I wrote a nice detailed post before, and then my mail program lost it for me... so here I go again... I've followed the same process with three different versions of asterisk, my local source copy from about 1 week ago CVS, current CVS from about 24 hours ago, and version 1.0.1, all three versions had identical results: I compiled/installed libpri, zaptel, asterisk I copied config file from my other working asterisk I customised zapata.conf to suit the new hardware TE410p + TDM31B I started asterisk I called an extension which 'does stuff' and then reaches a playback or background. That is where it stops working. asterisk just waits 'forever' until I hangup the channel, and I never hear the audio of the sound file. There is never any actual error message either. I also used the sample asterisk configs, with the same result. I also used the sample asterisk zapata.conf like this: aster0x asterisk # grep -v ^\; zapata.conf|grep -v ^$ [trunkgroups] [channels] immediate=no signalling = fxo_ks context=default channel => 125 I registered my sip phone (polycom ip 600) to this asterisk, and I also get no audio under the same circumstances) However, calling from sip to zap phones, I get two way audio Changing the playback on extension 500 (call digium) to a noop, I can then call digium over IAX and receive audio from their IVR... So, I am at a rather total loss as to why I do not receive audio from playback and background and voicemail on both sip and zap interfaces. Any assistance would be greatly appreciated. Regards, Adam
Hello,
I hope someone can help me with this. I have come across a few other
people who seem to have experienced this problem but the answer was
never posted.
I am trying to listen to voicemail by dialing 9999 for the main voice
mail menu...However I hear nothing. The Asterisk console says:
Executing VoiceMailMain ("SIP/2092-8370", "s2092") in new
stack
WARNING[31691]: file.c;550 ast_readaudio_callback: Failed to write frame
-- Playing 'vm-youhave' (language 'en')
== Spawn extension (test, 9999, 1) exited non-zero on 'SIP/2092-8370'
Can someone please shed some light on this?
Many Thanks,
Aisling.
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Hi. I have a A@H setup at my home. My problems is with phones outside my network. I call the extensions without a problem, it rings but when they answer I can't hear them and they can hear me. I set up in the SIP.CONF nat=yes I'm I missing any other setting or do I need a special switch that support asterisk. Thank you for your help. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
That was a bug fixed in Asterisk version 1.2.3 .... recently version 1.2.6 was released, so don't worry you can try the latest one without timing fears :D Alyed ---------------------------------------- Return-Path: <asterisk-users-bounces@lists.digium.com> Sat Apr 01 15:42:39 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Sat, 1 Apr 2006 15:42:39 -0700 Wierd timing - I'm struggling with exactly the same issue. My problem was with ZAP - ZAP. The phones ring, but no audio. Turns out there's a bug with the version I'm running. It has to do w/ the system date. When I changed my system date to 1-Jan-06, everything worked!! Here's what I found from another posting:>this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349 >change your system date to an older value. everything will work again.I'm hoping the bug is fixed in a more recent release build, but I haven't tried yet. Yours, Hugh On 4/1/06, Luis herrera wrote:> Hi. I have a A@H setup at my home. My problems is with > phones outside my network. I call the extensions > without a problem, it rings but when they answer I > can't hear them and they can hear me. > I set up in the SIP.CONF > nat=yes > > I'm I missing any other setting or do I need a special > switch that support asterisk. > Thank you for your help. > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060401/5f3b2bcc/attachment.htm
Hello,
My DIDs are hitting Directly to Asterisk Machine via SIP G729. From there I
am forwarding call to Cisco 3845 via SIP G729. And from Cisco calls are
terminating to my carriers via H323 G729.
DIDs------------> Asterisk ----------> Cisco 3845 --------> Carrier
Sip G729 Sip G729 H323 G729
Cisco is capable to convert calls from SIP to H323 and H323 to SIP.
My Problem is when a call hits to my Carrier I get no audio at all. Other
side gets ring but upon answering there is no audio.
Below is the output from CLI. I have installed G729 codec in system and its
working fine; there is no Firewall and NAT implementation in my scenario.
Called Cisco3800/99999999999999
Destroying call '0541efff076643ca2cab9c800fe8b2e2@Y.Y.Y.Y'
asterisk1*CLI>
<-- SIP read from X.X.X.X:50974:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK47392650;rport
From: "4169076956" <sip:4169076956@Y.Y.Y.Y>;tag=as2db17260
To: <sip:99999999999999@X.X.X.X>;tag=4B93E20-172
Date: Fri, 28 Jul 2006 20:13:13 GMT
Call-ID: 47c75ec474fe9df50eff04c50a7d34bf@Y.Y.Y.Y
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<-- SIP read from X.X.X.X:50974:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK47392650;rport
From: "4169076956" <sip:4169076956@Y.Y.Y.Y>;tag=as2db17260
To: <sip:99999999999999@X.X.X.X>;tag=4B93E20-172
Date: Fri, 28 Jul 2006 20:13:13 GMT
Call-ID: 47c75ec474fe9df50eff04c50a7d34bf@Y.Y.Y.Y
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:99999999999999@X.X.X.X:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 261
v=0
o=CiscoSystemsSIP-GW-UserAgent 5381 5741 IN IP4 X.X.X.X
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=audio 18068 RTP/AVP 18 101
c=IN IP4 X.X.X.X
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port X.X.X.X:18068
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263
|h263p), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100
(g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
-- SIP/Cisco3800-056c is making progress passing it to SIP/5060-08e53008
After few seconds call gets disconnect.
x.x.x.x Asterisk Box
y.y.y.y Cisco 3845 12.3T
[Cisco3845]
;disallow=all
allow=g729
dtmfmode=auto
host=y.y.y.y
insecure=very
sendrpid=yes
type=friend
Thanks