for far end camera control? Is there a way we can do this on SIP? The customer is using standard video phones with pbxnsip.
Regards
Ganesh
-
Hi,
We are running PBXNSIP version 3.0 on Linux centOS. The system was working fine and no changes were made. Its suddenly down and we have restarted it number of times but the service is not starting.
[root@cel-sip ~]# service httpd status
httpd (pid 2390 2388 2387 2386 2385 2384 2383 2382 2294) is running..
[root@cel-sip ~]# cd /etc/init.d
[root@cel-sip init.d]# ./pbxnsip restart
Stopping PBX:FAILED]
Starting PBX:
[root@cel-sip init.d]# ./pbxnsip status
pbxctrl is stopped
[root@cel-sip init.d]#
[root@cel-sip ~]# more /etc/init.d/pbxnsip
#!/bin/bash
#
# Init file for pbxnsip PBX
#
# Copyright © 2006 pbxnsip Inc., USA
#
# chkconfig: 2345 20 80
# description: SIP-based PBX
#
# processname: pbxctrl
# pidfile: /var/run/pbxctrl.pid
# source function library
. /etc/rc.d/init.d/functions
RETVAL=0
# Installation location
INSTALLDIR=/srv/pbx
PBX=$INSTALLDIR/pbxctrl
start()
{
echo -n "Starting PBX:"
$PBX --dir $INSTALLDIR
echo
RETVAL=1
}
stop()
{
echo -n "Stopping PBX:"
killproc $PBX -TERM
echo
RETVAL=1
}
case "$1" in
start)
start
;;
stop)
stop
;;
restart)
stop
start
;;
status)
status $PBX
RETVAL=$?
;;
*)
echo $"Usage: $0 {start|stop|restart|status}"
RETVAL=1
esac
exit $RETVAL
[root@cel-sip ~]#
Regards
Ganesh
-
We configured the outbound proxy and specified to use TLS (as Avaya only supports TLS on direct SIP trunk without Avaya SES). We were still unable to route calls between the two systems. Below is the logs captured. 59999 is the Avaya phone extension.
This link (http://www.avayausers.com/showthread.php?t=10700) says that "TCP Sip is supported, but UDP SIP is not supported without the Avaya Sip Server. (SES) Even then you still need a session border controller" (we are not using the Avaya SES here).
It also says SBC is required while using the SIP trunk on Avaya.
Since we are integrating this in the same network (LAN), do we need a SBC? Also I think SBC is inbuilt in PBXNSIP. Right?
Should i try using a SBC or is there any other settings we need to do from our side. Below is the logs captured from PBXNSIP.
[7] 2008/07/16 06:45:20: SIP Rx udp:192.168.38.21:12163:
INVITE sip:59999@192.168.38.21 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:79999@192.168.38.21:12163>
To: "59999"<sip:59999@192.168.38.21>
From: "79999"<sip:79999@192.168.38.21>;tag=20694b71
Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 423
v=0
o=- 6 2 IN IP4 192.168.38.21
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.38.21
t=0 0
m=audio 3622 RTP/AVP 107 119 100 106 0 105 98 8 3 101
a=alt:1 1 : qqzBZNsZ mIyNWmYl 192.168.38.21 3622
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
[7] 2008/07/16 06:45:20: UDP: Opening socket on port 49246
[7] 2008/07/16 06:45:20: UDP: Opening socket on port 49247
[8] 2008/07/16 06:45:20: Could not find a trunk (1 trunks)
[8] 2008/07/16 06:45:20: Using outbound proxy sip:192.168.38.21:12163;transport=udp because UDP packet source did not match the via header
[9] 2008/07/16 06:45:20: Resolve 50: udp 192.168.38.21 12163
[7] 2008/07/16 06:45:20: SIP Tx udp:192.168.38.21:12163:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21
From: "79999" <sip:79999@192.168.38.21>;tag=20694b71
To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412
Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.
CSeq: 1 INVITE
Content-Length: 0
[6] 2008/07/16 06:45:20: Sending RTP for MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.#6325fc9412 to 192.168.38.21:3622
[9] 2008/07/16 06:45:21: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 59999@192.168.38.21
[5] 2008/07/16 06:45:21: Dialplan New: Match 59999@192.168.38.21 to <sip:59999@192.168.38.20;user=phone> on trunk SIP
[8] 2008/07/16 06:45:21: Play audio_moh/noise.wav
[7] 2008/07/16 06:45:21: UDP: Opening socket on port 59432
[7] 2008/07/16 06:45:21: UDP: Opening socket on port 59433
[9] 2008/07/16 06:45:21: Resolve 51: url sip:192.168.38.20:5061;transport=tls
[9] 2008/07/16 06:45:21: Resolve 51: a tls 192.168.38.20 5061
[9] 2008/07/16 06:45:21: Resolve 51: tls 192.168.38.20 5061
[7] 2008/07/16 06:45:21: SIP Tx tls:192.168.38.20:5061:
INVITE sip:59999@192.168.38.20;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.38.21:1210;branch=z9hG4bK-48a3669a14150ffd1e6b3c48e9c5f659;rport
From: <sip:79999@localhost>;tag=5447
To: <sip:59999@192.168.38.20;user=phone>
Call-ID: ff5f3792@pbx
CSeq: 6271 INVITE
Max-Forwards: 70
Contact: <sip:79999@192.168.38.21:1210;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Type: application/sdp
Content-Length: 423
v=0
o=- 28504 28504 IN IP4 192.168.38.21
s=-
c=IN IP4 192.168.38.21
t=0 0
m=audio 59432 RTP/AVP 0 8 9 18 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UxcnvE8+sfevfgeIrnn35dXnjcuAf3Ikos1Dnk3f
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[9] 2008/07/16 06:45:21: Resolve 52: udp 192.168.38.21 12163
[7] 2008/07/16 06:45:21: SIP Tx udp:192.168.38.21:12163:
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21
From: "79999" <sip:79999@192.168.38.21>;tag=20694b71
To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412
Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.
CSeq: 1 INVITE
Contact: <sip:79999@127.0.0.1:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Type: application/sdp
Content-Length: 233
v=0
o=- 16981 16981 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 49246 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[6] 2008/07/16 06:45:21: Sending RTP for MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.#6325fc9412 to 127.0.0.1:3622
[7] 2008/07/16 06:45:21: SIP Tr udp:192.168.38.21:12163:
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21
From: "79999" <sip:79999@192.168.38.21>;tag=20694b71
To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412
Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.
CSeq: 1 INVITE
Contact: <sip:79999@127.0.0.1:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Type: application/sdp
Content-Length: 233
v=0
o=- 16981 16981 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 49246 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[7] 2008/07/16 06:45:22: SIP Rx udp:192.168.38.21:12163:
REGISTER sip:192.168.38.21 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-8e7bc46eb755e302-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>
To: "79999"<sip:79999@192.168.38.21>
From: "79999"<sip:79999@192.168.38.21>;tag=0948d273
Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ.
CSeq: 26 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0
[9] 2008/07/16 06:45:22: Resolve 53: udp 192.168.38.21 12163
[7] 2008/07/16 06:45:22: SIP Tx udp:192.168.38.21:12163:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-8e7bc46eb755e302-1---d8754z-;rport=12163;received=192.168.38.21
From: "79999" <sip:79999@192.168.38.21>;tag=0948d273
To: "79999" <sip:79999@192.168.38.21>;tag=11399c85b8
Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ.
CSeq: 26 REGISTER
Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>;expires=31
Content-Length: 0
[7] 2008/07/16 06:45:22: SIP Tr udp:192.168.38.21:12163:
SIP/2.0 183 Ringing
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21
From: "79999" <sip:79999@192.168.38.21>;tag=20694b71
To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412
Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.
CSeq: 1 INVITE
Contact: <sip:79999@127.0.0.1:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Type: application/sdp
Content-Length: 233
v=0
o=- 16981 16981 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 49246 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[7] 2008/07/16 06:45:41: Last message repeated 4 times
[5] 2008/07/16 06:45:41: SIP port accept from 192.168.38.14:24434
[7] 2008/07/16 06:45:48: SIP Rx udp:192.168.38.21:12163:
REGISTER sip:192.168.38.21 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-0e56b011be31d71c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>
To: "79999"<sip:79999@192.168.38.21>
From: "79999"<sip:79999@192.168.38.21>;tag=0948d273
Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ.
CSeq: 27 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0
[9] 2008/07/16 06:45:48: Resolve 54: udp 192.168.38.21 12163
[7] 2008/07/16 06:45:48: SIP Tx udp:192.168.38.21:12163:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-0e56b011be31d71c-1---d8754z-;rport=12163;received=192.168.38.21
From: "79999" <sip:79999@192.168.38.21>;tag=0948d273
To: "79999" <sip:79999@192.168.38.21>;tag=11399c85b8
Call-ID: OGY2MmIzNjUwY2VkYjFmM2IyMjVhYWI5MDYzNTEwYzQ.
CSeq: 27 REGISTER
Contact: <sip:79999@192.168.38.21:12163;rinstance=655b23b153707a14>;expires=29
Content-Length: 0
[7] 2008/07/16 06:45:51: Call ff5f3792@pbx#5447: Clear last INVITE
[9] 2008/07/16 06:45:51: Resolve 55: udp 192.168.38.21 12163
[7] 2008/07/16 06:45:51: SIP Tx udp:192.168.38.21:12163:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport=12163;received=192.168.38.21
From: "79999" <sip:79999@192.168.38.21>;tag=20694b71
To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412
Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.
CSeq: 1 INVITE
Contact: <sip:79999@127.0.0.1:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Length: 0
[7] 2008/07/16 06:45:51: SIP Rx udp:192.168.38.21:12163:
ACK sip:59999@192.168.38.21 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:12163;branch=z9hG4bK-d8754z-b414712c090b0d5c-1---d8754z-;rport
To: "59999" <sip:59999@192.168.38.21>;tag=6325fc9412
From: "79999"<sip:79999@192.168.38.21>;tag=20694b71
Call-ID: MzFkNGI1MGJjNmZmMjMyYTBhYzBiOWY5NmQzM2YxNDQ.
CSeq: 1 ACK
Content-Length: 0
[7] 2008/07/16 06:45:51: Other Ports: 1
[7] 2008/07/16 06:45:51: Call Port: ff5f3792@pbx#5447
[8] 2008/07/16 06:45:59: Hangup: Call ff5f3792@pbx#5447 not found
-
Sometimes the registration of phones goes off. If the pbxnsip service is restarted, the phones then gets registered. Below is the logs taken before the service was restarted.
[7] 2008/07/14 12:06:09:
SIP Rx udp:10.255.109.71:5060:
REGISTER sip:10.255.10.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.255.109.71:5060;branch=z9hG4bK8277290331884624686;rport
From: 9 <sip:9@10.255.10.41:5060>;tag=72856
To: 9 <sip:9@10.255.10.41:5060>
Call-ID: 156118939-2437221063@10.255.109.71
CSeq: 79 REGISTER
Contact: <sip:9@10.255.109.71:5060>
Max-Forwards: 70
Expires: 60
User-Agent: Voip Phone 1.0
Content-Length: 0
[9] 2008/07/14 12:06:09:
Resolve 1427946: aaaa udp 10.255.109.71 5060
[9] 2008/07/14 12:06:09:
Resolve 1427946: a udp 10.255.109.71 5060
[9] 2008/07/14 12:06:09:
Resolve 1427946: udp 10.255.109.71 5060
[7] 2008/07/14 12:06:09:
SIP Tx udp:10.255.109.71:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.255.109.71:5060;branch=z9hG4bK8277290331884624686;rport=5060
From: 9 <sip:9@10.255.10.41:5060>;tag=72856
To: 9 <sip:9@10.255.10.41:5060>;tag=01b3b5008d
Call-ID: 156118939-2437221063@10.255.109.71
CSeq: 79 REGISTER
Contact: <sip:9@10.255.109.71:5060>;expires=28
Content-Length: 0
[7] 2008/07/14 12:06:10:
SIP Rx udp:10.255.109.224:5060:
REGISTER sip:10.255.10.41 SIP/2.0
Via: SIP/2.0/UDP 10.255.109.224:5060;branch=z9hG4bK-6rcuu3t7ad56;rport
From: <sip:43@10.255.10.41>;tag=y2otc1x8wx
To: <sip:43@10.255.10.41>
Call-ID: 3c2670094baf-p6vrwsbpjxvf@10-255-109-224
CSeq: 155034 REGISTER
Max-Forwards: 70
Contact: <sip:43@10.255.109.224:5060;line=o9r6pvlu>;q=1.0
User-Agent: snom190-3.56y
P-NAT-Refresh: 15
Supported: gruu
Allow-Events: dialog
X-Real-IP: 10.255.109.224
WWW-Contact: <http://10.255.109.224:80>
WWW-Contact: <https://10.255.109.224:443>
Expires: 3600
Content-Length: 0
[9] 2008/07/14 12:06:10:
Resolve 1427947: aaaa udp 10.255.109.224 5060
[9] 2008/07/14 12:06:10:
Resolve 1427947: a udp 10.255.109.224 5060
[9] 2008/07/14 12:06:10:
Resolve 1427947: udp 10.255.109.224 5060
[7] 2008/07/14 12:06:10:
SIP Tx udp:10.255.109.224:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.255.109.224:5060;branch=z9hG4bK-6rcuu3t7ad56;rport=5060
From: <sip:43@10.255.10.41>;tag=y2otc1x8wx
To: <sip:43@10.255.10.41>;tag=090a661bac
Call-ID: 3c2670094baf-p6vrwsbpjxvf@10-255-109-224
CSeq: 155034 REGISTER
Contact: <sip:43@10.255.109.224:5060;line=o9r6pvlu>;expires=32
Content-Length: 0
[7] 2008/07/14 12:06:12:
SIP Rx udp:10.255.109.195:5060:
REGISTER sip:10.255.10.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.255.109.195:5060;branch=z9hG4bK1116514926326271659;rport
From: 2204 <sip:2204@10.255.10.41:5060>;tag=69916352
To: 2204 <sip:2204@10.255.10.41:5060>
Call-ID: 35631225-1727829012@10.255.109.195
CSeq: 80 REGISTER
Contact: <sip:2204@10.255.109.195:5060>
Max-Forwards: 70
Expires: 60
User-Agent: Voip Phone 1.0
Content-Length: 0
[9] 2008/07/14 12:06:12:
Resolve 1427948: aaaa udp 10.255.109.195 5060
[9] 2008/07/14 12:06:12:
Resolve 1427948: a udp 10.255.109.195 5060
[9] 2008/07/14 12:06:12:
Resolve 1427948: udp 10.255.109.195 5060
[7] 2008/07/14 12:06:12:
SIP Tx udp:10.255.109.195:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.255.109.195:5060;branch=z9hG4bK1116514926326271659;rport=5060
From: 2204 <sip:2204@10.255.10.41:5060>;tag=69916352
To: 2204 <sip:2204@10.255.10.41:5060>;tag=ee3bc096d6
Call-ID: 35631225-1727829012@10.255.109.195
CSeq: 80 REGISTER
Contact: <sip:2204@10.255.109.195:5060>;expires=32
Content-Length: 0
[7] 2008/07/14 12:06:13:
SIP Rx udp:10.255.104.164:5060:
REGISTER sip:10.255.10.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.255.104.164:5060;branch=z9hG4bK2418523389318324723;rport
From: 4236 <sip:4236@10.255.10.41:5060>;tag=135019576
To: 4236 <sip:4236@10.255.10.41:5060>
Call-ID: 451013859-679113950@10.255.104.164
CSeq: 80 REGISTER
Contact: <sip:4236@10.255.104.164:5060>
Max-Forwards: 70
Expires: 60
User-Agent: Voip Phone 1.0
Content-Length: 0
[9] 2008/07/14 12:06:13:
Resolve 1427949: aaaa udp 10.255.104.164 5060
[9] 2008/07/14 12:06:13:
Resolve 1427949: a udp 10.255.104.164 5060
[9] 2008/07/14 12:06:13:
Resolve 1427949: udp 10.255.104.164 5060
[7] 2008/07/14 12:06:13:
SIP Tx udp:10.255.104.164:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.255.104.164:5060;branch=z9hG4bK2418523389318324723;rport=5060
From: 4236 <sip:4236@10.255.10.41:5060>;tag=135019576
To: 4236 <sip:4236@10.255.10.41:5060>;tag=dd7e486634
Call-ID: 451013859-679113950@10.255.104.164
CSeq: 80 REGISTER
Contact: <sip:4236@10.255.104.164:5060>;expires=31
Content-Length: 0
[7] 2008/07/14 12:06:16:
SIP Rx udp:10.255.104.111:5060:
REGISTER sip:10.255.10.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.255.104.111:5060;branch=z9hG4bK11025564791679386;rport
From: 3166 <sip:3166@10.255.10.41:5060>;tag=135019576
To: 3166 <sip:3166@10.255.10.41:5060>
Call-ID: 2046212465-2842810015@10.255.104.111
CSeq: 79 REGISTER
Contact: <sip:3166@10.255.104.111:5060>
Max-Forwards: 70
Expires: 60
User-Agent: Voip Phone 1.0
Content-Length: 0
[9] 2008/07/14 12:06:16:
Resolve 1427950: aaaa udp 10.255.104.111 5060
[9] 2008/07/14 12:06:16:
Resolve 1427950: a udp 10.255.104.111 5060
[9] 2008/07/14 12:06:16:
Resolve 1427950: udp 10.255.104.111 5060
[7] 2008/07/14 12:06:16:
SIP Tx udp:10.255.104.111:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.255.104.111:5060;branch=z9hG4bK11025564791679386;rport=5060
From: 3166 <sip:3166@10.255.10.41:5060>;tag=135019576
To: 3166 <sip:3166@10.255.10.41:5060>;tag=a9e73364a9
Call-ID: 2046212465-2842810015@10.255.104.111
CSeq: 79 REGISTER
Contact: <sip:3166@10.255.104.111:5060>;expires=29
Content-Length: 0
[7] 2008/07/14 12:06:17:
SIP Rx udp:10.255.115.95:5060:
REGISTER sip:10.255.10.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.255.115.95:5060;branch=z9hG4bK1955631062293118923;rport
From: 2156 <sip:2156@10.255.10.41:5060>;tag=750122006
To: 2156 <sip:2156@10.255.10.41:5060>
Call-ID: 222317741-141084356@10.255.115.95
CSeq: 79 REGISTER
Contact: <sip:2156@10.255.115.95:5060>
Max-Forwards: 70
Expires: 60
User-Agent: Voip Phone 1.0
Content-Length: 0
[9] 2008/07/14 12:06:17:
Resolve 1427951: aaaa udp 10.255.115.95 5060
[9] 2008/07/14 12:06:17:
Resolve 1427951: a udp 10.255.115.95 5060
[9] 2008/07/14 12:06:17:
Resolve 1427951: udp 10.255.115.95 5060
[7] 2008/07/14 12:06:17:
SIP Tx udp:10.255.115.95:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.255.115.95:5060;branch=z9hG4bK1955631062293118923;rport=5060
From: 2156 <sip:2156@10.255.10.41:5060>;tag=750122006
To: 2156 <sip:2156@10.255.10.41:5060>;tag=c43cb57955
Call-ID: 222317741-141084356@10.255.115.95
CSeq: 79 REGISTER
Contact: <sip:2156@10.255.115.95:5060>;expires=30
Content-Length: 0
[7] 2008/07/14 12:06:17:
SIP Rx udp:10.255.109.70:5060:
REGISTER sip:10.255.10.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.255.109.70:5060;branch=z9hG4bK25280275672505723734;rport
From: 1233 <sip:1233@10.255.10.41:5060>;tag=496127447
To: 1233 <sip:1233@10.255.10.41:5060>
Call-ID: 261183116-1195921186@10.255.109.70
CSeq: 79 REGISTER
Contact: <sip:1233@10.255.109.70:5060>
Max-Forwards: 70
Expires: 60
User-Agent: Voip Phone 1.0
Content-Length: 0
[9] 2008/07/14 12:06:17:
Resolve 1427952: aaaa udp 10.255.109.70 5060
[9] 2008/07/14 12:06:17:
Resolve 1427952: a udp 10.255.109.70 5060
[9] 2008/07/14 12:06:17:
Resolve 1427952: udp 10.255.109.70 5060
[7] 2008/07/14 12:06:17:
SIP Tx udp:10.255.109.70:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.255.109.70:5060;branch=z9hG4bK25280275672505723734;rport=5060
From: 1233 <sip:1233@10.255.10.41:5060>;tag=496127447
To: 1233 <sip:1233@10.255.10.41:5060>;tag=4d22fcf143
Call-ID: 261183116-1195921186@10.255.109.70
CSeq: 79 REGISTER
Contact: <sip:1233@10.255.109.70:5060>;expires=31
Content-Length: 0
[4] 2008/07/14 12:06:19:
select returns error
[4] 2008/07/14 12:11:39:
Last message repeated 97657610 times
[5] 2008/07/14 12:11:39:
RTP Timeout on 28338137515610-209372138317486@10.255.104.164#307532807f
[9] 2008/07/14 12:11:39:
Resolve 1427953: url sip:4236@10.255.104.164:5060
[9] 2008/07/14 12:11:39:
Resolve 1427953: udp 10.255.104.164 5060
[7] 2008/07/14 12:11:39:
03712b05@pbx#504965278: Media-aware pass-through mode
[4] 2008/07/14 12:11:39:
select returns error
[4] 2008/07/14 12:11:39:
Last message repeated 19 times
[9] 2008/07/14 12:11:39:
Resolve 1427954: url sip:202.71.134.13:5060
[9] 2008/07/14 12:11:39:
Resolve 1427954: udp 202.71.134.13 5060
[4] 2008/07/14 12:11:39:
select returns error
[4] 2008/07/14 12:11:47:
Last message repeated 2309856 times
[5] 2008/07/14 12:11:47:
Call 03712b05@pbx#504965278: Last request not finished
[9] 2008/07/14 12:11:47:
Resolve 1427955: url sip:202.71.134.13:5060
[9] 2008/07/14 12:11:47:
Resolve 1427955: udp 202.71.134.13 5060
[8] 2008/07/14 12:11:47:
Hangup: Call 03712b05@pbx#504965278 not found
[4] 2008/07/14 12:11:47:
select returns error
-
We are integrating pbxnsip with Avaya system on SIP trunk. We have configured SIP trunk (gateway mode) on pbxnsip and Avaya. But unable to route calls both ways between the two system. Attached is the logs and wireshark traces captured.
4229 and 4431 is extension (Avaya phones) in Avaya. 2201 and 2202 is extensions (Snom phones) on pbxnsip. Avaya uses port 5061 and supports only TLS on SIP trunk.
Please let me know if you can get some information from these logs. Below is one more log taken from pbxnsip while trying to call Avaya phones.
[5] 2008/07/07 04:23:17: SIP port accept from 192.168.192.28:14935
[7] 2008/07/07 04:23:20: SIP Rx udp:192.168.136.36:2051:
REGISTER sip:192.168.192.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-81zc3zm6vbe4;rport
From: <sip:5203@192.168.192.50>;tag=3eez6gof7x
To: <sip:5203@192.168.192.50>
Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7
CSeq: 1033 REGISTER
Max-Forwards: 70
Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/6.5.13
Supported: gruu
Allow-Events: dialog
X-Real-IP: 192.168.136.36
WWW-Contact: <http://192.168.136.36:80>
WWW-Contact: <https://192.168.136.36:443>
Expires: 3600
Content-Length: 0
[9] 2008/07/07 04:23:20: Resolve 47: aaaa udp 192.168.136.36 2051
[9] 2008/07/07 04:23:20: Resolve 47: a udp 192.168.136.36 2051
[9] 2008/07/07 04:23:20: Resolve 47: udp 192.168.136.36 2051
[7] 2008/07/07 04:23:20: SIP Tx udp:192.168.136.36:2051:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-81zc3zm6vbe4;rport=2051
From: <sip:5203@192.168.192.50>;tag=3eez6gof7x
To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e
Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7
CSeq: 1033 REGISTER
Content-Length: 0
[7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053:
INVITE sip:4229@192.168.192.50;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport
From: <sip:2201@192.168.192.50>;tag=p10csqj5hf
To: <sip:4229@192.168.192.50;user=phone>
Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom300/6.5.13
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 345
v=0
o=root 576733664 576733664 IN IP4 192.168.25.103
s=call
c=IN IP4 192.168.25.103
t=0 0
m=audio 58152 RTP/AVP 18 4 0 8 3 9 2 101
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[8] 2008/07/07 04:23:26: Packet authenticated by transport layer
[7] 2008/07/07 04:23:26: UDP: Opening socket on port 52908
[7] 2008/07/07 04:23:26: UDP: Opening socket on port 52909
[8] 2008/07/07 04:23:26: Could not find a trunk (1 trunks)
[9] 2008/07/07 04:23:26: Using outbound proxy sip:192.168.25.103:2053;transport=tls because of flow-label
[9] 2008/07/07 04:23:26: Resolve 48: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053
From: <sip:2201@192.168.192.50>;tag=p10csqj5hf
To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48
Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE
CSeq: 1 INVITE
Content-Length: 0
[7] 2008/07/07 04:23:26: Set packet length to 20
[6] 2008/07/07 04:23:26: Sending RTP for 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE#1c9ec29c48 to 192.168.25.103:58152
[9] 2008/07/07 04:23:26: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 4229@192.168.192.50
[5] 2008/07/07 04:23:26: Dialplan New: Match 4229@192.168.192.50 to <sip:4229@192.168.192.28:5061;user=phone> on trunk SIP
[8] 2008/07/07 04:23:26: Play audio_moh/noise.wav
[7] 2008/07/07 04:23:26: UDP: Opening socket on port 49714
[7] 2008/07/07 04:23:26: UDP: Opening socket on port 49715
[9] 2008/07/07 04:23:26: Resolve 49: url sip:192.168.192.28:5061
[9] 2008/07/07 04:23:26: Resolve 49: udp 192.168.192.28 5061
[7] 2008/07/07 04:23:26: SIP Tx udp:192.168.192.28:5061:
INVITE sip:4229@192.168.192.28:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.192.50:5060;branch=z9hG4bK-017e34d40401d0870149413127470191;rport
From: <sip:2201@localhost>;tag=48313
To: <sip:4229@192.168.192.28:5061;user=phone>
Call-ID: 910d81bb@pbx
CSeq: 7463 INVITE
Max-Forwards: 70
Contact: <sip:2201@192.168.192.50:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Type: application/sdp
Content-Length: 294
v=0
o=- 56787 56787 IN IP4 192.168.192.50
s=-
c=IN IP4 192.168.192.50
t=0 0
m=audio 49714 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[7] 2008/07/07 04:23:26: Set packet length to 20
[9] 2008/07/07 04:23:26: Resolve 50: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 183 Ringing
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053
From: <sip:2201@192.168.192.50>;tag=p10csqj5hf
To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48
Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE
CSeq: 1 INVITE
Contact: <sip:2201@192.168.192.50:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 304
v=0
o=- 7292 7292 IN IP4 192.168.192.50
s=-
c=IN IP4 192.168.192.50
t=0 0
m=audio 52908 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[8] 2008/07/07 04:23:26: UDP: recvfrom receives ICMP message
[5] 2008/07/07 04:23:26: Connection refused on udp:192.168.192.28:5061
[6] 2008/07/07 04:23:26: Could not determine destination address on 49
[7] 2008/07/07 04:23:26: Call 910d81bb@pbx#48313: Clear last INVITE
[9] 2008/07/07 04:23:26: Resolve 51: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 500 Network Failure
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport=2053
From: <sip:2201@192.168.192.50>;tag=p10csqj5hf
To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48
Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE
CSeq: 1 INVITE
Contact: <sip:2201@192.168.192.50:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Length: 0
[7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053:
PRACK sip:2201@192.168.192.50:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-on4zow98h369;rport
From: <sip:2201@192.168.192.50>;tag=p10csqj5hf
To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48
Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
[8] 2008/07/07 04:23:26: Packet authenticated by transport layer
[9] 2008/07/07 04:23:26: Resolve 52: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:26: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-on4zow98h369;rport=2053
From: <sip:2201@192.168.192.50>;tag=p10csqj5hf
To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48
Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE
CSeq: 2 PRACK
Contact: <sip:2201@192.168.192.50:5061;transport=tls>
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Length: 0
[7] 2008/07/07 04:23:26: SIP Rx tls:192.168.25.103:2053:
ACK sip:4229@192.168.192.50;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-ee8owjl1jdd6;rport
From: <sip:2201@192.168.192.50>;tag=p10csqj5hf
To: <sip:4229@192.168.192.50;user=phone>;tag=1c9ec29c48
Call-ID: 3c267129f1b3-ssnraxa08a0l@snom300-0004132889BE
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1
Content-Length: 0
[8] 2008/07/07 04:23:26: Packet authenticated by transport layer
[7] 2008/07/07 04:23:26: Other Ports: 1
[7] 2008/07/07 04:23:26: Call Port: 910d81bb@pbx#48313
[7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053:
INVITE sip:4431@192.168.192.50;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport
From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw
To: <sip:4431@192.168.192.50;user=phone>
Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom300/6.5.13
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 347
v=0
o=root 1459444772 1459444772 IN IP4 192.168.25.103
s=call
c=IN IP4 192.168.25.103
t=0 0
m=audio 58646 RTP/AVP 18 4 0 8 3 9 2 101
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[8] 2008/07/07 04:23:30: Packet authenticated by transport layer
[7] 2008/07/07 04:23:30: UDP: Opening socket on port 52682
[7] 2008/07/07 04:23:30: UDP: Opening socket on port 52683
[8] 2008/07/07 04:23:30: Could not find a trunk (1 trunks)
[9] 2008/07/07 04:23:30: Using outbound proxy sip:192.168.25.103:2053;transport=tls because of flow-label
[9] 2008/07/07 04:23:30: Resolve 53: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053
From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw
To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863
Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE
CSeq: 1 INVITE
Content-Length: 0
[7] 2008/07/07 04:23:30: Set packet length to 20
[6] 2008/07/07 04:23:30: Sending RTP for 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE#6f162b9863 to 192.168.25.103:58646
[9] 2008/07/07 04:23:30: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 4431@192.168.192.50
[5] 2008/07/07 04:23:30: Dialplan New: Match 4431@192.168.192.50 to <sip:4431@192.168.192.28:5061;user=phone> on trunk SIP
[8] 2008/07/07 04:23:30: Play audio_moh/noise.wav
[7] 2008/07/07 04:23:30: UDP: Opening socket on port 52286
[7] 2008/07/07 04:23:30: UDP: Opening socket on port 52287
[9] 2008/07/07 04:23:30: Resolve 54: url sip:192.168.192.28:5061
[9] 2008/07/07 04:23:30: Resolve 54: udp 192.168.192.28 5061
[7] 2008/07/07 04:23:30: SIP Tx udp:192.168.192.28:5061:
INVITE sip:4431@192.168.192.28:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.192.50:5060;branch=z9hG4bK-f969c5b8969691bf078c04d44f93e63f;rport
From: <sip:2201@localhost>;tag=17880
To: <sip:4431@192.168.192.28:5061;user=phone>
Call-ID: 63864075@pbx
CSeq: 29415 INVITE
Max-Forwards: 70
Contact: <sip:2201@192.168.192.50:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Type: application/sdp
Content-Length: 292
v=0
o=- 2970 2970 IN IP4 192.168.192.50
s=-
c=IN IP4 192.168.192.50
t=0 0
m=audio 52286 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[7] 2008/07/07 04:23:30: Set packet length to 20
[9] 2008/07/07 04:23:30: Resolve 55: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 183 Ringing
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053
From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw
To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863
Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE
CSeq: 1 INVITE
Contact: <sip:2201@192.168.192.50:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 306
v=0
o=- 28977 28977 IN IP4 192.168.192.50
s=-
c=IN IP4 192.168.192.50
t=0 0
m=audio 52682 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[8] 2008/07/07 04:23:30: UDP: recvfrom receives ICMP message
[5] 2008/07/07 04:23:30: Connection refused on udp:192.168.192.28:5061
[6] 2008/07/07 04:23:30: Could not determine destination address on 54
[7] 2008/07/07 04:23:30: Call 63864075@pbx#17880: Clear last INVITE
[9] 2008/07/07 04:23:30: Resolve 56: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 500 Network Failure
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport=2053
From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw
To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863
Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE
CSeq: 1 INVITE
Contact: <sip:2201@192.168.192.50:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Length: 0
[7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053:
PRACK sip:2201@192.168.192.50:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-xb9k7e6khimn;rport
From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw
To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863
Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
[8] 2008/07/07 04:23:30: Packet authenticated by transport layer
[9] 2008/07/07 04:23:30: Resolve 57: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:30: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-xb9k7e6khimn;rport=2053
From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw
To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863
Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE
CSeq: 2 PRACK
Contact: <sip:2201@192.168.192.50:5061;transport=tls>
User-Agent: pbxnsip-PBX/2.1.6.2450
Content-Length: 0
[7] 2008/07/07 04:23:30: SIP Rx tls:192.168.25.103:2053:
ACK sip:4431@192.168.192.50;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-hs7l7aoujegq;rport
From: <sip:2201@192.168.192.50>;tag=o26x9ykeqw
To: <sip:4431@192.168.192.50;user=phone>;tag=6f162b9863
Call-ID: 3c26712f0c35-qmnzaj2w5mti@snom300-0004132889BE
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1
Content-Length: 0
[8] 2008/07/07 04:23:30: Packet authenticated by transport layer
[7] 2008/07/07 04:23:30: Other Ports: 2
[7] 2008/07/07 04:23:30: Call Port: 63864075@pbx#17880
[7] 2008/07/07 04:23:30: Call Port: 910d81bb@pbx#48313
[8] 2008/07/07 04:23:34: Hangup: Call 910d81bb@pbx#48313 not found
[7] 2008/07/07 04:23:36: SIP Rx udp:192.168.136.36:2051:
REGISTER sip:192.168.192.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-e5flcagayes3;rport
From: <sip:5203@192.168.192.50>;tag=cm2h1pdaec
To: <sip:5203@192.168.192.50>
Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7
CSeq: 1034 REGISTER
Max-Forwards: 70
Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/6.5.13
Supported: gruu
Allow-Events: dialog
X-Real-IP: 192.168.136.36
WWW-Contact: <http://192.168.136.36:80>
WWW-Contact: <https://192.168.136.36:443>
Expires: 3600
Content-Length: 0
[9] 2008/07/07 04:23:36: Resolve 58: aaaa udp 192.168.136.36 2051
[9] 2008/07/07 04:23:36: Resolve 58: a udp 192.168.136.36 2051
[9] 2008/07/07 04:23:36: Resolve 58: udp 192.168.136.36 2051
[7] 2008/07/07 04:23:36: SIP Tx udp:192.168.136.36:2051:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-e5flcagayes3;rport=2051
From: <sip:5203@192.168.192.50>;tag=cm2h1pdaec
To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e
Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7
CSeq: 1034 REGISTER
Content-Length: 0
[8] 2008/07/07 04:23:38: Hangup: Call 63864075@pbx#17880 not found
[7] 2008/07/07 04:23:45: SIP Rx tls:192.168.25.103:2053:
REGISTER sip:192.168.192.50 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-8ruwlxvmnsxh;rport
From: <sip:2201@192.168.192.50>;tag=zn5jibw3c8
To: <sip:2201@192.168.192.50>
Call-ID: 3c267013a604-k4l8r8hzrhkc@snom300-0004132889BE
CSeq: 5 REGISTER
Max-Forwards: 70
Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:5c780463-7a16-4199-bb5c-a029eae57121>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/6.5.13
Supported: gruu
Allow-Events: dialog
X-Real-IP: 192.168.25.103
WWW-Contact: <http://192.168.25.103:80>
WWW-Contact: <https://192.168.25.103:443>
Expires: 3600
Content-Length: 0
[8] 2008/07/07 04:23:45: Packet authenticated by transport layer
[9] 2008/07/07 04:23:45: Resolve 59: tls 192.168.25.103 2053
[7] 2008/07/07 04:23:45: SIP Tx tls:192.168.25.103:2053:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.25.103:2053;branch=z9hG4bK-8ruwlxvmnsxh;rport=2053
From: <sip:2201@192.168.192.50>;tag=zn5jibw3c8
To: <sip:2201@192.168.192.50>;tag=f9ed8b9df9
Call-ID: 3c267013a604-k4l8r8hzrhkc@snom300-0004132889BE
CSeq: 5 REGISTER
Contact: <sip:2201@192.168.25.103:2053;transport=tls;line=ff0r423h>;expires=178
Content-Length: 0
[5] 2008/07/07 04:23:46: SIP port accept from 192.168.192.28:14946
[7] 2008/07/07 04:23:51: SIP Rx udp:192.168.136.36:2051:
REGISTER sip:192.168.192.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-amoqwm4o901i;rport
From: <sip:5203@192.168.192.50>;tag=g4q9ekm9mp
To: <sip:5203@192.168.192.50>
Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7
CSeq: 1035 REGISTER
Max-Forwards: 70
Contact: <sip:5203@192.168.136.36:2051;line=vfooudzb>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:519aec13-5a5d-4f46-92d3-aa8451bb25aa>";audio;mobility="fixed";duplex="full";description="snom300";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
User-Agent: snom300/6.5.13
Supported: gruu
Allow-Events: dialog
X-Real-IP: 192.168.136.36
WWW-Contact: <http://192.168.136.36:80>
WWW-Contact: <https://192.168.136.36:443>
Expires: 3600
Content-Length: 0
[9] 2008/07/07 04:23:51: Resolve 60: aaaa udp 192.168.136.36 2051
[9] 2008/07/07 04:23:51: Resolve 60: a udp 192.168.136.36 2051
[9] 2008/07/07 04:23:51: Resolve 60: udp 192.168.136.36 2051
[7] 2008/07/07 04:23:51: SIP Tx udp:192.168.136.36:2051:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.136.36:2051;branch=z9hG4bK-amoqwm4o901i;rport=2051
From: <sip:5203@192.168.192.50>;tag=g4q9ekm9mp
To: <sip:5203@192.168.192.50>;tag=1ee70c8e7e
Call-ID: 3c26702d249f-1evyeo0n739p@snom300-0004132889B7
CSeq: 1035 REGISTER
Content-Length: 0
-
Customer is having MySQL 5 database for their office employees. They are planning to integrate the PBXNSIP IVR tree with this. The application is if an employee calls from outside to the office, the IVR tree of PBXNSIP directs the call to the database server. The employee then needs to enter his ID number and password for authentication. He will then get a series of options from the database IVR. For ex: If the employee has to reach the HR department and apply for a leave he will then enter the date for leave application. The HR department will then get an alert from the database server.
The complete employee portal is available in this database server. The administrator will add, modify and delete when required. The PBXNSIP has to integrate to the MySQL server. Will there any any problem/limit by doing this? How many calls can the PBXNSIP IVR handle simultaneously?
Daily CDR in MS-Excel/CSV file format
in Product/Site Comments
Posted
Hello,
I would like to know if there is a possibility of this feature being included in the new release.
Ganesh