One way audio

[BobbyJaffacake]

Hello.

 

We are testing snom one pbx (2011-4.2.0.3981 64bit 10 user trial) with OCS 2007R2 as a pbx for extensions and as gateway between ocs and ITSP (Draytel).

 

I absolutely love the software as its so tidy and straight foreward, but the only thing it wont properly pass a call from communicator to our itsp.

 

All outgoing calling from X-lite/sip phones works perfectly to our ITSP.

 

Communicator can call sip extensions fine and voicemail etc.

 

It only goes wrong when calling outside numbers when the reciever of the call does not answer straight away, ie after the first ring. The audio from communicator does not reach the recipient, but communicator can hear the recipient.

 

Ive spent a week looking into this and Im pulling my hair out. The settings are as the Wiki sets out, and I can get this to work with SipX no problem, however I dont want to use a Linux based system that is bulky compared to Snom One.

 

Ive changed our mediation server to use 2 nics, 1 nic. Pbx to use different outside connections. Different internet feeds and seperate connections to the mediation server.

Tried specifying the codecs and country codes.

Changed the timeouts plus loads more.

Whatever I do its always the same.

I have trawled this forum with some useful help but some links are back to the pbxnsip pages that dont exist any more.

All the servers are seperate, ie PBX seperate from Mediation.

 

I have spotted a issues in the log to do with "SIP TCP/TLS timeout on 10.0.0.21:2975, closing connection" but cannot find anything from that. See log below.

 

Any help would be greatly appreciated please.

 

Thanks Rob

 

 

[8] 2011/04/11 10:30:46: Received SIP connection 11 from 10.0.0.21:2975

[1] 2011/04/11 10:30:46: TCP: TOS could not be set, code 0

[9] 2011/04/11 10:30:46: UDP: Opening socket on 10.0.0.26:54194

[9] 2011/04/11 10:30:46: UDP: Opening socket on 10.0.0.26:54195

[5] 2011/04/11 10:30:46: Identify trunk (IP address and domain match) 2

[7] 2011/04/11 10:30:46: Set packet length to 20

[6] 2011/04/11 10:30:46: Sending RTP for 8fdc7ca8-3499-4f92-8b97-4c870ca2dfba to 10.0.0.21:61720, codec not set yet

[8] 2011/04/11 10:30:46: Call from an trunk 2

[8] 2011/04/11 10:30:46: Trunk OCS 2007r2@pbx.company.com has country code not set, area code not set

[9] 2011/04/11 10:30:46: Incoming: formatted From is = <sip:robert.xxxxxx@company.com>

[9] 2011/04/11 10:30:46: Incoming: formatted To is = <sip:0160xxxxxxx@10.0.0.26;user=phone>

[8] 2011/04/11 10:30:46: Trunk: Changing the user to 49

[8] 2011/04/11 10:30:46: To is <sip:0160xxxxxxx@10.0.0.26;user=phone>, user 0, domain 1

[8] 2011/04/11 10:30:46: From user 49

[8] 2011/04/11 10:30:46: Set the To domain based on From user 49@pbx.company.com

[7] 2011/04/11 10:30:46: set_codecs: for 8fdc7ca8-3499-4f92-8b97-4c870ca2dfba codecs "", codec_preference count 6

[8] 2011/04/11 10:30:46: Call state for call object 7: idle

[9] 2011/04/11 10:30:46: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 0160xxxxxxx@10.0.0.26

[5] 2011/04/11 10:30:46: Dialplan "Standard Dialplan": Match 0160xxxxxxx@10.0.0.26 to <sip:0160xxxxxxx@draytel.org;user=phone> on trunk draytel

[5] 2011/04/11 10:30:46: Using <sip:robert.xxxxxx@company.com> as redirect source address

[5] 2011/04/11 10:30:46: Charge user 49 for redirecting calls

[8] 2011/04/11 10:30:46: Play audio_moh/noise.wav

[9] 2011/04/11 10:30:46: UDP: Opening socket on 10.0.0.26:54756

[9] 2011/04/11 10:30:46: UDP: Opening socket on 10.0.0.26:54757

[7] 2011/04/11 10:30:46: set_codecs: for f0e36555@pbx codecs "", codec_preference count 6

[7] 2011/04/11 10:30:46: update_codec for f0e36555@pbx: codec_preference size 6, available codecs size 6

[9] 2011/04/11 10:30:46: Resolve 224: url sip:0160xxxxxxx@draytel.org;user=phone

[9] 2011/04/11 10:30:46: Resolve 224: naptr draytel.org

[9] 2011/04/11 10:30:46: Resolve 224: srv tls _sips._tcp.draytel.org

[9] 2011/04/11 10:30:46: Resolve 224: srv tcp _sip._tcp.draytel.org

[9] 2011/04/11 10:30:46: Resolve 224: srv udp _sip._udp.draytel.org

[9] 2011/04/11 10:30:46: Resolve 224: a udp draytel.org 5060

[9] 2011/04/11 10:30:46: Resolve 224: udp 217.14.132.183 5060

[7] 2011/04/11 10:30:46: Set packet length to 20

[7] 2011/04/11 10:30:46: update_codec for 8fdc7ca8-3499-4f92-8b97-4c870ca2dfba: codec_preference size 6, available codecs size 3

[6] 2011/04/11 10:30:46: Codec pcmu/8000 is chosen for call id 8fdc7ca8-3499-4f92-8b97-4c870ca2dfba

[9] 2011/04/11 10:30:46: Message repetition, packet dropped

[8] 2011/04/11 10:30:46: Answer challenge with username 8247176

[9] 2011/04/11 10:30:46: Resolve 226: udp 217.14.132.183 5060 udp:1

[9] 2011/04/11 10:30:46: Resolve 227: udp 217.14.132.183 5060 udp:1

[9] 2011/04/11 10:30:46: Message repetition, packet dropped

[9] 2011/04/11 10:30:49: Last message repeated 3 times

[7] 2011/04/11 10:30:49: Set packet length to 20

[6] 2011/04/11 10:30:49: Codec pcma/8000 is chosen for call id f0e36555@pbx

[6] 2011/04/11 10:30:49: Sending RTP for f0e36555@pbx to 77.240.48.216:18240, codec pcma/8000

[8] 2011/04/11 10:30:49: Call state for call object 7: alerting

[7] 2011/04/11 10:30:49: 8fdc7ca8-3499-4f92-8b97-4c870ca2dfba: RTP pass-through mode

[7] 2011/04/11 10:30:49: f0e36555@pbx: RTP pass-through mode

[6] 2011/04/11 10:30:54: SIP TCP/TLS timeout on 10.0.0.21:2975, closing connection

[8] 2011/04/11 10:30:54: Release SIP thread 11

[7] 2011/04/11 10:30:57: Call f0e36555@pbx: Clear last INVITE

[7] 2011/04/11 10:30:57: Set packet length to 20

[9] 2011/04/11 10:30:57: Resolve 229: url sip:217.14.132.183;lr=on;ftag=48979

[9] 2011/04/11 10:30:57: Resolve 229: udp 217.14.132.183 5060

[7] 2011/04/11 10:30:57: Determine pass-through mode after receiving response

[8] 2011/04/11 10:30:57: Call state for call object 7: connected

[9] 2011/04/11 10:30:57: Resolve 230: tcp 10.0.0.21 2975

[6] 2011/04/11 10:30:57: Response to 10.0.0.21:2975 must be sent over existing connection

[8] 2011/04/11 10:30:57: DNS: NAPTR draytel.org expired

[8] 2011/04/11 10:30:58: DNS: SRV _sips._tcp.draytel.org expired

[8] 2011/04/11 10:30:58: DNS: SRV _sip._tcp.draytel.org expired

[8] 2011/04/11 10:30:58: DNS: SRV _sip._udp.draytel.org expired

[9] 2011/04/11 10:30:58: Resolve 231: url sip:draytel.org

[9] 2011/04/11 10:30:58: Resolve 231: naptr draytel.org

[9] 2011/04/11 10:30:58: DNS: Request draytel.org from server 10.0.0.22

[8] 2011/04/11 10:30:58: DNS: Add NAPTR draytel.org (ttl=60)

[9] 2011/04/11 10:30:58: Resolve 231: naptr draytel.org

[9] 2011/04/11 10:30:58: Resolve 231: srv tls _sips._tcp.draytel.org

[9] 2011/04/11 10:30:58: DNS: Request _sips._tcp.draytel.org from server 10.0.0.22

[8] 2011/04/11 10:30:58: DNS: Add SRV _sips._tcp.draytel.org (ttl=60)

[9] 2011/04/11 10:30:58: Resolve 231: srv tls _sips._tcp.draytel.org

[9] 2011/04/11 10:30:58: Resolve 231: srv tcp _sip._tcp.draytel.org

[9] 2011/04/11 10:30:58: DNS: Request _sip._tcp.draytel.org from server 10.0.0.22

[8] 2011/04/11 10:30:58: DNS: Add SRV _sip._tcp.draytel.org (ttl=60)

[9] 2011/04/11 10:30:58: Resolve 231: srv tcp _sip._tcp.draytel.org

[9] 2011/04/11 10:30:58: Resolve 231: srv udp _sip._udp.draytel.org

[9] 2011/04/11 10:30:58: DNS: Request _sip._udp.draytel.org from server 10.0.0.22

[8] 2011/04/11 10:30:58: DNS: Add SRV _sip._udp.draytel.org (ttl=60)

[9] 2011/04/11 10:30:58: Resolve 231: srv udp _sip._udp.draytel.org

[9] 2011/04/11 10:30:58: Resolve 231: a udp draytel.org 5060

[9] 2011/04/11 10:30:58: Resolve 231: udp 217.14.132.183 5060

[9] 2011/04/11 10:30:58: Message repetition, packet dropped

[8] 2011/04/11 10:30:58: Answer challenge with username 8247176

[9] 2011/04/11 10:30:58: Resolve 232: udp 217.14.132.183 5060 udp:1

[9] 2011/04/11 10:30:58: Message repetition, packet dropped

[9] 2011/04/11 10:30:58: Last message repeated 2 times

[2] 2011/04/11 10:30:58: Trunk status draytel (3) changed to "200 OK" (Refresh interval 90 seconds)

[9] 2011/04/11 10:31:03: Resolve 233: aaaa udp 217.14.132.183 5060

[9] 2011/04/11 10:31:03: Resolve 233: a udp 217.14.132.183 5060

[9] 2011/04/11 10:31:03: Resolve 233: udp 217.14.132.183 5060

[7] 2011/04/11 10:31:03: 8fdc7ca8-3499-4f92-8b97-4c870ca2dfba: Media-aware pass-through mode

[8] 2011/04/11 10:31:03: Hangup: Call 13 not found

[9] 2011/04/11 10:31:03: Resolve 234: aaaa tcp 10.0.0.21 5060

[9] 2011/04/11 10:31:03: Resolve 234: a tcp 10.0.0.21 5060

[9] 2011/04/11 10:31:03: Resolve 234: tcp 10.0.0.21 5060

[8] 2011/04/11 10:31:03: Received SIP connection 12 from 10.0.0.21:5060

[1] 2011/04/11 10:31:03: TCP: TOS could not be set, code 0

[8] 2011/04/11 10:31:03: Hangup: Call 13 not found

[7] 2011/04/11 10:31:03: Call 8fdc7ca8-3499-4f92-8b97-4c870ca2dfba: Clear last request

[5] 2011/04/11 10:31:03: BYE Response: Terminate 8fdc7ca8-3499-4f92-8b97-4c870ca2dfba

[Vodia PBX]

Hard to say from that log, try to get the SIP packets in the log as well.

 

 

You could also try to set the registration interval on the PBX to a very short duration, at least for TCP (like 20 seconds). Maybe this helps to avoid the TCP disconnects; I agree that could be the problem. The Communication has some weired bugs like it cannot deal with Expiry times; anyhow those problems should be better visible if we can see the SIP packets. Unless you are using TLS, a PCAP trace might also be useful.

[BobbyJaffacake]

Hi. Thanks for the prompt response.

 

Ive attached the wireshark file of the conversation from the Snom server. (192.168.1.149) while an error occurred.

 

Im not sure you set the registration interval but Ill look.

 

I hope this can help.

mocsmed-snom.zip

[pbx support]

After the call was connected on the outbound leg, PBX did not send 200 OK to the OCS. That was bit interesting.

 

Could you try to set the "Ringback" setting on the "OCS 2007r2" trunk to "Message 180" instead of the default "Media" and see if the call gets connected properly?

 

Also, I saw an ICMP destination unreachable error too on the trace file.

[Vodia PBX]

The problem is that the TCP connection does not register anything. Unless something is registered on the connection, the PBX keep the connection alive only for 8 seconds and then disconnects the TCP. Then when it wants to send the 200 Ok, the connection is already gone.

 

 

In order to solve this problem, we need to give you another version with a fix. Please contact support (private message to pbx_support in this forum will do) to spin you a new build, please indicate the OS that you are using.

[BobbyJaffacake]

Hi. Thanks for the suggestions, Ive tried changing the Ringback with no success. Ive also contacted pbxsupport to see if they can help with a new build.

 

I also tried different ports for the TCP side in case of conflicts and binding issues but still the same results.

 

We shall see, however Im on holidays for just over a week now so we shall see.

[BobbyJaffacake]

Hi again.

 

Well pbxsupport promptly responded within a few days with a patched version, and Im glad to say that it works perfectly now. The version they provided is 2011-4.2.1.4009 (Win64).

I have to say a big thanks to the developers and support team at Snom and all their help as you have saved me so much time.

 

Thank you again and keep up the good work.