No progress tones on transferred call

Report
Question

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:

Asterisk Users 3.2 years ago 0 Answers

Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

Report
Question

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…

Asterisk Users 3.2 years ago 7 Answers

No extension found ?

Report
Question

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: < ---…

Asterisk Users 3.3 years ago 5 Answers

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

Report
Question

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

Asterisk Users 3.3 years ago 2 Answers

Can someone tell me what is this issue ?

Report
Question

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 >>
>

Asterisk Users 3.5 years ago 0 Answers

audio , Failing due to no acceptable offer found

Report
Question

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
>…

Asterisk Users 3.6 years ago 1 Answer

Asterisk Rewrites "From" Header

Report
Question

Asterisk cannot act as a proxy, it is a B2BUA. If you want to make its behavior appear to be a proxy, there are a number of things you can do, but it will never just ‘pass along’ headers from an incoming INVITE to an outgoing INVITE. When Asterisk rewrites the from header when CallerID(num-pres)=prohib_screen is set, acts exactly as it should. How else…

Asterisk Tips 3.6 years ago 0 Answers

SIP trunk call initiated as Anonymous@anonymous.invalid

Report
Question

I have a Grandstream HT-502 device connected to my Asterisk PBX. It is
configured not to place anonymous calls, and from my mostly layman
reading of the invitation that the device sends, it should not be
anonymous. However, the Asterisk PBX sends an anonymous invitation to
our SIP trunk provider. Can anyone explain why? The two INVITE packets follow. The devices sends the following INVITE: INVITE sip:2223334444@pbx.xxxxx.com SIP/2.0
Via: SIP/2.0/UDP 192.168.9.197:46538;branch=z9hG4bK526774101;rport
From: "222333555" ;tag=2072922124
To:
Call-ID: 1082640776-46538-3@BJC.BGI.J.BJH
CSeq: 21 INVITE
Contact: "222333555"
Authorization:…

Asterisk Users 3.6 years ago 0 Answers

video mail is not store

Report
Question

Hi, I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based. On the Mercuro setup video mail is stored and retrieved properly while with Android based setup video mail is not stored (audio is through). Both the client use H.264 codec with following sdp information: Android Based Client SDP Parameters v=0
o=- 1325786904 1325786904 IN IP4 172.16.130.47
s=Polycom RealPresence
c=IN IP4 172.16.130.47
b=AS:1920
t=0 0
a=sendrecv
m=audio 3230 RTP/AVP 118 115…

Asterisk Users 3.6 years ago 1 Answer

asterisk 1.8 codec negotiation

Report
Question

Hi. I am using asterisk 1.8 and everything was working fine when I was
at svn 342661. I then upgraded to vrsion 349339 and discovered the
following problem -- one of the end points is a freeswitch box which
offers a number of codecs, including PCMU. However, when I tried to
make a call I got a 488 response and a message "multiple audio streams
not supported" in the log. Is this by design? I found an issue 18859, but that referenced where
the end point offered both regular rtp and srtp. But it…

Asterisk Users 3.6 years ago 4 Answers