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