[Asterisk-Dev] SS7 for *
Michael Baird
mike at tc3net.com
Sat Oct 2 11:32:12 MST 2004
Yes, I would be interested in more information about SS7 support in
asterisk also.
Regards
Michael Baird
> Hi Steve,
>
> Steve Wrote:
> >If you want a GPL one, carry one. However, if you are prepared to pay,
> >SS7 for * is now working.
>
> Is there anybody I can contact regarding this? I am prepared to pay.
>
>
> Yours,
> Hadi.
>
> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of asterisk-dev-request at lists.digium.com
> Sent: Friday, October 01, 2004 1:59 PM
> To: asterisk-dev at lists.digium.com
> Subject: Asterisk-Dev Digest, Vol 3, Issue 1
>
> Send Asterisk-Dev mailing list submissions to
> asterisk-dev at lists.digium.com
>
> 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
> asterisk-dev-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-dev-owner at lists.digium.com
>
> When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Dev digest..."
>
>
> Today's Topics:
>
> 1. qcall CDR problem (VoIP)
> 2. Re: ast_waitstream_full: Wait failed (Resource temporarily
> unavailable) (Dinesh Nair)
> 3. Re: Call control problems from Java (Miroslav Nachev)
> 4. Re: Use the Meetme application with another module than
> USB-UHCI (Arkadi Shishlov)
> 5. RE: Call control problems from Java (Zac Wolfe)
> 6. Re: Wish List / Brain Storm from AstriCon (Steve Underwood)
> 7. Re: Wish List / Brain Storm from AstriCon (Steve Underwood)
> 8. How is DTMF relayed? (Andreas Sikkema)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 1 Oct 2004 12:16:41 +0800
> From: "VoIP" <voip at er21.com>
> Subject: [Asterisk-Dev] qcall CDR problem
> To: "'Asterisk Developers Mailing List'"
> <asterisk-dev at lists.digium.com>
> Message-ID: <ER-DNScKVOz9b1tvmC20000001a at er-dns.er21.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi, I am trying to use qcall to do callback service. And put the qcall file as follows,
> ------------------------------------
> SIP/2001 at b.com 1001 at a.com SIP/3001 at c.com 0
> ------------------------------------
> Qcall first initiate a call to 2001 at b.com and wait for his answer, when 2001 at b.com answers the call, Qcall connects to 3001 at c.com. So 2001 and 3001
> can talk to each other. Here we set the caller id to 1001 at a.com.
>
> Actually there are 2 legs in this call. One is 2001 at b.com, and the other is 3001 at c.com. I found CDR only recorded call from 1001 at a.com to 3001 at c.com but no such CDR from 1001 at a.com to 2001 at b.com.
>
> If 2001 at b.com and 3001 at c.com are both PSTN numbers, then it should be a big problem to charge the callback users.
>
> Although this is a user level questions, I doubt the CDR mechanism should have problem.
>
> Regds,
> KK
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Fri, 01 Oct 2004 12:19:21 +0800
> From: Dinesh Nair <dinesh at alphaque.com>
> Subject: Re: [Asterisk-Dev] ast_waitstream_full: Wait failed (Resource
> temporarily unavailable)
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <415CDAC9.3000708 at alphaque.com>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> On 01/10/2004 02:55 Gil Kloepfer said the following:
> > I think I identified this problem a while back. If you guys would
> > like me to repost this, I can...otherwise look for the message
> > I sent with subject: Oddities in asterisk/say.c
>
> excellent, gil. my first observation of the problem was voicemail hanging
> up on me as well, which i then tracked down to the SayNumber application,
> and further back to ast_waitstream_full().
>
> your solution mirrors the observation (and solution) i posted as well.
>
> so, based on your experience, it'd be ok to modify that snippet of code to
>
> if (audiofd > -1 && ctrlfd > -1) in all places that it occurs in say.c ?
>
> (the rest of the functions are just different language handling functions,
> so theoretically it should be alright)
>
> --
> Regards, /\_/\ "All dogs go to heaven."
> dinesh at alphaque.com (0 0) http://www.alphaque.com/
> +==========================----oOO--(_)--OOo----==========================+
> | for a in past present future; do |
> | for b in clients employers associates relatives neighbours pets; do |
> | echo "The opinions here in no way reflect the opinions of my $a $b." |
> | done; done |
> +=========================================================================+
>
>
> ------------------------------
>
> Message: 3
> Date: Fri, 1 Oct 2004 10:10:09 +0200
> From: Miroslav Nachev <miro at space-comm.com>
> Subject: Re: [Asterisk-Dev] Call control problems from Java
> To: Asterisk Developers Mailing List
> <asterisk-dev at lists.digium.com>(E-mail)
> Message-ID: <05766962.20041001101009 at space-comm.com>
> Content-Type: text/plain; charset=us-ascii
>
> Dear Zac,
>
> This is Great. We have ideas to replace most of the Asterisk
> Services (modules) with Java too, but we haven't time yet. But in case
> that you are start this job we will help you.
> Because our idea was to use some Modules Convention which to
> support independet language support what do you think to make first
> some working plan and then to start working?
> We are ready to share this hard job with you. Please say how can we
> help you?
>
>
> Best Regards,
> Miroslav Nachev
>
> First the good news: I've just released a Java (JNI) bridge to Asterisk
> called, JAsterisk (ya I got a little creative with the name). It's
> available at (http://sourceforge.net/projects/jasterisk/) and is pre-alpha.
> It's currently distributed as a patch on the 1.0.0 source release but I hope
> to have a version available soon that doesn't require any modifications to
> Asterisk code.
>
> Anyway, almost all Asterisk functionality is available and working fine from
> Java. Almost...
>
> I have some strange issues that I'm dealing with that hopefully won't seem
> so strange to someone out there:
>
> 1. Call comes in, channel is routed to the "Safi" (Safi Systems is my
> company's name) application which simply fires a callback to the JNI
> notifying a Jasterisk application of it's presence, and loops while waiting
> for channel hangup (like the wait_for_hangup function in app_dial.c). The
> Jasterisk application then routes the call to "Jo Blo" at some address, by
> executing the Asterisk "Dial" application in a new Java thread. When Jo
> answers the call, caller and callee are connected and everything seems to be
> OK. However, if Jo then tries to transfer the call (using '#' for Zaptel
> or the 'Transfer' function for IAX channels), the caller and callee are
> disconnected and the call dies. Why? If the caller calls Jo Blo directly
> (as per usual) instead of being routed through the Jasterisk app, the
> transfer works fine. Similar issues are noticed when parking redirected
> calls.
>
> 2. pbx.c builtin functions "Transfer" and "Goto" don't work in Jasterisk
> apps. The channel structure's members are correctly updated but the
> extension or context switch never happens. What am I missing here?
>
> This software is in it's very early stages and it's my goal to make it the
> de-facto Java-->Asterisk interface. I'm very open to suggestions and if a
> better approach is revealed I'll happily adjust or rewrite the thing from
> scratch. Eventually I'd like to be able to forgo the Asterisk dialplan
> entirely (except maybe for an exten => _.,1,Safi() catch-all), and handle
> all call routing from Java, perhaps using a JTAPI interface or similar.
>
> Progress is encouraging so far -- I've already deployed some pretty complex
> production IVR's using this package and they're all stable and running fine.
> However, these call routing related issues have me a little worried that I
> may have taken a wrong turn somewhere.
>
> Any ideas or recommendations? TIA
>
> Zac Wolfe
> Safi Systems LLC
> zacw at safisys.com
>
>
>
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
> ------------------------------
>
> Message: 4
> Date: Fri, 1 Oct 2004 10:54:15 +0300
> From: "Arkadi Shishlov" <arkadi at mebius.lv>
> Subject: Re: [Asterisk-Dev] Use the Meetme application with another
> module than USB-UHCI
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <20041001075415.GB7055 at mebius.lv>
> Content-Type: text/plain; charset=us-ascii
>
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer
>
>
> ------------------------------
>
> Message: 5
> Date: Fri, 1 Oct 2004 03:49:41 -0700
> From: "Zac Wolfe" <zacw at safisys.com>
> Subject: RE: [Asterisk-Dev] Call control problems from Java
> To: "'Miroslav Nachev'" <miro at space-comm.com>, "'Asterisk Developers
> Mailing List (E-mail)'" <asterisk-dev at lists.digium.com>
> Message-ID: <000901c4a7a4$5bf05a10$6601a8c0 at c910133a2>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi Miroslav,
>
> I'd *love* to have some help on this project but lets see if our goals are
> in sync.
>
> My overall goal is to develop a Java-based VXML/CCXML engine for Asterisk.
> Achieving this requires that I have complete call control from Java.
> Optimally, we could then do away with the Asterisk Dialplan entirely and do
> everything via VXML/CCXML. The AGI interface is OK but for my application I
> need fine-grained event-driven processes where we can track each call from
> setup to tear-down and everything in between.
>
> I've started building a JTAPI-like framework that sits on top of JAsterisk
> that handles all the events and provides a high-level framework for building
> telephony apps. But, as I described in my previous email, I'm dealing with
> some nagging problems that I need to fix before I can complete this project.
>
> You described the "module convention". Is it the Asterisk application API
> you're referring to (ex app_dial,app_meetme,...)? If so, I think Jasterisk
> may be able to do what you're talking about already (or with some minor
> additions). BTW, The version of JAsterisk i'm working on now allows you to
> start the JVM from Asterisk or to start Asterisk from the JVM (which is the
> way it's set up currently).
>
> Anyway, lets explore some more and see if we can work together on this.
>
> I hope I'm making sense -- it's 4am where I am.
>
> Zac
>
> -----Original Message-----
> From: Miroslav Nachev [mailto:miro at space-comm.com]
> Sent: Friday, October 01, 2004 1:10 AM
> To: Asterisk Developers Mailing List (E-mail)
> Subject: Re: [Asterisk-Dev] Call control problems from Java
>
>
> Dear Zac,
>
> This is Great. We have ideas to replace most of the Asterisk
> Services (modules) with Java too, but we haven't time yet. But in case
> that you are start this job we will help you.
> Because our idea was to use some Modules Convention which to
> support independet language support what do you think to make first
> some working plan and then to start working?
> We are ready to share this hard job with you. Please say how can we
> help you?
>
>
> Best Regards,
> Miroslav Nachev
>
> First the good news: I've just released a Java (JNI) bridge to Asterisk
> called, JAsterisk (ya I got a little creative with the name). It's
> available at (http://sourceforge.net/projects/jasterisk/) and is pre-alpha.
> It's currently distributed as a patch on the 1.0.0 source release but I hope
> to have a version available soon that doesn't require any modifications to
> Asterisk code.
>
> Anyway, almost all Asterisk functionality is available and working fine from
> Java. Almost...
>
> I have some strange issues that I'm dealing with that hopefully won't seem
> so strange to someone out there:
>
> 1. Call comes in, channel is routed to the "Safi" (Safi Systems is my
> company's name) application which simply fires a callback to the JNI
> notifying a Jasterisk application of it's presence, and loops while waiting
> for channel hangup (like the wait_for_hangup function in app_dial.c). The
> Jasterisk application then routes the call to "Jo Blo" at some address, by
> executing the Asterisk "Dial" application in a new Java thread. When Jo
> answers the call, caller and callee are connected and everything seems to be
> OK. However, if Jo then tries to transfer the call (using '#' for Zaptel
> or the 'Transfer' function for IAX channels), the caller and callee are
> disconnected and the call dies. Why? If the caller calls Jo Blo directly
> (as per usual) instead of being routed through the Jasterisk app, the
> transfer works fine. Similar issues are noticed when parking redirected
> calls.
>
> 2. pbx.c builtin functions "Transfer" and "Goto" don't work in Jasterisk
> apps. The channel structure's members are correctly updated but the
> extension or context switch never happens. What am I missing here?
>
> This software is in it's very early stages and it's my goal to make it the
> de-facto Java-->Asterisk interface. I'm very open to suggestions and if a
> better approach is revealed I'll happily adjust or rewrite the thing from
> scratch. Eventually I'd like to be able to forgo the Asterisk dialplan
> entirely (except maybe for an exten => _.,1,Safi() catch-all), and handle
> all call routing from Java, perhaps using a JTAPI interface or similar.
>
> Progress is encouraging so far -- I've already deployed some pretty complex
> production IVR's using this package and they're all stable and running fine.
> However, these call routing related issues have me a little worried that I
> may have taken a wrong turn somewhere.
>
> Any ideas or recommendations? TIA
>
> Zac Wolfe
> Safi Systems LLC
> zacw at safisys.com
>
>
>
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Fri, 01 Oct 2004 19:40:35 +0800
> From: Steve Underwood <steveu at coppice.org>
> Subject: Re: [Asterisk-Dev] Wish List / Brain Storm from AstriCon
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <415D4233.3050507 at coppice.org>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Steven Sokol wrote:
>
> >37) R2 Integration (along with SS7 and QSIG).
> >
> >
> R2 support has now been released as GPL code. SS7 is now working at a
> test site in Germany, though is unlikely to become GPL any time soon. It
> is currently carrying small scale non-fee paying traffic, while its
> stability is being assessed. This is a fresh implementation of SS7,
> owing nothing to implementations like OpenSS7.
>
> >[...]
> >
> >39) Move from GLIB C to MicroLib C for better support of embedded systems.
> >
> >
> Shouldn't that be support uLIBC as well, rather than instead? Excellent
> idea. Threading seems to screw up when using uLIBC right now. That seems
> to be the only thing stopping me from running a stripped down * on
> Linksys boxes.
>
> >40) T.38 fax support over Asterisk (pass-through).
> >
> >
> Passthrough should be possible using the T38modem code in H.323. Some
> work is needed, but I doubt its that much.
>
> >41) What happened to SpanDSP? Steve Underwood. Had to drop the project.
> >May pick it back up again.
> >
> >
> Eh? It has been in continuous active development. The April release has
> been pretty stable, so I haven't had cause to keep releasing small
> increments. A pre-release of the next major step was made available for
> download this week. I suggest only serious testers play with it.
>
> >42) Steve Underwood may be working on R2? Working on Class 1 and 2 support
> >in his Span DSP. He is looking for support (financial and/or technical).
> >
> >
> R2 is released. Class 1 is being worked on, at a low priority. Class 2
> is pointless, as HylaFax can do all that stuff for you. I am not looking
> for financial support, though a MacLaren F1 would be nice :-). If anyone
> wants to push the class 1 implementation forward, they are most welcome to.
>
> >[...]
> >SNMP support for Asterisk.
> >
> >
> That would be valuable.
>
> >Voice quality monitoring?
> >
> >
> That is hard, and the standardised methods - the ones most people will
> accept - seem to have some IP issues. I tend to think most of the
> results they achieve are rather bogus, anyway.
>
> >Frame slip warnings and more across SNMP. Essentially alarm forwarding as
> >SNMP.
> >
> >
> An *excellent* idea. However, I remain unconvinced the current software
> actually detects slips properly. I am implementing the ITU BERT specs
> right now, to try to provide a proper end-to-end platform for this. The
> T1/E1 framer chips support BERT tests, but a test to the application
> level is what is really needed.
>
> Regards,
> Steve
>
>
>
> ------------------------------
>
> Message: 7
> Date: Fri, 01 Oct 2004 19:50:58 +0800
> From: Steve Underwood <steveu at coppice.org>
> Subject: Re: [Asterisk-Dev] Wish List / Brain Storm from AstriCon
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <415D44A2.8040203 at coppice.org>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Steven Sokol wrote:
>
> >10) IBM's open-source speech recognition software
> >
> >
> >
> IBM haven't released any. What they released is just a few components
> related to recognition.
>
> >11) Intel's soft-DSP chips for some advanced management of media streams.
> >
> >
> What do you mean by Intel soft-DSP chips?
>
> >12) Dynamic routing protocol - ARP (Asterisk Routing Protocol). Dynamic
> >extension - (ENUM).
> >
> >
> ARP is a bad name, since ARP is already a core networking protocol.
>
> >13) Voice frame size available in chunk sizes beyond 20ms.
> >
> >
> Do you mean within *, or only for batching in larger packets. The former
> seems bad. The latter a good idea.
>
> >[...]
> >
> >15) Certification program for Asterisk - hardware and software. Additional
> >approvals in other markets.
> >
> >
> This is an interesting area, and will probably require lockdown of
> modules, so they cannot be modified without raising an alarm "NON
> APPROVED CODE RUNNING" or somesuch. The ISDN4Linux guys came up with
> something like that, which has been acceptable to the approvals body in
> the EU.
>
> >16) IAX2 standards track - RFC submission
> >
> >
> A good idea.
>
> >17) Assembler coding of codec handlers for improved performance.
> >
> >
> A wacky idea :-\ Making use of SIMD can help in some areas, though.
> Algorithmic changes usually get you the biggest gains, but you might hit
> some patent issues (no, not US only software patents).
>
> >[...]
> >22) QSIG - Help people migrate to Asterisk. (Q931 can help cover some
> >additional channel functions).
> >
> >
> Someone has done some work on QSIG in an extended libpri, but put it
> aside. It might be resurrected.
>
> >34) Additional video codecs. More than H261, H263.
> >
> >35) SS7 Support for Asterisk. Malcolm will set up a mailing list. A new
> >development group is forming. Email: asterisk-ss7 at flanet.net
> >
> >
> If you want a GPL one, carry one. However, if you are prepared to pay,
> SS7 for * is now working.
>
> Regards,
> Steve
>
>
>
> ------------------------------
>
> Message: 8
> Date: Fri, 1 Oct 2004 13:59:07 +0200
> From: "Andreas Sikkema" <andreas.sikkema at ritstele.com>
> Subject: [Asterisk-Dev] How is DTMF relayed?
> To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> Message-ID: <34F1B1EDB3E7B04C9A282FE3537FC49F141A3E at mail.ritstele.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> I'm looking into DTMF relaying of Asterisk, specifically the
> RFC 2833 relaying, but I'm not sure I understand where
> the outgoing messages are sent.
>
> In rtp.c ast_rtp_read() calls process_rfc2833(), which can
> call send_dtmf(). From there the results of send_dtmf() are
> returned back to ast_rtp_read().
>
> When ast_rtp_read() receives this result, it checks if
> rtp->callback is set and calls the callback function if
> available.
>
> I think this callback can be a function that really sends
> the mesage to the other party, but I'm not sure. Can
> someone enlighten me?
>
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
--
Michael Baird <mike at tc3net.com>
More information about the asterisk-dev
mailing list