The clue in the log is "no authority found". Something in the
configuration at the other end doesn't match the configuration at this
end - almost certainly the username and password.
Why are you including the IP address when dialling the trunk? If your
peers are set up with IP addresses (which t
Michelle Dupuis escribió:
> You're not alone...we never found the cause of this (rare) occurance...
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
> Sent: Sunday, May 31, 2009 8:58 PM
> To
my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go
to 1.6 voice mail. it says
== Using SIP RTP CoS mark 5
-- Executing [4...@sip:1] Dial("SIP/312-09f9a720",
"IAX2/trun...@147.120.203.98/4567,10,t") in new stack
-- Called trun...@147.120.203.98/4567
[Jun 1 11
Asterisk 1.4.25 does work with Zaptel.
On 05/31/2009 07:46 PM, bilal ghayyad wrote:
> Hi All;
>
> I discovered that Asterisk 1.4.25 does no thave zapata.conf, any advise? Does
> it mean that Asterisk 1.4.25 no more support for zaptel and it works only
> with dahdi? So, what is the latest Asteris
You're not alone...we never found the cause of this (rare) occurance...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Sunday, May 31, 2009 8:58 PM
To: Asterisk Users List
Subject: [asteris
On Sunday 31 May 2009 13:36:26 Philipp Kempgen wrote:
> Tzafrir Cohen schrieb:
> > How useful are the equivalent safe_asterisk scripts?
>
> There's no real reason to treat asterisk differently but then again
> I haven't seen Apache or MySQL crash very often but I did see some
> versions of Asterisk
Thanks. I wonder do I need to reload it if I am using
realtime/database? I have to change the accountcode during the call
so it is not possible to do it if reload is needed.
On Fri, May 29, 2009 at 9:35 PM, Tarek Sawah wrote:
> accountcode is a setting you add to your SIP peer.. so it doesn't r
Hi All;
I was using since one year Asterisk 1.4.19.2 and zaptel 1.4.10.1 and they were
working fine via SIP, IAX and Digium fxo and fxs ports.
Suddenly just before 2 or 3 days, the voice become garbage like robot when I
place a call from the SIP Phone (which is in a country and the Asterisk bo
Hi All;
I discovered that Asterisk 1.4.25 does no thave zapata.conf, any advise? Does
it mean that Asterisk 1.4.25 no more support for zaptel and it works only with
dahdi? So, what is the latest Asterisk version that is working with zaptel?
Regards
Bilal
_
On Sun, May 31, 2009 at 4:40 PM, David Backeberg wrote:
> So for me, first patching, then upgrading when main-lined DAHDI came out.
> Plus upgrading to 1.6.0.1, NOT using talker optimization.
> Plus the other things I mentioned about disabling vad and lengthening
> the interval for looking for tal
On Sun, May 31, 2009 at 4:20 PM, David Backeberg wrote:
> On Sun, May 31, 2009 at 3:51 PM, sean darcy wrote:
>> David Backeberg wrote:
>>>
>>> You don't say the kind of call you're making, but if you're using
>>> MeetMe() I have more advice regarding voice quality with conference
>>> rooms.
>>
>>
ATA A --> Asterisk --> ATA B
The ATA supports T.38 Intelbras, I tested it with other manufacturers of ATA
with T.38 and also had the same problem.
2009/5/31 David Backeberg
> On Sat, May 30, 2009 at 5:38 PM, Daviramos Roussenq Fortunato
> wrote:
> > I'm having problems in tramissão a fax usi
On Sun, May 31, 2009 at 3:51 PM, sean darcy wrote:
> David Backeberg wrote:
>>
>> You don't say the kind of call you're making, but if you're using
>> MeetMe() I have more advice regarding voice quality with conference
>> rooms.
>>
>
> I don't know about the OP, I'd sure appreciate any advice rega
David Backeberg wrote:
>
> You don't say the kind of call you're making, but if you're using
> MeetMe() I have more advice regarding voice quality with conference
> rooms.
>
I don't know about the OP, I'd sure appreciate any advice regarding
voice quality with MeetMe(). When we have 2 -3 inter
On Sat, May 30, 2009 at 5:38 PM, Daviramos Roussenq Fortunato
wrote:
> I'm having problems in tramissão a fax using T.38.
> My scenario is:
> Asterisk 1.6.0.5
> 2 ATA of Intelbras 2210.
> ReceiveFAX in the asterisk.
>
> Unable to fax when it is a ATA to another user on the Asteris
Tzafrir Cohen schrieb:
> How useful are the equivalent safe_asterisk scripts?
To be honest I still haven't decided if asterisk should be auto-
respawned by something like safe_asterisk or inittab[1] or launchd[2]
or upstart[3] or Service Management Facility[4] or whatever the
various launchers a
On Sat, May 30, 2009 at 02:35:43PM -0400, Nathanial A. Byrnes wrote:
> Hello,
>I am working on a trixbox based system with a TDM410P connected to 3
> phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN
> with some polycom and Aastra SIP phones. In general everything works.
jonas kellens wrote:
> On my TDM410P pci-card I have an hardware echo cancellation module
> (Digium VPMADT032 EC Modul).
> I have set 'echocancel=yes' in my chan_dahdi.conf to activate this
> hardware module.
>
> Do I now have 2 echo cancellers that are activated ? A software echo
> canceller a
Tzafrir Cohen schrieb:
> On Sun, May 31, 2009 at 02:47:51PM +0200, Philipp Kempgen wrote:
>> OTOH it might be a nice thing to build this functionality into
>> Asterisk itself which could then even call these scripts on
>> asterisk -rx 'restart now', asterisk -rx 'reload' etc.
>
> For those you ca
On Sun, May 31, 2009 at 02:47:51PM +0200, Philipp Kempgen wrote:
> Tzafrir Cohen schrieb:
> > On Sat, May 23, 2009 at 04:43:52PM +0200, Philipp Kempgen wrote:
> >> Tzafrir Cohen schrieb:
> >> > On Fri, May 22, 2009 at 01:33:59PM +0200, Philipp Kempgen wrote:
> >> >> Does anybody think it would make
Tzafrir Cohen schrieb:
> On Sat, May 23, 2009 at 04:43:52PM +0200, Philipp Kempgen wrote:
>> Tzafrir Cohen schrieb:
>> > On Fri, May 22, 2009 at 01:33:59PM +0200, Philipp Kempgen wrote:
>> >> Does anybody think it would make sense for /etc/init.d/asterisk
>> >> to run /etc/asterisk/startup.d/*.sh o
thanks for replying,
I'll give it a try
On Sun, May 31, 2009 at 1:24 PM, Rob Hillis wrote:
> Sounds like you're looking at the wrong variable. You should be looking
> at CALLERID(num).
>
> peace keeper wrote:
> > Hi there,
> >
> > I am using the Asterisk as the PBX, and need to know the caller
Hi All;
I am looking to start develop an Softphone that has messanger feature (voice
and text, who is online also), anyone can advise for the best link to start
with it, so they have open source for softphone that we can start on it from
there?
Any advise?
Regards
Bilal
Hi All;
I discover that there is a relation between voice quality and the network
settings and configuration on the Asterisk machine.
For example, in the sip.conf, if I set the
localnet=xxx.xxx.xxx.xxx/yyy.yyy.yyy.zzz wrong, then this will effect on the
quality of the voice to certain level,
Sounds like you're looking at the wrong variable. You should be looking
at CALLERID(num).
peace keeper wrote:
> Hi there,
>
> I am using the Asterisk as the PBX, and need to know the caller ID for
> the incoming call,
> but when I show the caller Id, it gives the Zaptel channel that
> recieves
Hi people!
I am looking for a h.323 implementation guide for asterisk. I looked in
the doc folder of the latest asterisk source distribution and I didn't
fund anything acording to this subject.
If you guys could give me any advise, I would thank you.
Tamer
_
Hi there,
I am using the Asterisk as the PBX, and need to know the caller ID for the
incoming call,
but when I show the caller Id, it gives the Zaptel channel that recieves the
inbound calls in the asterisk,
Am I missing some configuration !
what should I do to be able to exteract the real calle
On Sat, 2009-05-30 at 23:15 -0300, Daniel Bareiro wrote:
>
> I was testing calling from my cell phone to an analog telephone and if the
> other person hangs before I do it, I see that in the my cell phone the call
> even continues persisting so that if the person of the other endpoint take the
>
On my TDM410P pci-card I have an hardware echo cancellation module
(Digium VPMADT032 EC Modul).
I have set 'echocancel=yes' in my chan_dahdi.conf to activate this
hardware module.
Do I now have 2 echo cancellers that are activated ? A software echo
canceller and a hardware echo canceller ??
Form
Hello!
My DID provider has multiple IPs addresses that is sends packets from. How
to do associate more that on IP address to a sip device in sip.conf (or any
other ideas)?
Thanks,
Elliot
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