Johannes van Hulst
2004-Aug-30 11:23 UTC
[Asterisk-Users] Redirect SIP calls to the SIP provider sipgate.de
I have an asterisk server and I am trying to set the server up as a redirect server of all my internet SIP phones. My Asterisk server as his own internet IP address. At this moment I can make international calls to a IAX provider but I am now trying to setup a SIP provider as well And I get the following error -- Executing Dial("SIP/t10002-4666", "SIP/0031201234567@sipprov|120") in new stack -- Called 0031201234567@sipprov Aug 30 15:11:23 NOTICE[159484848]: chan_sip.c:6643 handle_response: Failed to authenticate on INVITE to '"Han Xten" <sip:t10001@200.179.001.11>;tag=as6a15a27f' == Spawn extension (homephone, 0031201234567, 1) exited non-zero on 'SIP/t10002-4666' Aug 30 15:11:32 WARNING[159484848]: chan_sip.c:675 retrans_pkt: Maximum retries exceeded on call 537024d80840b3144f54435220e25db8@200.179.001.11 for seqno 104 (Non-critical Request) How can I setup an SIP account, so that a sip phone can connect to the SIP provider (sipgate.de) Without traveling trough my asterisk server. For example I have: Asterisk server ip 200.179.001.11 IP phone 200.188.001.12 And a provider on sipgate.de. Now I would like to use the asterisk server as a registration off call server without handling the SIP packages. Is this possible? Greetings Han -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040830/39756d80/attachment.htm