* You are viewing Posts Tagged ‘rport’

Error SIP/2.0 488 Not acceptable here

Hello,

a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk 1.8.13.0 – NAT – Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my landline phone
number from Sipgate (not [my sip id]@sipgate.de).

My sip.conf including the codec restrictions looks like this (I left out my
local sip account)

[general]
> port=5060
> bindaddr=0.0.0.0
> context=other
> language=de
> allowguest=no
>
> qualify=no
> disallow=all
> allow=alaw
> allow=ulaw
> allow=g729
> allow=gsm
> allow=slinear
> srvlookup=yes
>
> register => : @sipgate.de/
>
>
>
> [sipgate]
> type=friend
> insecure=invite
> nat=yes
> username=

> fromuser=

> fromdomain=sipgate.de
> secret= > host=sipgate.de
> qualify=yes
> canreinvite=no
> dtmfmode=rfc2833
> context = from_external_voip_provider
>

The relevant part from my full asterisk log /var/log/asterisk/full
including the 488 Not acceptable here error message:

[Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:
> < --- SIP read from UDP:217.10.79.9:5060 --->
> INVITE sip:@192.168.5.11:5060 SIP/2.0
> Record-Route:
> Record-Route:
> Record-Route:
> Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0
> Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0
> Via: SIP/2.0/UDP 217.10.79.9:5060
> ;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse
> Via: SIP/2.0/UDP 192.168.0.8:2048
> ;received=;branch=z9hG4bK-un6p0cm50qse;rport=2048
> From: “sipgate.de” @sipgate.de>;tag=8cgn1bajqb
> To: @sipgate.de;user=phone>
> Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> CSeq: 2 INVITE
> Max-Forwards: 67
> Contact:
> @:2048;line=swnt2d3t>;reg-id=1
> X-Serialnumber: 000413251D76
> User-Agent: snom300/8.7.3.7
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 522
> P-Asserted-Identity: @sipgate.de>
>
> v=0
> o=root 269390684 269390684 IN IP4 192.168.0.8
> s=call
> c=IN IP4 217.10.77.20
> t=0 0
> m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 G726-32/8000
> a=rtpmap:108 AAL2-G726-32/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> a=direction:active
> a=nortpproxy:yes
> < ------------->
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: — (25 headers 21 lines) —
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to 217.10.79.9:5060(NAT)
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as basis
> request – 4fdf703d880d-ywqwnfbbj1h7
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found peer ‘sipgate’ for
> ‘‘ from 217.10.79.9:5060
> [Jun 18 20:15:26] VERBOSE[1164] netsock2.c: == Using SIP RTP CoS mark 5
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 9
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 0
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 8
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 3
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 99
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 108
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 18
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format 101
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> G722 for ID 9
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> PCMU for ID 0
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> PCMA for ID 8
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> GSM for ID 3
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> G726-32 for ID 99
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> AAL2-G726-32 for ID 108
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> G729 for ID 18
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description format
> telephone-event for ID 101
> [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP, but
> they responded without it!
> [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:
> < --- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
> SIP/2.0 488 Not acceptable here
> Via: SIP/2.0/UDP 217.10.79.9:5060
> ;branch=z9hG4bK8f5c.48627b3.0;received=217.10.79.9;rport=5060
> Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0
> Via: SIP/2.0/UDP 217.10.79.9:5060
> ;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse
> Via: SIP/2.0/UDP 192.168.0.8:2048
> ;received=;branch=z9hG4bK-un6p0cm50qse;rport=2048
> From: “sipgate.de” @sipgate.de>;tag=8cgn1bajqb
> To: @sipgate.de;user=phone>;tag=as6364b798
> Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> CSeq: 2 INVITE
> Server: Asterisk PBX 1.8.13.0~dfsg-1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Content-Length: 0
>

I am having problems to see to what “488 Not acceptable here” relates to?
What is not acceptable? Is it maybe about

> [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP, but
> they responded without it!

and not a codec problem?

I am not sure if this is relevant and if it really shows the working
codecs, bot for completeness the outputs of “core show codecs” and “core
show translation” follow:

> core show codecs
> Disclaimer: this command is for informational purposes only.
> It does not indicate anything about your configuration.
> INT BINARY HEX TYPE NAME
> DESCRIPTION
>
> ———————————————————————————–
> 1 (1 < < 0) (0x1) audio g723
> (G.723.1)
> 2 (1 < < 1) (0x2) audio gsm
> (GSM)
> 4 (1 < < 2) (0x4) audio ulaw
> (G.711 u-law)
> 8 (1 < < 3) (0x8) audio alaw
> (G.711 A-law)
> 16 (1 < < 4) (0x10) audio g726aal2
> (G.726 AAL2)
> 32 (1 < < 5) (0x20) audio adpcm
> (ADPCM)
> 64 (1 < < 6) (0x40) audio slin (16
> bit Signed Linear PCM)
> 128 (1 < < 7) (0x80) audio lpc10
> (LPC10)
> 256 (1 < < 8) (0x100) audio g729
> (G.729A)
> 512 (1 < < 9) (0x200) audio speex
> (SpeeX)
> 1024 (1 < < 10) (0x400) audio ilbc
> (iLBC)
> 2048 (1 < < 11) (0x800) audio g726
> (G.726 RFC3551)
> 4096 (1 < < 12) (0x1000) audio g722
> (G722)
> 8192 (1 < < 13) (0x2000) audio siren7
> (ITU G.722.1 (Siren7, licensed from Polycom))
> 16384 (1 < < 14) (0x4000) audio siren14
> (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
> 32768 (1 < < 15) (0x8000) audio slin16 (16
> bit Signed Linear PCM (16kHz))
> 65536 (1 < < 16) (0x10000) image jpeg
> (JPEG image)
> 131072 (1 < < 17) (0x20000) image png
> (PNG image)
> 262144 (1 < < 18) (0x40000) video h261
> (H.261 Video)
> 524288 (1 < < 19) (0x80000) video h263
> (H.263 Video)
> 1048576 (1 < < 20) (0x100000) video h263p
> (H.263+ Video)
> 2097152 (1 < < 21) (0x200000) video h264
> (H.264 Video)
> 4194304 (1 < < 22) (0x400000) video mpeg4
> (MPEG4 Video)
> 8388608 (1 < < 23) (0x800000) video unknown
> (unknown)
> 16777216 (1 < < 24) (0x1000000) video unknown
> (unknown)
> 33554432 (1 < < 25) (0x2000000) text unknown
> (unknown)
> 67108864 (1 < < 26) (0x4000000) text red
> (T.140 Realtime Text with redundancy)
> 134217728 (1 < < 27) (0x8000000) text t140
> (Passthrough T.140 Realtime Text)
> 268435456 (1 < < 28) (0x10000000) text unknown
> (unknown)
> 536870912 (1 < < 29) (0x20000000) text unknown
> (unknown)
> 1073741824 (1 < < 30) (0x40000000) (unk) unknown
> (unknown)
> 2147483648 (1 < < 31) (0x80000000) (unk) unknown
> (unknown)
> 4294967296 (1 < < 32) (0x100000000) audio g719
> (ITU G.719)
> 8589934592 (1 < < 33) (0x200000000) audio speex16
> (SpeeX 16khz)
> 17179869184 (1 < < 34) (0x400000000) audio unknown
> (unknown)
> 34359738368 (1 < < 35) (0x800000000) audio unknown
> (unknown)
> 68719476736 (1 < < 36) (0x1000000000) audio unknown
> (unknown)
> 137438953472 (1 < < 37) (0x2000000000) audio unknown
> (unknown)
> 274877906944 (1 < < 38) (0x4000000000) audio unknown
> (unknown)
> 549755813888 (1 < < 39) (0x8000000000) audio unknown
> (unknown)
> 1099511627776 (1 < < 40) (0x10000000000) audio unknown
> (unknown)
> 2199023255552 (1 < < 41) (0x20000000000) audio unknown
> (unknown)
> 4398046511104 (1 < < 42) (0x40000000000) audio unknown
> (unknown)
> 8796093022208 (1 < < 43) (0x80000000000) audio unknown
> (unknown)
> 17592186044416 (1 < < 44) (0x100000000000) audio unknown
> (unknown)
> 35184372088832 (1 < < 45) (0x200000000000) audio unknown
> (unknown)
> 70368744177664 (1 < < 46) (0x400000000000) audio unknown
> (unknown)
> 140737488355328 (1 < < 47) (0x800000000000) audio testlaw
> (G.711 test-law)
> 281474976710656 (1 < < 48) (0x1000000000000) video unknown
> (unknown)
> 562949953421312 (1 < < 49) (0x2000000000000) video unknown
> (unknown)
> 1125899906842624 (1 < < 50) (0x4000000000000) video unknown
> (unknown)
> 2251799813685248 (1 < < 51) (0x8000000000000) video unknown
> (unknown)
> 4503599627370496 (1 < < 52) (0x10000000000000) video unknown
> (unknown)
> 9007199254740992 (1 < < 53) (0x20000000000000) video unknown
> (unknown)
> 18014398509481984 (1 < < 54) (0x40000000000000) video unknown
> (unknown)
> 36028797018963968 (1 < < 55) (0x80000000000000) video unknown
> (unknown)
> 72057594037927936 (1 < < 56) (0x100000000000000) video unknown
> (unknown)
> 144115188075855872 (1 < < 57) (0x200000000000000) video unknown
> (unknown)
> 288230376151711744 (1 < < 58) (0x400000000000000) video unknown
> (unknown)
> 576460752303423488 (1 < < 59) (0x800000000000000) video unknown
> (unknown)
> 1152921504606846976 (1 < < 60) (0x1000000000000000) video unknown
> (unknown)
> 2305843009213693952 (1 < < 61) (0x2000000000000000) video unknown
> (unknown)
> 4611686018427387904 (1 < < 62) (0x4000000000000000) video unknown
> (unknown)
>

> core show translation
> Translation times between formats (in microseconds) for one
> second of data
> Source Format (Rows) Destination Format (Columns)
>
> g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
> speex ilbc g726 g722 siren7 siren14 slin16 g719 speex16 testlaw
> g723 – – – – – – – -
> – – – – – – – – – – -
> gsm – – 2 2 10001 2 1 20001 -
> 90001 – 10001 2 – – 70001 – 130001 2
> ulaw – 10001 – 1 10001 2 1 20001 -
> 90001 – 10001 2 – – 70001 – 130001 2
> alaw – 10001 1 – 10001 2 1 20001 -
> 90001 – 10001 2 – – 70001 – 130001 2
> g726aal2 – 20000 10001 10001 – 10001 10000 30000 -
> 100000 – 20000 10001 – – 80000 – 140000 10001
> adpcm – 10001 2 2 10001 – 1 20001 -
> 90001 – 10001 2 – – 70001 – 130001 2
> slin – 10000 1 1 10000 1 – 20000 -
> 90000 – 10000 1 – – 70000 – 130000 1
> lpc10 – 20000 10001 10001 20000 10001 10000 – -
> 100000 – 20000 10001 – – 80000 – 140000 10001
> g729 – – – – – – – -
> – – – – – – – – – – -
> speex – 20000 10001 10001 20000 10001 10000 30000
> – – – 20000 10001 – – 80000 – 140000 10001
> ilbc – – – – – – – -
> – – – – – – – – – – -
> g726 – 10001 2 2 10001 2 1 20001 -
> 90001 – – 2 – – 70001 – 130001 2
> g722 – 20000 10001 10001 20000 10001 10000 30000 -
> 100000 – 20000 – – – 10000 – 70000 10001
> siren7 – – – – – – – -
> – – – – – – – – – – -
> siren14 – – – – – – – -
> – – – – – – – – – – -
> slin16 – 170000 160001 160001 170000 160001 160000 180000 -
> 250000 – 170000 10000 – – – – 60000 160001
> g719 – – – – – – – -
> – – – – – – – – – – -
> speex16 – 180000 170001 170001 180000 170001 170000 190000 -
> 260000 – 180000 20000 – – 10000 – – 170001
> testlaw – 10001 2 2 10001 2 1 20001 -
> 90001 – 10001 2 – – 70001 – 130001 -
>
>
Thank you very much for any hint on this!

Best regards
Stefan

No progress tones on transferred call

Asterisk 1.4
We are experiencing an issue on transfers where no progress tones are heard
by the caller:
1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF).
1595 answers
2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears
MoH.
3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595
hears no ringing
When xfer is pressed and the extension is dialled:
U 203.89.001.001:5060 -> 121.98.001.001:1034
INVITE sip:1CDF0F4AFFFF@192.168.1.72:5060 SIP/2.0..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e;rport..From: “C Allerid”
;tag=as72616c50..To:
..Contact:
..Call-ID:
59ba10300b9b8cb5684eba2368c90a77@203.89.001.001..CSeq: 102
INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 05 Jun 2012
08:05:02 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO..Supported: replaces..Content-Type:
application/sdp..Content-Length: 262….v=0..o=root 3031 3031 IN IP4
203.89.001.001..s=session..c=IN IP4 203.89.001.001..t=0 0..m=audio 13728
RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:1010-16..a=ptime:20..a=sendrecv..
U 121.98.001.001:1034 -> 203.89.001.001:5060
SIP/2.0 100 Trying..To: ..From: “C
Allerid” ;tag=as72616c50..Call-ID:
59ba10300b9b8cb5684eba2368c90a77@203.89.001.001..CSeq: 102 INVITE..Via:
SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Server:
Cisco/SPA508G-7.4.9c..Content-Length: 0….
U 121.98.001.001:1034 -> 203.89.001.001:5060
SIP/2.0 180 Ringing..To:
;tag=53e23c5265d60f06i0..From: “C
Allerid”
;tag=as72616c50..Call-ID:59ba10300b9b8cb5684eba2368
c90a77@203.89.001.001..CSeq: 102 INVITE..Via: SIP/2.0/UDP
203.89.001.001:5060;branch=z9hG4bK5286810e..Contact: “$USER”
..Server:
Cisco/SPA508G-7.4.9c..Content-Length: 0….
After transfer is pressed the second time there is no further SIP messages
with

Asterisk CLI

Upgrade from version 1.6.24 to 1.8.12 – Retransmission timeout error

Hi list,

we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
version and face the following problem: one of our peer
(voicetrading.com) doesn’t accept our calls anymore, we receive a
timeout error “Packet timed out after 32000ms with no response”.

Switching back to 1.6 make things working again!

In sip.conf we have nat=no, peer conf is:

[myPeerDef]
type=peer
host=111.111.1.111
context=honeypot

insecure=invite

directmedia=no

disallow=all

allow=ulaw,alaw

dtmfmode=inband

We aren’t registered, our IP is authorized by their system.

Debug of sessions (222.222.22.22 is our server 111.111.1.111 is their)

Working one with 1.6:

Audio is at 222.222.22.22 port 26002
Adding codec 0×4 (ulaw) to SDP
Adding codec 0×8 (alaw) to SDP
Reliably Transmitting (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666@111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
Max-Forwards: 70
From: “TOOTAi” ;tag=as52190c5c
To:
Contact:
Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:10:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 284043376 284043376 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26002 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Can’t make Asterisk send authentication to remote peer on INVITE

This is a really simple problem that I just can’t get to work. There
are two Asterisk servers with the following sip user and peer. When a
call is attempted, Asterisk is not sending authentication details in
response to the 401. Note, if the secret is blank on 172.16.0.2 test,
the INVITE succeeds.

on 172.16.0.2:

[test]
type=friend
secret=abcde
host=dynamic
context=demo

on 172.16.0.1 :

[natty]
type=peer
host=172.16.0.2
fromuser=test
secret=abcde

originate SIP/natty/1234568 extension 200
== Using SIP RTP CoS mark 5
Audio is at 172.16.0.1 port 19486
Adding codec 0×2 (gsm) to SDP
Adding codec 0×4 (ulaw) to SDP
Adding codec 0×8 (alaw) to SDP
Adding non-codec 0×1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.0.2:5060:
INVITE sip:1234568@172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
Max-Forwards: 70
From: “asterisk” ;tag=as1689b2b6
To:
Contact:
Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Date: Sat, 14 Apr 2012 09:10:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1594270426 1594270426 IN IP4 172.16.0.1
s=Asterisk PBX 1.6.2.9-2ubuntu2.1
c=IN IP4 172.16.0.1
t=0 0
m=audio 19486 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=ptime:20
a=sendrecv

SIP and NAT best practices in Asterisk

What should I do in order to to be as secure as possible and with “clean” logs?

Well, for an article about Asterisk security best practices, consider reading this article.

About SIP and NAT best practices, in short, the simplest answer is to always use ‘nat=yes’ (or at least ‘nat=force_rport’ in recent versions of Asterisk that support it), until you come across a SIP endpoint that fails to work properly with that setting. If you do come across such an endpoint, try hard to get it to work with that setting; if you can’t, then set ‘nat=no’ for that endpoint, and understand that the endpoint’s name could be discoverable using the attack methods previously disclosed.

If the endpoint’s configuration is suitably locked down (permit/deny, for example) this may not be a concern for you. If it’s not locked down (for example, if it has to register to your Asterisk server from random locations), then the next step would be to seriously consider requesting that the user of that endpoint consider switching to some other SIP endpoint.

Finally, to date, the only endpoints that have been identified that do not work with Asterisk’s ‘rport’ handling forced upon them are Cisco phones. In that case, you could consider using a Digium IP phone in order to take the most out of your Asterisk Installation.

 

(Thanks to Kevin P. Fleming for his elaborated answers on the Asterisk Mailing List)

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 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 0×2 (gsm) to SDP
>
> Adding codec 0×4 (ulaw) to SDP
>
> Adding codec 0×8 (alaw) to SDP
>
> Adding non-codec 0×1 (telephone-event) to SDP
>
> Reliably Transmitting (NAT) to 217.14.138.127:5065:
>
> INVITE sip:00918885268942@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” ;tag=as2f61c90c
>
> To:
>
> Contact:
>
> Call-ID: 3cd12da658b42c10186c01ed3a7d21a7@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@voipon
>
> Retransmitting #1 (NAT) to 217.14.138.154:5060:
>
> INVITE sip:00918885268942@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”
;tag=as2f61c90c
>
> To:
>
> Contact:
>
> Call-ID: 3cd12da658b42c10186c01ed3a7d21a7@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@sip.voipon.co.uk‘ in 32000 ms (Method:
> INVITE)
>
> — SIP/voipon-00000014 is circuit-busy
>
> Scheduling destruction of SIP dialog ‘
> 3cd12da658b42c10186c01ed3a7d21a7@sip.voipon.co.uk‘ 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-: virbhati@gmail.com
> Skype id:- virbhati2
>
>