Re: [asterisk-users] Asterisk OUtbound IVR Recording

nO , How to make an out bout call and have a dialplan and record the same . I got it from the VOIP Wiki . Thanks Mahesh From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext Jayson Baker Sent: Sunday, October 10, 2010

Re: [asterisk-users] Asterisk 1.8: Warning messages in CLI while putting a SIP-Call on hold

Am Mittwoch, den 06.10.2010, 15:11 +0200 schrieb Karsten Wemheuer: Hi, while testing current release candidate 1.8.0-rc2 I stumbled on a weird behavior. I did not find any hints in the archives or at the bug tracker. Two SIP-Clients are connected (both on the local net, no NAT). The RTP

[asterisk-users] Quintum Tenor AX and Echo

I have a Quintum AX Tenor gateway sending calls to Asterisk from BT analogue lines connected to FXO. The agents hear an echo on their side but incoming callers hear the conversation fine. I can't seem to find the problem. Anyone seen this issue before? p style=margin: 0; padding: 0;

Re: [asterisk-users] Modifying cid.cid_name in app_parkandannounce.c

Am 10.10.10 15:46, schrieb dotnetdub: Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf,

Re: [asterisk-users] Quintum Tenor AX and Echo

On 10/11/2010 09:07 AM, Antonio Berrios wrote: I have a Quintum AX Tenor gateway sending calls to Asterisk from BT analogue lines connected to FXO. The agents hear an echo on their side but incoming callers hear the conversation fine. I can't seem to find the problem. Anyone seen this

[asterisk-users] Quintum Tenor AX and Echo

Let's try this again. I have a Quintum AX Tenor gateway sending calls to Asterisk from BT analogue lines connected to FXO. The agents hear an echo on their side but incoming callers hear the conversation fine. I can't seem to find the problem. Anyone seen this issue before? p style=margin:

[asterisk-users] About Action Originate

I use the action Originate，i want the called first ringing，the called answer,callee ringing.it can achieve? Best regards, justhinker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] OpenR2

On Mon, Oct 11, 2010 at 01:08:50AM -0300, Flavio Miranda wrote: Hi all, Is it Openr2 supported by asterisk 1.6.2 without pach instalation ? Yes. Provided you have libopenr2. I am a little bit confuse about that. My asterisk 1.6.2 show me the following warning: Unknown signalling method

Re: [asterisk-users] Setting up realtime config.

On Sun, Oct 10, 2010 at 06:29:27PM -0600, Mike Diehl wrote: Asterisk replied: Peer test not found. So it looks like I'm missing something pretty basic. I would suggest to check extconfig.conf. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) --

Re: [asterisk-users] About Action Originate

You need to create a dialplan context to achieve it and then access it using originate. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-11 5:54 AM, 施铁泉 justhin...@gmail.com wrote: I use the action Originate，i want the called first ringing，the called answer,callee ringing.it can achieve?

Re: [asterisk-users] SPA-2102 sending local IP instead of WAN IP in SIP packets

On 10/09/2010 01:34 PM, bruce bruce wrote: And that is exactly what is done on the device: Nat=yes but Asterisk still sees the SIP packet coming in to register with a local IP an so it responds to a local IP which doesn't even exist on the Asterisk network. This is what frustrates me that it's

[asterisk-users] Call Failed Audio

Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika --

[asterisk-users] iax2 users calls limit for outgoing / incoming

Dear All, I want set call limit for IAX2 users at the time incoming and outgoing, Please help me how i can set call limit as asterisk support for SIP users. -- Thanks Regards, M. Asif Raza -- _ --

Re: [asterisk-users] OpenR2

Thanks for while!! I will do that! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Mon, 11 Oct 2010 12:06:04 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] OpenR2 On Mon, Oct 11,

Re: [asterisk-users] Asterisk OUtbound IVR Recording

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jayson Baker Sent: Saturday, October 09, 2010 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk OUtbound IVR Recording

Re: [asterisk-users] iax2 users calls limit for outgoing / incoming

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mian Asif Sent: Monday, October 11, 2010 7:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] iax2 users calls limit for outgoing / incoming Dear All, I want

[asterisk-users] Create channel bank with TDMoE

Hello, I want to create channel bank in this case: channel bank |-| | FXS,FXO-TDMoE--|--Asterisk |-| How can it? --

Re: [asterisk-users] Create channel bank with TDMoE

On Mon, Oct 11, 2010 at 06:18:24PM +0330, Karim Davoodi wrote: Hello, I want to create channel bank in this case: channel bank |-| | FXS,FXO-TDMoE--|--Asterisk |-|

Re: [asterisk-users] Create channel bank with TDMoE

Hi, Here in Brazil we've got a company named CIANET. They do exactly you want. I am engineer and work on it with them. http://www.cianet.ind.br/pt/channel_bank.php Thank you Luis A P Barbosa 2010/10/11 Karim Davoodi karimdavo...@gmail.com Hello, I want to create channel bank in this case:

Re: [asterisk-users] Create channel bank with TDMoE

Karim Davoodi wrote: Hello, I want to create channel bank in this case: channel bank |-| | FXS,FXO-TDMoE--|--Asterisk |-| How can it?

Re: [asterisk-users] Call Failed Audio

Sorry this is a list for the Asterisk GUI Project. I think you may have better luck on the FreePBX list / forums. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux *

Re: [asterisk-users] Setting up realtime config.

On Monday 11 October 2010 4:34:46 am Stefan Tichy wrote: On Sun, Oct 10, 2010 at 06:29:27PM -0600, Mike Diehl wrote: Asterisk replied: Peer test not found. So it looks like I'm missing something pretty basic. I would suggest to check extconfig.conf. That's where the problem was; I

Re: [asterisk-users] Unable to find a codec translation path from ulaw|h261 to slin

Never mind... I mistakenly interpreted codec_a_mu.so as some sort of universal translator between ulaw, alaw, and slin. When I loaded the rest of the modules, it worked like a champ. Mike. On Sunday 10 October 2010 6:57:30 pm Mike Diehl wrote: I'm doing some final check-outs before

Re: [asterisk-users] TDM 400p and Noise on the line

Hi I wonder if anyone has any sugestions I have had a TDM400 for a couple of years, and I have always had problems with noise on the line, so tonight I have been doing some research and have found that if I load the CPU dahdi_test has almost perfect results and no noise dahdi_test

[asterisk-users] don't leave meetme conf if key pressed

Hi @ all, what is the best way to to use features like MeetmeCount without leaving the conference. I use Meetme(,X) and MEETME_EXIT_CONTEXT=context, but the problem is that the caller leave the Conference :( Is it possible to press a key, without this obstacle? Thanx for your answers Daniel

[asterisk-users] Second time Parking issue

Hello All, I have a issue with park and pickup feature. I have asterisk 1.4.35 branch, Here is the scenario for the park and pickup. I have changed parking feature with *72 for my asterisk in features.conf. When i have inbound call it comes to one extension or ring group and then I put that

Re: [asterisk-users] Second time Parking issue

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of max.asterisk Sent: Monday, October 11, 2010 3:21 PM To: Asterisk Developers Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Second

[asterisk-users] MWI Assistance

Hi, I'm struggling to get the MWI set up on a few Polycom phones. The setup is like this. I've got a few phones in the context called [company2_phones] and I've got a few mailboxes in the voicemail context [company2]. Therefore, for each entry in sip.conf (i'm actually using sip realtime if

Re: [asterisk-users] user number in conference

On Tue, 12 Oct 2010, Daniel Knoll wrote: i forgot to ask, how can i get the user number from a caller he is in a conference, i don't find a variable to us this for the current channel. Only the command meetme list roomnr shows the usernumber, but i can't use this output. If you use AMI in

Re: [asterisk-users] MWI Assistance

I'm struggling to get the MWI set up on a few Polycom phones. Sorted. From voip-info. http://www.voip-info.org/wiki/view/Asterisk+RealTime The database peers/users are not kept in memory. These are only loaded when we have a call and then deleted, so there's no support for NAT keep-alives

[asterisk-users] SIP and ANI

Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different