Jeremy Salmon
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Posts posted by Jeremy Salmon
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Hi,
I just installed snom ONE on a hight availability cluster.
My configuration
A shared IP between 2 box (ex : 192.168.1.83)
Box 1 (192.168.1.81) - Linux Debian 6.x - snom ONE with /etc/init.d/snomONE updated to make it LSB compliant - DRBD for HDD replication to Box 2 - Corosync + Pacemaker for failover management - Dhcpd
Box 2 (192.168.1.82) - Linux Debian 6.x - snom ONE with /etc/init.d/snomONE updated to make it LSB compliant - DRBD for HDD replication to Box 1 - Corosync + Pacemaker for failover management - Dhcpd
Patton gateway to connect to Telco.
Everything work well with a Grey (free) édition.
Services toggle between Box 1 and Box 2 in case of fail of network or anything else.
Phones and Gateway are registered to 192.168.1.83.
But now I want to activate a Blue licence.
Is there a solution for this type of solution or should I install 2 licenses?
Regards,
Jeremy
PS : A tutorial will be published quickly on http://blog.snom.ma.
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Ok.
I plugged an USB driver and :
apt-get update apt-get install usbmount reboot
First USB Key plugged is mounted on /mnt/usb/
mv /usr/local/snomONE/recordings /mnt/usb/ ln -snf /mnt/usb/recordings/ /usr/local/snomONE/recordings mv /usr/local/snomONE/tftp /mnt/usb/ ln -snf /mnt/usb/tftp/ /usr/local/snomONE/tftp
I also
rm -rf /usr/local/snomONE/audio_en/
(I use french language)
Thanks for your help !
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Thanks for your EXTREMELY quick answer !!!
Is there a method to automatically conserve space on the disk ?
Check if the file system is full (you can see that in the status screen of the PBX). If thats is the case, remove some files and make a backup from the web interface. Dont just reboot, as then your config might really get screwed up.
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Hi,
I have a recurring problem on several snom SOHO.
Sometimes I lose the configuration. The weird thing is that I lose a few pieces (eg extensions or license).
It's very embarrassing ... and this happens to me on several (about 6) SOHO.
Is there a BUG ?
(the same thing happening on SheevaPlug installed with snom ONE)
Regards,
Jeremy
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Hi Mr X,
Thanks for your response.
I know the Failover mecanism of trunk. My problem is when I have no credit on a SIM it's not a fault .... No busy tone or anything else ... Just a voice with "you have no credit to make this call ...."
I use failover to pass on 2nd, 3rd SIM, ... when previous is busy.
I think I have to dev a small plugin :
- Make a sum of total call duration on trunk_x between 01/mm/yyy and now()
- Check if SIM duration > total call duration
- Update xml of dialplan to remove trunk if not
I think I have to put it on a cron ....
or can I launght an app on each hangup ?
Thanks
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Hi,
I have 4 GSM Gateway with 36 hours of communication on each SIM.
GSM gateway are connected to a Patton SN4114 with one trunk for each port (TRUNK_GSM_1, TRUNK_GSM_2, ...)
I search a solution to make "load balancing" between each Trunk because I can't detect when I have no credit on a SIM card ...
Idea 1 : make call cycling (call 1 on trunk 1, call 2 on trunk 2, etc ...) from snom ONE (not from the Patton)
Idea 2 : use an "outbound credit" on trunk not extension ... Is it possible ?
Other idea or solution ?
Thanks in advance,
Regards,
Jeremy
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Great ! Thanks ...
Not the same philosophy with Asterisk ... That's why I have not thought of that
Many thanks,
Jeremy
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hmmm ... Thanks for this idea ...
But not working for me.
I have 9 DID for 9 geographic area ...
In asterisk I wrote Set(CALLERID(num)=33144675643 in my dialplan to rewrite the outgoing callerid ...
In this post you talked about dialplan replacement parameters : http://forum.snomone.com/index.php?/topic/3412-dialplan-replacement/
What is parameter 1,2,3 ???
The last solution is to subscribe 9 trunk to my provider ... but it's expensive .... 2 euros for a DID, 12 euros for a trunk ...
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Hi All,
I have a trunk with multiple DID (ex : 33144736315, 33356764576, ...)
I want to select the outgoing DID from Dialplan ... For example when I call a 335xxxxxxxx I want to use the local 33356764576.
How can I play with replacement to make this ?
Thanks !
JS
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For info firmware version of the DIR300 is ME_2.01.
Under "Advanced Setup", "NAT" I have ALG. By default it's activated.
I turned off "SIP Enabled". It seem to work fine.
Thanks for your patience !!!
Just a last question : why SIP Provider IPPI.fr worked and OVH no ? For info OVH have a CIRPACK and Ippi an Asterisk.
Regards,
JS
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It's a DLINK Dir300. But I tried with two others.
Here my Trunk settings:
# Trunk 3 in domain societe4.topsystem.be
Name: OVH_TRUNK
Type: register
To: sip
RegPass: ********
Direction:
Disabled:
Global: false
Display:
RegAccount: 0033184190197
RegRegistrar: sip.ovh.net
RegKeep:
RegUser: 0033184190197
Icid:
Require:
OutboundProxy: sip.ovh.net
Ani:
DialExtension:
Prefix:
Trusted: false
AcceptRedirect: false
RfcRtp: false
Analog: false
SendEmail:
UseUuid: false
Ring180: false
Failover: never
Privacy: false
Glob:
RequestTimeout:
Codecs:
CodecLock: true
Expires: 3600
FromUser:
Tel: true
TranscodeDtmf: false
AssociatedAddresses:
InterOffice: false
DialPlan:
Colines:
DialogPermission:
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Thanks for your response
192.168.1.1 is my gateway IP.
I have 4 snom 320 in my LAN connected to my snom ONE (192.168.1.13). I can make internal call without problem.
I use an another SIP Provider (ippi.fr) on this server without problem.
I have just sound problem with this OVH provider.
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Hi All,
I have a trunk to OVH SIP Provider (ovh.fr).
Trunk is registered but when I place a call, I ear just the beginning of the first ring and after nothing else ...
I played with trunk option, sip replacement list, ... without success
snom ONE IP : 192.168.1.13
snom320 IP : 192.168.1.12
OVH sip server : 91.121.129.17
Called number : 0972101112
Here my SIP log :
=====
[5] 2011/12/28 10:28:32: Last message repeated 6 times
[5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:
INVITE sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1
X-Serialnumber: 000413318D7F
P-Key-Flags: keys="3"
User-Agent: snom320/8.4.18
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 522
v=0
o=root 469490837 469490837 IN IP4 192.168.1.12
s=call
c=IN IP4 192.168.1.12
t=0 0
m=audio 62652 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fpsUzc8Bo1L9ecyoUesYEHcSpeeJL9NY265vLFVc
a=rtpmap:9 g722/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv
[5] 2011/12/28 10:28:32: Last message repeated 2 times
[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport=2081
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 1 INVITE
Content-Length: 0
[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport=2081
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 1 INVITE
User-Agent: snom-PBX/2011-4.2.0.3981
WWW-Authenticate: Digest realm="societe4.topsystem.be",nonce="1a6d0303a5486a12117e695eaaaf8663",domain="sip:0972101112@societe4.topsystem.be;user=phone",algorithm=MD5
Content-Length: 0
[5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:
ACK sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1
Proxy-Require: buttons
Content-Length: 0
[5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:
INVITE sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1
X-Serialnumber: 000413318D7F
P-Key-Flags: keys="3"
User-Agent: snom320/8.4.18
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest username="400",realm="societe4.topsystem.be",nonce="1a6d0303a5486a12117e695eaaaf8663",uri="sip:0972101112@societe4.topsystem.be;user=phone",response="1f49712e3eb260cf6b2c006d031e9218",algorithm=MD5
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 522
v=0
o=root 469490837 469490837 IN IP4 192.168.1.12
s=call
c=IN IP4 192.168.1.12
t=0 0
m=audio 62652 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fpsUzc8Bo1L9ecyoUesYEHcSpeeJL9NY265vLFVc
a=rtpmap:9 g722/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv
[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 2 INVITE
Content-Length: 0
[5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:
INVITE sip:0972101112@91.121.129.17;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8e16178196f7d0955897860527ca4960;rport
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>
Call-ID: 0f4287ed@pbx
CSeq: 12053 INVITE
Max-Forwards: 70
Contact: <sip:0033184190197@192.168.1.13:5060;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/2011-4.2.0.3981
Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes
Content-Type: application/sdp
Content-Length: 323
v=0
o=- 801070725 801070725 IN IP4 192.168.1.13
s=-
c=IN IP4 192.168.1.13
t=0 0
m=audio 60346 RTP/AVP 0 8 3 2 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 2 INVITE
Contact: <sip:400@192.168.1.13:5060;transport=tcp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 298
v=0
o=- 1825812257 1825812257 IN IP4 192.168.1.13
s=-
c=IN IP4 192.168.1.13
t=0 0
m=audio 57150 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=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2011/12/28 10:28:32: SIP Rx udp:91.121.129.17:5060:
SIP/2.0 407 authentication required
Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8e16178196f7d0955897860527ca4960
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a364c-25053b736
Call-ID: 0f4287ed@pbx
CSeq: 12053 INVITE
Contact: <sip:0972101112@41.141.84.105:5060;user=phone>
Proxy-Authenticate: Digest realm="sip.ovh.net", nonce="001a360d6b5d399f4797a37b635627ac", opaque="0010abd225acd41", stale=false, algorithm=MD5
server: Cirpack/v4.42j (gw_sip)
Allow: UPDATE, REFER, INFO
Content-Length: 0
[5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:
ACK sip:0972101112@91.121.129.17;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8e16178196f7d0955897860527ca4960;rport
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a364c-25053b736
Call-ID: 0f4287ed@pbx
CSeq: 12053 ACK
Max-Forwards: 70
Content-Length: 0
[5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:
INVITE sip:0972101112@91.121.129.17;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>
Call-ID: 0f4287ed@pbx
CSeq: 12054 INVITE
Max-Forwards: 70
Contact: <sip:0033184190197@192.168.1.13:5060;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/2011-4.2.0.3981
Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes
Proxy-Authorization: Digest realm="sip.ovh.net",nonce="001a360d6b5d399f4797a37b635627ac",response="16890c1b4a1beaf825a0f2e3fd1234fc",username="0033184190197",uri="sip:0972101112@91.121.129.17;user=phone",opaque="0010abd225acd41",algorithm=MD5
Content-Type: application/sdp
Content-Length: 323
v=0
o=- 801070725 801070725 IN IP4 192.168.1.13
s=-
c=IN IP4 192.168.1.13
t=0 0
m=audio 60346 RTP/AVP 0 8 3 2 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:
PRACK sip:400@192.168.1.13:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-ckw8xvotqm55;rport
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 3 PRACK
Max-Forwards: 70
Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1
RAck: 1 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons
Content-Length: 0
[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:
SIP/2.0 200 Ok
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-ckw8xvotqm55;rport=2081
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 3 PRACK
Contact: <sip:400@192.168.1.13:5060;transport=tcp>
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Length: 0
[5] 2011/12/28 10:28:32: SIP Rx udp:91.121.129.17:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>
Call-ID: 0f4287ed@pbx
CSeq: 12054 INVITE
Contact: <sip:41.141.84.105:5060>
server: Cirpack/v4.42j (gw_sip)
Allow: UPDATE, REFER, INFO
Content-Length: 0
[5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090
Call-ID: 0f4287ed@pbx
CSeq: 12054 INVITE
Contact: <sip:41.141.84.105:5060>
server: Cirpack/v4.42j (gw_sip)
Allow: UPDATE, REFER, INFO
Content-Length: 0
[5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:
SIP/2.0 183 Media change
Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090
Call-ID: 0f4287ed@pbx
CSeq: 12054 INVITE
Contact: <sip:41.141.84.105:5060>
server: Cirpack/v4.42j (gw_sip)
Allow: UPDATE, REFER, INFO
Content-Type: application/sdp
Content-Length: 262
v=0
o=cp10 132506811328 132506811330 IN IP4 192.168.1.1
s=SIP Call
c=IN IP4 192.168.1.1
t=0 0
m=audio 7072 RTP/AVP 0 8 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-05b06979574af555d61e176134a2f476
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1452189312
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-07824-001a1068-6632914c5
Call-ID: ad9d1f30@pbx
CSeq: 18918 INVITE
Contact: <sip:41.141.84.105:5060>
p-asserted-identity: <sip:0972101112@91.121.129.17;user=phone>
server: Cirpack/v4.42j (gw_sip)
Allow: UPDATE, REFER, INFO
Content-Type: application/sdp
Content-Length: 262
v=0
o=cp10 132506808498 132506808500 IN IP4 192.168.1.1
s=SIP Call
c=IN IP4 192.168.1.1
t=0 0
m=audio 7070 RTP/AVP 0 8 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[5] 2011/12/28 10:28:34: SIP Rx tcp:192.168.1.12:2081:
CANCEL sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 2 CANCEL
Max-Forwards: 70
Reason: SIP;cause=487;text="Request terminated by user"
Proxy-Require: buttons
Content-Length: 0
[5] 2011/12/28 10:28:34: SIP Tx tcp:192.168.1.12:2081:
SIP/2.0 200 Ok
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 2 CANCEL
Contact: <sip:400@192.168.1.13:5060;transport=tcp>
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Length: 0
[5] 2011/12/28 10:28:34: SIP Tx tcp:192.168.1.12:2081:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 2 INVITE
Contact: <sip:400@192.168.1.13:5060;transport=tcp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Length: 0
[5] 2011/12/28 10:28:34: SIP Tx udp:91.121.129.17:5060:
CANCEL sip:0972101112@91.121.129.17;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>
Call-ID: 0f4287ed@pbx
CSeq: 12054 CANCEL
Max-Forwards: 70
Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes
Content-Length: 0
[5] 2011/12/28 10:28:34: SIP Rx tcp:192.168.1.12:2081:
ACK sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport
From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn
To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710
Call-ID: 3c267cc8ba57-s7m7z0rilz9z
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1
Proxy-Require: buttons
Content-Length: 0
[5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>
Call-ID: 0f4287ed@pbx
CSeq: 12054 CANCEL
server: Cirpack/v4.42j (gw_sip)
Content-Length: 0
[5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:
SIP/2.0 487 Session canceled
Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090
Call-ID: 0f4287ed@pbx
CSeq: 12054 INVITE
Contact: <sip:41.141.84.105:5060>
server: Cirpack/v4.42j (gw_sip)
Allow: UPDATE, REFER, INFO
Content-Length: 0
[5] 2011/12/28 10:28:34: SIP Tx udp:192.168.1.1:5060:
ACK sip:41.141.84.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport
Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090
Call-ID: 0f4287ed@pbx
CSeq: 12054 ACK
Max-Forwards: 70
Contact: <sip:0033184190197@192.168.1.13:5060;transport=udp>
Remote-Party-ID: <sip:0033184190197@91.121.129.17>;party=calling;screen=yes
Content-Length: 0
[5] 2011/12/28 10:28:34: INVITE Response 487 Session canceled: Terminate 0f4287ed@pbx
[5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:
ACK sip:192.168.1.13:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKb094afeb726b070e8ab8f4e7222edbb0
Via: SIP/2.0/UDP 41.141.84.105:5060;branch=z9hG4bKc0e36b0d24e61351467780d2975f9369
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090
Call-ID: 0f4287ed@pbx
CSeq: 12054 ACK
Contact: <sip:0033184190197@41.141.84.105>
max-forwards: 68
remote-party-id: <sip:0033184190197@91.121.129.17>;party=calling;screen=yes
Content-Length: 0
[5] 2011/12/28 10:28:35: SIP Rx udp:91.121.129.17:5060:
SIP/2.0 487 Session canceled
Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39
Record-Route: <sip:siproxd@192.168.1.1:5060;lr>
From: <sip:0033184190197@91.121.129.17>;tag=1119664557
To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090
Call-ID: 0f4287ed@pbx
CSeq: 12054 INVITE
Contact: <sip:41.141.84.105:5060>
server: Cirpack/v4.42j (gw_sip)
Allow: UPDATE, REFER, INFO
Content-Length: 0
=====
Any ideas ?
I thing sound disappear just after the "183 Media Change" ....
Thanks in advance,
Jeremy
snom ONE on Linux : High Availability
in Best Practices
Posted
But with "virtual machine" there is only ONE machine. Where is the failover if the machine completely crash ?