[asterisk-users] Help_video call not run

- Forwarded Message - From: "Durgesh Mishra" To: "asterisk-users" Sent: Wednesday, December 21, 2011 10:36:06 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Help_video call not run Hi all In sip.conf i take as [general] videosupport=yes  

[asterisk-users] OT - Which switch to play with LLDP-MED

Hi, I would like to play with LLDP-MED in a lab, and specifically, to test phone provisionning and auto-configuration (assign phones to VLANs, ...). Eight 10/100 PoE ports would be enough for me. Which model would you recommend ? Regards. -- _

Re: [asterisk-users] Help_video call not run

Hi what is the format of the file you are trying to play with exact codec info. On Tue, Dec 20, 2011 at 19:17, Durgesh Mishra < durgesh.mis...@rancoretech.com> wrote: > Hi all > > > > In sip.conf > > i take as > > [general] > > videosupport=yes > > > >; then UDPT

[asterisk-users] queue not skipping ringing phone

I have a queue that distributes calls among 3 phones. When a phone is in use (including on hold), queue skips that device and sends the call to the next available one as expected. On the other hand, if a call comes in while one of the phones is ringing, the queue doesn't seem to recognize that

[asterisk-users] Help_video call not run

Hi all In sip.conf i take as [general] videosupport=yes [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=adpcm allow=h263p allow=h264 allow=h263 [pho

Re: [asterisk-users] Use different local IP for each SIP trunk

Externip support per device in sip.conf http://edvina.net/products/edvx/ 2011/12/20 giovanni.v > Il 20/12/2011 6.07, Anton Kvashenkin ha scritto: > > you can add exterin= in sip.conf for each trunk >> > > I think this can be used only in [general] section not on peers > definition; also useful

Re: [asterisk-users] sendvoicemail=yes not quite working

On Tue, Dec 20, 2011 at 7:03 PM, M Maki wrote: > I have a system working great with the exception of the sendvoicemail=yes > voicemail.conf option. I can not figure out what I am missing or have > configured wrong... > > > While in voicemail after selecting 3 for advanced options, then 5 to leave

Re: [asterisk-users] sendvoicemail=yes not quite working

On Tue, Dec 20, 2011 at 8:53 PM, Todd Routhier wrote: > > > On Tue, Dec 20, 2011 at 7:03 PM, M Maki wrote: > >> I have a system working great with the exception of the sendvoicemail=yes >> voicemail.conf option. I can not figure out what I am missing or have >> configured wrong... >> >> >> While

[asterisk-users] sendvoicemail=yes not quite working

I have a system working great with the exception of the sendvoicemail=yes voicemail.conf option. I can not figure out what I am missing or have configured wrong... While in voicemail after selecting 3 for advanced options, then 5 to leave a message I am directed to the correct mailbox. But af

Re: [asterisk-users] Populate CDR issues

Hi Mike, I've tried updating my CDR's via the h exten but with no success. I've tried with both endbeforehexten=no and endbeforehexten=yes (in cdr.conf) but the value refused to appear in my CDR (even though I see the Set() application being executed in the console under the h exten). Thank you

Re: [asterisk-users] Limit # of inbound calls on SIP trunk

Well freepbx has that in the gui you should read the tool tips Read the trunk limit tooltip -Original Message- From: Steve Edwards Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 20 Dec 2011 12:16:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: Ast

Re: [asterisk-users] Call pickup on Asterisk 1.8 and Polycom IP550s?

For what it's worth, the phone is getting enough information. The first call works fine - it's the second call that never triggers the pickup screen, though it does cause the lamp to blink for that line. It's like the phone understands "ringing" but not "busy+ringing". I'm tempted to say it's

Re: [asterisk-users] GOIP GSM to SIP Gateway?

I would think it would be better to set a variable for each user and then have a single context with something like: _NXX,1,Dial(SIP/${WhatToUse}/${EXTEN}) Or something like this. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 20, 2011, at 1:03 PM, John Kini

Re: [asterisk-users] GOIP GSM to SIP Gateway?

On Tue, Dec 20, 2011 at 12:39 PM, Matt wrote: > > Is there anyway (short of defining dial an 8 from this phone for this > trunk to this SIM and a 9 from this phone for a trunk to this SIM) to > get it to use certain SIM cards when calls are made from certain > phones? > > You could define multipl

Re: [asterisk-users] OOH323 config file

On 11-12-20 11:21 AM, Carlos Chavez wrote: Just a warning to people trying to use ooh323 with Asterisk 1.8.7. The example config file that comes with asterisk is called chan_ooh323.conf when it actually should be named ooh323.conf for it to work. Sent me into a panic when I was trying t

Re: [asterisk-users] Limit # of inbound calls on SIP trunk

Un-top-posting... On Mon, 19 Dec 2011, Douglas Mortensen wrote: I have a system with FreePBX, and as far as I can tell it does not provide a means to limit the number of simultaneous inbound calls on a SIP trunk. Therefore I suspect that I’ll need to do some manual dialplan manipulation.

Re: [asterisk-users] GOIP GSM to SIP Gateway?

Hello It is possible but how do you have the dialplan ? In your dial plan you can do that Regards On Dec 20, 2011 2:40 PM, "Matt" wrote: > Hi, > Has anyone here any experiencing with linking an Asterisk PBX to a > GOIP GSM to SIP Gateway? We've got inbound calls from the GSM network > working

[asterisk-users] GOIP GSM to SIP Gateway?

Hi, Has anyone here any experiencing with linking an Asterisk PBX to a GOIP GSM to SIP Gateway? We've got inbound calls from the GSM network working properly, however, outbound calls seem to randomly choose a SIM line to use. Is there anyway (short of defining dial an 8 from this phone for this t

Re: [asterisk-users] Limit # of inbound calls on SIP trunk

Excellent. Do you think these functions would enable me to create rules based on both the concurrent # of inbound and/or outbound calls, or only total # of concurrent calls (agnostic to call direction being inbound vs. outbound)? Thanks, - Doug Mortensen Network Consultant Impala Networks P: 505

[asterisk-users] OOH323 config file

Just a warning to people trying to use ooh323 with Asterisk 1.8.7. The example config file that comes with asterisk is called chan_ooh323.conf when it actually should be named ooh323.conf for it to work. Sent me into a panic when I was trying to install an H323 link to an Avaya server and

Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

Eric Wieling wrote: Polycom (r) UC Software: Configuration File Conversion Utility\ On the pagehttp://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html And for those of us without Windows, this utility appears to work fine under wine. Doug

Re: [asterisk-users] PITCH_SHIFT()

On 20/12/11 01:15 AM, John Jolly wrote: In Leif Madsen's AstriCon 2010 talk titled "5 Things You Didn't Know Asterisk Could Do

Re: [asterisk-users] India Telecom regulations

How can we get thise license? Who do we have to pay. Nick. On Tue, Dec 20, 2011 at 9:52 AM, khalid touati wrote: > Thank you Raj, > I hope it will soon require no license as I heard there is a project to > change this law, for now I believe I will recommend our office in India to > go for li

Re: [asterisk-users] India Telecom regulations

Thank you Raj, I hope it will soon require no license as I heard there is a project to change this law, for now I believe I will recommend our office in India to go for license (to bridge to PSTN). Thanks once more for your help! 2011/12/19 Raj Mathur (राज माथुर) > On Tuesday 20 Dec 2011, khalid

Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

Polycom (r) UC Software: Configuration File Conversion Utility\ On the page http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium

Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

Out of curiosity, what is "the Polycom script"? I obviously haven't moved from 3.2.x firmware yet. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Friday, December 16, 2011 4:45 PM To: Aster

[asterisk-users] Help_video call not run

Hi all In sip.conf i take as [general] videosupport=yes    ; then UDPTL will flow to the remote device [phone1] type=friend host=dynamic context= employees disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw allow=alaw allow=ad

Re: [asterisk-users] File Convert

On Tue, Dec 20, 2011 at 05:34:46PM +0530, Gopalakrishnan N wrote: > Hi users, > > I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file > to G729 using file convert, but I am facing error as follows, > > file convert /tmp/welcome.gsm /tmp/welcome.g729 > Failed to convert /tm

[asterisk-users] Asterisk Sip Media Call Type

Hi all, I am trying to make a SIP Video and Audio Call, Now when I add at the Asterisk the video Support and the right codec whether I make Audio or Video Call from my clients the Call will be received as Video Call, so the problem is if I make from one client Audio or Video Call it will be reci

[asterisk-users] File Convert

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/

Re: [asterisk-users] Use different local IP for each SIP trunk

Il 20/12/2011 6.07, Anton Kvashenkin ha scritto: you can add exterin= in sip.conf for each trunk I think this can be used only in [general] section not on peers definition; also useful only when asterisk is behind nat. Not? -- _

Re: [asterisk-users] Problem with Atxfer for the calling party [SOLVED]

As explained in the posts before, this tread was solved. Thanks. On Tue, 2011-12-13 at 17:07 -0200, Antonio Modesto wrote: > On Tue, 2011-12-13 at 16:35 -0200, Roberto Linck wrote: > > > Hi Antonio, > > > > > > I'd never had used extensions.ael but in extensions.conf, using > > Macro I alway

Re: [asterisk-users] Use different local IP for each SIP trunk

Hello, Douglas Mortensen wrote: With that said, then it appears that the only way that I can have multiple trunks setup with them is to have asterisk use a different IP for all of the SIP & RTP traffic for each given trunk. Essentially I would setup multiple IP addresses on my eth0 interface. Is