Your Server Voipon isn't responding- See if internet is working fine, or your Voipon sip trunk/peer is registered OK?
On Fri, Feb 3, 2012 at 5:53 PM, virendra bhati <[email protected]> wrote: > Call is not routing from server to destination. > > > app8*CLI> console dial 00918885268942 > > [Feb 3 06:01:15] NOTICE[28124]: console_video.c:133 console_video_start: > voice only, console video support not present > > -- Executing [00918885268942@default:1] Answer("Console/dsp", "") in > new stack > > << Console call has been answered >> > > -- Executing [00918885268942@default:2] Dial("Console/dsp", "SIP/ > 00918885268942@voipon") in new stack > > == Using SIP RTP CoS mark 5 > > Audio is at 10.30.131.136 port 12556 > > Adding codec 0x2 (gsm) to SDP > > Adding codec 0x4 (ulaw) to SDP > > Adding codec 0x8 (alaw) to SDP > > Adding non-codec 0x1 (telephone-event) to SDP > > Reliably Transmitting (NAT) to 217.14.138.127:5065: > > INVITE sip:[email protected]:5065;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 10.30.131.136:5060;branch=z9hG4bK5388007b;rport > > Max-Forwards: 70 > > From: "asterisk" <sip:[email protected]>;tag=as2f61c90c > > To: <sip:[email protected]:5065;user=phone> > > Contact: <sip:[email protected]> > > Call-ID: [email protected] > > CSeq: 102 INVITE > > User-Agent: Asterisk PBX 1.6.2.21 > > Date: Fri, 03 Feb 2012 06:01:16 GMT > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > > Supported: replaces, timer > > Content-Type: application/sdp > > Content-Length: 313 > > > > v=0 > > o=root 1850926672 1850926672 IN IP4 10.30.131.136 > > s=Asterisk PBX 1.6.2.21 > > c=IN IP4 10.30.131.136 > > t=0 0 > > m=audio 12556 RTP/AVP 3 0 8 101 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > > > --- > > -- Called 00918885268942@voipon > > Retransmitting #1 (NAT) to 217.14.138.154:5060: > > INVITE sip:[email protected]:5065;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 10.30.131.136:5060;branch=z9hG4bK5388007b;rport > > Max-Forwards: 70 > > From: "asterisk" <sip:[email protected]>;tag=as2f61c90c > > To: <sip:[email protected]:5065;user=phone> > > Contact: <sip:[email protected]> > > Call-ID: [email protected] > > CSeq: 102 INVITE > > User-Agent: Asterisk PBX 1.6.2.21 > > Date: Fri, 03 Feb 2012 06:01:16 GMT > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > > Supported: replaces, timer > > Content-Type: application/sdp > > Content-Length: 313 > > > > Scheduling destruction of SIP dialog ' > [email protected]' in 32000 ms (Method: > INVITE) > > -- SIP/voipon-00000014 is circuit-busy > > Scheduling destruction of SIP dialog ' > [email protected]' in 32000 ms (Method: > INVITE) > > == Everyone is busy/congested at this time (1:0/1/0) > > -- Executing [00918885268942@default:3] NoOp("Console/dsp", > "**CONGESTION**") in new stack > > > -- > > Thanks and regards > > Virendra Bhati > +91-8885268942 > Software Engineer > E-mail-: [email protected] > Skype id:- virbhati2 > >
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