[asterisk-dev] asterisk-dev Digest, Vol 40, Issue 6
chetan jha
chetang.jha at gmail.com
Sat Nov 3 12:33:22 CDT 2007
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, asterisk-dev-request at lists.digium.com <
asterisk-dev-request at lists.digium.com> wrote:
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> 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 <asterisk at ivrtechgroup.com>
> Subject: Re: [asterisk-dev] Asterisk SIP Channels Bridge
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <472C10CE.4090503 at ivrtechgroup.com>
> 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 <brianc at palaver.net>
> Subject: Re: [asterisk-dev] Trunk: Can't get any verbosity
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <472C10BA.1020504 at palaver.net>
> 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" <mayankmathur at tetrain.com>
> Subject: Re: [asterisk-dev] Asterisk SIP Channels Bridge
> To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> Message-ID: <50536.122.160.8.206.1194071368.squirrel at mail.tetrain.com>
> 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 <asterisk at ivrtechgroup.com>
> Subject: Re: [asterisk-dev] Asterisk SIP Channels Bridge
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <472C25F5.9010304 at ivrtechgroup.com>
> 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" <sergeevvictor at gmail.com>
> 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" <asterisk-dev at lists.digium.com>
> Message-ID:
> <b2cbe8e70711030457t88107bi251ea2a27d5c8e18 at mail.gmail.com>
> 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
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> ------------------------------
>
> Message: 6
> Date: Sat, 3 Nov 2007 09:13:02 -0500
> From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
> 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 <asterisk-dev at lists.digium.com>
> Message-ID: <200711030913.02566.tilghman at mail.jeffandtilghman.com>
> 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 <tilghman at mail.jeffandtilghman.com>
> Subject: Re: [asterisk-dev] Trunk: Can't get any verbosity
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <200711030917.17876.tilghman at mail.jeffandtilghman.com>
> 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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