<div>Hi Eric,</div>
<div> </div>
<div>If you could provide me with some more details </div>
<ul>
<li>which soft phone are you using ?</li>
<li>if you can take a ethereal trace of conference call and send it across to analyze whats wrong</li>
<li>for MoH did you install mpg123 </li></ul>
<div>Regards </div>
<div>Chetan Jha<br><br> </div>
<div><span class="gmail_quote">On 11/3/07, <b class="gmail_sendername"><a href="mailto:asterisk-dev-request@lists.digium.com">asterisk-dev-request@lists.digium.com</a></b> <<a href="mailto:asterisk-dev-request@lists.digium.com">
asterisk-dev-request@lists.digium.com</a>> wrote:</span></div>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Send asterisk-dev mailing list submissions to<br> <a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com
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asterisk-dev-owner@lists.digium.com</a><br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of asterisk-dev digest..."<br><br><br>Today's Topics:<br><br> 1. Re: Asterisk SIP Channels Bridge (Asterisk)
<br> 2. Re: Trunk: Can't get any verbosity (Brian Capouch)<br> 3. Re: Asterisk SIP Channels Bridge (Mayank Mathur)<br> 4. Re: Asterisk SIP Channels Bridge (Asterisk)<br> 5. Re: LUA in Asterisk. was [svn-commits] tilghman: trunk r88250
<br> - in /trunk: ./ build_tools/ configs/ include/asterisk...<br> (Victor Sergeev)<br> 6. Re: LUA in Asterisk. was [svn-commits] tilghman: trunk r88250<br> - in /trunk: ./ build_tools/ configs/ include/asterisk...
<br> (Tilghman Lesher)<br> 7. Re: Trunk: Can't get any verbosity (Tilghman Lesher)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Sat, 03 Nov 2007 02:10:22 -0400
<br>From: Asterisk <<a href="mailto:asterisk@ivrtechgroup.com">asterisk@ivrtechgroup.com</a>><br>Subject: Re: [asterisk-dev] Asterisk SIP Channels Bridge<br>To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">
asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a href="mailto:472C10CE.4090503@ivrtechgroup.com">472C10CE.4090503@ivrtechgroup.com</a>><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>Hi Mayank,
<br><br>Yes, I am trying to conference 2 users in through SIP. I am not using<br>any Digium card and calls come in from the carrier via SIP.<br><br>First caller would call in and be placed on hold. And I have the unique<br>
name of the channel saved in the database. And the subscriber will get a<br>text message indicating that you have a call. And if the subscriber<br>wants to talk to the original caller who is still on hold, he/she will<br>
call into the system and the system would bridge both calls together. I<br>am getting sporadic results with the bridging. And while the original<br>caller is on hold , music on hold will not play most of the time.<br><br>
I have read that this is a pretty simple feature to do if we use a PRI.<br><br>thanks<br><br>Eric Lee<br><br><br>Mayank Mathur wrote:<br>> hi<br>> ru looking to do Conferencing b/w users thru SIP / just want 2<br>> simultaneous users to get connected thru SIP ??
<br>> And what Prob ru facing ??<br>> Let me know whether if i can help u out .<br>><br>><br>><br>>> Hi there,<br>>><br>>> I'm trying to bridge 2 SIP channels together via AGI script. The first
<br>>> caller would call in and be placed on hold and the second caller would<br>>> call in and both the calls gets connected together.<br>>><br>>> But I am having problem with the second caller finding the first channel.
<br>>><br>>> Can someone point me to the right direction?<br>>><br>>> thanks<br>>><br>>> Eric<br>>><br>>> _______________________________________________<br>>> --Bandwidth and Colocation Provided by
<a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br>>><br>>> asterisk-dev mailing list<br>>> To UNSUBSCRIBE or update options visit:<br>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">
http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>>><br>>><br>><br>><br>><br><br><br><br><br>------------------------------<br><br>Message: 2<br>Date: Sat, 03 Nov 2007 02:09:39 -0400<br>From: Brian Capouch <
<a href="mailto:brianc@palaver.net">brianc@palaver.net</a>><br>Subject: Re: [asterisk-dev] Trunk: Can't get any verbosity<br>To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com
</a>><br>Message-ID: <<a href="mailto:472C10BA.1020504@palaver.net">472C10BA.1020504@palaver.net</a>><br>Content-Type: text/plain; charset=us-ascii; format=flowed<br><br>Russell Bryant wrote:<br>> Brian Capouch wrote:
<br>><br>>>. . . it's doing some waaaay funky things<br>>>to the CLI that never used to happen--putting it into reverse video, and<br>>>then coloring up some but not all of the output.<br>>>
<br>>>On my terminals reverse video is pretty hard to read, so IMO it would be<br>>>better if it behaved the way it used to . . .<br>><br>><br>> Ah, that may be from this change. Try reverting it.<br>
><br>> ------------------------------------------------------------------------<br>> r86119 | tilghman | 2007-10-17 12:06:47 -0500 (Wed, 17 Oct 2007) | 3 lines<br>><br>> Support color on certain platforms, even when started at boot (before TERM is set)
<br>> Closes issue #9048<br>> ------------------------------------------------------------------------<br>><br><br>Any way we could get the patch reverted in the primary trunk feed?<br><br>I have to patch it each time I build, and I don't think "color by
<br>default" is correct, is it?<br><br>Thanks.<br><br>b.<br><br>--<br>This message has been scanned for viruses and<br>dangerous content by MailScanner, and is<br>believed to be clean.<br><br><br><br><br>------------------------------
<br><br>Message: 3<br>Date: Sat, 3 Nov 2007 11:59:28 +0530 (IST)<br>From: "Mayank Mathur" <<a href="mailto:mayankmathur@tetrain.com">mayankmathur@tetrain.com</a>><br>Subject: Re: [asterisk-dev] Asterisk SIP Channels Bridge
<br>To: "Asterisk Developers Mailing List" <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a href="mailto:50536.122.160.8.206.1194071368.squirrel@mail.tetrain.com">
50536.122.160.8.206.1194071368.squirrel@mail.tetrain.com</a>><br>Content-Type: text/plain;charset=iso-8859-1<br><br><br>Dear Eric<br>I Got your reqm. but im not able to understood why ur using AGI scripting<br>to place a call or even to do conferencing in Asterisk using SIP.
<br>R all your users accessing same Asterisk Server ??<br>We don;t req PRI if all your users r using Public IP and can access Server.<br>U can directly place a conference using Asterisk's Inbuilt features.<br><br><br>
> Hi Mayank,<br>><br>> Yes, I am trying to conference 2 users in through SIP. I am not using<br>> any Digium card and calls come in from the carrier via SIP.<br>><br>> First caller would call in and be placed on hold. And I have the unique
<br>> name of the channel saved in the database. And the subscriber will get a<br>> text message indicating that you have a call. And if the subscriber<br>> wants to talk to the original caller who is still on hold, he/she will
<br>> call into the system and the system would bridge both calls together. I<br>> am getting sporadic results with the bridging. And while the original<br>> caller is on hold , music on hold will not play most of the time.
<br>><br>> I have read that this is a pretty simple feature to do if we use a PRI.<br>><br>> thanks<br>><br>> Eric Lee<br>><br>><br>> Mayank Mathur wrote:<br>>> hi<br>>> ru looking to do Conferencing b/w users thru SIP / just want 2
<br>>> simultaneous users to get connected thru SIP ??<br>>> And what Prob ru facing ??<br>>> Let me know whether if i can help u out .<br>>><br>>><br>>><br>>>> Hi there,<br>>>>
<br>>>> I'm trying to bridge 2 SIP channels together via AGI script. The first<br>>>> caller would call in and be placed on hold and the second caller would<br>>>> call in and both the calls gets connected together.
<br>>>><br>>>> But I am having problem with the second caller finding the first<br>>>> channel.<br>>>><br>>>> Can someone point me to the right direction?<br>>>><br>>>> thanks
<br>>>><br>>>> Eric<br>>>><br>>>> _______________________________________________<br>>>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--
</a><br>>>><br>>>> asterisk-dev mailing list<br>>>> To UNSUBSCRIBE or update options visit:<br>>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev
</a><br>>>><br>>>><br>>><br>>><br>>><br>><br>><br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">
http://www.api-digital.com--</a><br>><br>> asterisk-dev mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev
</a><br>><br><br><br>--<br><br>Regards,<br>Mayank Mathur<br><br><br><br><br>------------------------------<br><br>Message: 4<br>Date: Sat, 03 Nov 2007 03:40:37 -0400<br>From: Asterisk <<a href="mailto:asterisk@ivrtechgroup.com">
asterisk@ivrtechgroup.com</a>><br>Subject: Re: [asterisk-dev] Asterisk SIP Channels Bridge<br>To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>>
<br>Message-ID: <<a href="mailto:472C25F5.9010304@ivrtechgroup.com">472C25F5.9010304@ivrtechgroup.com</a>><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>Hi Mayank,<br><br>thanks again for the quick response.
<br><br>We are an IVR service bureau. The reason we need AGI script is because<br>we have a lot of toll free numbers that gets point to different agi<br>script. And each agi script are an IVR application by itself. And all
<br>call traffic comes into a SIP router and it gets routed to the<br>appropriate asterisk server. So essentially each asterisk server could<br>be running multiple ivr/AGI script that does it's own thing. But we are<br>
just not getting consistent results in merging the 2 SIP channels<br>together for this particular application.<br><br>thanks.<br><br>Eric Lee<br><br><br><br><br><br>Mayank Mathur wrote:<br>> Dear Eric<br>> I Got your reqm. but im not able to understood why ur using AGI scripting
<br>> to place a call or even to do conferencing in Asterisk using SIP.<br>> R all your users accessing same Asterisk Server ??<br>> We don;t req PRI if all your users r using Public IP and can access Server.<br>
> U can directly place a conference using Asterisk's Inbuilt features.<br>><br>><br>><br>>> Hi Mayank,<br>>><br>>> Yes, I am trying to conference 2 users in through SIP. I am not using<br>
>> any Digium card and calls come in from the carrier via SIP.<br>>><br>>> First caller would call in and be placed on hold. And I have the unique<br>>> name of the channel saved in the database. And the subscriber will get a
<br>>> text message indicating that you have a call. And if the subscriber<br>>> wants to talk to the original caller who is still on hold, he/she will<br>>> call into the system and the system would bridge both calls together. I
<br>>> am getting sporadic results with the bridging. And while the original<br>>> caller is on hold , music on hold will not play most of the time.<br>>><br>>> I have read that this is a pretty simple feature to do if we use a PRI.
<br>>><br>>> thanks<br>>><br>>> Eric Lee<br>>><br>>><br>>> Mayank Mathur wrote:<br>>><br>>>> hi<br>>>> ru looking to do Conferencing b/w users thru SIP / just want 2
<br>>>> simultaneous users to get connected thru SIP ??<br>>>> And what Prob ru facing ??<br>>>> Let me know whether if i can help u out .<br>>>><br>>>><br>>>><br>>>>
<br>>>>> Hi there,<br>>>>><br>>>>> I'm trying to bridge 2 SIP channels together via AGI script. The first<br>>>>> caller would call in and be placed on hold and the second caller would
<br>>>>> call in and both the calls gets connected together.<br>>>>><br>>>>> But I am having problem with the second caller finding the first<br>>>>> channel.<br>>>>>
<br>>>>> Can someone point me to the right direction?<br>>>>><br>>>>> thanks<br>>>>><br>>>>> Eric<br>>>>><br>>>>> _______________________________________________
<br>>>>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br>>>>><br>>>>> asterisk-dev mailing list<br>>>>> To UNSUBSCRIBE or update options visit:
<br>>>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>>>>><br>>>>><br>>>>><br>>>>
<br>>>><br>>> _______________________________________________<br>>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br>>><br>>> asterisk-dev mailing list
<br>>> To UNSUBSCRIBE or update options visit:<br>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>>><br>>><br>>
<br>><br>><br><br><br><br><br>------------------------------<br><br>Message: 5<br>Date: Sat, 3 Nov 2007 13:57:56 +0200<br>From: "Victor Sergeev" <<a href="mailto:sergeevvictor@gmail.com">sergeevvictor@gmail.com
</a>><br>Subject: Re: [asterisk-dev] LUA in Asterisk. was [svn-commits]<br> tilghman: trunk r88250 - in /trunk: ./ build_tools/ configs/<br> include/asterisk...<br>To: "Asterisk Developers Mailing List" <
<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID:<br> <<a href="mailto:b2cbe8e70711030457t88107bi251ea2a27d5c8e18@mail.gmail.com">b2cbe8e70711030457t88107bi251ea2a27d5c8e18@mail.gmail.com
</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Russell Bryant wrote:<br><br>SVN commits to the Digium repositories wrote:<br><br>Author: tilghman<br>Date: Fri Nov 2 10:36:34 2007<br>New Revision: 88250
<br><br>URL: <a href="http://svn.digium.com/view/asterisk?view=rev&rev=88250">http://svn.digium.com/view/asterisk?view=rev&rev=88250</a><br>Log:<br>Add pbx_lua as a method of doing extensions<br><br><br><br>This is quite a significant addition. Please add it to CHANGES ...
<br><br>That's a great feature!<br><br>Does it mean that Digium decided to replace AEL with LUA?<br>It seems there'll be no sense to use AEL anymore if you can do the same in<br>real programming language.<br><br>It's strange that such major feature was added without any discussion with
<br>development community (Nov 1 patch submitted, next day it is in the trunk).<br>Recently was discussed a topic about release cycle. IMO Asterisk should have<br>a roadmap for every release and such kind of addition should be planned and
<br>announced to community.<br><br>Victor<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: <a href="http://lists.digium.com/pipermail/asterisk-dev/attachments/20071103/ac2cf3ce/attachment.html">
http://lists.digium.com/pipermail/asterisk-dev/attachments/20071103/ac2cf3ce/attachment.html</a><br><br>------------------------------<br><br>Message: 6<br>Date: Sat, 3 Nov 2007 09:13:02 -0500<br>From: Tilghman Lesher <
<a href="mailto:tilghman@mail.jeffandtilghman.com">tilghman@mail.jeffandtilghman.com</a>><br>Subject: Re: [asterisk-dev] LUA in Asterisk. was [svn-commits]<br> tilghman: trunk r88250 - in /trunk: ./ build_tools/ configs/
<br> include/asterisk...<br>To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a href="mailto:200711030913.02566.tilghman@mail.jeffandtilghman.com">
200711030913.02566.tilghman@mail.jeffandtilghman.com</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>On Saturday 03 November 2007 06:57:56 Victor Sergeev wrote:<br>> That's a great feature!
<br>><br>> Does it mean that Digium decided to replace AEL with LUA?<br><br>No.<br><br>> It seems there'll be no sense to use AEL anymore if you can do the same in<br>> real programming language.<br><br>Are you saying that users shouldn't have a choice?
<br><br>> It's strange that such major feature was added without any discussion with<br>> development community (Nov 1 patch submitted, next day it is in the trunk).<br><br>Here's your chance. Discuss.<br><br>
> Recently was discussed a topic about release cycle. IMO Asterisk should<br>> have a roadmap for every release and such kind of addition should be<br>> planned and announced to community.<br><br>We don't roadmap, because we have no idea what code will be submitted to us
<br>during each development cycle. Asterisk is strongly community-oriented as to<br>its direction. Submissions of new code are always welcome.<br><br>We did not know LUA was coming, but once it arrived, we added it. If you want
<br>to submit a properly licensed implementation of the dialplan in another<br>language, go right ahead.<br><br>--<br>Tilghman<br><br><br><br>------------------------------<br><br>Message: 7<br>Date: Sat, 3 Nov 2007 09:17:17 -0500
<br>From: Tilghman Lesher <<a href="mailto:tilghman@mail.jeffandtilghman.com">tilghman@mail.jeffandtilghman.com</a>><br>Subject: Re: [asterisk-dev] Trunk: Can't get any verbosity<br>To: Asterisk Developers Mailing List <
<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Message-ID: <<a href="mailto:200711030917.17876.tilghman@mail.jeffandtilghman.com">200711030917.17876.tilghman@mail.jeffandtilghman.com
</a>><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>On Saturday 03 November 2007 01:09:39 Brian Capouch wrote:<br>> Russell Bryant wrote:<br>> > Brian Capouch wrote:<br>> >>. . . it's doing some waaaay funky things
<br>> >>to the CLI that never used to happen--putting it into reverse video, and<br>> >>then coloring up some but not all of the output.<br>> >><br>> >>On my terminals reverse video is pretty hard to read, so IMO it would be
<br>> >>better if it behaved the way it used to . . .<br>> ><br>> > Ah, that may be from this change. Try reverting it.<br>> ><br>> > ------------------------------------------------------------------------
<br>> > r86119 | tilghman | 2007-10-17 12:06:47 -0500 (Wed, 17 Oct 2007) | 3<br>> > lines<br>> ><br>> > Support color on certain platforms, even when started at boot (before<br>> > TERM is set) Closes issue #9048
<br>> > ------------------------------------------------------------------------<br>><br>> Any way we could get the patch reverted in the primary trunk feed?<br>><br>> I have to patch it each time I build, and I don't think "color by
<br>> default" is correct, is it?<br><br>Reverted in trunk.<br><br>--<br>Tilghman<br><br><br><br>------------------------------<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by
<a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev
</a><br><br>End of asterisk-dev Digest, Vol 40, Issue 6<br>*******************************************<br></blockquote><br>