[asterisk-users] Getting calee audio in Asterisk (real time)

Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio -> audio in to jack, audio out from jack -> current Asterisk

[asterisk-users] Forwarding inbound mobiles

We have a need for up to a dozen UK mobile numbers to be forwarded to a UK landline. I know that I can just forward them, but was wondering if anyone knew of any deals / contracts with a UK mobile operator that would lessen the cost. At the moment we are looking at going with Vodafone . Thanks J

[asterisk-users] Transfer calls using ##

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 exte

[asterisk-users] Problems with Asterisk 1.6.2.1 working in Realtime with PostgreSQL

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? Tha

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

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 w

Re: [asterisk-users] Productivity Suite on Polycom IP7000

- Original Message - From: "Watkins, Bradley" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, May 04, 2010 4:50 PM Subject: Re: [asterisk-users] Productivity Suite on Polycom IP7000 > > >> -Original Message- >> From: asterisk-users-boun...@lists.di

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

> 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 S

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

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

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

> 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 Asteris

Re: [asterisk-users] Productivity Suite on Polycom IP7000

> -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

[asterisk-users] Productivity Suite on Polycom IP7000

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 i

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

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 Coloc

Re: [asterisk-users] Asterisk 1.6.2.7 Now Available

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 iss

Re: [asterisk-users] working example of t38 fax w/ 1.6.2?

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 mailto:sean

Re: [asterisk-users] sending T.38 fax negotiation problem

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 wro

[asterisk-users] problem with ringinuse=no, queue members receive randomly two calls

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 settin

Re: [asterisk-users] sending T.38 fax negotiation problem

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 t

Re: [asterisk-users] Code in extensions.conf to leave a voice mailin another PBX ?!

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 Discussi

Re: [asterisk-users] client-server encryption

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

[asterisk-users] Asterisk 1.4.31 Now Available

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 w

[asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server - it is robbery!

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-users] Asterisk 1.6.2.7 Now Available

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 possible

[asterisk-users] Asterisk 1.6.1.19 Now Available

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 half

[asterisk-users] Asterisk 1.6.0.27 Now Available

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 half

Re: [asterisk-users] Interesting email project.

- 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 > > . with the time date and CID included in the > > email. I know how to code some b

Re: [asterisk-users] client-server encryption

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 succ

[asterisk-users] client-server encryption

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 w

Re: [asterisk-users] Code in extensions.conf to leave a voice mail in another PBX ?!

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 > Hi Guys, > Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine. > Peder

[asterisk-users] queue members

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

Re: [asterisk-users] Interesting email project.

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 > . 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.

[asterisk-users] Problem with AMI Originate

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

Re: [asterisk-users] Interesting email project.

whats censored UIN? [VoIP] On Tue, May 4, 2010 at 8:00 AM, mike mosier wrote: > wow thanks guys. Ill try it out. > > Respectfully > Michael D Mosier > Ftoc Certified > > On May 4, 2010 1:36 AM, wrote: > > Hello Mike, > > > On 05-04-2010 06:18, mike mosier wrote: > > > When DID 713xxx is di

Re: [asterisk-users] Interesting email project.

wow thanks guys. Ill try it out. Respectfully Michael D Mosier Ftoc Certified On May 4, 2010 1:36 AM, 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 => _713X.,1,System(/web/html/icq.php [Vo

Re: [asterisk-users] sending T.38 fax negotiation problem

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 ab

Re: [asterisk-users] sending T.38 fax negotiation problem

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

Re: [asterisk-users] working example of t38 fax w/ 1.6.2?

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 > Miguel Amez wrote: > > Hi Sean, > > > > Do you know about t38mod

Re: [asterisk-users] sending T.38 fax negotiation problem

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 thi

Re: [asterisk-users] Reading the CDR

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 d

[asterisk-users] Check if extension loaded over AMI

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

[asterisk-users] DSCP QoS value in YeaLink phone settings

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 in

Re: [asterisk-users] Asterisk and Patton

On Fri, 30 Apr 2010 18:52:46 +0200, Philipp von Klitzing 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 trunk and 4 company in our office, I

Re: [asterisk-users] Channel failover

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