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i have an issue with my calls being made via ASTERISK PBX running on linux CENTOS 6.2.

The use case is that a call is triggered from /var/spool/asterisk/outbound/

The caller is dialed The dialplan executes :

Answer()
Wait(1.5)
Set(Timestamp=$<someformat)
Record(.../<filename>.wav,0,0,y)
HangUp()

My SIP trunk provder is nextiva. What i have noticed from the wireshark trace is that nextiva sends a SIP: BYE request just before the call is dropped.

I attack the wireshark trace for reference:

536 110.28522   192.168.0.236   208.73.146.95   SIP/SDP Request: INVITE sip:0116590224650@208.73.146.95, with session description
537 110.477662  208.73.146.95   192.168.0.236   SIP Status: 100 Trying
538 110.491041  208.73.146.95   192.168.0.236   SIP Status: 407 Proxy Authentication Required
539 110.491738  192.168.0.236   208.73.146.95   SIP Request: ACK sip:0116590224650@208.73.146.95
540 110.491833  192.168.0.236   208.73.146.95   SIP/SDP Request: INVITE sip:0116590224650@208.73.146.95, with session description
541 110.685694  208.73.146.95   192.168.0.236   SIP Status: 100 Trying
551 117.480397  208.73.146.95   192.168.0.236   SIP/SDP Status: 183 Session Progress, with session description
554 120.407182  208.73.146.95   192.168.0.236   SIP/SDP Status: 200 OK, with session description
555 120.407495  192.168.0.236   208.73.146.95   SIP Request: ACK sip:0116590224650@208.73.146.95:5060;transport=udp
556 121.40902   192.168.0.236   208.73.146.95   RTP PT=ITU-T G.711 PCMU, SSRC=0xE5D7E61, Seq=39878, Time=160 
557 121.429117  192.168.0.236   208.73.146.95   RTP PT=ITU-T G.711 PCMU, SSRC=0xE5D7E61, Seq=39879, Time=320 
558 SSRC=0x17D1D704, Seq=64350, Time=1164450752 
2152    151.356593  208.73.146.95   192.168.0.236   RTP PT=ITU-T G.711 PCMU, 
SSRC=0x17D1D704, Seq=64351, Time=1164450912 
.
.
.
.

2153    151.376572  208.73.146.95   192.168.0.236   RTP PT=ITU-T G.711 PCMU, SSRC=0x17D1D704, Seq=64352, Time=1164451072 
2156    151.409798  192.168.0.236   208.73.146.95   RTCP    Receiver Report   Source description   
2157    151.497917  208.73.146.95   192.168.0.236   SIP Request: BYE sip:706955271@192.168.0.236:5060
2158    151.498195  192.168.0.236   208.73.146.95   SIP Status: 200 OK
2164    152.125251  192.168.0.236   208.73.146.95   SIP Request: REGISTER 

Has anyone else had familiar issues?

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Are you sure that the called party didn't just hang up? –  Michael Hampton Jul 22 '13 at 21:14
    
Yes it is not that case. I manage to solve it, problem was in the juniper firewall provided by my ISP. –  Qumar Jul 23 '13 at 7:56
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1 Answer 1

up vote 1 down vote accepted

In cases where the BYE is being sent from downstream like that, I always open a ticket with my provider and inquire as to why they sent the BYE. Very often it's another ULC (underlying carrier) they'll need to open the ticket with in order to resolve it. Sometimes it's even further downstream than that. Provide your CallIDs and PCAPs and it shouldn't be an issue for them to track it down.

share|improve this answer
    
Hi Matt! Thanks yes i opened a ticket. I have since rectified the issue. The problem is my SIP provider ( Nextiva) said that they had issues with juniper fire walls, we were using a juniper fire wall provided by my ISP provider. We did a work around the juniper fire wall now everything works as it should. Thanks for re affirming my thoughts that it could not be a config error on my end! –  Qumar Jul 23 '13 at 7:55
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