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
___
-- Bandwidth and Colocation Provided by
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
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
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
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
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 the
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,
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
thanks for replying,
I'll give it a try
On Sun, May 31, 2009 at 1:24 PM, Rob Hillis r...@hillis.dyndns.org 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
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 on start like
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 sense for
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 can mostly
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 and 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.
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
On Sat, May 30, 2009 at 5:38 PM, Daviramos Roussenq Fortunato
daviramo...@gmail.com 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
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 internal
On Sun, May 31, 2009 at 3:51 PM, sean darcy seandar...@gmail.com 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
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 dbackeb...@gmail.com
On Sat, May 30, 2009 at 5:38 PM, Daviramos Roussenq Fortunato
daviramo...@gmail.com wrote:
I'm
On Sun, May 31, 2009 at 4:20 PM, David Backeberg dbackeb...@gmail.com wrote:
On Sun, May 31, 2009 at 3:51 PM, sean darcy seandar...@gmail.com 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
On Sun, May 31, 2009 at 4:40 PM, David Backeberg dbackeb...@gmail.com 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
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
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
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 tareksa...@hotmail.com wrote:
accountcode is a setting you add to your SIP
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 crash
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:
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 Asterisk
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
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