Cyprus VoIP wrote:
Thank you for your answer. The 'internal extension' is indeed a T.38
capable device that works perfectly when connected directly to the
Proxy/ITSP.
As you said, the key to debugging/resolving this issue is the logger. I
wasn't aware of this file. this is what I have
;tag=as0cae0b**
see the last part this is what that i want to change here in from it should
be some CLI
thanks
Masood
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Message: 24
Date: Fri, 4 Dec 2009 11:32:59 +0500
From: Masood Ahmed masoo...@gmail.com
Subject: [asterisk-users
Set 'canreinvite=no' on all applicable peers?
I tried with yes and no. No difference. I'm almost certain it's related
to the Keeping RTP active during T.38 session issue.
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2009/11/19 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Nov 19, 2009 at 07:01:13AM +0100, Olivier wrote:
Hi,
I'm using a revision 6822-enabled Dahdi-Tools (see
https://issues.asterisk.org/view.php?id=13897) with a Junghanns QuadBRI.
This patch has now been merged into the trunk of
Hi!
What version of spandsp is recommended to use when u compile
asterisk-trunk?
Best regards
MAGNUS BENNGRD ___
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Cyprus VoIP wrote:
So, I enabled the full logger, and the strange thing I see is this message:
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session
It seems that this might be the reason Asterisk initiates a reINVITE
with voice codecs, after connecting the 2 parties.
Hi,
I'm using revision 6822 of Dahdi Tools.
# dahdi_hardware
pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P
# asterisk -rx dahdi show version
DAHDI Version: 2.2.0.2 Echo Canceller: OSLEC
# cat /etc/dahdi/genconf_parameters
...
pri_termtype
SPAN/1 TE
Olivier schrieb:
2009/12/4 Olivier oza-4...@myamail.com
Has someone successfully used this QUEUE_VARIABLES() function (in
1.6.2-rc7) ?
A previous question about it remainded unanswered (
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466).
On 12/04/2009 06:54 PM, Magnus Benngård wrote:
Hi!
What version of spandsp is recommended to use when u compile
asterisk-trunk?
The next one, or if that hasn't been released yet, the current one.
Steve
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2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
2009/12/4 Olivier oza-4...@myamail.com
Has someone successfully used this QUEUE_VARIABLES() function (in
1.6.2-rc7) ?
A previous question about it remainded unanswered (
I'd like to put a phone in a special context, where a test is made on its
business hours, then if so, proceed to the normal context to do whatever it
does with outgoing and local calls.
I've tried, just to go from one context to the next:
[specialoutgoing]
exten = _X.,1,noop(This is a special
Cyprus VoIP wrote:
So, I enabled the full logger, and the strange thing I see is this message:
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session
It seems that this might be the reason Asterisk initiates a reINVITE
with voice codecs, after connecting the 2 parties.
On Fri, Dec 04, 2009 at 09:58:40PM +0800, Steve Underwood wrote:
On 12/04/2009 06:54 PM, Magnus Benngård wrote:
Hi!
What version of spandsp is recommended to use when u compile
asterisk-trunk?
The next one, or if that hasn't been released yet, the current one.
Specifically?
--
On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote:
Hi,
I'm using revision 6822 of Dahdi Tools.
# dahdi_hardware
pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P
# asterisk -rx dahdi show version
DAHDI Version: 2.2.0.2 Echo Canceller: OSLEC
# cat
Cyprus VoIP wrote:
If it's not related, why does Asterisk send again INVITE messages to
both parties? How can this be prevented? I don't see more debug data
prior to the new INVITE.
It's probably because you are using 1.6.1.9; that release (and older)
had a 'feature' that allowed automatic
Olivier schrieb:
2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
How can can you get current queue's length (ie maxlen) or waiting call
number from dialplan ?
Set(err=${QUEUE_VARIABLES(techsupport)});
Verbose(1,maxlen: ${QUEUEMAX});
2009/12/4 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote:
Hi,
I'm using revision 6822 of Dahdi Tools.
# dahdi_hardware
pci::05:06.0 wcb4xxp+ d161:b410 Digium Wildcard B410P
# asterisk -rx dahdi show version
DAHDI
On Fri, Dec 04, 2009 at 04:06:29PM +0100, Olivier wrote:
2009/12/4 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote:
Hi,
I'm using revision 6822 of Dahdi Tools.
# dahdi_hardware
pci::05:06.0 wcb4xxp+ d161:b410 Digium
Hello again,
Adding more information:
Core show channels:
Channel Location State Application(Data)
DAHDI/4-1s...@national_mobile:1 Rsrvd(None)
DAHDI/1-1s...@national_mobile:1 Rsrvd(None)
Dahdi show channels:
It's probably because you are using 1.6.1.9; that release (and older)
had a 'feature' that allowed automatic switching back to audio from T.38
if one of the endpoints sent an audio packet. It turns out that wasn't a
good idea, and it's been removed... but in later versions. You'll have
to
VoIP Users Conference begins in about 30 minutes to discuss the use of
VoIP on social networks like Facebook. If you have any interest in
this (or maybe you customers do?) please join us
IRC anytime: #vuc on Freenode
SIP see http://vuc.me for all the URI and PSTN numbers
Skype:vuc.me or
Trying to configure IAX for use
I think I have everything set right. But my IAX phone wont connect.
When I run wireshark I'm seeing this
Note if above screenshot from wireshark does not show here is a link for
it: http://img402.imageshack.us/i/tempe.jpg/
I've tried a variety
[r...@voip ~]# asterisk -V
Asterisk 1.6.1.11
When using the above version with IMAP VoiceMail integration when I leave a
message my SNOM360 it shows 2 message waiting; yet when running voicemail show
users from the Asterisk CLI it correctly reports 1.
It would appear that when the VM is
Hi,
Running 1.4.26.1 here. I have installed TE420B card in my server, and
followed the appropriate steps (as far as I know to configure it). This
TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling
type.
When I dial out, I get this message:
Dec 4 11:37:31]
2009/12/4 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Dec 04, 2009 at 04:06:29PM +0100, Olivier wrote:
2009/12/4 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote:
Hi,
I'm using revision 6822 of Dahdi Tools.
#
Forget it, found my issues. I have been looking for hours, but as soon as I
write this I find it. dahdi-channels.conf wasn't included in
chan_dahdi.conf.
That being said, I have other issues now, but at least that one is fixed.
Regards,
Mike
From:
On 4 Dec 2009, at 16:37, James A. Shigley wrote:
egg*CLI iax2 reload
== Parsing '/etc/asterisk/iax.conf': == Found
== Parsing '/etc/asterisk/users.conf': == Found
[Dec 4 10:17:36] NOTICE[6080]: chan_iax2.c:11087 set_config:
Ignoring bindport on reload
[Dec 4 10:17:36]
2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
How can can you get current queue's length (ie maxlen) or waiting call
number from dialplan ?
Set(err=${QUEUE_VARIABLES(techsupport)});
192.168.16.3 is my desk
17.140 is *
192.168.16.0/21 is the subnet (255.255.248.0)
Firewall isn't an issue here, that I can see for sure.
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.51,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Ok, check if it is actually listening using netstat?
Steve
On 4 Dec 2009, at 17:17, James A. Shigley wrote:
192.168.16.3 is my desk
17.140 is *
192.168.16.0/21 is the subnet (255.255.248.0)
Firewall isn't an issue here, that I can see for sure.
James Shigley
Monroe Telephone
Following up on this if I leave a second message then the WMI count goes to 4.
When I check the voicemail directory on the server I see :-
[r...@voip 1001]# ls -lR
.:
total 20
drwxr-xr-x 2 root root 4096 Dec 4 17:49 INBOX
drwxr-xr-x 2 root root 4096 Oct 8 21:02 Old
drwxr-xr-x 2 root root
as soon as I delete the two messages I receive in the console :-
[Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP Warning:
Unknown message data: 1 EXPUNGE
[Dec 4 17:52:43] WARNING[11673]: app_voicemail.c:2358 mm_log: IMAP Warning:
Unknown message data: 1 EXPUNGE
Best
magnus, simple answer: just use the latest version available. and if
something is not working inside the t.30/t.38 protocol, try the latest
spanpshot: http://www.soft-switch.org/downloads/snapshots/spandsp/?C=M;O=Dand
if something i still not working, give a good description how to
reproduce the
Hi,
I am facing terrible issue regarding no audio/voice on both sides. I am
using g729 codec on two machines and carrier also supports g729 codec. I can
see the RTP traffic flowing but there is no audio.
Call is going from Server 1 to Server 2. I can see the established SIP
channels on Server but
Hi,
I'm having alot of trouble understanding how to use dialplans for outgoing
calls on Dahdi.
Context : I have 3 TI spans, so 69 voice channels and three D channels
(24,48,72). This is on a TE420B from Digium, if it matters.
Here are my (apparently simple) questions in no
Can any Digium E1 cards be used for split data/voice use?
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Simple explanation for #1; it's dial tech/port/number. Dahdi/3 would open
DAHDI port 3 for an outgoing call.
For #2, you should be using core show channels instead of dahdi show
channels. Dsc shows the lines that are available to asterisk, csc shows the
ones in use.
_
From:
Thanks a lot. That helped.
As for #2, dahdi show channels still lists channel 71 (in my particular
case) even though it is in use (core show channels shows it being used).
It's just that the extension is empty in dahdi show channels.
i.e.:
Chan Extension Context Language
On my * 1.6.0.13 box I see this:
dahdi show channels
Chan Extension Context Language MOH InterpretBlocked
State
pseudononesaiden default
In Service
1 415111 from-outsideen default
Dear Xavier;
Actually I beleive you put me in the right channel, but for me realm is
something new to be used. I did not try it at all before. I read some about it,
but still I am not familiar with it
If you can help me in the realm, I will appreciate this:
1) What is the relation between the
we have a similar problem. When we try to make two skype-calls at a time,
only one of them has working audio. For this to happen, both calls must be
ringing at the same time. Does anyone know how to fix this?
I have fixed this issue and it will be in the 1.0.7 release which is currently
in
Thank you, at least I am getting the same thing.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Friday, December 04, 2009 16:37
To: Asterisk Users Mailing List - Non-Commercial
Hi,
I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk
1.4 using realtime architecture. Extensions are defined in realtime database
and dial plan is in AEL. I am able to correctly setup hints in the dialplan,
but they don't work. Did some research and found out that
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