p800aul
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Posts posted by p800aul
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Please could you send me your patton configuration file. I have the same problem with the same issue. I find your snom file config usefull but I still have no reaction with the trunk.
Could you explain me step by step haow to connect both devices.
Thanks in advance, Etienne.
Here's my running config you will need to edit it to work with your setup, unless the snom one pbx is at 192.168.1.13 on your network. edit at interface sip IF_SIP_1 and interface sip IF_SIP_2.
As far as step by step you load this file in to the patton (after the edit) setup a trunk as per the previous post and it should work, as it does for me. If you can not get it to work and you are sure it's the patton give patton tech support a call they are very good indeed and can remotly set the thing up for you, if you are really stuck. Snom one should just work as setup above, if it doesn't publish your config and if i cannot see whats wrong someone else will (i'm new aswell)
I assume you have a dialplan?
regards
Paul
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You may want to check this post out
http://forum.snom.com/index.php?showtopic=5804
Worth reading it all
I now have it working by the way
Regards
Paul
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Hi
This is how our 4112 2 line fxo is setup on a trunk. Notes are (in brackets)
I hope it helps
if you need my running config of the patton let me know and i'll post it.
Regards
Paul
# Trunk 5 in domain localhost
Name: Patton
Type: register
To: sip (nothing set)
RegPass: ******** (nothing set)
Direction: (Inbound outbound)
Disabled: false
Global: false
Display:
RegAccount:
RegRegistrar: 192.168.1.XXX (the IP address of the Patton)
RegKeep:
RegUser:
Icid:
Require:
OutboundProxy: 192.168.1.XXX (the IP address of the Patton)
Ani:
DialExtension: 72 (the Hunt group)
Prefix:
Trusted: false
AcceptRedirect: false
RfcRtp: false
Analog: false
SendEmail:
UseUuid: false
Ring180: false
Failover: only_5xx
Privacy: false
Glob:
RequestTimeout:
Codecs:
CodecLock: true
Expires: 3600
FromUser:
Tel: true
TranscodeDtmf: false
AssociatedAddresses:
InterOffice: false
DialPlan:
Colines:
DialogPermission:
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Whow, so that means you need a new firmware for the gateway? Or was is a Patton config option?
Config, i was told originally that not having a line attached to the second FXO port wouldn't matter, it seems it does! They made a couple of changes to my original config the settings on the FXO interface so it will recognize a longer or shorter tone break and see this as a disconnect and go back on-hook to be ready to accept another call.
Secondly, disabled cyclic routing in your hunt group. Since I only have one interface working at the moment, the hunt will now try the first interface over and over. Apparently this is easy change back when I need that 2nd interface in routing.
Out of interest for everyone, here is my running config for a Patton 4112 with only one pstn line attached, I'm in the UK.
Regards Paul
#----------------------------------------------------------------#
# #
# SN4112/JO/EUI #
# R5.2 2009-01-14 H323 SIP FXS FXO #
# 2011-01-26T07:26:33 #
# SN/00A0BA0609C0 #
# Generated configuration file #
# #
#----------------------------------------------------------------#
cli version 3.20
webserver port 80 language en
sntp-client
sntp-client server primary 194.35.252.7 port 123 version 4
sntp-client server secondary 194.164.127.5 port 123 version 4
sntp-client local-clock-offset
system
ic voice 0
low-bitrate-codec g729
profile ppp default
profile call-progress-tone defaultDialtone
play 1 1000 450 -6
profile call-progress-tone defaultAlertingtone
play 1 1000 450 -13
pause 2 5000
profile call-progress-tone defaultBusytone
play 1 300 450 -7
pause 2 300
profile call-progress-tone defaultReleasetone
play 1 300 450 -7
pause 2 300
profile call-progress-tone defaultCongestiontone
play 1 300 450 -7
pause 2 300
profile tone-set default
profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
fax transmission 1 relay t38-udp
fax transmission 2 bypass g711alaw64k
profile pstn default
profile sip default
profile aaa default
method 1 local
method 2 none
context ip router
interface IF_IP_LAN
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
interface IF_IP_WAN
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
context ip router
route 0.0.0.0 0.0.0.0 192.168.1.1 0
context cs switch
digit-collection timeout 2
interface sip IF_SIP_1
bind context sip-gateway GW_SIP_ALL_LINES
route call dest-service HUNT_FXO
remote 192.168.1.13 5060
early-connect
early-disconnect
address-translation outgoing-call request-uri user-part fix 10015 host-part to-header target-param none
interface sip IF_SIP_2
bind context sip-gateway GW_SIP_ALL_LINES
route call dest-service HUNT_FXO
remote 192.168.1.13 5060
early-connect
early-disconnect
address-translation outgoing-call request-uri user-part fix 10016 host-part to-header target-param none
interface fxo IF_FXO_1
route call dest-interface IF_SIP_1
loop-break-duration min 60 max 5000
disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 1
mute-dialing
interface fxo IF_FXO_2
route call dest-interface IF_SIP_2
loop-break-duration min 100 max 500
disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 1
mute-dialing
service hunt-group HUNT_FXO
drop-cause normal-unspecified
drop-cause no-circuit-channel-available
drop-cause network-out-of-order
drop-cause temporary-failure
drop-cause switching-equipment-congestion
drop-cause access-info-discarded
drop-cause circuit-channel-not-available
drop-cause resources-unavailable
route call 1 dest-interface IF_FXO_1
route call 2 dest-interface IF_FXO_2
context cs switch
no shutdown
authentication-service AS_ALL_LINES
username 10015 password Z+ApY8PXmFjMRxFr04ls2w== encrypted
username 10016 password c7k7vrPq2MMY+mdxPJS6aQ== encrypted
location-service LS_ALL_LINES
identity 10015
identity 10016
context sip-gateway GW_SIP_ALL_LINES
interface LAN
bind interface IF_IP_LAN context router port 5060
context sip-gateway GW_SIP_ALL_LINES
no shutdown
port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_LAN router
no shutdown
port fxo 0 0
use profile fxo gb
encapsulation cc-fxo
bind interface IF_FXO_1 switch
no shutdown
port fxo 0 1
use profile fxo gb
encapsulation cc-fxo
bind interface IF_FXO_2 switch
shutdown
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here you go
by the way Patton fixed it
Microsoft Windows XP [Version 5.1.2600]
© Copyright 1985-2001 Microsoft Corp.
C:\Documents and Settings\paul>route print
===========================================================================
Interface List
0x1 ........................... MS TCP Loopback interface
0x2 ...00 24 1d a0 7f 65 ...... Realtek PCIe FE Family Controller - Packet Sched
uler Miniport
===========================================================================
===========================================================================
Active Routes:
Network Destination Netmask Gateway Interface Metric
0.0.0.0 0.0.0.0 192.168.1.1 192.168.1.13 20
127.0.0.0 255.0.0.0 127.0.0.1 127.0.0.1 1
192.168.1.0 255.255.255.0 192.168.1.13 192.168.1.13 20
192.168.1.13 255.255.255.255 127.0.0.1 127.0.0.1 20
192.168.1.255 255.255.255.255 192.168.1.13 192.168.1.13 20
224.0.0.0 240.0.0.0 192.168.1.13 192.168.1.13 20
255.255.255.255 255.255.255.255 192.168.1.13 192.168.1.13 1
Default Gateway: 192.168.1.1
===========================================================================
Persistent Routes:
None
Regards
Paul
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If the PBX puts the public address in the SIP packet although it is sent in the LAN, then that is not okay. You will have the "hairpinning NAT" problem. You can fix this by changing the route on the server, that's why I was asking what the route on the server looks like.
So is the routing in Matt's post ok for me?
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Hi pbxnsip
Thanks for your reply.
I've been back on to patton as the issue seems to be a 502 from that trunk
SIP/2.0 502 Bad GatewayVia: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043
I've run a debug on the patton and this hopefully will give them a clue as to why this is happening. It only happens on alternate calls i.e. a call to 123321 goes through, hang up, call to 123321 busy tone (502), hang up, call to 123321 goes through, this is regardless of time between the calls.
The rest of the system works i have the pbx on the dmz and as it's a simple system i set it up using this from Matt post xx.xx.xx.xx is the public
when i get a solution or not
i'll come backRegards
Paul
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Hi
Any help with this would be great.
I have a patton 4112 2 x fxo gateway on Snom One.
When making a call via this gateway the call will either connect or give a busy tone alternately, this behaviour is consistent, i.e. call to 01246123123 call rings and works fine, hang up, call 01246123123 line busy, hang up, call 01246123123 call rings and works fine and so on.....
I have had the config of the patton checked by patton and they don’t see any issues which could cause this behaviour, we tried changing a few things on the patton with no effect.
The logs etc are below along with the trunk set up.
Thanks for any help
regards
Paul
the trunk is setup:
# Trunk 5 in domain localhost
Name: Patton
Type: register
To: sip
RegPass: ********
Direction:
Disabled: false
Global: false
Display:
RegAccount:
RegRegistrar: 192.168.1.200
RegKeep:
RegUser:
Icid:
Require:
OutboundProxy: 192.168.1.200
Ani:
DialExtension: 44
Prefix:
Trusted: false
AcceptRedirect: false
RfcRtp: false
Analog: false
SendEmail:
UseUuid: false
Ring180: false
Failover: only_5xx
Privacy: pai
Glob:
RequestTimeout:
Codecs:
CodecLock: true
Expires: 3600
FromUser:
Tel: true
TranscodeDtmf: false
AssociatedAddresses:
InterOffice: false
DialPlan:
Colines: 2
DialogPermission:
Log from a succesful call
[9] 2011/01/23 22:32:47:
[7] 2011/01/23 22:32:50:
INVITE sip:263016@192.168.1.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport
Route: <sip:192.168.1.13;lr>
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>
Call-ID: m6j5w1sdsk
CSeq: 22153 INVITE
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.2.42-a
Content-Type: application/sdp
Content-Length: 398
v=0
o=root 1565728340 1565728341 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 52578 RTP/AVP 0 8 18 3 9 2 10 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:VUyNQYRiVjnhewe7vV1qF+eJ9VtmWTiW1RZOyQT6|2^31
a=sendrecv
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[7] 2011/01/23 22:32:50:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22153 INVITE
Content-Length: 0
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[7] 2011/01/23 22:32:50:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22153 INVITE
User-Agent: snom-PBX/4.2.0.3974
WWW-Authenticate: Digest realm="192.168.1.13",nonce="380850f72b1790232735c52375b1d44a",domain="sip:263016@192.168.1.13;user=phone",algorithm=MD5
Content-Length: 0
[7] 2011/01/23 22:32:50:
ACK sip:263016@192.168.1.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport
Route: <sip:192.168.1.13;lr>
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22153 ACK
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Content-Length: 0
[7] 2011/01/23 22:32:50:
INVITE sip:263016@192.168.1.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport
Route: <sip:192.168.1.13;lr>
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>
Call-ID: m6j5w1sdsk
CSeq: 22154 INVITE
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.2.42-a
Authorization: Digest realm="192.168.1.13",nonce="380850f72b1790232735c52375b1d44a",response="6929aee06e8ccb51bbe0ab176106929d",username="45",uri="sip:263016@192.168.1.13;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 398
v=0
o=root 1565728340 1565728341 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 52578 RTP/AVP 0 8 18 3 9 2 10 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:VUyNQYRiVjnhewe7vV1qF+eJ9VtmWTiW1RZOyQT6|2^31
a=sendrecv
[8] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[7] 2011/01/23 22:32:50:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22154 INVITE
Content-Length: 0
[8] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[7] 2011/01/23 22:32:50:
INVITE sip:263016@192.168.1.200;user=phone SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport
From: "Forty Five" <sip:45@localhost;user=phone>;tag=81
To: <sip:263016@192.168.1.200;user=phone>
Call-ID: f5058ca5@pbx
CSeq: 8184 INVITE
Max-Forwards: 70
Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/4.2.0.3974
P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone>
Content-Type: application/sdp
Content-Length: 327
v=0
o=- 58077 58077 IN IP4 xx.xx.xx.xx
s=-
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 63112 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=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[9] 2011/01/23 22:32:50:
[7] 2011/01/23 22:32:50:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22154 INVITE
Contact: <sip:45@xx.xx.xx.xx:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/4.2.0.3974
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 324
v=0
o=- 43827 43827 IN IP4 xx.xx.xx.xx
s=-
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 54492 RTP/AVP 0 8 9 2 3 96
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:96 telephone-event/8000
a=fmtp:96 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[7] 2011/01/23 22:32:50:
PRACK sip:45@xx.xx.xx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-0tvq9n;rport
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22155 PRACK
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
RAck: 1 22154 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Content-Length: 0
[9] 2011/01/23 22:32:50:
[7] 2011/01/23 22:32:50:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-0tvq9n;rport=4043;received=192.168.1.1
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22155 PRACK
Contact: <sip:45@xx.xx.xx.xx:5060>
User-Agent: snom-PBX/4.2.0.3974
Content-Length: 0
[7] 2011/01/23 22:32:51:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport=5060;received=192.168.1.13
From: "Forty Five" <sip:45@localhost;user=phone>;tag=81
To: <sip:263016@192.168.1.200;user=phone>
Call-ID: f5058ca5@pbx
CSeq: 8184 INVITE
Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26
Content-Length: 0
[9] 2011/01/23 22:32:51:
[7] 2011/01/23 22:32:54:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport=5060;received=192.168.1.13
From: "Forty Five" <sip:45@localhost;user=phone>;tag=81
To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773
Call-ID: f5058ca5@pbx
CSeq: 8184 INVITE
Contact: <sip:263016@192.168.1.200:5060>
Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26
Supported: replaces
Content-Type: application/sdp
Content-Length: 221
v=0
o=MxSIP 0 57 IN IP4 192.168.1.200
s=SIP Call
c=IN IP4 192.168.1.200
t=0 0
m=audio 4976 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[7] 2011/01/23 22:32:54:
[9] 2011/01/23 22:32:54:
[9] 2011/01/23 22:32:54:
[7] 2011/01/23 22:32:54:
ACK sip:263016@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-291e98728a7e6723bd58601166585878;rport
From: "Forty Five" <sip:45@localhost;user=phone>;tag=81
To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773
Call-ID: f5058ca5@pbx
CSeq: 8184 ACK
Max-Forwards: 70
Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp>
P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone>
Content-Length: 0
[9] 2011/01/23 22:32:54:
[9] 2011/01/23 22:32:54:
[9] 2011/01/23 22:32:54:
[7] 2011/01/23 22:32:54:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22154 INVITE
Contact: <sip:45@xx.xx.xx.xx:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/4.2.0.3974
Content-Type: application/sdp
Content-Length: 324
v=0
o=- 43827 43827 IN IP4 xx.xx.xx.xx
s=-
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 54492 RTP/AVP 0 8 9 2 3 96
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:96 telephone-event/8000
a=fmtp:96 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[7] 2011/01/23 22:32:54:
ACK sip:45@xx.xx.xx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-zaqp37;rport
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22154 ACK
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Content-Length: 0
[8] 2011/01/23 22:32:59:
[9] 2011/01/23 22:32:59:
[7] 2011/01/23 22:33:00:
BYE sip:45@xx.xx.xx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-ce9s8k;rport
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22156 BYE
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Supported: 100rel, replaces, norefersub
Content-Length: 0
[9] 2011/01/23 22:33:00:
[7] 2011/01/23 22:33:00:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-ce9s8k;rport=4043;received=192.168.1.1
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196
Call-ID: m6j5w1sdsk
CSeq: 22156 BYE
Contact: <sip:45@xx.xx.xx.xx:5060>
User-Agent: snom-PBX/4.2.0.3974
Content-Length: 0
[9] 2011/01/23 22:33:00:
[9] 2011/01/23 22:33:00:
[7] 2011/01/23 22:33:00:
BYE sip:263016@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-ac3b9d3338c98fa09f1c607eeeca5213;rport
From: "Forty Five" <sip:45@localhost;user=phone>;tag=81
To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773
Call-ID: f5058ca5@pbx
CSeq: 8185 BYE
Max-Forwards: 70
Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp>
P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone>
Content-Length: 0
And the log from a failed call made stright after the above.
INVITE sip:263016@192.168.1.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport
Route: <sip:192.168.1.13;lr>
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>
Call-ID: o77i5idc96
CSeq: 12645 INVITE
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.2.42-a
Content-Type: application/sdp
Content-Length: 398
v=0
o=root 1833475499 1833475500 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 56684 RTP/AVP 0 8 18 3 9 2 10 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZT6rmB8cjbvcKGdGKF5E1VXrx1ZBnP7/04nGSRA7|2^31
a=sendrecv
[9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:64380
[9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:64381
[9] 2011/01/23 22:31:51: Resolve 7529: aaaa udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7529: a udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7529: udp 192.168.1.2 4043
[7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12645 INVITE
Content-Length: 0
[9] 2011/01/23 22:31:51: Resolve 7530: aaaa udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7530: a udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7530: udp 192.168.1.2 4043
[7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12645 INVITE
User-Agent: snom-PBX/4.2.0.3974
WWW-Authenticate: Digest realm="192.168.1.13",nonce="c9244e6dab13e1f55b84ee9830031c0f",domain="sip:263016@192.168.1.13;user=phone",algorithm=MD5
Content-Length: 0
[7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.2:4043:
ACK sip:263016@192.168.1.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport
Route: <sip:192.168.1.13;lr>
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12645 ACK
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Content-Length: 0
[7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.2:4043:
INVITE sip:263016@192.168.1.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport
Route: <sip:192.168.1.13;lr>
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>
Call-ID: o77i5idc96
CSeq: 12646 INVITE
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.2.42-a
Authorization: Digest realm="192.168.1.13",nonce="c9244e6dab13e1f55b84ee9830031c0f",response="70b900d2b3d9130bd5d678d3f7985945",username="45",uri="sip:263016@192.168.1.13;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 398
v=0
o=root 1833475499 1833475500 IN IP4 192.168.1.2
s=-
c=IN IP4 192.168.1.2
t=0 0
m=audio 56684 RTP/AVP 0 8 18 3 9 2 10 96
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZT6rmB8cjbvcKGdGKF5E1VXrx1ZBnP7/04nGSRA7|2^31
a=sendrecv
[8] 2011/01/23 22:31:51: Tagging request with existing tag
[9] 2011/01/23 22:31:51: Resolve 7531: aaaa udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7531: a udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7531: udp 192.168.1.2 4043
[7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12646 INVITE
Content-Length: 0
[8] 2011/01/23 22:31:51: Set the To domain based on From user 45@localhost
[9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:53478
[9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:53479
[9] 2011/01/23 22:31:51: Resolve 7532: url sip:192.168.1.200
[9] 2011/01/23 22:31:51: Resolve 7532: udp 192.168.1.200 5060
[7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.200:5060:
INVITE sip:263016@192.168.1.200;user=phone SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport
From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580
To: <sip:263016@192.168.1.200;user=phone>
Call-ID: ed6c53a8@pbx
CSeq: 27475 INVITE
Max-Forwards: 70
Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/4.2.0.3974
P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone>
Content-Type: application/sdp
Content-Length: 327
v=0
o=- 28432 28432 IN IP4 xx.xx.xx.xx
s=-
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 53478 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=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[9] 2011/01/23 22:31:51: Resolve 7533: aaaa udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7533: a udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7533: udp 192.168.1.2 4043
[7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12646 INVITE
Contact: <sip:45@xx.xx.xx.xx:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/4.2.0.3974
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 324
v=0
o=- 16374 16374 IN IP4 xx.xx.xx.xx
s=-
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 64380 RTP/AVP 0 8 9 2 3 96
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:96 telephone-event/8000
a=fmtp:96 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.1:4043:
PRACK sip:45@xx.xx.xx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-4jmwo3;rport
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12647 PRACK
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
RAck: 1 12646 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Content-Length: 0
[9] 2011/01/23 22:31:51: Resolve 7534: udp 192.168.1.1 4043
[7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.1:4043:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-4jmwo3;rport=4043;received=192.168.1.1
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12647 PRACK
Contact: <sip:45@xx.xx.xx.xx:5060>
User-Agent: snom-PBX/4.2.0.3974
Content-Length: 0
[7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.200:5060:
SIP/2.0 502 Bad Gateway
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport=5060;received=192.168.1.13
From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580
To: <sip:263016@192.168.1.200;user=phone>;tag=3034763092
Call-ID: ed6c53a8@pbx
CSeq: 27475 INVITE
Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26
Content-Length: 0
[7] 2011/01/23 22:31:51: Call ed6c53a8@pbx: Clear last INVITE
[9] 2011/01/23 22:31:51: Resolve 7535: url sip:192.168.1.200
[9] 2011/01/23 22:31:51: Resolve 7535: udp 192.168.1.200 5060
[7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.200:5060:
ACK sip:263016@192.168.1.200;user=phone SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport
From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580
To: <sip:263016@192.168.1.200;user=phone>;tag=3034763092
Call-ID: ed6c53a8@pbx
CSeq: 27475 ACK
Max-Forwards: 70
Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp>
P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone>
Content-Length: 0
[5] 2011/01/23 22:31:51: INVITE Response 502 Bad Gateway: Terminate ed6c53a8@pbx
[9] 2011/01/23 22:31:51: Resolve 7536: aaaa udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7536: a udp 192.168.1.2 4043
[9] 2011/01/23 22:31:51: Resolve 7536: udp 192.168.1.2 4043
[7] 2011/01/23 22:31:52: SIP Tx udp:192.168.1.2:4043:
SIP/2.0 502 Bad Gateway
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12646 INVITE
Contact: <sip:45@xx.xx.xx.xx:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/4.2.0.3974
Content-Length: 0
[8] 2011/01/23 22:31:52: Hangup: Call 101 not found
[7] 2011/01/23 22:31:52: SIP Rx udp:192.168.1.2:4043:
ACK sip:263016@192.168.1.13;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport
Route: <sip:192.168.1.13;lr>
From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx
To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866
Call-ID: o77i5idc96
CSeq: 12646 ACK
Max-Forwards: 70
Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>"
Content-Length: 0
[7] 2011/01/23 22:31:57: SIP Rx udp:192.168.1.2:4043:
REGISTER sip:192.168.1.13 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-90ywmz;rport
From: "Paul Stead" <sip:44@192.168.1.13>;tag=s8ozvp
To: "Paul Stead" <sip:44@192.168.1.13>
Call-ID: ulydh2y8@snom
CSeq: 3435 REGISTER
Max-Forwards: 70
Contact: <sip:44@192.168.1.2:4043;transport=udp;line=j5sjvx>;reg-id=1;+sip.instance="<urn:uuid:249f54b0-67ba-445c-8433-55ee8f3a7b1a>"
Supported: path, outbound, gruu
User-Agent: snom-m9/9.2.42-a
Authorization: Digest realm="192.168.1.13",nonce="10ceb016de3d4209ddadda412473a800",response="9afc717ab4f9d532a315f8378102e9f7",username="44",uri="sip:192.168.1.13",algorithm=MD5
Expires: 354
Content-Length: 0
[9] 2011/01/23 22:31:57: Resolve 7537: aaaa udp 192.168.1.2 4043
[9] 2011/01/23 22:31:57: Resolve 7537: a udp 192.168.1.2 4043
[9] 2011/01/23 22:31:57: Resolve 7537: udp 192.168.1.2 4043
[7] 2011/01/23 22:31:57: SIP Tx udp:192.168.1.2:4043:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-90ywmz;rport=4043
From: "Paul Stead" <sip:44@192.168.1.13>;tag=s8ozvp
To: "Paul Stead" <sip:44@192.168.1.13>;tag=49ef1d8f34
Call-ID: ulydh2y8@snom
CSeq: 3435 REGISTER
Contact: <sip:44@192.168.1.2:4043;transport=udp;line=j5sjvx>;expires=352
Require: outbound
Supported: path
Content-Length: 0
-
The news is the Patton was de bricked and with the help here it's now working with the Snom one PBX.
Much joy here
Ta
Regards
Paul
-
get a professional partner help you get up and going (yes, a little more complex at first) with snom ONE and you will run very smooth ongoing.
Yep agree trying to find one any suggestions UK Derbyshire / South Yorkshire?
Oh and by the way Snom say, when asked about professional partner help, come on the course and take the exam
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That portech guide is not for Patton fxo.
I've started writting a book on setting up a snom ONE system (including everything a real life system will neeD). It includes a chapter on setting up Patton gateway step by step. I put the book on pause because snom was writting a book/manual as well. If there is a need I will continue it.
(these books get out of date VERY quickly. that chapter has quite a few new developements since i wrote it. although it is still valid)
I would like to add that your experience with patton is not normal: our experience with their product is that it is first rate. We have sold a lot of Patton gateways for 3CX systems and pbxnsip/snom ONE. No failures (other than technicians frying them plugging hot things into them. ;-) We replace a lot of grandstream gateways with pattons to made all kinds of issues go away.
Their tech support has been FIRST RATE. And very consistly first rate.
Also, patton gateways have come up as the most recommended gateway among win/pbx administrators:
Hi
I'm sure you are right Sir, but brick like it is, although I am sure it will be up and running later today when support get around to it. The reason I bought it was that it was first rate. I understand that the Portech is not a Patton but at least it gives me (a newbie) a clue on how to setup trunk for it (I think)
The 3cx to be honest looks much easier to set up, I’ve watched a the video which seems to be, set up a pstn gateway, tell it it’s a Patton, the 3cx gives you the txt file to upload to the Patton and away it goes. My point with Snom One is that there is little or no help apart from you guys on how to set these things up. That said I want to persevere with Snom as I have bought their phones, but I am finding this very hard work indeed.
Sir, I am very grateful for all of your help so far, if I could have sight of any of the ‘book’ which you think could help me that would be most helpful. All I am trying at this stage is as follows
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- Snom one PBX
- Viop Trunk (currently working in a basic form)
- 2 pstn lines via a Patton 4112 (not set up at all yet)
- 1 x Remote office (not working trying to use x-lite at the moment looking at putting the pbx dmz side, will want a hard phone at sometime, found your review of the 300 most helpful thanks)
- Snom m9 phones (working very well with the PBX and VIOP trunk)
Clearly that will not be the end of it but the above would be a start
Regards
Paul
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yes, the remote phone can be behind NAT.
(in some odd circumstances the remote firewall can be tricky. but you should be able to take a phone /laptop into a wifi hotspot at a cafe and it should work. does for me)
Thank the Lord
And thanks for your help
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Yes. If you want to talk to the PBX, you need a routable address and the PBX must be able to advertize this address to the phones. Sounds trivial, but it's the core of the problem.
Just so I'm clear the DMZ (public address) is at the PBX end only. The remote office extension (Xlite - Snom 300) can be behind a firewall with nats?
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Well, NAT is probably the biggest problem with VoIP. Until IPv6 is ready, we need to deal with more or less dirty workarounds. Check out http://kiwi.pbxnsip.com/index.php/Office_with_private_and_public_IP_addresses for a typical scenario. If you want to register phones from the Internet, you do need to have a routable IP address ("public" IP address); all the workarounds with port forwarding etc are extremly difficult and instable so consider putting one interface of the PBX host on a public IP.
I'll give it a shot I assume unless i do this i will have issues even if i use a snom 300 type phone remotely?
Thanks
Paul
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I can understand that frustration well
. It is not only VoIP also other products are getting so complex you need a PhD to get them working (cell phones, cars, home automation to to name a few).Agreed(My 8 year old thinks that I'm second only to Lee Westwood golfing wise so at least I'm good at something
) Looking at the PDF it seems that it would only take two minutes to represent this as a How To for any FXO,GSM type gateway. It could even have the new and correct name on (Snom rather than pbxnsip
.)Anyway does the PDF represent a good 'how to' for getting a patton 4112 2 x fxo working on a snom one pbx?
thanks for your interest in my issue
regards
Paul
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Make sure you don't use STUN on X-lite, which is a permanent (and unneccessary) source for trouble. Also, we are coming out with a updated version of the PBX this week, if you can backup and update your PBX and see if that fixes any issues:
Win32 – http://pbxnsip.com/download/pbxctrl-2011-4.2.0.3974.exe
Win64 – http://pbxnsip.com/download/pbxctrl-2011-4.2.0.3974_64bit.exe
Debian - http://pbxnsip.com/download/pbxctrl-debian4.0-2011-4.2.0.3974
Centos32 - http://pbxnsip.com/download/pbxctrl-centos5-2011-4.2.0.3974
Centos64 - http://pbxnsip.com/download/pbxctrl-centos5-2011-4.2.0.3974_64bit
SuSe - http://pbxnsip.com/download/pbxctrl-suse10-2011-4.2.0.3974
Sheeva – http://pbxnsip.com/download/pbxctrl-sheeva-2011-4.2.0.3974
You can also try the snom m9 soft phone, which should work without trouble (http://forum.pbxnsip.com/index.php?/topic/4140-snom-m9-soft-phone/).
OK I updated the PBX and it still doen't work.
Lets try and start at the begining
Which ports do I need to be open at the remote office router for this to work, I check these to start with and move on if needed?
Thanks for your input so far
Regards
Paul
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Hi
This PDF seems like just the thing I've been looking for, one of the most frustrating thing about Snom One for a newbie like me (I believe I have a broad understanding of technology) is that the whole voip thing appears to be a black art. This seems to be true even when taking to the experts, I phoned Patton(USA 1 hour)yesterday regarding a locked up 4112 and following instructions from the tech guy, we bricked the unit (I await further instructions via email from them).
The point of this rant is the question, are there any "how to's" anywhere like the PDF discussed here if so where are they, if not I think it may be a good idea to have some. Once I have my 4112 backup I think I can use this PDF to help me set it up on the Snom One unless someone out there (a black art master)knows different.
Regards
Paul
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Paul,
I wonder if you have something wrong with the workstation in question?
i had tested the soft snom m9 and didn't notice those items. (could be i missed it)
But considering your having issues with both xlite/m9 soft i would suggest perhaps trying a cleaner pc if that is any question in your mind.
The two issues are different, aren’t they?
I did try the M9 soft on a clean note book and had the same issues, that is the M9 software not very responsive and resource hungry. I don’t have the note book with me today so cannot try it again from this remote office. You will also notice I said that the M9 is taking 50% of the CPU at rest (idle) the X-Lite is 0% - 1% I would suggest that this is therefore nothing to do with the pc I’m working on. In addition it allows me to open the browser once to insert the initial details but once I select save that’s when it freezes. So I can’t even change the settings.
As I am new to this stuff if someone has their settings for a M9 (or X-Lite) to a snom one remote pbx I will be delighted to try those inserting my servers ip address. If you are using another soft phone (I assume you may be as you’ve only tested the M9) tell me which one and I’ll try that.
thanks for your help so far
regards
Paul
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Make sure you don't use STUN on X-lite, which is a permanent (and unneccessary) source for trouble. Also, we are coming out with a updated version of the PBX this week, if you can backup and update your PBX and see if that fixes any issues:
Win32 – http://pbxnsip.com/download/pbxctrl-2011-4.2.0.3974.exe
Win64 – http://pbxnsip.com/download/pbxctrl-2011-4.2.0.3974_64bit.exe
Debian - http://pbxnsip.com/download/pbxctrl-debian4.0-2011-4.2.0.3974
Centos32 - http://pbxnsip.com/download/pbxctrl-centos5-2011-4.2.0.3974
Centos64 - http://pbxnsip.com/download/pbxctrl-centos5-2011-4.2.0.3974_64bit
SuSe - http://pbxnsip.com/download/pbxctrl-suse10-2011-4.2.0.3974
Sheeva – http://pbxnsip.com/download/pbxctrl-sheeva-2011-4.2.0.3974
You can also try the snom m9 soft phone, which should work without trouble (http://forum.pbxnsip.com/index.php?/topic/4140-snom-m9-soft-phone/).
Thanks
I'm not (as far as I can see)using a stun.
I will try the update later to see if that resolves the issue.
I've just tried the M9 soft again taking off the X-Lite and it reaffirmed my first experience of it (hence trying X-Lite). Once I had given the 1st identity a setup in the web interface, the web interface froze so did the M9. In addition, at rest (no calls) the process m9Soft2.exe takes 50% of the CPU. X-Lite takes either 0% or 1%. This seems to screw everything else including the browser.
Go figure
Unless you think I'm doing something wrong with the M9?
Regards
Paul
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Hi
I've managed to get the x-lite to register to the snom one pbx at a remote office, the X-Lite can make trunk calls and extensions at the remote office can call the X-Lite. Making a call to an extension at the remote office is where I’m stuck, upon dialling the extension, X-Lite says the call is connected without ringing. If I then hang up on the X-Lite, when i check the PBX call logs it says the call is still connected and requires me to disconnect it at the PBX.
I am clearly missing something, any clues?
Regards
Paul
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Hi
I assume there is a way of pulling (i.e. using) the domain address book from a snom one pbx to the M9 dect phones.
I can find an guidance on this can it be done?
Regards
Paul
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open ports:
Standard:
Port: 5060 / UDP (SIP-Signalisierung)
Port: 5004 / UDP (RTP, Voice) <- here open a range of ports (5000-5020) that helps me!
Port: 10000 UDP (STUN) <- I think unnecessary
Ok now working with more help from a tech support guy help.
In addtion to the above the following ports need to be opened 49152-64512 UDP inbound and outbound.
In addition there was a DNS issue, sipgate.co.uk was not resolving correctly on the computer snom ONE was located and after this was changed to the IP address for sipgate.co.uk it worked
Thanks for all your help and hopfully the thread will help others
Regards
Paul
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sorry but whats your question? do you want to use sipgate as a trunk?
If yes and you cant connect snom one with sipgate check if your firewall is on.
You just have to input your given information into snome one. today I just change from 3cx to snome one and use sipgate (even its sipgate.de not .co.uk ;-) )
By that i guess you mean that i have to open ports on my router. I figured that as I didn't for 3cx so i didn't for Snom One, if i do which ones?
An easy answer to my question would be a screen print of your trunk setup (as you are using SipGate) with your details obscured to protect your account so i can check mine are the same along with the ports you've opened on your router. Or a copy from your text based edit window
Thanks for you input so far
Regards
Paul
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Hi
Sorry to be a pain testing Snom One and while i got 3cx to setup without difficulty i like some of the options available on Snom One so I am now trying this.
Using the following information (not real of course) what goes where in the trunk setup screen I've tried every which way to get it to work and unless i am missing another and further configuration i can not get it to work. needless to say I've looked for a "how to" i don't see one.
SIP account data
This account data must be used to configure your SIP device. sipgate provides detailed information for customers to configure their SIP devices to function correctly.
More information can be found here.
SIP-ID: 12345678
SIP password: GFFDRGG
Status: offline
Nickname: No name was set Edit
--------------------------------------------------------------------------------
SIP Server data
This account data must be used to configure your SIP-device. sipgate provides detailed information for customers to configure their SIP devices to function correctly.
More information can be found here.
Registry: sipgate.co.uk
Proxy: sipgate.co.uk
STUN: stun.sipgate.net:10000
NTP: ntp.sipgate.net
i assume the trunk is all i need to setup the extensions are working after about one and one half an hours. You would think that snom m9 and snom one would just work out of the box
By the way they sipgate DON'T provide detailed information for customers to configure their snom one SIP devices to function correctly.
thanks
regards
Paul
i'll come back
. It is not only VoIP also other products are getting so complex you need a PhD to get them working (cell phones, cars, home automation to to name a few).
.)
'Incoming anonymous calls: Ask for Name' only available on Extension Level
in Hunt Group Setup
Posted
Was this ever done, if so could you point me to it. If not what's the best workround?
thanks
Regards
Paul