Andrew Herdman
2005-May-07 19:32 UTC
[Asterisk-Users] Problem Dialing out via external SIP account.
Hi all, saw a few messages here, and read the part on the wiki on using asterisk to dial out via another SIP service provider, who incidently is also using Asterisk. First the details; PHONE1 Extension: 2002002001 IP Address: 192.168.128.25 ASTERISK1 Extension: 1111111111 IP Address: ASTERISK1 ASTERISK2 IP Address: ASTERISK2 Destination PSTN Extension: 2222222222 (Information changed to hide reality) 222222222 can call 1111111111 and that rings 2002002001 and when I answer, all is well, RTP streams in both directions work fine. 2002002001 can call 2222222222 and 2222222222 rings with CID from 1111111111, so far so good. Picking up the phone, no connection between the phones is made. PHONE1 indicates that the call is attempting to connect, 2222222222 says the call is up. After 60 seconds, 2002002001 gives up and disconnects (or maybe it's ASTERISK1 or ASTERISK2). Then 2222222222 also disconnects at the same time. (Signalling works). Apologies for attaching a zip file, but I included my sip.conf and extensions.conf which are quire small, but I have also included an Ethereal text dump of all the SIP conversation packets which is 132k, and well 8k is better for everyone. Remember folks, don't open a zip without up to date antivirus. (Or no Windows :) Thanks for any help or suggestions to try out. Andrew -------------- next part -------------- A non-text attachment was scrubbed... Name: 2005-05-07_SIP_Dial_001.zip Type: application/octet-stream Size: 8001 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050507/447715a3/2005-05-07_SIP_Dial_001.obj