No progress tones on transferred call

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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 3.1 years ago 0 Answers

Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

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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 ?

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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

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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 ?

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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