[asterisk-users] Can someone tell me what is this issue ?

[virendra bhati]

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 at default:1] Answer("Console/dsp", "") in
new stack

 << Console call has been answered >>

    -- Executing [00918885268942 at default:2] Dial("Console/dsp",
"SIP/00918885268942 at 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:00918885268942 at sip.voipon.co.uk: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:7476849 at sip.voipon.co.uk>;tag=as2f61c90c

To: <sip:00918885268942 at sip.voipon.co.uk:5065;user=phone>

Contact: <sip:7476849 at 10.30.131.136>

Call-ID: 3cd12da658b42c10186c01ed3a7d21a7 at sip.voipon.co.uk

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 at voipon

Retransmitting #1 (NAT) to 217.14.138.154:5060:

INVITE sip:00918885268942 at sip.voipon.co.uk: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:7476849 at sip.voipon.co.uk>;tag=as2f61c90c

To: <sip:00918885268942 at sip.voipon.co.uk:5065;user=phone>

Contact: <sip:7476849 at 10.30.131.136>

Call-ID: 3cd12da658b42c10186c01ed3a7d21a7 at sip.voipon.co.uk

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 '
3cd12da658b42c10186c01ed3a7d21a7 at sip.voipon.co.uk' in 32000 ms (Method:
INVITE)

    -- SIP/voipon-00000014 is circuit-busy

Scheduling destruction of SIP dialog '
3cd12da658b42c10186c01ed3a7d21a7 at sip.voipon.co.uk' in 32000 ms (Method:
INVITE)

  == Everyone is busy/congested at this time (1:0/1/0)

    -- Executing [00918885268942 at default:3] NoOp("Console/dsp",
"**CONGESTION**") in new stack


-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbhati at gmail.com
Skype id:- virbhati2
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