thr3ads.net - asterisk users - [asterisk-users] Can someone tell me what is this issue ? [Feb 2012]

If this information is useful, please help other people find it:
Share via:

virendra bhati

2012-Feb-03 12:53 UTC

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

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20120203/e5388d1b/attachment.htm>

Sammy Govind

2012-Feb-03 13:07 UTC

head link

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

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 <virbhati at gmail.com>
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 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
>
>-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20120203/c9379c85/attachment.htm>

asterisk users - Feb 2012 - Can someone tell me what is this issue ?

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

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