On Tue, 2009-10-06 at 22:11 -0700, Kirill 'Big K' Katsnelson wrote:
On 091001 0406, Mindaugas Kezys wrote:
We had many problems with IAX2, changing to SIP solved them all.
Let me paste link to wise-words which clearly illustrates our experience:
Hi,
Dial(ALSA/hw:0,0);
- false syntax
Dial(ALSA/hw:0,0);
-
WARNING[1292]: channel.c:3914 ast_request: No channel type registered for
'ALSA'
WARNING[1292]: app_dial.c:1528 dial_exec_full: Unable to create channel of
type 'ALSA' (cause 66 - Channel not implemented)
Do you have another way to
- Trevor Peirce tpei...@digitalcon.ca wrote:
| --[ UxBoD ]-- wrote:
| Please how do I stop the following ???
|
| Asterisk ended with exit status 127
| Asterisk died with code 127.
| Automatically restarting Asterisk.
| mpg123: no process killed
|
|
| You figure out why asterisk is
Hi
Could you possibly have your providers set up in your sip.conf then
handle your outgoing calls via an AGI where you check the number of
channels in use (core show channels) and then route the call through one
of the providers depending on how many channels are in use?
Ish
Eric Chamberlain
B.Masoud @ SH wrote:
China too wide, but regardless! How is asterisk take care such situation?
-regardless- I think I pointed it out already, please do some research,
simply googling for asterisk + reinvite leads you here:
http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite
--
Iván
hin lee wrote:
Doug,
I have tested both ends and got the same results. I was able to using
FOP to drag to Conferences, just not the Parking Lot. Another strange
thing I found is this:
I'm not familiar with how Freepbx handles things, and I've personally
never used FOP for parking or on
What you are looking for is probably a HOBIC (Hotel Billing Information
Center) interface for asterisk, there was a similar thread in this list
five years ago, I came across this some time ago and I did not find a
working implementation.
The links are dead, but you may found it of interest:
Hi
I have a site that has asterisk install with a tdm410 one port is
connected to a pstn that is used as a backup outbound line when/if the
internet/voip is unavailable.
Currently my dial plan for this line is to ignore it, I just basically
do a
s,1,noop
s,n,wait (60)
s,n,hangup
what I would
Thanks,
What if I want to group a TDM2400 into 3 groups, r0/0 to r0/7 , r1/8 to
r1/15 , r2/16 to r2/23
How to do that?
Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: Monday,
On Wed, Oct 07, 2009 at 01:24:01PM +0300, B.Masoud @ SH wrote:
Thanks,
What if I want to group a TDM2400 into 3 groups, r0/0 to r0/7 , r1/8 to
r1/15 , r2/16 to r2/23
How to do that?
DAHDI channel numbers are 1-based and not 0-based. HEnce you should
probably use 1-8, 9-16 and 17-24
group =
2009/10/7 Leif Madsen leif.mad...@asteriskdocs.org
Please test some of the newly released RCs, such as 1.6.1.7-rc2 or
1.6.2.0-rc3
(depending on the branch you're using).
Leif.
Hi Leif,
We've just followed your advice and installed 1.6.1.7-rc2.
Before this, we had lines (3 times in 4 hours
I would do something like this:
s/041100,1,answer()
s,1,noop
s/041100,2,background(message)
s/041100,3,wait(10,m)
s/041100,4,playback(vm-goodbye)
s/041100,5,hangup
1/041100,1,goto(specialcontext,s,1)s,n,wait(60)
s,n,hangup
This would play message to let you know you are
A number of our clients has such issues. What we suggest for escalation is
to do a blind transfer to a second-level queue, so that the logging is
correct and even if second-line support cannot handle the call immediately,
you get the functionality and the logging.
Just my two euro cents,
l.
any interest in it?
I'm evauating to add this feature but before to do that i'd like to
know if there is some other approach that can avoid some developement.
Regards
On Wed, Sep 30, 2009 at 12:48 PM, nik600 nik...@gmail.com wrote:
Dear all
is it possible to handle a queue using a programmed
Hi, I realise this is probably the wrong list for such a question, but I
need a pointer to somewhere I can get some feedback on experience of
(business class) voip providers for the UK?
Situation is that we are currently with Gradwell and use them for an
inbound/outbound single line for a
I have a block of DID's that I ported to Vitelity about 7 days ago. The
problem is if a POTS caller dials into the system, his dtmf is not heard
at READ() or Background() while a prompt is played. After the prompt is
finished, then dtmf is heard. I've been working with their support, but
it
As a follow up and unfortunately, I must say now that upgrading to
1.6.1.7-rc2 didn't help.
We downgraded to a 1.6.1.0 with which we never met the problem we're facing
now.
___
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Use a GoToIf based on the callerid number.
- Original Message -
From: Alex Samad a...@samad.com.au
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 07, 2009 12:21
Subject: [asterisk-users] system cmd + fax line
Hi
I have a site that has asterisk install with a tdm410 one
- Original Message -
From: Trevor Peirce tpei...@digitalcon.ca
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 06, 2009 23:14
Subject: Re: [asterisk-users] MPG123 Dying
--[ UxBoD ]-- wrote:
Please how do I stop the
If you use Iphones to call in, the regular MOH application renders
sporatically. MPG123 allows you to increase/decrease the volume.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: Wednesday,
- Original Message -
From: Barton Fisher bpvoip...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 07, 2009 18:25
Subject: [asterisk-users] DTMF Issues
I have a block of DID's that I ported to Vitelity
- Original Message -
From: Ed W li...@wildgooses.com
To: Asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Wednesday, October 07, 2009 17:22
Subject: [asterisk-users] Need provider recommendations for the UK
Hi, I realise this is probably the wrong list for such a
Hello
We have launched another Gadget Option (Call Extension Button) to use with
Asterisk:
For instance:
extensions.conf:
[callButtonRoute]
exten = 444,1,Dial(SIP/18874...@10.0.0.1,90,t)
exten = 444,2,Hangup
sip.conf:
[callButton]
type=peer
username=callButton
secret=1qa2ws3ed
Looking for Genuine VPS Server for 250 ports on Rent.
Anyone can help ?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users
AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI
(single span). I'm sure I just have something goofed up in the
dialplans? I have a bunch of Polycom 331 IP phones connecting to the
server. I can dial the other extensions in the system fine and I can
dial long distance
Ben Schorr wrote:
AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI
(single span). I’m sure I just have something goofed up in the
dialplans? I have a bunch of Polycom 331 IP phones connecting to the
server. I can dial the other extensions in the system fine and I can
Perhaps send it as 10 digits or 1+?
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Wednesday, October 07, 2009 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Can dial
Thanks Cary,
If I send as 10 digits (dialing the local area code) I get an error from
the phone co that I've dialed an incorrect number. Unfortunately I live
in Hawaii where the 808 area code covers the whole state but is treated
differently on neighbor-islands.
So if I dial 8085551212 I
No, unfortunately not, our local area code is split among multiple
islands. You have to dial it to reach a neighbor island. If you dial it
with a local number it tries to find that number on a neighbor island
(then usually fails).
I'm trying to find out from them if they want us to dial
Please follow up on the issue tracker at http://issues.asterisk.org
Thanks!
Leif.
Olivier wrote:
As a follow up and unfortunately, I must say now that upgrading to
1.6.1.7-rc2 didn't help.
We downgraded to a 1.6.1.0 with which we never met the problem we're
facing now.
I have seen this message stopped sounds while I am watching asterisk
debug:
-- Called 9/0532828384
-- Call accepted by 192.168.10.220 (format ulaw)
-- Format for call is ulaw
-- IAX2/9-69 stopped sounds
-- IAX2/9-69 answered SIP/xxx.xxx.xxx.xxx-b7d009a0
What does it
One of our users recently had a powerfail while connected to our meetme
gateway. (Asterisk 1.4.17 on debian 4.0)
Through the course of it, asterisk never hung up. His system came back
up, and started sending ICMP port unreachables, but the stream went on,
flooding him with silence media stream
On 091007 0001, Hans Witvliet wrote:
Obviously, it is an outdated document that should be revised.
It stated:
Now sip supports more than asterisk sip supports. SIP RFC requires tcp
support for example, yet its not in trunk yet.
This feature is already in the 1.6-branch, production. You
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