Re: [asterisk-users] SIP trunk problem: Message option 200 (heartbeat)

[Joshua C. Colp]

On Wed, Sep 4, 2019, at 6:01 AM, bilal ghayyad wrote:
> Hello;
> 
> I am facing a trouble with the SIP service provider, they are saying 
> that there is a problem related to message option 200 (the heartbeat), 
> so what is required to add this for the sip configuration? Below is my 
> sip debug trace log with the them and the sip peer configuration:

OPTIONS is treated as if it were an INVITE, so it looks up the extension in the 
dialplan. The following shows what extension and context:

[Sep 4 12:42:20] Looking for s in trunkinbound (domain 10.240.147.26)

If you add an "s" extension to the "trunkinbound" context it should then 
respond 200 OK.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] SIP trunk problem: Message option 200 (heartbeat)

[bilal ghayyad]

Hello;
I am facing a trouble with the SIP service provider, they are saying that there 
is a problem related to message option 200 (the heartbeat), so what is required 
to add this for the sip configuration? Below is my sip debug trace log with the 
them and the sip peer configuration:


[Sep  4 12:42:18]<>

[Sep  4 12:42:18]Scheduling destruction of SIP 
dialog'kntaydtnxawwiyky4yaadanied4lk4ys@139.110.215.10' in 32000 ms (Method: 
OPTIONS)

[Sep  4 12:42:19]Really destroying SIP dialog 
'xidaktlblveesdlbtkbyyexelaiibtvx@139.110.215.10'Method: OPTIONS

[Sep  4 12:42:20]

[Sep  4 12:42:20] <--- SIP read fromUDP:10.215.110.139:5060 --->

[Sep  4 12:42:20] OPTIONS sip:10.240.147.26:5060SIP/2.0

[Sep  4 12:42:20] Via: 
SIP/2.0/UDP10.215.110.139:5060;branch=z9hG4bKxlaisbketikainbbedltysdla;Role=3;Hpt=8e78_16;TRC=-;pth=0;X-HwDim=4

[Sep  4 12:42:20] Call-ID:aecdkcavavdticeydtswaewesttbbad4@139.110.215.10

[Sep  4 12:42:20] From:;tag=yaekiyny

[Sep  4 12:42:20] To: 

[Sep  4 12:42:20]CSeq: 1 OPTIONS

[Sep  4 
12:42:20]Contact:;expires=65535

[Sep  4 12:42:20]Accept: application/sdp

[Sep  4 12:42:20]Max-Forwards: 70

[Sep  4 12:42:20]Content-Length: 0

[Sep  4 12:42:20]

[Sep  4 12:42:20] <->

[Sep  4 12:42:20] --- (10 headers 0 lines) ---

[Sep  4 12:42:20] Sending to 10.215.110.139:5060(NAT)

[Sep  4 12:42:20]Looking for s in trunkinbound (domain 10.240.147.26)

[Sep  4 12:42:20]

[Sep  4 12:42:20] <--- Transmitting (NAT) to10.215.110.139:5060 --->

[Sep  4 12:42:20] SIP/2.0 404 Not Found

[Sep  4 12:42:20] Via: 
SIP/2.0/UDP10.215.110.139:5060;branch=z9hG4bKxlaisbketikainbbedltysdla;Role=3;Hpt=8e78_16;TRC=-;pth=0;X-HwDim=4;received=10.215.110.139;rport=5060

[Sep  4 12:42:20] From: ;tag=yaekiyny

[Sep  4 12:42:20] To:;tag=as7efcc39d

[Sep  4 12:42:20]Call-ID: aecdkcavavdticeydtswaewesttbbad4@139.110.215.10

[Sep  4 12:42:20]CSeq: 1 OPTIONS

[Sep  4 12:42:20]Server: Asterisk PBX 13.24.1-vici

[Sep  4 12:42:20]Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY, INFO,PUBLISH, MESSAGE

[Sep  4 12:42:20]Supported: replaces, timer

[Sep  4 12:42:20]Accept: application/sdp

[Sep  4 12:42:20]Content-Length: 0

And below is the sip peer configuration:
[ooredoo]type=friendhost=10.215.110.139bindport=5060dtmfmode=autocontext=trunkinboundcanreinvite=nodisallow=allallow=ulawallow=alawallow=g729allow=gsmtrustrpid=yesnat=force_rport,comediainsecure=invite,port

So what is needed to resolve this [Sep  4 12:42:20] SIP/2.0 404 Not Found ?What 
is needed to be added for the sip peer configuration?
RegardsBilal-- 
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