Not sure if the setting is the same in Callweaver as it is in Asterisk, but in sip.conf you'll either want to put an entry such as: externip 1.2.3.4 or externhost my.host
Right now your sip server is sending out a private address as its interface to send back packets too and you need it to hand off an address that is the public interface of your router. Tom -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gerald Cox Sent: Friday, February 15, 2008 2:21 PM To: Users Mailing List - Non-Commercial Discussion Subject: [Callweaver-users] Problems with callweaver handling VoIP calls I got callweaver setup with a few SIP phones, an FXO phone, and Qwest as a VoIP provider. Direct SIP calls to the callweaver from an external IP works great. Calls from SIP phone to SIP phone in callweaver works great. The t.38 part of callweaver works great. When calling through VoIP, I cannot hear what's being said on the regular phone that dialed our 8xx number that's directed straight to a SIP phone on our callweaver server, but I can hear, on either phone, what is being said on the SIP phone. DTMF info is not passed into callweaver from the regular phone dialed/dialing via VoIP. I experience the exact same problems when placing an outgoing call as I do when getting an incoming call. The only difference is, when getting an incoming call, it drops the call after about 30 seconds. It never drops the call on outgoing calls. Also, on incoming calls, the call doesn't get dropped/closed when the regular phone, that dialed callweaver through VoIP, gets hung up. The only log info that I could think of that might help follows: Pulled from the callweaver CLI when placing an incoming call to our 8xx #: Retransmitting #6 (no NAT) to 65.115.130.151:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 65.115.130.151:5060;branch=z9hG4bK0olmbr107go0ab0ko780.1;received=65.115.130 .151 From: <sip:[EMAIL PROTECTED]:5060;isup-oli=0;affo6642p801=Affo6642P801-62p 8g55gnns02>;tag=SD1kj3e01-gK07346d05 To: <sip:[EMAIL PROTECTED]:5060>;tag=as1c61fbbd Call-ID: SD1kj3e01-988bc3631bb5f1e8413ded042ff878df-v300050 CSeq: 11694 INVITE User-Agent: CallWeaver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 218 v=0 o=root 25882 25882 IN IP4 192.168.1.160 s=session c=IN IP4 192.168.1.160 t=0 0 m=audio 10344 RTP/AVP 0 100 a=rtpmap:0 PCMU/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - I get a, about, 6 or 7 of those messages before it drops the call and spits out these messages: Feb 15 06:14:07 WARNING[3075787664]: chan_sip.c:1554 retrans_pkt: Maximum retries exceeded on transmission SD1kj3e01-988bc3631bb5f1e8413ded042ff878df-v300050 for seqno 11694 (Critical Response) Feb 15 06:14:07 WARNING[3075787664]: chan_sip.c:1576 retrans_pkt: Hanging up call SD1kj3e01-988bc3631bb5f1e8413ded042ff878df-v300050 - no reply to our critical packet. == Spawn extension (menuOptions, s, 13) exited non-zero on 'SIP/5060-09cea5c8' Any ideas or other places to look for problems is much appreciated. _______________________________________________ Callweaver-users mailing list [email protected] http://lists.callweaver.org/mailman/listinfo/callweaver-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.6/1280 - Release Date: 2/15/2008 9:00 AM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.6/1280 - Release Date: 2/15/2008 9:00 AM _______________________________________________ Callweaver-users mailing list [email protected] http://lists.callweaver.org/mailman/listinfo/callweaver-users
