Hello All! There is description of my problem with Asteriks below. Asteriks CLI says: "File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call" Sip debug on the server gives the next: Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.26:5060 From: <sip:8523@192.168.0.24;user=phone>;tag=106403508 To: <sip:500@192.168.0.24;user=phone>;tag=as0771c6f9 Call-ID: 3296035458@192.168.0.26 CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: <sip:500@192.168.0.22> Content-Type: application/sdp Content-Length: 209 8523 is Cisco ATA-186 The sip.conf content: - - - - - [cisco8523] type=friend username=8523 secret=test nat=no host=dynamic canreinvite=no qualify=300 defaultip=192.168.0.26 - - - - - Why I place a call to Asteriks. I hear some invitation but connection brokes when retransmit exceed. Could anyone give some advice or solution. Thanks in advance -- Best regards Vlad
That means that asterisk is sending SIP messages but gets no response from the device. Martin On Thu, 3 Jul 2003 vk@akcecc.net wrote:> Hello All! > > There is description of my problem with Asteriks below. > Asteriks CLI says: > "File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call" > > Sip debug on the server gives the next: > > Retransmitting #5 (no NAT): > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.0.26:5060 > From: <sip:8523@192.168.0.24;user=phone>;tag=106403508 > To: <sip:500@192.168.0.24;user=phone>;tag=as0771c6f9 > Call-ID: 3296035458@192.168.0.26 > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Contact: <sip:500@192.168.0.22> > Content-Type: application/sdp > Content-Length: 209 > > 8523 is Cisco ATA-186 > > The sip.conf content: > - - - - - > [cisco8523] > type=friend > username=8523 > secret=test > nat=no > host=dynamic > canreinvite=no > qualify=300 > defaultip=192.168.0.26 > - - - - - > > Why I place a call to Asteriks. I hear some invitation but connection brokes > when retransmit exceed. > > Could anyone give some advice or solution. > Thanks in advance > > -- > Best regards > Vlad > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Thank you for your responce I have also Pingtel SIP phone I have encounter with the same problem on it. (Disconnect after some reinvite messages) But that I put in the Pingtel configuration (SIP_SESSION_REINVITE_TIMER) to 1200 and I have no disconnection after it at all. So it seems that the problem with ATA devices, they are transmitting reinvite messages. So, my question is how could I turn off reinvite messages from ATA client? because directive like: "canreinvite=no" are not doing anything. On Thu, 3 Jul 2003 09:23:17 -0500 (CDT), Martin Pycko wrote> That means that asterisk is sending SIP messages but gets no > response from the device. > > Martin >> On Thu, 3 Jul 2003 vk@akcecc.net wrote: > > > Hello All! > > > > There is description of my problem with Asteriks below. > > Asteriks CLI says: > > "File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call" > > > > Sip debug on the server gives the next: > > > > Retransmitting #5 (no NAT): > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP 192.168.0.26:5060 > > From: <sip:8523@192.168.0.24;user=phone>;tag=106403508 > > To: <sip:500@192.168.0.24;user=phone>;tag=as0771c6f9 > > Call-ID: 3296035458@192.168.0.26 > > CSeq: 1 INVITE > > User-Agent: Asterisk PBX > > Contact: <sip:500@192.168.0.22> > > Content-Type: application/sdp > > Content-Length: 209 > > > > 8523 is Cisco ATA-186 > > > > The sip.conf content: > > - - - - - > > [cisco8523] > > type=friend > > username=8523 > > secret=test > > nat=no > > host=dynamic > > canreinvite=no > > qualify=300 > > defaultip=192.168.0.26 > > - - - - - > > > > Why I place a call to Asteriks. I hear some invitation but connection brokes > > when retransmit exceed. > > > > Could anyone give some advice or solution. > > Thanks in advance > > > > -- > > Best regards > > Vlad > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- Best regards Vlad