Hi Ryan,
Are the SIP and RTP ports are randomly selected or there are specific ports
for these? Unchecking
random port selection option on the device/softphone may help.
--Vivek
On 11/10/07, Ryan Newington [EMAIL PROTECTED] wrote:
Hi Luki,
Thanks for your advice. I've checked the firewall
Hi folks
I've tried asking this in private mail for quite some time, but sadly
got no reply. So I end up needing to raise the question here.
Sangoma's s setup process includes a small patch to Zaptel. I have some
technical reservations with that patch. It seems that under certain
circumstances
Hi Vivek,
The SIP port is set to the standard port 5060. The RTP ports as far as I know
are random ephemeral ports between 63000 and 64000.
I can change the port range on the media server, asterisk and the device, but
neither seems to help.
My diagram below is probably misleading. The RTP
Hello,
My situation is that , I can't make calls with asterisk, but with x-lite
works fine. Asterisk shows , that successfully registers with another SIP
server, asterisk sends invite, gets trying, and after 30 secs asterisk gets
408 Request timeout. And as I said , with x-lite no problems. I
Jonn R Taylor wrote:
There are alot of option for handeling faxes. One is to use iaxmodem
and hylafax. This option works the best.
Completely agree - we've been using such a setup for almost a year now.
/Per Jessen, Zürich
--
http://www.spamchek.com/ - your spam is our business.
On Sat, 10 Nov 2007 23:05:47 -0600, Eric \ManxPower\ Wieling
[EMAIL PROTECTED] wrote:
Why not use ${UNIQUEID}?
It's not listed in ATFT, even 2nd ed, so I didn't know about it.
Seems like ${UNIQUEID} is generated with each new call, and includes
an extension:
-- Executing [EMAIL
Perhaps you could try to change the Agent name in SIP.conf (replace asterisk
by another one, like Xlite).
Be sure that you asterisk server has the same configuration than your Xlite
client , look at default expirey as an example. My SIP provider need some
custom parameters to accept calls.
It didn't help, I've tested it before...
2007/11/11, Yann JOUANIN [EMAIL PROTECTED]:
Perhaps you could try to change the Agent name in SIP.conf (replace
asterisk by another one, like Xlite).
Be sure that you asterisk server has the same configuration than your
Xlite client , look at
hi,
What is the useragent that you have specified in the sip.conf? if
you specified useragent=asterisk change that that to something like
useragent=eyebeam or leave it empty.
On Nov 11, 2007 4:08 PM, Giedrius Augys [EMAIL PROTECTED] wrote:
Hello,
My situation is that , I can't make calls
Hi Ryan,
I was just wondering if they need to be according rtp.conf. ( or you may
need to modify rtp.conf)
Regards,
Vivek
On 11/11/07, Ryan Newington [EMAIL PROTECTED] wrote:
Hi Vivek,
The SIP port is set to the standard port 5060. The RTP ports as far as I
know are random ephemeral
On Sunday 11 November 2007 01:38, Tzafrir Cohen wrote:
On Sat, Nov 10, 2007 at 01:40:23PM -0600, Eric ManxPower Wieling wrote:
Thomas Winter wrote:
Hi,
Iam running debian etch on thecus n2100 (Xscale 80219)
I do not have MoH because standard mpg123 gives only loud noise.
I can
I've have been changed useragent too, but not helped
2007/11/11, Gopal krishnan [EMAIL PROTECTED]:
hi,
What is the useragent that you have specified in the sip.conf? if
you specified useragent=asterisk change that that to something like
useragent=eyebeam or leave it empty.
On Nov 11,
Hello
I would like to know if it is possible to have multiple PBXs implemented
in one Asterisk box.
I have different companies using my Asterisk server (remotely) and I
don't want them to be calling each other.
I want to create different profiles in which my clients can only see its
own PBX.
Tzafrir Cohen wrote:
Hi folks
I've tried asking this in private mail for quite some time, but sadly
got no reply. So I end up needing to raise the question here.
Sangoma's s setup process includes a small patch to Zaptel. I have some
technical reservations with that patch. It seems that
On Nov 11, 2007 12:40 AM, Vincent wrote:
Thanks, but it won't do, as I need to get the exact filename so
I can send an e-mail pointing to the file later in the script :-/
you can generate your own name using a combo of
STRFTIME() CALLERID() CDR() ( and RAND() if you like )
You need to look at the files in /path/to/src/asterisk/doc (or /docs, I
don't recall) there is much information there, including a file named
README.variables (1.2) or channelvariables.txt (1.4).
Vincent wrote:
On Sat, 10 Nov 2007 23:05:47 -0600, Eric \ManxPower\ Wieling
[EMAIL PROTECTED]
I'm using Asterisk 1.4.13, the latest released version. The linux platform
is FC7.
I setup my Asterisk server to use IMAP storage. Dovecot is the IMAP
server. Its storing messages perfectly--no problems.
I should also mention that I'm using MySQL for real-time configuration.
That must be
On Nov 11, 2007 1:07 PM, Steve Totaro wrote:
[...]
Your company (Xorcom) is a direct competitor of Sangoma,
is that correct? I rarely answer questions or give competitors
ideas that may come back and hurt my business.
with all due respect, you are being presumptive projecting
your
(clarification)
On Nov 11, 2007 1:07 PM, Steve Totaro wrote:
[...]
Your company (Xorcom) is a direct competitor of Sangoma,
is that correct? I rarely answer questions or give competitors
ideas that may come back and hurt my business.
with all due respect, you are being presumptive
On Sun, Nov 11, 2007 at 12:07:04PM -0500, Steve Totaro wrote:
Just out of curiosity, I have yet to see any issues with Sangoma cards
and the way they ride on top (and patch) the Zaptel drivers. This
personal dataset is around one hundred productions boxes.
Two questions:
1. Have you
does anyone know how to stop the clicking sound that happens at the end
of a playback() command?
is it something I can do in the recording?
I looked in the 'book' but there was only a 'j' option...
thanks,
daveC
___
--Bandwidth and Colocation
Hi, when you want to post a new message to the list, please don't reply
to an existing message. Start a new message to the list's address.
As for your question: see an inline answer:
On Sun, Nov 11, 2007 at 01:07:16PM -0500, dave cantera wrote:
does anyone know how to stop the clicking sound
On Sunday 11 November 2007 11:07:04 Steve Totaro wrote:
Tzafrir Cohen wrote:
Sangoma's s setup process includes a small patch to Zaptel. I have some
technical reservations with that patch. It seems that under certain
circumstances it may cause unexpected behavior when used with other
I would recommed to convert that to gsm format
http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk
On 11/11/07, Michael Schwartz [EMAIL PROTECTED] wrote:
I'm using Asterisk 1.4.13, the latest released version. The linux platform
is FC7.
I setup my
Hi,
On Nov/09/2007, Tony Plack wrote:
Try setting acpi=off in your boot options for the kernel.
Before read your mail, I did noacpi (I guess that is the same,
/proc/interrupts file changed). But without any luck. I also tried noht
(no hyper threading, I think...)
--
Carles Pina i Estany
Hello,
On Nov/10/2007, Tzafrir Cohen wrote:
On Fri, Nov 09, 2007 at 04:59:37PM -0600, Tony Plack wrote:
The thing is that this works, but
The performance of the box becomes really bad.
It seems that the problem, at least in my case is that the HPET timer from
the cpu does not
On Sun, Nov 11, 2007 at 08:51:40PM +0100, Carles Pina i Estany wrote:
I also tried using bristuff 0.3y, 0.3s, etc. (is it 0.3 bristuff when
Asterisk is 1.2.X?). Always without any result :-(
Latest bristuff for 1.2 is y-k . See http://bristuff.org/ .
However the bristuff Zaptel patch has no
- Original Message -
From: Baji Panchumarti [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 11, 2007 12:43 PM
Subject: Re: [asterisk-users] sangoma zaptel patches
(clarification)
On Nov 11, 2007
I got the cause of the problem. I set canreinvite=yes and the
mentioned error gone.
On Nov 10, 2007 12:27 AM, Steve Davies [EMAIL PROTECTED] wrote:
On 11/9/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Steve Davies wrote:
I would hazard that it is the port number of '0' that is
Hi Vivek,
I'm not sure what you mean, could you explain further?
Regards
Ryan
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vivek Shrivastava
Sent: Monday, 12 November 2007 1:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP
well i think rtp port range is defined in rtp.conf and correct me if i am
wrong, these ports must be opened/forwarded to communicate.
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Let me know if you need more information.
Thanks,
Vivek
On 11/11/07, Ryan
As another suggested the Sangoma cards should work.
However we need someone to write a frontend to Steve Underwood's
wonderful spanDSP library. This will allow us a T38 gateway of sorts
meaning you can connect a Linksys ATA using T.38 and we can say that
(assuming your fax machine strictly
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