* You are viewing Posts Tagged ‘pcmu’

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 0x4 (ulaw) to SDP
Adding codec 0x8 (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

No extension found ?

Hi

I have a small problems with incoming call.

I have a peer actually configured for outcall :

sip.conf:

[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=port,invite
context=incoming

This SIP account work for outgoing call. when i want receive call from
this sipplier, i have a “extension not found”.

In extensions.conf for incoming:

[incoming]
exten => _X.,1,Dial(IAX2/VoIP/${EXTEN},180,rt)

in dialplan show incoming, no problems i see the dialplan.

when i call, i have:

< --- SIP read from UDP://84.xx.xx.72:5060 —>
INVITE sip:331NUMNOFOUND@78.IPOFMYSERVER:5060 SIP/2.0
Record-Route:
Record-Route:
Record-Route:

Record-Route:

Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: “+331MYCLID”
;tag=2RUVP51HBW30000E1D00001u0K4NFQC0QNAN31
To:

Call-ID:
60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Contact:
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Max-Forwards: 67
P-Asserted-Identity:

Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value=”4f924d2c1e20abe1d@172.16.20.119″
X-PSN-Trunk: ME

v=0
o=- 18406958643964291255 1 IN IP4 172.16.21.11
s=session
c=IN IP4 84.xx.xx.34
t=0 0
m=audio 64296 RTP/AVP 8 18 4 0 105 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=fmtp:4 bitrate=6.3
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes

< ------------->

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 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 (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

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 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@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
>
>

audio , Failing due to no acceptable offer found

On 01/28/2012 10:22 AM, Din Assegaf wrote:
> Hi All,
>
> I’m trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
>
> But when making A Call from SIP Client, I got cli Warning … and no
> call has been made.
>
> My Sip Client is using lib java peers client http://peers.sourceforge.net/
> with standard codec PCMU/PCMA
>
> [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp:
> Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8 101
> [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:9029 process_sdp: Failing
> due to no acceptable offer found
>
> the strange thing is when using asterisk 1.6, is normal,
> when using asterisk 1.8.x and using another client like Ekiga is normal too,

The error message is misleading; you are having this problem because the
‘m’ line in the SDP with the ‘audio’ offer has a port number of 0
(zero)., which means it is not an active media stream offer. It does not
make any sense for the SDP in an INVITE for a new call to have an m-line
with a port number of zero.

I’ll improve the error message so that this sort of situation won’t be
as confusing in the future.