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User disconnects call


lintentech
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Hi all, i've been using the service of a 3rd party hosted VoIP and thought I'd have a try with by own PBX. As all our phones are Snom's the most logical choice was the Snom ONE pbx!

 

Now all has been going fine but i've got a problem calling one of the extenstion, it rings, i can answer it but some times there is just no audio and the other phone just remains on "183 Session Progress" I then get an email sayiny snom PBX: User disconnects call:

 

Rx: udp:86.28.219.71:1217 (1164 bytes)

INVITE sip:1001@77.240.1.41;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.0.5:1217;branch=z9hG4bK-j3zinuhxiti1;rport

From: <sip:1002@77.240.1.41>;tag=4wtli7le39

To: <sip:1001@77.240.1.41;user=phone>

Call-ID: 3c26823595d8-dtbbpvekty9n

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:1002@192.168.0.5:1217;line=8vhw1eju>;reg-id=1

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom370/7.3.30

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Content-Type: application/sdp

Content-Length: 450

 

v=0

o=root 467526577 467526577 IN IP4 192.168.0.5

s=call

c=IN IP4 192.168.0.5

t=0 0

m=audio 59200 RTP/AVP 0 8 9 99 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZjUEK2WhjujAVLxYtuQnWpV4eImHuRSd9X0wY7yW

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:99 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

 

Any one got any pointers for me please? It always seems to be ext 1002 calling 1001 - I don't *think* i've had the issues 1001 calling 1002.

 

Dan

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Was that the complete email or you posted just the initial part?

 

Extension to extension calling should not have any audio issues, unless the phone is remote and something in the remote network is blocking(NAT/firewall etc) some messages.

If both phones are on the same network, just try to reset the phone and see if the problem goes away.

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Was that the complete email or you posted just the initial part?

 

Extension to extension calling should not have any audio issues, unless the phone is remote and something in the remote network is blocking(NAT/firewall etc) some messages.

If both phones are on the same network, just try to reset the phone and see if the problem goes away.

 

That was the complete email yes, I'll post some more from the phone log the next time it occurs. It's hosted system so yes there it could be NAT/firewall but it's very intermittent and only seems to be effecting a particular extension.

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What you see in the attachment of the email is the other side of the call (the one that causes the issue). So it makes sense to see only the INVITE.

 

With 99.99 % probabilty this is a firewall issue, maybe an issue with the UDP packet size, maybe the firewall has a built-in SIP ALG that does more hard than it helps. I would use TLS transport layer (plug and play!), then the firewall has no chance to mess with the packets and it will probably work.

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