Re: [Asterisk-Users] ser+asterisk problem
On 19/05/05, Kamran Ahmad [EMAIL PROTECTED] wrote: hello I am using ser with asterisk asterisk on 5070 (on back end) ser on 5060 (on front end) i am getting all requests at asterisk. i tried by changing asterisk port bindport=5090 but still getting all requests from sjphone at asterisk. can any one tell what is the reason Did you restart Asterisk - that's a complete restart, not just a 'reload' Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk problem
Welesley Sibelson Dias wrote: Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI -- Executing Dial(SIP/16008-3d17, SIP/16007SIP/16006|20|tr) in new stack -- Called 16007 -- Called 16006 -- SIP/16007-8c24 is ringing -- SIP/16007-8c24 answered SIP/16008-3d17 -- Attempting native bridge of SIP/16008-3d17 and SIP/16007-8c24 Mar 30 13:53:11 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Mar 30 13:53:12 WARNING[1125685952]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 8 (Response) =3D=3D Spawn extension (sip, 1000, 1) exited non-zero on 'SIP/16008-3d17' Jadylson da Rocha Passos Bomfim I know of a GrandStream bug which generates a wrong ack to the 200 OK asterisk sends on connecting. SER drops this ack and asterisk drops the call, as it should. This is fixed in latest firmware image. Kind regards, Geert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Asterisk problem
Geert Nijpels wrote: -- Attempting native bridge of SIP/16008-3d17 and SIP/16007-8c24 Mar I know of a GrandStream bug which generates a wrong ack to the 200 OK asterisk sends on connecting. SER drops this ack and asterisk drops the call, as it should. This is fixed in latest firmware image. At a guess it looks like the bridging is happening to a NAT'd SIP connection and doesn't like the non-routable IPs, stick the following line in your sip.conf for the phone notransfer=yes and see if that fixes your problem... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
