On 1/22/08, Michael J. Liberatore [EMAIL PROTECTED] wrote:
I do have queues set up but I would have to setup queues for all calls
then, even from other inside the office calls. Cause if I disable call
waiting, wouldn't that be the same as saying maximum sip connections to
the phone = 1?
Or
office where there are half of the number of
receptionists that are reeally needed.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Davies
Sent: Monday, January 21, 2008 9:09 AM
To: Asterisk Users Mailing List - Non-Commercial
On 1/22/08, Steve Johnson [EMAIL PROTECTED] wrote:
Hi list,
There are many Polycom experts on this list -- hopefully someone has a
solution.
With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk
causes the Polycom 601 phones to start dumping these messages to the
CLI.
On Jan 22, 2008 12:22 PM, Steve Davies [EMAIL PROTECTED] wrote:
Based on some rapid checks, 7.1.30 firmware behaves in exactly the same way.
Cheers,
Steve
As a follow up, I just spoke with our UK snom distributor, EFL, and
they are discussing this with snom already. It seems that there has
I assume that you've already updated the firmware to 2.1.2 - There
were several problems and crash-bugs with previous firmware versions,
but I have found 2.1.2 to be relatively stable.
Regards,
Steve
___
-- Bandwidth and Colocation Provided by
Hi,
Given that the Digium transcoding card has no external connections
(AFAIK), it strikes me that it would suit a mini-PCI slot very well.
Does such a beast exist, or is it likely to? Am I correct in assuming
that this is a Digium-only product, and there is no OEM equivalent
generic board out
2009/10/16 Ishfaq Malik i...@pack-net.co.uk:
Brent Davidson wrote:
We have several offices running Asterisk version 1.4.20.1, and OSLEC
with Rhino R4FXO-EC and one running a Digium TDM800P card for interface
to analog lines. All offices are running Snom 300 phones. Phones all
have static
2009/10/16 Richard Kenner ken...@gnat.com:
I sent a query about this before, but have some further information and am
hoping somebody has a suggestion as to what to try next to debug this.
I'm using an Asterisk box primarily for MeetMe conferencing. There are
two sources: TDM via two Q.SIG
Hi,
First a confession - The box in question is a 1.2.35 box, so this may
be solved in a newer version as I know the JB code is all hugely
changed, but... It may be worth checking into.
Scenario:
- IAX outbound call from Asterisk, which rings okay.
- Remote end sends ANSWER, which we
On 26 January 2010 04:21, Joel Lansden j...@digitalparadise.net wrote:
Greetings all.
First off, thank you for your time on this. I have spent literally 12 hours
searching every forum and article I can find, and I’m going cross-eyed, so I
need to bother everyone with this.
I am running *
On 16 February 2010 19:51, Danny Dias ing.diasda...@gmail.com wrote:
Hello My friends,
Today my asterisk stop working and i could see the following messags in
/var/log/asterisk/messages at the time that asterisk stop working:
[Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version? Is there a patch?
Thank you!
Doug
According to my experimentation, Polycom VVX1500 phones work with
all versions of Asterisk as far
On 18 February 2010 00:14, Michael Graves mgra...@mstvp.com wrote:
On Wed, 17 Feb 2010 17:12:01 +, Steve Davies wrote:
On 17 February 2010 16:56, asterisk aster...@nbsvoice.com wrote:
Can anyone tell if asterisk and Polycom VVX1500 work with video yet?
If so what version
On 19 February 2010 15:28, Steve Davies davies...@gmail.com wrote:
[snip]
I just upgraded to the new bootblock and 3.2.2 firmware, and these
phones will now talk video to other devices. Nothing in the changelogs
indicates why, but there is a definite jump up from the previous
release
On 22 February 2010 15:59, Will Payne w...@teambadger.co.uk wrote:
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like
On 22 February 2010 16:18, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
looking for your valued input on suitable suggestions for high quality VoIP
DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and
looking to a new manufacturer.
--
Thanks, Phil
We use the snom
Hi,
Can anyone recommend a Premicell device that they have used
successfully with Asterisk.
It would need to connect to an ISDN2e port on the PABX and take 2 SIMs
- It will largely be used for inbound mobile calls from the GSM
network into the Asterisk box. It would be useful if the ISDN2 port
On 28 February 2010 15:28, Gordon Henderson gordon+aster...@drogon.net wrote:
On Sun, 28 Feb 2010, LATEEF, IRFAN (ATTSI) wrote:
Gordon ,
Are you referring to Femto Cells ??
No - devices like the Portech boxes - they take SIM card(s) and present
each one as a SIP interface.
Although I
Hi,
I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap
:) I was wondering if someone could point me at 3 things that I appear
to have lost?
1) ZapEC(off) - Is there an equivalent dialplan command to request no
EC on a channel before dialling in DAHDI?
2) rxfax(file.tiff) - I
Hi,
Can anyone suggest a way of doing the following in Asterisk 1.6.2 - I
do not think it can be done trivially using TRANSFER_CONTEXT.
What I want is for the TRANSFER_CONTEXT for all technologies to be the
same as the initial context defined in the configuration of the device
initiating the
On 15 April 2010 14:11, Jared Smith jsm...@digium.com wrote:
On Thu, 2010-04-15 at 13:59 +0100, Steve Davies wrote:
Let us then assume that the contexts are configured in the config files as:
IAX/1234: context=external
SIP/100: context=default
SIP/101: context=superuser
SIP
Hi,
I am migrating some billing code from 1.2 to 1.6 cdr output. Mostly
this is not too bad, but I have a scenario where some data appears to
be lost
Call from SIP/100 to SIP/200, but the SIP/200 device is programmed to
send a redirect to extension 1234. chan_sip creates a
Local/1...@context
On 16 April 2010 02:53, Jared Smith jsm...@digium.com wrote:
You'll need to play around with variable inheritance to get it set right. If
you define a variable with a single underscore (_TRANSFER_CONTEXT in my
example), it'll get inherited by the next spawned channel, but go no further.
On 15 April 2010 18:11, Steve Davies davies...@gmail.com wrote:
Hi,
I am migrating some billing code from 1.2 to 1.6 cdr output. Mostly
this is not too bad, but I have a scenario where some data appears to
be lost
Call from SIP/100 to SIP/200, but the SIP/200 device is programmed to
send
Hi,
For IAX there is a fairly clear description of the authentication
process for inbound calls. A similar SIP document used to exist on the
voip-info wiki, but since 1.6.2 has a number of changes, I was
wondering how different (if at-all) 1.6 authentication might be in SIP
over 1.2. or 1.4
Hi,
If I am expecting too much here, please just tell me so, but I was
under the impression that this was put into 1.6.x
I have 2 types of SIP devices. For argument's sake, let us say that
one type of device can talk G722 and ALAW, and the other only talks
ALAW. I have directmedia=yes.
Calls
On 26 May 2010 15:59, Mike l...@virtutel.ca wrote:
Hi,
This is a bit off-topic, but still related to telephony. Is there a
barebones TAPI driver that exists that would allow me to call up a command
line with, as parameter, the number to dial.
There is a command-line tool dialer.exe that
On 26 June 2010 22:08, Ryan Wagoner rswago...@gmail.com wrote:
I have Polycom phones that support the g722 codec. Adding allow=g722
to the [general] section of sip.conf works great and I can make calls
between the phones using g722. However Asterisk is negotiating g722
for calls going out my
On 21 July 2010 10:59, MohammedShehzad pmh...@gmail.com wrote:
I have been facing an issue that voice is getting distorted sometimes in
MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f
On 26 July 2010 17:27, Zarko Zivanovic outlaw...@gmail.com wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
No success. Anybody please help!
-Original Message-
From:
Hi,
I had the following odd behaviour in Asterisk 1.2 - We are migrating
to 1.6, and I will re-test ASAP, though it is quite hard to replicate,
but I am curious to know whether it is a known IAX issue in 1.2.
We had 2 users in iax.conf:
[user1]
username=user1
secret=secret1
context=context1
On 28 July 2010 17:32, Tilghman Lesher tles...@digium.com wrote:
On Wednesday 28 July 2010 06:49:01 Steve Davies wrote:
[snip] to avoid repetition below
I don't see a 'type' argument to either of the above, so neither of these
would at all be used. That said, you're assuming that the deny
Hi,
I am happy with the usual GDB backtrace methods and so forth, but have
an issue that I cannot work out how to trace on 1.6.2.10.
If I use either the Bridge() app, or the manager Action: Bridge() in a
certain scenario (Basically to bridge 2 SIP channels, like an attended
transfer, resulting
On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:
Steve Davies schrieb:
I need suggestions please on how to determine where it is locking, and why.
Many thanks,
Steve
hello,
have you allready tried strace ?
you could just easily start asterisk with this command:
strace
On 24 August 2010 08:07, Stefan Schmidt s...@sil.at wrote:
Steve Davies schrieb:
On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:
hello,
have you allready tried strace ?
you could just easily start asterisk with this command:
strace asterisk -
Yes, I tried
On 24 August 2010 14:34, Steve Davies davies...@gmail.com wrote:
On 24 August 2010 08:07, Stefan Schmidt s...@sil.at wrote:
Steve Davies schrieb:
On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:
hello,
have you allready tried strace ?
you could just easily start asterisk
On 25 August 2010 08:22, Matt Riddell li...@venturevoip.com wrote:
On 25/08/10 7:20 PM, Tilghman Lesher wrote:
I really thought that the canary should have sounded if Asterisk got in
a loop - or maybe that only happens with high priority?
The canary only runs in high priority mode, and it's
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro) or
Dial(chan1chan2,,M(macro)) and use the macro to update the call
recording filename. But, the macro runs
On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro
On 11 September 2010 20:36, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
On 09/09/10 17:59, Steve Davies wrote:
On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name
Hi,
We have a user who is putting large call volumes through Asterisk, and
wants to BLF monitor up to 90 extensions. We are struggling to find a
handset that can keep up with Asterisk :)
1) Is there a handset that will do this?
2) Is there a different (standard) way to send BLF and allow
On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings by agent that took the call) here but since our asterisk call
recording is a separate server to the ones dealing with queues I'll be
On 13 September 2010 11:43, Olivier oza_4...@yahoo.fr wrote:
2010/9/13 Steve Davies davies...@gmail.com
[snip]
Our test involves about 10 BLF-NOTIFY messages per second to each
handset with a 5-second pause every 5 seconds. This will either crash
or render unusable all of the following
On 13 September 2010 12:16, Stefan Schmidt s...@sil.at wrote:
Hello,
Am 13.09.10 11:56, schrieb Steve Davies:
Hi,
We have a user who is putting large call volumes through Asterisk, and
wants to BLF monitor up to 90 extensions. We are struggling to find a
handset that can keep up
On 13 September 2010 16:58, Carlos Chavez cur...@telecomabmex.com wrote:
On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote:
On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings
On 13 September 2010 19:12, Cassius Smith cass...@cassius.org wrote:
Steve
I have 64 channels being monitored with an SPA962 with two SPA932
sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very
happy with this. Latest firmware is a must.
HTH
Cassius Smith
Any chance you
Hi,
We have a scenario where we need multiple discrete SIP trunks (peers)
from/to a single endpoint. Because the authentication system starts by
matching IP address, it only ever matches on one of the SIP peer
entries, and ignores the others. This is documented behaviour and as
such is correct. I
On 7 October 2010 10:10, Stefan Schmidt s...@sil.at wrote:
Am 07.10.10 10:52, schrieb Steve Davies:
Hi,
snipped
Hello,
i just want to say something about point 4 which comes to my mind about
security.
4) I am not sure whether it is worth dropping through and testing auth
against other
On 22 October 2010 14:24, Miguel Molina mmol...@millenium.com.co wrote:
I think the OP is asking for the old MoH sound (fpm-world-mix, etc) that
came with asterisk. I wonder why the change from the fpm sounds to the
opsound ones, it was a licensing issue?
I think the original 'fpm' files were
On 18 November 2010 17:43, Mike l...@net-wall.com wrote:
I tried thator I think I did something similar, but that may or may not
apply (depending on my understanding of parking lots)
Here is my relevant contexts. The SIP phones are registered under this
context:
[some_context]
I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
Setting a group requires an argument (group name)
But the syntax is shown as: Syntax: GROUP([category])
The [category] square brackets indicate to me an optional parameter,
which contradicts the error.
Verison 1.6 is
On 24 November 2010 10:12, Steve Davies davies...@gmail.com wrote:
I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
Setting a group requires an argument (group name)
But the syntax is shown as: Syntax: GROUP([category])
The [category] square brackets indicate to me
On 25 November 2010 13:02, bayardo.sanc...@gmail.com wrote:
The proble is dialplan I configure fine
--
Sent from my BlackBerry®
VoIP, Windows/Linux Administration and Network Management
US Numbers: 561-886-0664
Nicaragua Mobile: +505.8488.6876
-Original Message-
From: Stefan
Hi,
Has anybody else noticed that MOH does not play on parked calls in
1.6.2.14? Or is it just my setup? MOH seems to work in every other
respect (Call Held or in-Queue), but when a call is parked, the logs
show MOH being started, but the parked party hears nothing.
The verbose logs show the
On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Davies
Sent: 07
On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote:
On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
Hi,
On one of our asterisk systems that is quite busy, we are seeing the
following from 'netstat -s':
Udp:
17725210 packets received
36547 packets to unknown port received.
44017 packet receive errors
17101174 packets sent
RcvbufErrors: 44017 --- this
When
On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
On one of our asterisk systems that is quite busy, we are seeing the
following from 'netstat -s':
Udp:
17725210 packets received
36547 packets to unknown port received.
44017 packet receive errors
17101174
On 10 December 2010 17:21, Shaun Ruffell sruff...@digium.com wrote:
On 12/10/2010 11:02 AM, Steve Davies wrote:
On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
On one of our asterisk systems that is quite busy, we are seeing the
following from 'netstat -s':
Udp
On 10 December 2010 17:33, Steve Davies davies...@gmail.com wrote:
On 10 December 2010 17:21, Shaun Ruffell sruff...@digium.com wrote:
On 12/10/2010 11:02 AM, Steve Davies wrote:
On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
On one of our asterisk systems
On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote:
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either the Asterisk server
On 22 December 2010 12:44, Gilles codecompl...@free.fr wrote:
Hello
I have an Asterisk 1.4 server and two XLite softphones, where
Asterisk and the local XLite phone are located in a LAN behind a NAT
router, and the remote XLite phone is located elsewhere on the Net
behind its own NAT
On 7 December 2010 17:47, Steve Davies davies...@gmail.com wrote:
On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote:
On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message
On 24 December 2010 14:40, Administrator TOOTAI ad...@tootai.net wrote:
Hi,
We had 2 asterisk 1.4 connected together in iax, all was fine. One of them
was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38
When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But
On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote:
Hi Again,
I thought I had this sorted, but it appears that in a clean
environment I did not in fact fix it. There appears to be a bit of a
contradiction.
1) In 1.6.2.x, musiconhold requires DAHDI (which we have)
2) In 1.6.2
On 24 December 2010 15:44, Steve Davies davies...@gmail.com wrote:
On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote:
Hi Again,
I thought I had this sorted, but it appears that in a clean
environment I did not in fact fix it. There appears to be a bit of a
contradiction.
1
On 13 January 2011 16:28, Jonas Kellens jonas.kell...@telenet.be wrote:
I actually found this :
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
But a second question :
how can I know how long a caller stayed inside the queue untill it was
answered by a member ??
The
Hi,
The following IAX config (slightly edited) causes an issue for me in
version 1.6.2.16.1, where my CDR data is unreliable.
[user1]
type=friend
auth=md5
accountcode=user1
notransfer=yes
context=context1
host=10.0.0.250
username=user1
secret=secret1
disallow=all
allow=alaw
[user2]
type=friend
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote:
At 12:12 PM 2/15/2011, you wrote:
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls,
On 16 February 2011 10:13, Peter den Hartog peterdenhar...@gmail.com wrote:
I'm running Asterisk 1.6 and was wondering if anybody have a workig barge
in solution running.
I was thinking of using chanspy, but i would like that the original call
would be dropped, and the new call would be the
*Bump* No takers? Perhaps no-one else thinks this is a bug?
Regards,
Steve
On 7 February 2011 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
The following IAX config (slightly edited) causes an issue for me in
version 1.6.2.16.1, where my CDR data is unreliable.
[user1]
type=friend
On 10 March 2011 11:17, Ishfaq Malik i...@pack-net.co.uk wrote:
Just fixed our problem with
directmedia=no
but this only applies if your extensions are behind a nat
Ish
There are several reasons why directmedia=no might be the correct
configuration.
1) NAT - probably the most common
Hi,
Short version:
Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA
indication into a DAHDI/q.931 ALERTING signal when your ISDN provider
does not pass early media on receipt of an PROGRESS(8) indication?
Long version:
I have an Asterisk 1.6.2.18-rc1 based system with a
On 2 April 2011 09:46, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list,
I often see the following in my message log :
[Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found
[Apr 2 08:15:01]
From my observations, if a video capable device starts the call in
non-video mode, it is never able to add video to the channel? Is this
correct, or am I missing something?
It looks as if the codec 'jointcapability' is calculated at the start
of the call, and can never be added to (with
On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote:
Any ideas on why callers who call into my customer's SIP trunk are not
hearing a ringback tone? I had this on one other asterisk system, and wound
up needing to set progressinband=yes in the SIP trunk config.
I have set
: 505.327.7300
.
-Original Message-
From: Steve Davies [mailto:davies...@gmail.com]
Sent: Thursday, April 07, 2011 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No ringback even though progressinband=yes is
set
On 7 April 2011 17:02
On 15 April 2011 13:02, Vlasis Hatzistavrou vh...@kinetix.gr wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have
Hi,
Can anyone let me know how I can enable video (h.263) on SIP, but if a
video call is passed over IAX, it will remove the video and pass the
audio only.
What I tried was:
SIP - videosupport=yes
- disallow=all
- allow=alaw
- allow=h263
IAX - disallow=all
- allow=alaw
On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Cassius Smith
Sent: Friday, May 06, 2011 11:23 AM
To: Asterisk Users Mailing List -
On 20 May 2011 16:16, Ishfaq Malik i...@pack-net.co.uk wrote:
On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote:
On 11-05-20 09:37 AM, Ishfaq Malik wrote:
Do many people use this?
Is it reliable and safe?
It may still work, but that code is quite old, and I'm not even sure it's
On 24 May 2011 10:43, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
On 05/24/2011 11:02 AM, Steve Davies wrote:
[snip]
I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most
recent version is on Github, and is not that old. In fact that reminds
me that I really must upload
On 1 June 2011 15:10, randall rand...@songshu.org wrote:
On 06/01/2011 03:55 PM, randall wrote:
On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still
Hi,
Since raising this ticket about broken CDR data:
https://issues.asterisk.org/jira/browse/ASTERISK-17826
I have been researching how CDR records work in various circumstances.
CEL will do most things that people want, but that does not change
that CDR records are likely to persist into
On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote:
Interesting thing is when i reload sip.conf i got MWI lamp working on
polycom 501
But its not working when anyone leave voicemail. Do you know its some
timeout or polling setting in sip.conf ?
Still my question is my my
On 9 June 2011 15:49, satish patel satish...@hotmail.com wrote:
Date: Wed, 8 Jun 2011 18:15:14 +0100
From: davies...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote:
Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?
https://sites.google.com/site/webrtc/ilbc-freeware
It seems to require that the Google iLBC licence document is on the
box, but that otherwise it is free-to use by
On 22 June 2011 17:14, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
On 06/22/2011 03:32 PM, Steve Davies wrote:
Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?
https://sites.google.com/site/webrtc
On 22 June 2011 17:09, marvin horst fivehor...@gmail.com wrote:
I want to use extension numbers that begin with the # key in my dialplan,
but I can't get my Aastra phone (6731i) to transmit the # key to asterisk.
It works fine for the * key.
I've tried numerous Local Dial Plan patterns in the
On 9 July 2011 12:34, randulo rand...@randulo.com wrote:
Go ahead and lambast me for this post, it isn't specific to Asterisk, but:
G+ has only been open at all for a week and I already am chatting with
over 200 people who are into VoIP, Asterisk and all the rest of the
stuff we here care
On Saturday, 9 July 2011, Gordon Henderson gordon+aster...@drogon.net wrote:
On Sat, 9 Jul 2011, Steve Davies wrote:
On 9 July 2011 12:34, randulo rand...@randulo.com wrote:
Go ahead and lambast me for this post, it isn't specific to Asterisk, but:
G+ has only been open at all for a week
On 18 July 2011 12:03, Lee Archer lee.arc...@thebigword.com wrote:
Hi, is anyone else having problems with the reload command crashing Asterisk
1.6.2.19? I’ve run a few tests and 1.6.2.18.2 doesn’t have this problem but
1.6.2.19 after a few reloads is just dumping and restarting.
Thanks
On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote:
Hello guys
I need some help to do works FAX using SIP, anybody know the secret to
this? Have asterisk 1.6.
Thanks!!
--
Enviado do meu celular
Eduardo Carpes
E-mail: car...@bsd.com.br
www.freebsd.org
The magic sauce that you
On 18 July 2011 13:00, Lee Archer lee.arc...@thebigword.com wrote:
Hi Steve, I think it's related to my ODBC connection. I started with a
random hang where it looked ODBC related which led me to try a few things.
Reloading the config a few times is causing core dumps which 1.6.2.18.2 just
On 18 July 2011 14:05, Lee Archer lee.arc...@thebigword.com wrote:
Seems to be an already reported problem but since no more fixes for 1.6
it's back to 1.6.2.18.2 for me.
https://issues.asterisk.org/jira/browse/ASTERISK-18103
Regards
Lee
If it is a regression introduced in 1.6.2.19, then
On 14 August 2011 08:36, Eric Wieling ewiel...@nyigc.com wrote:
I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing.
Below is a dialplan snippet and the resulting CLI output. This is running in
an 'h' extension.
Noop(DIALSTATUS=${DIALSTATUS})
Hi,
Is the following possible in some way? I want to have several SIP
providers able to send me calls, each provider may send calls into
many possible DDIs. Each provider has a cluster of servers, but is
unable to authenticate with me, so the following would be a sort of
pseudo-code sip.conf
On 5 October 2011 10:21, Nasir Iqbal na...@ictinnovations.com wrote:
You can do this by an AMI based transfer (Redirect) to Local channel, and
then in local channel's dialplan you need to add your desired custom sip
header followed by a dial command.
Nasir Iqbal
ICT Innovations
On 17 October 2011 11:01, gincantalupo gincantal...@fgasoftware.com wrote:
Hi,
found where the problem is.I tried with a Grandstream phone and it
works!!!
The problem is my (crappy) Snom phone.
I'm investigating the probhope to find the cause asap.
FYI: snom firmware 7.3.30 is
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