fre, 15.02.2008 kl. 12.21 -0700, skrev Gerald Cox: > 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=Affo6642P801-62p8g55gnns02>;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
If the phone is NAT you need to enable NAT traversal, either with STUN or other means. If CallWeaver is NAT you need to enable stun in sip.conf. Try again with these settings enabled and see if it helps. _______________________________________________ Callweaver-users mailing list [email protected] http://lists.callweaver.org/mailman/listinfo/callweaver-users
