Asmaa Ahmed
2013-Sep-19 07:54 UTC
[asterisk-users] The call is established but without exchanged voice packets
Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see thischan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration[general]context=internalallowguest=noallowoverlap=nobindport=5060bindaddr=0.0.0.0srvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=<IP>[7001]type=friendhost=dynamicsecret=123context=internal[7002]type=friendhost=dynamicsecret=456context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here!http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130919/69f6dfe1/attachment.htm>
Salman Zafar
2013-Sep-19 08:14 UTC
[asterisk-users] The call is established but without exchanged voice packets
Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed <asabatgirl at hotmail.com>wrote:> Hello, > > I am trying to make my first call on Asterisk to succeed. I have Asterisk > 1.8.10.1 running on Ubuntu machine. > The configuration is quite simple just for my first test, Trying to have a > call between two X-lite sipphone. The subscribers succeeded to register and > the call is established, but still no voice can be heard, and lead the > call to be disconnected after! By checking the logs, I can see this > chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on > transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 > (Critical Response) > > Here's my simple sip configuration > [general] > context=internal > allowguest=no > allowoverlap=no > bindport=5060 > bindaddr=0.0.0.0 > srvlookup=no > disallow=all > allow=ulaw > alwaysauthreject=yes > canreinvite=no > nat=yes > session-timers=refuse > externip=<IP> > > [7001] > type=friend > host=dynamic > secret=123 > context=internal > > [7002] > type=friend > host=dynamic > secret=456 > context=internal > > A snoop capture for my call is uploaded in the following link. I wonder > if there is any missing configuration or plugin need to be set here! > > http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 <http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992> > Thanks. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Regards ************************** Muhammad Salman *************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130919/c03a7f03/attachment.htm>
Matthew J. Roth
2013-Sep-19 15:57 UTC
[asterisk-users] The call is established but without exchanged voice packets
Asmaa Ahmed wrote:> > > I am trying to make my first call on Asterisk to succeed. I have > Asterisk 1.8.10.1 running on Ubuntu machine. > > The configuration is quite simple just for my first test, Trying to > have a call between two X-lite sipphone. The subscribers succeeded > to register and the call is established, but still no voice can be > heard, a nd lead the call to be disconnected after! By checking the > logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission > timeout reached on transmission > Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 > (Critical Response)The SIP trace you provided breaks down as follows: X-Lite Asterisk --------------- ------------------------------- INVITE(No Auth) ---> <--- 401 Unauthorized ACK ---> INVITE(Auth) ---> <--- 100 Trying <--- 200 OK <--- 200 OK (Retransmitted 10 Times) <--- BYE OK ---> This shows that the three-way handshake (INVITE/200 OK/ACK) used to establish SIP sessions is not completed because Asterisk never receives an ACK from X-Lite. After retransmitting the 200 OK 10 times Asterisk gives up and disconnects the call.> Here's my simple sip configuration > [general] > context=internal > allowguest=no > allowoverlap=no > bindport=5060 > bindaddr=0.0.0.0 > srvlookup=no > disallow=all > allow=ulaw > alwaysauthreject=yes > canreinvite=no > nat=yes > session-timers=refuse > externip=<IP>