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Calls going in to hold mode during a call.


Jeremy Isherwood
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Okay, I have a SNOM ONE, with 10 extensions 4 SIP trunks (Spitfire) and one Patton BRI. Phones are either 870 or 821.

 

Most of the time when you dial an external number the phone goes quite, no ringing and when you look at the handset display it show your line and a separate box for the out going line in (CALLBACK)

 

When the call answers the call is connect to the handset.

 

The attachéd image shows handset state when this fault happens.

 

Also during a call, it suddenly changes, you can't hear the other party, you have to press the outside line to pick the call back up. I have noticed that happens a lot when another extension finishes a call.

Log shows

 

192.XXX.XXX.XXX is internal IP

XXX.XXX.XXX.XXX is external IP

 

[6] 2011/12/09 14:39:58: Sending RTP for 4bb5263c351c-1joeogyn0org to 192.XXX.XXX.XXX:53916, codec not set yet

[6] 2011/12/09 14:39:59: Codec pcmu/8000 is chosen for call id 4bb5263c351c-1joeogyn0org

[6] 2011/12/09 14:40:02: Codec pcmu/8000 is chosen for call id 0ca6f9b7@pbx

[6] 2011/12/09 14:40:02: Sending RTP for 0ca6f9b7@pbx to 192.XXX.XXX.XXX:57058, codec pcmu/8000

[5] 2011/12/09 14:40:02: INVITE Response 487 Request Terminated: Terminate acd39328@pbx

[5] 2011/12/09 14:40:34: BYE Response: Terminate 4bb5263c351c-1joeogyn0org

[6] 2011/12/09 14:40:39: Sending RTP for 6804293c8796-3tmsy24ggk3t to 192.XXX.XXX.XXX:63022, codec not set yet

[5] 2011/12/09 14:40:39: Dialplan "FUE-DP1": Match 01527000000@192.XXX.XXX.XXX to <sip:01527000000@XXX.XXX.XXX.XXX;user=phone> on trunk Spitfire

[6] 2011/12/09 14:40:39: Codec pcmu/8000 is chosen for call id 6804293c8796-3tmsy24ggk3t

[6] 2011/12/09 14:40:40: Codec pcmu/8000 is chosen for call id 1ce12315@pbx

[6] 2011/12/09 14:40:40: Sending RTP for 1ce12315@pbx to XXX.XXX.XXX.XXX:63832, codec pcmu/8000

[5] 2011/12/09 14:41:40: BYE Response: Terminate 1ce12315@pbx

[6] 2011/12/09 14:45:01: Sending RTP for 71b6263cbbe6-bd0tdvkcy08x to 192.XXX.XXX.XXX:57572, codec not set yet

[5] 2011/12/09 14:45:01: Dialplan "FUE-DP1": Match 01527000000@192.XXX.XXX.XXX to <sip:01527000000@XXX.XXX.XXX.XXX;user=phone> on trunk Spitfire

[6] 2011/12/09 14:45:01: Codec pcmu/8000 is chosen for call id 71b6263cbbe6-bd0tdvkcy08x

[6] 2011/12/09 14:45:01: Codec pcmu/8000 is chosen for call id 654b2079@pbx

[6] 2011/12/09 14:45:01: Sending RTP for 654b2079@pbx to XXX.XXX.XXX.XXX:58724, codec pcmu/8000

[6] 2011/12/09 14:45:23: Sending RTP for 8705293c9707-70w6p5go569z to 192.XXX.XXX.XXX:53722, codec not set yet

[6] 2011/12/09 14:45:23: Codec pcmu/8000 is chosen for call id 8705293c9707-70w6p5go569z

[6] 2011/12/09 14:45:24: Last message repeated 2 times

[6] 2011/12/09 14:45:24: Received DTMF 0

post-15034-0-46397300-1323444762_thumb.jpg

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Are they using BroadSoft trunks? We recently found a bug there when BS sends an IP address 0.0.0.0 and asks the PBX to propose a new SDP. Because of the 0.0.0.0 the PBX believes the call is on hold, and that causes issues with the PBX.

 

What is a broadsoft trunk?

 

Not sure what you mean. A clearer explanation please.

 

Thanks.

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BroadSoft provides SIP technology (www.broadsoft.com), there are many service providers are using their technology. Maybe you service provider is one of them. Looking at the SIP pakcet coming from them will show you if that is that is the case. If you like, please copy & paste it here.

 

What am I looking for in the log file? Do you have an example?

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[8] 2011/12/09 20:51:46: Incoming call: Request URI sip:441213587860@192.168.250.66:5060;transport=udp;line=c81e728d, To is <sip:441213587890@85.258.143.17>

[8] 2011/12/09 20:51:46: Set the To domain based on To user 100@192.168.250.66

[5] 2011/12/09 20:51:46: Using <sip:01924605950@217.153.128.46;user=phone> as redirect source address

[8] 2011/12/09 20:51:46: Answer challenge with username 441213587860

[7] 2011/12/09 20:51:46: Call ce0b2f61@pbx: Clear last request

[7] 2011/12/09 20:51:46: Call 083f2d8c@pbx: Clear last request

[7] 2011/12/09 20:51:46: Call b9797867@pbx: Clear last request

[8] 2011/12/09 20:51:53: Hangup: Call 505 not found

[7] 2011/12/09 20:51:53: Call ce0b2f61@pbx: Clear last request

[7] 2011/12/09 20:51:53: Call ce0b2f61@pbx: Clear last INVITE

[5] 2011/12/09 20:51:53: INVITE Response 487 Request Terminated: Terminate ce0b2f61@pbx

[7] 2011/12/09 20:51:53: Call 083f2d8c@pbx: Clear last request

[7] 2011/12/09 20:51:53: Call 083f2d8c@pbx: Clear last INVITE

[5] 2011/12/09 20:51:53: INVITE Response 487 Request Terminated: Terminate 083f2d8c@pbx

[7] 2011/12/09 20:51:53: Call a3e874e7@pbx: Clear last request

[7] 2011/12/09 20:51:53: Call b9797867@pbx: Clear last request

[7] 2011/12/09 20:51:53: Call b9797867@pbx: Clear last INVITE

[5] 2011/12/09 20:51:53: INVITE Response 487 Request Terminated: Terminate b9797867@pbx

[7] 2011/12/09 20:51:53: Call a3e874e7@pbx: Clear last INVITE

[5] 2011/12/09 20:51:53: INVITE Response 487 Request terminated: Terminate a3e874e7@pbx

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Well the SIP trace looks actually okay on the trunk side. How stable is your Internet connection? How does the MOS graph look on this trunk? If you get low scores, this looks like there is a problem with the suitability of the Internet access for VoIP calls. If you want, you can also run a Wireshark trace for calls to give you the ultimate answer if the connection is okay or not.

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