When I place an outbound call via my Cisco Sip devices 7960 and ATA using iconnect or nikotel as my SIP LD provider, the call connects and then disconnects after a few seconds. When the call is placed from an analog extension via the digium tdm40b it works fine. I have looked at the Debug but an unable to interpret the results. Does anyone have any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031203/6ef4e8b8/attachment.htm
Actually an update here.. there is no audio between any of the sip phones -----Original Message----- From: Kevin [mailto:Asterisk@gtcus.com] Sent: Thursday, December 04, 2003 12:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Outbound SIP Call When I place an outbound call via my Cisco Sip devices 7960 and ATA using iconnect or nikotel as my SIP LD provider, the call connects and then disconnects after a few seconds. When the call is placed from an analog extension via the digium tdm40b it works fine. I have looked at the Debug but an unable to interpret the results. Does anyone have any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031203/092ca39b/attachment.htm
Perhaps its a problem with iptables , do I need iptables? -----Original Message----- From: Kevin [mailto:Asterisk@gtcus.com] Sent: Thursday, December 04, 2003 12:16 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Outbound SIP Call Actually an update here.. there is no audio between any of the sip phones -----Original Message----- From: Kevin [mailto:Asterisk@gtcus.com] Sent: Thursday, December 04, 2003 12:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Outbound SIP Call When I place an outbound call via my Cisco Sip devices 7960 and ATA using iconnect or nikotel as my SIP LD provider, the call connects and then disconnects after a few seconds. When the call is placed from an analog extension via the digium tdm40b it works fine. I have looked at the Debug but an unable to interpret the results. Does anyone have any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031203/c18b8133/attachment.htm