Can you provide the trace of the entire call?

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 05/14/2014 01:54 PM, Maqbul Khan - Reform InfoTech wrote:
Hi Razvan,

Thanks for your email, here is the trace:

User Datagram Protocol, Src Port: 58131 (58131), Dst Port: sip (5060)
Session Initiation Protocol (CANCEL)
     Request-Line: CANCEL sip:[email protected] SIP/2.0
         Method: CANCEL
         Request-URI: sip:[email protected]
             Request-URI User Part: 1234
             Request-URI Host Part: 192.168.1.29
         [Resent Packet: False]
     Message Header
         Via: SIP/2.0/UDP
192.168.1.40:58131;rport;branch=z9hG4bKPjuplWbA-fyp8Ih6HLWuR6lJKzF.oe3U1a
             Transport: UDP
             Sent-by Address: 192.168.1.40
             Sent-by port: 58131
             RPort: rport
             Branch: z9hG4bKPjuplWbA-fyp8Ih6HLWuR6lJKzF.oe3U1a
         Max-Forwards: 70
         From: "John"
<sip:[email protected]>;tag=HeCdsPoUumnQBNAzwbcJUylJDbBcfAp0
             SIP Display info: "John"
             SIP from address: sip:[email protected]
                 SIP from address User Part: 918866443340
                 SIP from address Host Part: 192.168.1.29
             SIP from tag: HeCdsPoUumnQBNAzwbcJUylJDbBcfAp0
         To: <sip:[email protected]>
             SIP to address: sip:[email protected]
                 SIP to address User Part: 1234
                 SIP to address Host Part: 192.168.1.29
         Call-ID: HUeOvf2nNpD20DrZepuaIidfKsbJWWXv
         CSeq: 10981 CANCEL
             Sequence Number: 10981
             Method: CANCEL
         Route: <sip:192.168.1.29;transport=udp;lr>
             Route URI: sip:192.168.1.29;transport=udp;lr
                 Route Host Part: 192.168.1.29
                 Route URI parameter: transport=udp
                 Route URI parameter: lr
         User-Agent: CSipSimple_lt023g-16/r2353
         Content-Length:  0


Maqbul A Khan
Reform InfoTech
*email*: [email protected] <mailto:[email protected]>
*mobile*: +91 99988 97686
*skype*: maqbul.a

On 12-May-2014, at 1:50 pm, Răzvan Crainea <[email protected]
<mailto:[email protected]>> wrote:

Hi, Maqbul!

Can you provide a trace for this scenario?

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 05/12/2014 09:39 AM, Maqbul - Reform InfoTech wrote:
Hi,

I’m using PJSIP Library for Android client, when call is initiated by A
Party, and before B Party answers the call, if A Party sends CANCEL
request, OpenSIPS does neither send ACK nor forward it to B Party.

This works fine if we do testing with X-Lite or Zoiper.  Please let me
know what could be the issue?

Maqbul A Khan
Reform InfoTech
*email*: [email protected] <mailto:[email protected]>
*mobile*: +91 99988 97686
*skype*: maqbul.a



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