Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID 713xxx is dialed send an email to mmos...@xxx.com. with the
time date and CID included in the email. I know how to code some but am
looking for the best way to do this.
something like this?
exten =
On 4 May 2010, at 03:44, Jack Bates wrote:
We recently got VoIP, so when we make a call, Asterisk should first try
to make the call with VoIP, but in case either our VoIP or our internet
service are down, Asterisk should then try to make the call with our old
school analog phone line
Well,
On Fri, 30 Apr 2010 18:52:46 +0200, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
As I said, you could think about creating 4 different SIP gateways on the
Patton with 4 differing SIP ports. I don't know if the Patton will handle
4 gateways - but it might.
We have 4
Hello list,
I need to set Voice QoS and SIP QoS for YeaLink. The possible values are
0 ~ 63.
With Grandstream I can fill in DiffServ 46, which is EF. That's what I want.
With Snom I fill in 184, which corresponds to EF or DSCP 46 (according
to their wiki)
But what value do I want to fill
Hello list,
I was wondering if there is a way to see if a given piece of dialplan is
loaded through AMI.
I have seen the GetConfig command, but it seems to expect a file name to
retrieve, and I don't necessarily know that (as it could be down the line bu
multiple levels of #includes from the main
Dan Journo wrote:
- you could also consider the M() option to Dial together with the CDR
userfield for logging whatever channel variable make sense to you
I'll see if I can sort it out with that.
- have you looked at the destination channel in the CDR?
The destination
Hi,
I'm experiencing the same problem with t38modem and hylafax.
My problem is that on the re-Invite phase it syncs lower than 2400 bpps and
the connection hangs on the second page.
Could you please post here the patch for asterisk 1.6.2.4 or even indicate
which is the trunk of asterisk where
App_fax? I didn't hear about that. What's that?
Could you please explain that a little bit better?
I'm experiencing some troubles with T38modem and would like to solve on the
better way.
regards,
Miguel Amez
2010/5/4 sean darcy seandar...@gmail.com
Miguel Amez wrote:
Hi Sean,
Do you
On 05/04/2010 06:30 AM, Miguel Amez wrote:
I'm experiencing the same problem with t38modem and hylafax.
My problem is that on the re-Invite phase it syncs lower than 2400 bpps
and the connection hangs on the second page.
The patch I'm talking about won't affect t38modem and Hylafax usage at
Kevin P. Fleming wrote:
On 05/04/2010 06:30 AM, Miguel Amez wrote:
I'm experiencing the same problem with t38modem and hylafax.
My problem is that on the re-Invite phase it syncs lower than 2400 bpps
and the connection hangs on the second page.
The patch I'm talking about won't
wow thanks guys. Ill try it out.
Respectfully
Michael D Mosier
Ftoc Certified
On May 4, 2010 1:36 AM, ad...@3a.hu wrote:
Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID 713xxx is dialed send an email to mmos...@x...
something like this?
exten =
whats censored UIN? [VoIP]
On Tue, May 4, 2010 at 8:00 AM, mike mosier trixbo...@gmail.com wrote:
wow thanks guys. Ill try it out.
Respectfully
Michael D Mosier
Ftoc Certified
On May 4, 2010 1:36 AM, ad...@3a.hu wrote:
Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID
Hello there,
How to retrieve the failure reason when calling AMI Originate with
Async = 0?
The system seems to return the following no matter what:
[Response] = Error
[Message] = Originate failed
How to determine if the number was busy, invalid, etc?
Would I have to run AMI Originate
mike mosier wrote:
Hey all.
My boss asked me to implement the following
When DID 713xxx is dialed send an email to mmos...@xxx.com
mailto:mmos...@xxx.com. with the time date and CID included in the
email. I know how to code some but am looking for the best way to do this.
Sorry I
Hi,
ZAP/DAHDI extension 3210 calls an Asterisk queue 4050 with one SIP agent 4053
added via -QueueAdd(4050, Local/4...@from-internal/n, 1) (not via
agents.conf).
SIP extension 4053 rings, answers and then decides to blind-transfer to
ZAP/DAHDI extension 3666.
The show queue command still
Hi Guys,
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
is anyone familiar with?
2010/4/29 khalid touati khalidtou...@gmail.com
Hi Guys,
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's
Hi,
I'm trying to set up a secure VoIP channel between a Windows softphone client
and an Asterisk 1.6... server running with OpenBSD. By secure I mean to
prevent any man in the middle to reconstitute any vocal exchange nor
sender/addressee/any header data/ of the VoIP call (in first step, I
Hi,
On 05-04-2010 18:46, isca...@free.fr wrote:
- Create a VPN using OpenVPN
= impossible for me , i'm not admin of the Windows system.
this is a bad thing, but the vpn concept might work after all. have you
considered a pptp/l2tp/ipsec vpn? AFAIK on the client side, you may
succeed
- Original Message -
mike mosier wrote:
Hey all.
My boss asked me to implement the following
When DID 713xxx is dialed send an email to mmos...@xxx.com
mailto:mmos...@xxx.com. with the time date and CID included in the
email. I know how to code some but am looking for
The Asterisk Development Team has announced the release of Asterisk 1.6.0.27.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Note that support for the 1.6.0 and 1.6.1 branches are moving to security fixes
only, scheduled for the first
The Asterisk Development Team has announced the release of Asterisk 1.6.1.19.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Note that support for the 1.6.0 and 1.6.1 branches are moving to security fixes
only, scheduled for the first
The Asterisk Development Team has announced the release of Asterisk 1.6.2.7.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.7 resolves several issues reported by the
community, and would have not been
All,
Thanks for the suggestions, but the system is a plan non-sip, non-ip,
non pri setup. It's pretty much a closed box setup.
And the prices for the card and support are robbery - which is why we
aren't going to go with another setup like that. While it has been
reliable - I don't think
The Asterisk Development Team has announced the release of Asterisk 1.4.31.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.31 resolves several issues reported by the
community, and would have not been possible
Iscario-
I'm trying to set up a secure VoIP channel between a Windows softphone
client
and an Asterisk 1.6... server running with OpenBSD. By secure I mean to
prevent any man in the middle to reconstitute any vocal exchange nor
sender/addressee/any header data/ of the VoIP call (in first
See if this helps
http://www.voipuser.org/forum_topic_3921.html
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Tuesday, May 04, 2010 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi Kevin.
That sounds marvellous!
Maybe some of my problems come from that issue, so tomorrow's revision of
1.6.2.7 could have a solution.
Facing another problem, as I told you, I'm experiencing some troubles with
t38modem's configuration, and I would like to know if you had have
experience with
Dear all
on a debian amd64 i've installed (from source) asterisk 1.4.30
On the system we have in average 50 concurrent calls in queue and 40
sip members.
I'm experiencing an apparently random problem:
sometimes some users receive 2 calls from asterisk, apparently
ignoring the ringinuse=no
To make it clear, the change was merged to the 1.6.2 branch recently, and will
not be in 1.6.2.7 as those releases candidates were made a couple of weeks ago.
The changes will be available in the next set of release candidates, slated to
be 1.6.2.8-rc1 sometime this week.
Leif.
Miguel Amez
On 5/4/2010 7:32 AM, Miguel Amez wrote:
App_fax? I didn't hear about that. What's that?
Could you please explain that a little bit better?
I'm experiencing some troubles with T38modem and would like to solve on
the better way.
regards,
Miguel Amez
2010/5/4 sean darcy seandar...@gmail.com
On 5/4/2010 1:59 PM, Asterisk Development Team wrote:
The Asterisk Development Team has announced the release of Asterisk 1.6.2.7.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.7 resolves several issues
sean darcy wrote:
If I'm reading the ChangeLog correctly 1.6.2.7 = 1.6.2.7-rc3. Right?
Correct -- all releases are a direct copy of the last release candidate (in
nearly all cases anyways).
Leif.
--
_
-- Bandwidth and
Has anyone here ever actually truly successfully gotten a Polycom IP7000 to
take a productivity suite license and enabled the bonus features like 4-way
calling, recording etc? It ALWAYS works perfectly with ALL of our
Soundpoint IP 5/6xx phones, but NEVER for our IP7000s.
I just want to know
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Karl Fife
Sent: Tuesday, May 04, 2010 5:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Productivity Suite
The Asterisk Development Team has announced the release of Asterisk 1.6.2.7.
What version of Skype for Asterisk works with this release?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Richard Kenner wrote:
The Asterisk Development Team has announced the release of Asterisk 1.6.2.7.
What version of Skype for Asterisk works with this release?
Should be the latest available on the Digium downloads site. It says version
1.6.2.0 but I've been using Skype for Asterisk on my
Should be the latest available on the Digium downloads site. It says
version 1.6.2.0 but I've been using Skype for Asterisk on my 1.6.2
branch for quite some time (I just updated it last week).
Hmm. So was I until it abruptly stopped working. It started again when
I went back to an older SVN
- Original Message -
From: Watkins, Bradley bradley.watk...@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, May 04, 2010 4:50 PM
Subject: Re: [asterisk-users] Productivity Suite on Polycom IP7000
-Original
Richard Kenner wrote:
Should be the latest available on the Digium downloads site. It says
version 1.6.2.0 but I've been using Skype for Asterisk on my 1.6.2
branch for quite some time (I just updated it last week).
Hmm. So was I until it abruptly stopped working. It started again when
I
Hi Anyone,
I have a server with asterisk 1.6.2.1 working in Realtime with PostgreSQL, but
I'm having problems when happened any error in a table, for example,
automatically this error stop the Asterisk.
Has a way to configure the DB that when happened any problem don't stop the
asterisk?
I have a question about the blind transfer
using ##. This works great on our cordless phone, but there have been
occasions that we can't transfer using ##. I was able to reproduce the
issue by doing the following:
1) Call in from the outside line,
2) Ask the operator to transfer me to an
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