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Inbound call to SIP Uri


global_s
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Hi everyone,

i'm having issues trying to setup an incoming trunk with my provider Voxbone.

 

Voxbone redirect calls to a SIP uri specified by the user. So in their system I inserted EXTnumber@SnomoneIPaddress

 

I setup an inbound trunk as SIP gateway, leaving evry setting blank (voxbone has multiple ip addresses) with no dialplan.

 

When calling the number, the call gets to snomone, the system answer and I can see in the logs that it is connected and active. However I don't hear any audio on the external number. Audio gets passed to the snomone

 

If I try to call the same extension from another extension (who's is not in the same network as the pbx) the audio is fine. The issue is with phone calls coming from voxbone.

 

RTP range i specified from 10000 to 10999, no other active call in the system; ports are ok (the extension can communicate with the system). Codec used is G711

No dial plan at all in my system

 

What is the problem here?

 

Here's the log file

[5] 2012/07/18 17:23:25:	SIP Rx udp:81.201.83.45:5060:
INVITE sip:8877@176.254.24.10 SIP/2.0
Call-ID: R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45
CSeq: 102 INVITE
From: "3902XXXXXXX" <sip:3902XXXXXXX@voxbone.com>;tag=75020
To: <sip:8877@176.254.24.10>
Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKa2c394812378151dc99c6b011eb44f0b
Max-Forwards: 69
Content-Type: application/sdp
Contact: <sip:3902XXXXXXX@81.201.83.45:5060;transport=udp>
User-Agent: Vox Callcontrol
Content-Length: 311

v=0
o=root 32412 32412 IN IP4 81.201.83.48
s=session
c=IN IP4 81.201.83.48
t=0 0
m=audio 10004 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[9] 2012/07/18 17:23:25:	UDP: Opening socket on 10.252.0.8:10870
[9] 2012/07/18 17:23:25:	UDP: Opening socket on 10.252.0.8:10871
[9] 2012/07/18 17:23:25:	UDP: Opening socket on [::]:10870
[9] 2012/07/18 17:23:25:	UDP: Opening socket on [::]:10871
[5] 2012/07/18 17:23:25:	Identify trunk (domain name match) 4
[5] 2012/07/18 17:23:25:	SIP Rx udp:81.201.83.45:5060:
INVITE sip:8877@176.254.24.10 SIP/2.0
Call-ID: R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45
CSeq: 102 INVITE
From: "3902XXXXXXX" <sip:3902XXXXXXX@voxbone.com>;tag=75020
To: <sip:8877@176.254.24.10>
Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKa2c394812378151dc99c6b011eb44f0b
Max-Forwards: 69
Content-Type: application/sdp
Contact: <sip:3902XXXXXXX@81.201.83.45:5060;transport=udp>
User-Agent: Vox Callcontrol
Content-Length: 311

v=0
o=root 32412 32412 IN IP4 81.201.83.48
s=session
c=IN IP4 81.201.83.48
t=0 0
m=audio 10004 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[9] 2012/07/18 17:23:25:	Resolve 153: aaaa udp 81.201.83.45 5060
[9] 2012/07/18 17:23:25:	Resolve 153: a udp 81.201.83.45 5060
[9] 2012/07/18 17:23:25:	Resolve 153: udp 81.201.83.45 5060
[5] 2012/07/18 17:23:25:	SIP Tx udp:81.201.83.45:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKa2c394812378151dc99c6b011eb44f0b
From: "3902XXXXXXX" <sip:3902XXXXXXX@voxbone.com>;tag=75020
To: <sip:8877@176.254.24.10>;tag=f4788da59d
Call-ID: R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45
CSeq: 102 INVITE
Content-Length: 0

[7] 2012/07/18 17:23:25:	Set packet length to 20
[6] 2012/07/18 17:23:25:	Sending RTP for R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45 to 81.201.83.48:10004, codec not set yet
[8] 2012/07/18 17:23:25:	Call from a trunk 4
[8] 2012/07/18 17:23:25:	Trunk Voxbone@176.254.24.10 has country code not set, area code not set
[9] 2012/07/18 17:23:25:	Incoming: formatted From is = "3902XXXXXXX" <sip:3902XXXXXXX@voxbone.com;user=phone>
[9] 2012/07/18 17:23:25:	Incoming: formatted To is = <sip:8877@176.254.24.10;user=phone>
[9] 2012/07/18 17:23:25:	Incoming: formatted URI is = sip:8877@176.254.24.10;user=phone
[8] 2012/07/18 17:23:25:	To is <sip:8877@176.254.24.10;user=phone>, user 31, domain 1
[8] 2012/07/18 17:23:25:	To user 8877
[5] 2012/07/18 17:23:25:	Domain trunk Voxbone@176.254.24.10 sends call to 8877 in domain 176.254.24.10
[8] 2012/07/18 17:23:25:	Set the To domain based on To user 8877@176.254.24.10
[8] 2012/07/18 17:23:25:	Call state for call object 32: idle
[8] 2012/07/18 17:23:25:	Call state for call object 32: connected
[8] 2012/07/18 17:23:25:	Play space10
[7] 2012/07/18 17:23:25:	set_codecs: for R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45 codecs "0 8 18", codec_preference count 3
[7] 2012/07/18 17:23:25:	Set packet length to 20
[9] 2012/07/18 17:23:25:	update_codecs for R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45: adding codec pcmu/8000 to available list
[9] 2012/07/18 17:23:25:	update_codecs for R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45: adding codec pcma/8000 to available list
[9] 2012/07/18 17:23:25:	update_codecs for R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45: codec_preference size 3, available codecs size 3
[6] 2012/07/18 17:23:25:	Codec pcmu/8000 is chosen for call id R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45
[9] 2012/07/18 17:23:25:	Resolve 154: aaaa udp 81.201.83.45 5060
[9] 2012/07/18 17:23:25:	Resolve 154: a udp 81.201.83.45 5060
[9] 2012/07/18 17:23:25:	Resolve 154: udp 81.201.83.45 5060
[5] 2012/07/18 17:23:25:	SIP Tx udp:81.201.83.45:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKa2c394812378151dc99c6b011eb44f0b
From: "3902XXXXXXX" <sip:3902XXXXXXX@voxbone.com>;tag=75020
To: <sip:8877@176.254.24.10>;tag=f4788da59d
Call-ID: R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45
CSeq: 102 INVITE
Contact: <sip:8877@176.254.24.10:5060;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
Content-Type: application/sdp
Content-Length: 275

v=0
o=- 1611230459 1611230459 IN IP4 176.254.24.10
s=-
c=IN IP4 176.254.24.10
t=0 0
m=audio 10870 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=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[6] 2012/07/18 17:23:25:	Codec pcmu/8000 is chosen for call id R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45
[5] 2012/07/18 17:23:25:	SIP Rx udp:81.201.83.45:5060:
ACK sip:8877@176.254.24.10:5060;transport=udp SIP/2.0
Call-ID: R725636W5BB2LP7TIWNUXXPAUU@81.201.83.45
CSeq: 102 ACK
From: "3902XXXXXXX" <sip:3902XXXXXXX@voxbone.com>;tag=75020
To: <sip:8877@176.254.24.10>;tag=f4788da59d
Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bK5e4614b091a3f5f2ff253aec99790b95
Max-Forwards: 69
Contact: <sip:3902XXXXXXX@81.201.83.45:5060;transport=udp>
User-Agent: Vox Callcontrol
Content-Length: 0

[8] 2012/07/18 17:23:27:	Play audio_en/co_welcome_conference.wav
[7] 2012/07/18 17:23:28:	Received RFC4733 DTMF on codec 101
[6] 2012/07/18 17:23:28:	Received DTMF 8
[6] 2012/07/18 17:23:28:	Received DTMF 0

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Hmm. Everything looks good from the log. Usually audio problems happen the other direction. I assume the audio files are installed, especiall the audio_en/co_welcome_conference.wav file and it is readable.

 

The only idea that I have is to install Wireshark and make a PCAP trace. Then we might find out what is going on with the RTP and if packets are being sent to the provider.

 

P.S. Please try to upgrade to 4.5 (Epsilon), for a new installation that is definitevely a good idea. E.g. 4.5 has the Wireshark PCAP utility already included if you are running Linux.

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Hmm. Everything looks good from the log. Usually audio problems happen the other direction. I assume the audio files are installed, especiall the audio_en/co_welcome_conference.wav file and it is readable.

 

The only idea that I have is to install Wireshark and make a PCAP trace. Then we might find out what is going on with the RTP and if packets are being sent to the provider.

 

P.S. Please try to upgrade to 4.5 (Epsilon), for a new installation that is definitevely a good idea. E.g. 4.5 has the Wireshark PCAP utility already included if you are running Linux.

 

Thanks for the reply.

I upgraded to 4.5 last version but the issue remains.

I tried to find the wireshark utility on the web pages, but I guess it is just in terminal right? I haven't found any reference to it on the KB.

 

anyway, even odder, Voxbone competitor, DIDWW works perfectly with my current configuration.

Seems like the snom is not sending packets to voxbone (or voxbone discard them) but I cannot figure out why.

 

Do you have a link with the explanation on how to wireshark from version 4.5?

 

Thanks for the help. It is very appreciated

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This is strande, indeed. Wireshark can be run as seperate tool (www.wireshark.org), it is free of charge. This will help us finding out if the stream to the provider really goes out and how the packets look like. The tool that we have added in 4.5 is something similar, but has limits e.g. on the selection of the Ethernet interface and it only works in Linux. It is probably easier to just quickly install a standalone tool instead.

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This is strande, indeed. Wireshark can be run as seperate tool (www.wireshark.org), it is free of charge. This will help us finding out if the stream to the provider really goes out and how the packets look like. The tool that we have added in 4.5 is something similar, but has limits e.g. on the selection of the Ethernet interface and it only works in Linux. It is probably easier to just quickly install a standalone tool instead.

 

As for my issue, I think it is something with Voxbone; another number bought from them is working fine with the same configuration. I opened a ticket and we will see.

 

As for the utility, I would love to know how to use the tool already included. First because my pbx runs on a virtualized debian machine, seconds because i don't want to install any other tool that could interfere withe the pbx stability, plus there is no stable port of wireshark for debian.

 

Looking forward to hear from you.

 

Thanks

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As for the utility, I would love to know how to use the tool already included. First because my pbx runs on a virtualized debian machine, seconds because i don't want to install any other tool that could interfere withe the pbx stability, plus there is no stable port of wireshark for debian.

 

In Linux, you dont have to worry about stabilty. Wireshark is only running when you start it and it does not install anything in the network stack (not like in Windows). You can even run tcpdump instead of wireshark, and then use wireshark on another machine as viewer.

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In Linux, you dont have to worry about stabilty. Wireshark is only running when you start it and it does not install anything in the network stack (not like in Windows). You can even run tcpdump instead of wireshark, and then use wireshark on another machine as viewer.

 

 

Ok, I'll try that for future references.

My original issue was because codec mismatch.

 

Voxbone and Snom one didn't send the right codec to each other and the communication got stuck. How come that they failed to agree on a common codec?

 

voxbone offered g711a g771u g729

snom one offered g711u g771a

 

voxbone sent messages in g711a

snom one sent messages in g711u

 

Issue solved by disabiling g711a in Voxbone

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