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 hints
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-Commer
> 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
i
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
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
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 M
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: aster
Can any Digium E1 cards be used for split data/voice use?
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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 particular
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
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 pr
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 Regards
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 4096
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 Tele
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
Webportal,WebA
2009/12/4 Philipp Kempgen
> Olivier schrieb:
> > 2009/12/4 Philipp Kempgen
> >> 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:
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] NOTI
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: asterisk-users-boun...@lists.d
2009/12/4 Tzafrir Cohen
> On Fri, Dec 04, 2009 at 04:06:29PM +0100, Olivier wrote:
> > 2009/12/4 Tzafrir Cohen
> >
> > > On Fri, Dec 04, 2009 at 02:42:18PM +0100, Olivier wrote:
> > > > Hi,
> > > >
> > > > I'm using revision 6822 of Dahdi Tools.
> > > >
> > > > # dahdi_hardware
> > > > pci::
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] WARNI
[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 tempora
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 o
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 skype:ld.v
> 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
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:
C
On Fri, Dec 04, 2009 at 04:06:29PM +0100, Olivier wrote:
> 2009/12/4 Tzafrir Cohen
>
> > 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 Wil
2009/12/4 Tzafrir Cohen
> 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:
Olivier schrieb:
> 2009/12/4 Philipp Kempgen
>> 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});
>>Verbose(1,waiting ca
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 automati
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 /e
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?
-
> 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
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
2009/12/4 Philipp Kempgen
> Olivier schrieb:
> > 2009/12/4 Olivier
>
> >> 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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Olivier schrieb:
> 2009/12/4 Olivier
>> 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).
http://lists.digium.com/pipermail/asterisk-
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
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 parti
Hi!
What version of spandsp is recommended to use when u compile
asterisk-trunk?
Best regards
MAGNUS BENNGRD ___
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2009/11/19 Tzafrir Cohen
> 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 DAHDI.
>
> >
> >
> 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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5;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57
>> 804d...@ip1..to <804d...@117.20.20.234..to>: ..
>> *
>> *From: "asterisk";tag=as0cae0b**
>>
>> see the last part this is what that i want to change here in from it
>> should
>>
hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57
> 804d...@ip1..to <804d...@117.20.20.234..to>: ..
> *
> *From: "asterisk";tag=as0cae0b**
>
> see the last part this is what that i want to change here in from it should
> be some CLI
>
> thanks
> Masood
> 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
2009/12/4 Olivier
> Hello,
>
> Has someone successfully used this QUEUE_VARIABLES() function (in
> 1.6.2-rc7) ?
> I tried to use it as I'm using SIPPEER() but without success.
>
> A previous question about it remainded unanswered (
> http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/
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