Am Freitag, den 01.12.2006, 20:41 + schrieb David Bath:
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF
On Fri, Dec 01, 2006 at 10:55:24AM +0200, Roman Yeryomin wrote:
On Thursday 30 November 2006 21:49, Tzafrir Cohen wrote::
On Thu, Nov 30, 2006 at 07:19:14PM +0200, Roman Yeryomin wrote:
Hello!
I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 --
all give the
Am Mittwoch, den 29.11.2006, 20:48 -0500 schrieb Andrew Joakimsen:
Does anyone know where to source the Siemens Gigaset phones in North
America? I called 1-800-SIEMENS and was told the Gigaset range is no
longer marketed here since a few years ago. How far from being FCC
compliant is the DECT
Hi
On Fri, Dec 01, 2006 at 01:43:20AM -0500, Matthew Rubenstein wrote:
On Thu, 2006-11-30 at 17:56 -0700,
[EMAIL PROTECTED] wrote:
Message: 18
Date: Fri, 1 Dec 2006 00:56:10 +0200
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] zaptel compilation problems with linux
Am Freitag, den 01.12.2006, 17:57 -0500 schrieb Mark Price:
Hi,
On 12/1/06, Mark Price [EMAIL PROTECTED] wrote:
hi,
I am using asterisk 1.2.10.
I am trying to send sip links in asterisk voicemail, so that
users can easily reply to emails.
On Thu, Nov 30, 2006 at 09:10:59PM -0500, Doug Crompton wrote:
I do a ./configure successfully but when I try doing a 'make' I get
error 1 - menuselect
What am I doing wrong?
Please post a complete trace. The real error message should be a bit
above the error message from make. BTW: you
Hi All,
First, Edwin thanks for the suggestion in the previous email about
Regex. This unfortunately did not work... I believe it was correctly
evaluation the true condition (i.e. I got the same behaviour).
Anselm, thanks! This way does do it. I believe you must be correct -
the variables are
Hi all. I have an IAX trunk between 2 Asterisk servers. Everything is
working correctly dialing between the servers as well as through the
PSTN (a T1 connected to one of the servers).
The second Asterisk server routes all calls to the PSTN via the first
server. Calls to local 10-digit, and
Dave Morrow wrote:
My long distance provider requires that a billing code be entered
after dialing a long distance call. From the directly attached
Asterisk server, these calls work when the user enters their PIN after
dialing. From the second server (connected via an IAX trunk), I never
Unfortunately, the codes are private for the individual.
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-3020
Fax: (519) 451-6615
Lead, follow or get out of the way!
This message has originated from
Dave Morrow wrote:
Unfortunately, the codes are private for the individual.
Then I would suggest that you prompt the user for that code, before the
actual dial.
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve
H.interesting thought. Not sure how to do it though...
I found this this morning. I think it might be the answer I seek
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum=2
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
Hi, Anselm
On 12/2/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Freitag, den 01.12.2006, 17:57 -0500 schrieb Mark Price:
Hi,
On 12/1/06, Mark Price [EMAIL PROTECTED] wrote:
hi,
I am using asterisk 1.2.10.
I am trying to send sip links in asterisk
Hi, I've been trying to implement this [1] on 1.2.13 and whilst my twiddlings
seem to work, I just wanted confirmation that I'm not doing something really
stupid which could cause a segfault under certain conditions.
My chan_queue.c addition is this one line:
ast_queue_log(queue,
Hello All,
I'm having problems connecting Asterisk to Telco in Poland (using E1). The
telco guys are saying that the RING message is missing.
How can I make Asterisk to send the RING message? Does anyone have any samples
of zaptel and zapata for Poland?
Best Regards,
Alex
On 09:48, Sat 02 Dec 06, Dave Morrow wrote:
H.interesting thought. Not sure how to do it though...
I found this this morning. I think it might be the answer I seek
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum=2
Probably yeah.
The r option in the
Hi guys,
Here is a bit more detailed information of my problem:
If I connect Asterisk PBX to the Polish telco via E1, I don't get any red
alarms or anything. The line seems to be fine and the inbound calls are also
accepted by the Asterisk. However, whenever I try to make an outbound call, the
JS == Jon Schøpzinsky [EMAIL PROTECTED] writes:
JS I would just guess that the PCI bus would be pretty busy, with 3
JS T1 cards. Couldn't that be a problem? Jon
A T1 is less than 2Mbps. The PCI bus can just about handle 1Gbps
ethernet. That's a LOT of T1's.
/Benny
AMH == Anselm Martin Hoffmeister [EMAIL PROTECTED] writes:
AMH Well, perhaps the IF hinders evaluation from happening? It is
AMH by far not as elegant, but you could try
AMH exten=123456,1,GotoIf($[${REGEX(^0..)} = 1]?2:3)
AMH exten=123456,2,Set(CALLERID(num)=44${CALLERID(num):1})
AMH
Hello All,
I'm having problems connecting Asterisk to Telco in Poland (using E1). The
telco guys are saying that the RING message is missing.
How can I make Asterisk to send the RING message? Does anyone have any
samples of zaptel and zapata for Poland?
Best Regards,
Alex
which telco
If I'm not mistaken (that's how I was told), the inbound calls are managed by
Telekomunikacija Polska, and outbound calls are managed by Profuturo.
- Original Message
From: Bartosz Jozwiak [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Oh... that's an interesting idea Benny. I didn't realize you could use
TO/FROM type syntax in the dialplan...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benny
Amorsen
Sent: 02 December 2006 17:28
To: asterisk-users@lists.digium.com
Subject:
This is what PRI debug says on problematic call:
Protocol Discriminator: Q.931 (8) len=42
Call Ref: len= 2 (reference 3/0x3) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability:
Speech (0)
I've got a PAP2 that I've got working with asterisk. At the moment, its
configured so that when a phone is picked up on it, it connects to Asterisk.
My hope is that I can let Asteirsk handle the entire dialplan, including
dial tone generation. What would my context in extenstions.conf look like
Hi Scott,
I have direct contact with a support person from Grandstream.
I will ask him about that and tell you what did he say as soon as possible.
Please just wait.
Regards
Claudemir
On 11/30/06, Scott Keagy [EMAIL PROTECTED] wrote:
So I've got phones with ancient firmware, and the
Saturday, December 2, 2006, 6:16:25 PM, Benny Amorsen wrote:
JS == Jon Schopzinsky [EMAIL PROTECTED] writes:
JS I would just guess that the PCI bus would be pretty busy, with 3
JS T1 cards. Couldn't that be a problem? Jon
A T1 is less than 2Mbps. The PCI bus can just about handle 1Gbps
We've had a few people complain that the beep before leaving a
voicemail is not loud enough and too short. Does anybody have a
recorded beep that they can share, that is a little louder and a little
longer? We've had this box in production for 2+ years, so I hate to
mess with the gain on the
Hi!
I am having problems with rxfax. When receiving a fax (on a Zap channel from
a te110p), I see on the console:
Dec 2 18:49:22 WARNING[31532]: channel.c:2341 set_format: Unable to find a
codec translation path from unknown to unknown
Dec 2 18:49:22 WARNING[31532]: app_rxfax.c:311
On Sat, 2006-12-02 at 09:53 -0700,
[EMAIL PROTECTED] wrote:
Date: Sat, 2 Dec 2006 11:51:37 +0200
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] zaptel compilation problems with linux
2.6.19
To: Asterisk-Users asterisk-users@lists.digium.com
Message-ID: [EMAIL
On 12/1/06, Tomer Horn [EMAIL PROTECTED] wrote:
Michel R Vaillancourt wrote:
Tomer Horn wrote:
Hello list,
I am curious here if anybody here got an experience connecting Avaya
to Asterisk using H323 / T1. I am completely lack of experience with
Avaya and I wanna know if anybody here has
On 11/30/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Thursday 23 November 2006 11:44, Heidi Mendoza wrote:
We're looking at using 4 or 8 port T1 cards with echo cancellation and are
evaluating brands to go with. We know that Sangoma has excellent solutions
especially when it comes to
you can find an example on the wiki here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+dial
On 12/1/06, Nigel J. Terry [EMAIL PROTECTED] wrote:
I posted this a week ago and have had no response. Can someone tell me if I
am asking a stupid question, i.e. is the answer either obvious or
Hi Nigel,
If I understand your question correctly, you can accomplish what you need
in Trixbox/FreePBX by having your calls answered by a queue. When the
caller is in this queue, he will hear music on hold until the call is
answered by an agent. When the agent answers the call a recorded
I am doing this already. I assume you are using a 'batphone' dialplan on
the pap2 that places calls on asterisk into the 's' extension.
The asterisk feature you want is 'DISA' (Direct Inward System Access - I
think). My sip.conf has the pap2 coming into context 'ata_in', so my
asterisk dialplan
Well, I can't pretend to know how other people use it, but perhaps an example
of how I use it would be helpful.
Most of the sites that I maintain have a pair of boxes that are being
loadbalanced (by UltraMonkey: www.ultramonkey.org), so I have no particular
way of knowing who is registered
James Harper wrote:
I am doing this already. I assume you are using a 'batphone' dialplan on
the pap2 that places calls on asterisk into the 's' extension.
In the telephone industry, called a house phone
If anyone knows how to tell DISA to give a different dialtone then I'd love to
know!
James Harper wrote:
I am doing this already. I assume you are using a 'batphone'
dialplan on
the pap2 that places calls on asterisk into the 's' extension.
In the telephone industry, called a house phone
If anyone knows how to tell DISA to give a different dialtone then
I'd
love to
James Harper wrote:
Yes, sorry, I should have been more specific. Within the same dialplan I'd like
to be able to present different dialtones with DISA. Internaldialton, and
'external' dialtone.
Thanks
James
I suspect that would involve some digging into the source, given that
there is
take a look on Audacity program is opensource and has the option Generate
Beep, then just add some Gain as you want...
On 12/2/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
We've had a few people complain that the beep before leaving a
voicemail is not loud enough and too short. Does
Hello,
Can anyone comment on the success of AMD/NVMachineDetect in an
Australian setting? What kind of hit/miss ratio can we expect on a good
quality g711 IAX tunk?
Does the region even matter? I'm really not sure if these applications
are tailored to a US/UK machines and VM services.
Thanks for the help James! Not long after I sent the email I came across
other instructions for using DISA exactly how you suggested. Works like a
charm!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Harper
Sent: Saturday, December 02, 2006 9:00
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