[asterisk-users] Off-topic: SIP DTMF most supported method
Hi, I know it is a bit off-topic, but I'd like to ask the community what is the current most supported way to deal with DTMF? I'm looking for an all-SIP system and I'm mostly interested in the end devices support of the different methods (DTMF in-band audio, DTMF RTP telephony events packets, SIP INFO, ...) Thanks in advance. Cesc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and multicast RTP
Hi, I would need to bridge a SIP call with a multicast RTP channel. Both sides are receiving and transmitting RTP. Googling, I saw that an app_rtppage, which was in the SVN for a while and its not there anymore. It did, I think, only partly what I need (it sent from SIP to the mcast ... not the other way around), but it was a start. Any idea how to do this? I also could use ser/opensips/openser/kamailio with rtpproxy (does rtpproxy support this? it would in any case be a complex modification, I think). But my current setup is based on asterisk, so I'd rather not move it from there or install new apps. Thanks a bunch! Cesc -- Forwarded message -- From: Cesc Santa [EMAIL PROTECTED] Date: Fri, Nov 28, 2008 at 3:26 PM Subject: Asterisk RTP pager To: [EMAIL PROTECTED] Hi, I came across your RTPpage application and just made me very happy. If I may, some questions. * With which Asterisk versions has it been tested? is it in the official tree? * What I'd like to do is to link this RTPpage with incoming SIP calls ... so that all RTP from SIP is dumped to the multicast RTP and viceversa. Is that possible with this application? Thanks for your time, Cesc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] off-topic: Avaya 46xx, release 032207 ... help
Hi, I am trying to use an Avaya 4602 phone, which I just updated from a very old SIP software to the latest I could find on avaya's site (032207). The upgrade went fine and it gets registered on the Asterisk server. Now, a couple of glitches, though. - The phone's web server is not working ... so I have no easy way to configure it. It used to work with the old release of the software. I get on the firefox browser a connection has been reset error message. - Avaya admin guide keeps mentioning all the commands you can enter via the keyboard on the phone ... but they don't work for me ... (the MUTE + numbers combination). Any ideas? the web browser problem is the most annoying one. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip ... codec conversion matrix
inline ... On 8/10/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 10 Aug 2007, Cesc Santa wrote: Hi, I have asterisk 1.2.18. Installed from binary or compiled by yourself? I compiled it myself ... I just took a peak at the command: show translation and I saw that I can only convert from/to ulaw, ulaw, gsm and slin. No speex, no ilbc ... do I need a license or compile something extra? The G723, 726 and 729 ... I need a license, is that it? one for all of them? or for each? How do I get them to work? not just pass-through ... I need conversion. Speex and ilibc will be compiled into asterisk automatically. It is on my systems (Debian) I can use them ... but cannot do translation ... only pass-through ... it complains that there is no function to convert to internal codec (pcm?) G726 should also be compiled in as standard - it's supplied with asterisk. no license to use it needed? These may not be free enough for your Linux distribution, so that might be why you don't have them if you're running a binary installation. Eg. iLbc isn't free enough for standard Debian. well, i use debian, but not the asterisk binary distro ... i compiled myself from 1.2.18 sources G723 and G729 are patent encumbered and so you need licenses to run the software. You can get G729 licenses from Digium, but I don't know about G723... ok ... i will contact digium for g729 license ... anyone knows about g723? So maybe you need to compile your own rather than run a binary version? I did :( ;) Thanks! Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip ... codec conversion matrix
Hi, I have asterisk 1.2.18. I just took a peak at the command: show translation and I saw that I can only convert from/to ulaw, ulaw, gsm and slin. No speex, no ilbc ... do I need a license or compile something extra? The G723, 726 and 729 ... I need a license, is that it? one for all of them? or for each? How do I get them to work? not just pass-through ... I need conversion. Thanks a lot! Cesc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip ... codec conversion matrix
inline On 8/10/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 10 Aug 2007, Cesc Santa wrote: I can use them ... but cannot do translation ... only pass-through ... it complains that there is no function to convert to internal codec (pcm?) G726 should also be compiled in as standard - it's supplied with asterisk. no license to use it needed? None at all. ok ... i guess i will have to try harder ;) These may not be free enough for your Linux distribution, so that might be why you don't have them if you're running a binary installation. Eg. iLbc isn't free enough for standard Debian. well, i use debian, but not the asterisk binary distro ... i compiled myself from 1.2.18 sources OK. This is what I do too! I have Debian Sarge on my development box. So it's strange you don't have iLBC and speex. There is a debian package for speex: On my development box: bob:~# dpkg -l | grep speex ii libspeex-dev 1.1.6-2The Speex Speech Codec ii libspeex1 1.1.6-2The Speex Speech Codec ii speex 1.1.6-2The Speex Speech Codec this is probably one thing i am missing ... i will check but I have no iLBC packages - they appear to be in the asterisk source... bob:~# asterisk -rx 'show translation' Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 3 2 1 7 -3119 ulaw - 4 - 1 3 2 1 7 -3119 alaw - 4 1 - 3 2 1 7 -3119 g726 - 4 2 2 - 2 1 7 -3119 adpcm - 4 2 2 3 - 1 7 -3119 slin - 3 1 1 2 1 - 6 -3018 lpc10 - 5 3 3 4 3 2 - -3220 g729 - - - - - - - - - - - speex - 5 3 3 4 3 2 8 - -20 ilbc - 6 4 4 5 4 3 9 -33 - so I get everything apart from g723 and g729. I didn't do anything special to the makefile, etc. (other than set the i586 flag as my target box needs it) and this is definitely what I would expect to have ... knowing someone has it ... i'll get to work :) G723 and G729 are patent encumbered and so you need licenses to run the software. You can get G729 licenses from Digium, but I don't know about G723... ok ... i will contact digium for g729 license ... anyone knows about g723? Do you actually need g729? I'd not spend the money on it unless you really needed it. I've not heard of anyone using or offering g723... And I have to say; you're not missing much by not having iLBC or speex. The CPU overhead is significant and the voice quality is somewhat dubious :) Unless you absolutely really desperately need to compress the data stream to squeeze out every ounch of bandwidth, then I'd really not use them. If you need to compress, start with G726, then move to GSM, and then if you need better audio quality, buy g729 licenses. Gordon I agree with you ... actually, I am interested in G729 (and I am definitely getting a few licenses). G723 I also got a customer who would like to see it working in my product ... so I'll check it out. As for Speex and iLBC, it was some collateral damage ... I was looking at the transcoding and knowing these codecs were/are free, I found it strange that I could not translate them ... I'll fix that. Even it is just for the fun. Many thanks! Cesc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip ... codec conversion matrix
On 8/10/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Aug 10, 2007 at 11:35:32AM +0200, Cesc Santa wrote: Hi, I have asterisk 1.2.18. I just took a peak at the command: show translation and I saw that I can only convert from/to ulaw, ulaw, gsm and slin. No speex, no ilbc ... do I need a license or compile something extra? The G723, 726 and 729 ... I need a license, is that it? one for all of them? or for each? Do you have the modules codec_speex.so ? codec_ilbc.so ? Are they loaded? No special license is needed. I just checked ... I have the codec_ilbc (as well _g729, _g726 ), but not codec_speex. I guess I need to get the libspeex-dev package so it compiles. And then I checked my configuration of asterisk ... modules.conf ... loaded all the needed modules (codec_, format_) ... and voila! it works. Now I have translation between all codecs, except G729 and G723 (and of course, speex, which I don't have the module for). Now, G729 I can get a licence from Digium. Anyone know about G723? Do I need a license? I know it is not very used, but my costumer requires it ... Cesc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Refer ... rejected?
Hi, I have asterisk 1.2.18. I am trying to get asterisk to react to an (out of dialog) REFER ... see below. I get a 603 (no dialog) ... and in the code (sip.conf:3277) a comment being able but not supporting it?? Any pointers would be great ... is it a configuration option? REFER sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.200.135;branch=z9hG4bKcf61.5bbef4d3.0 To: sip:[EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=50315E7E56FF CSeq: 1 REFER Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: SER 0.9.6 Contact: sip:[EMAIL PROTECTED] Refer-To: sip:[EMAIL PROTECTED] Referred-By: sip:[EMAIL PROTECTED] --- (11 headers 0 lines) --- Transmitting (no NAT) to 192.168.200.135:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP 192.168.200.135;branch=z9hG4bKcf61.5bbef4d3.0;received=192.168.200.135 From: sip:[EMAIL PROTECTED];tag=50315E7E56FF To: sip:[EMAIL PROTECTED];tag=as34a5041e Call-ID: [EMAIL PROTECTED] CSeq: 1 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 with SIP and H.323
Hi, I need an Asterisk with G729 support. Preference is with Asterisk 1.2(.18), but if not possible, then it can be 1.4. Question is, can I enable G729 for both protocols? do the H323 implementation allow it? I found the codec support for H323 in 1.2.18 very poor ... only got u/a-law to work ... not even GSM. Would the Digium G729 license be good both for SIP and H323? Cesc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.2 from svn ... lock on shutdown
Hi, I hope this gets picked up by some bug marshall ... I have downloaded (yesterday) the 1.2 branch from svn ... When running: asterisk -c loaded modules: [modules] autoload=no load = pbx_functions.so load = pbx_config.so load = codec_a_mu.so load = format_pcm_alaw.so load = codec_ulaw.so load = codec_alaw.so load = format_pcm.so load = func_uri.so ;required by app_dial and chan_sip load = res_features.so load = app_dial.so ;playback and echo apps ... load = app_playback.so load = app_echo.so load = codec_gsm.so load = format_gsm.so load = format_wav_gsm.so load = chan_h323.so load = chan_sip.so load = chan_local.so When I do: stop now asterisk hangs up, but locks: *CLI stop now Beginning asterisk shutdown Executing last minute cleanups Asterisk cleanly ending (0). I attached gdb to the locked process: 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 (gdb) bt #0 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 #1 0xb793f304 in std::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::operator++ () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #2 0xb79881a0 in std::__distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #3 0xb79881cb in std::distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #4 0xb7989ee6 in std::_Rb_treePString, std::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase*, std::_Select1ststd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* , std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #5 0xb7989f20 in std::mapPString, PFactoryOpalMediaFormat, PString::WorkerBase*, std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #6 0xb7989f5a in PFactoryOpalMediaFormat, PString::Unregister_Internal () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #7 0xb7989f9d in PFactoryOpalMediaFormat, PString::Unregister () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #8 0xb7989fc9 in OpalPluginMediaFormat::~OpalPluginMediaFormat () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #9 0xb748bea1 in PAbstractList::RemoveAt () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #10 0xb74892e1 in PCollection::RemoveAll () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #11 0xb7489e25 in PAbstractList::DestroyContents () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #12 0xb7490152 in PContainer::Destruct () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #13 0xb791ca57 in PAbstractList::~PAbstractList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #14 0xb79755c9 in PListOpalMediaFormat::~PList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #15 0xb79828e7 in H323PluginCodecManager::~H323PluginCodecManager () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #16 0xb7e0d4f0 in exit () from /lib/tls/libc.so.6 #17 0x080bddd7 in quit_handler (num=135324439, nice=3, safeshutdown=1, restart=0) at asterisk.c:945 #18 0x080be019 in handle_shutdown_now (fd=1, argc=2, argv=0xb830) at asterisk.c:1104 #19 0x0809811b in ast_cli_command (fd=1, s=0x8151900 \001) at cli.c:1364 #20 0x080c0d93 in main (argc=2, argv=0xbd84) at asterisk.c:1019 (gdb) Regards, Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: h323 problem with asterisk 1.2.18
Hi guys, I had the same problem ... and then remembered that my asterisk 1.2.9.1 compiled just fine ... So, i tried that Makefile ... and voila! :) See attached patch ... Cesc On 5/8/07, nik600 [EMAIL PROTECTED] wrote: On 5/7/07, nik600 [EMAIL PROTECTED] wrote: i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/ export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH but when i go to: cd asterisk-1.2.18/channels/h323/ and do a make opt: [EMAIL PROTECTED]:/data/programmi/asterisk_1.2.18/asterisk-1.2.18/channels/h323# make opt make: *** No rule to make target `opt'. Stop. why? where am i wrong? i've also tried the last version of pwlib and openh323, but without fixing the problem thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser i've also tried supported version Open H.323 version v1.17.1, PWLib v1.9.0 but.. it doesn't compile. It seems to be a problem with makefile -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users asterisk.1.2.18.svn63330.h323.patch Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 from svn ... lock on shutdown
Hi, I will add the report ... though I find the system a bit cumbersome for sporadic users like me. Oh, and you are right ... without chan_h323 asterisk shuts down just fine. Regards, Cesc On 5/8/07, Joshua Colp [EMAIL PROTECTED] wrote: Cesc wrote: Hi, I hope this gets picked up by some bug marshall ... Eep! Filing a bug is best instead of email it here for future reference... I attached gdb to the locked process: 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 (gdb) bt #0 0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6 #1 0xb793f304 in std::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::operator++ () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #2 0xb79881a0 in std::__distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #3 0xb79881cb in std::distancestd::_Rb_tree_iteratorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #4 0xb7989ee6 in std::_Rb_treePString, std::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase*, std::_Select1ststd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* , std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #5 0xb7989f20 in std::mapPString, PFactoryOpalMediaFormat, PString::WorkerBase*, std::lessPString, std::allocatorstd::pairPString const, PFactoryOpalMediaFormat, PString::WorkerBase* ::erase () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #6 0xb7989f5a in PFactoryOpalMediaFormat, PString::Unregister_Internal () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #7 0xb7989f9d in PFactoryOpalMediaFormat, PString::Unregister () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #8 0xb7989fc9 in OpalPluginMediaFormat::~OpalPluginMediaFormat () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #9 0xb748bea1 in PAbstractList::RemoveAt () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #10 0xb74892e1 in PCollection::RemoveAll () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #11 0xb7489e25 in PAbstractList::DestroyContents () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #12 0xb7490152 in PContainer::Destruct () from /usr/lib/libpt_linux_x86_r.so.1.9.2 #13 0xb791ca57 in PAbstractList::~PAbstractList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #14 0xb79755c9 in PListOpalMediaFormat::~PList () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #15 0xb79828e7 in H323PluginCodecManager::~H323PluginCodecManager () from /usr/lib/libh323_linux_x86_r.so.1.17.3 #16 0xb7e0d4f0 in exit () from /lib/tls/libc.so.6 #17 0x080bddd7 in quit_handler (num=135324439, nice=3, safeshutdown=1, restart=0) at asterisk.c:945 #18 0x080be019 in handle_shutdown_now (fd=1, argc=2, argv=0xb830) at asterisk.c:1104 #19 0x0809811b in ast_cli_command (fd=1, s=0x8151900 \001) at cli.c:1364 #20 0x080c0d93 in main (argc=2, argv=0xbd84) at asterisk.c:1019 (gdb) This is definitely an issue with chan_h323 and OpenH323. If you don't load chan_h323 can you then shut down fine? If so please file a bug on bugs.digium.com and the individual who looks after that stuff will look at it. Thanks! Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H323 to H323 bridging ... failed ... also with chan_local
Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten = 111, 1, Dial(chan_h323, H323/[EMAIL PROTECTED])(in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the call to phone 2 is setup (completed) and when trying to proceed with call from phone1, asterisk stops: *CLI -- Executing Dial(H323/ip$192.168.1.100:1894/4096, H323/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] May 7 11:29:22 WARNING[845]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/wave-1(-2033656) to H323/ip$192.168.1.100:1894/4096(-2033656) -- H323/wave-1 answered H323/ip$192.168.1.100:1894/4096 May 7 11:29:22 WARNING[845]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/ip$192.168.1.100:1894/4096(-2033656) to H323/wave-1(-2033656) May 7 11:29:22 WARNING[845]: app_dial.c:1586 dial_exec_full: Had to drop call because I couldn't make H323/ip$192.168.1.100:1894/4096 compatible with H323/wave-1 == Spawn extension (h323_default, 811, 1) exited non-zero on 'H323/ip$192.168.1.100:1894/4096' I have tried with both phones individually, and both are asterisk-compatible with H323. Bridging works if the originating call is SIP, for example. But if I try H323 with H323, it's a nono. Am I doing something wrong? do I need to set up some parameter? I thought about using chan_local, but I came across this: *CLI -- Executing Dial(H323/ip$192.168.1.100:1940/4096, local/[EMAIL PROTECTED]/n) in new stack May 7 11:31:47 WARNING[860]: channel.c:2512 ast_request: No translator path exists for channel type local (native -1) to -2033656 May 7 11:31:47 NOTICE[860]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'local' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Wait(H323/ip$192.168.1.100:1940/4096, 1) in new stack -- Executing Playback(H323/ip$192.168.1.100:1940/4096, /etc/asterisk/sounds/pbx-invalid) in new stack -- Playing '/etc/asterisk/sounds/pbx-invalid' (language 'en') Thanks in advance! Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] isdn cross-over ...
Hi, I am sure this is a bit off topic, but maybe the people here have the knowledge. Quick question: I have two ISDN S/T phones. What is the quickest way to test them (call one from the other)? can i make an isdn cross-over cable, taking the correct pinning, of course? What i need is to avoid the need for an NT connection (via a PBX). If the above is not possible ... where can I buy a cheap, small, simple ISDN PBX with at least two NT ports, so that i can connect my two phones and call each other? Tks! Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] isdn cross-over ...
Thanks for the link.The VConsole equipment seems to be a close match to what i am looking for, just quite expensive ... but i hope that the company i work for will pay for it :) I see many options here. It came to my knowledge that we have a PSTN + 1 ISDN pbx ... a small quattrovox box. How can I connect the 2 phones in that one ISDN? and, can i call one from the other in this fashion? Cesc On 10/9/06, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Oct 09, 2006 at 10:55:51AM -0500, Ejay Hire wrote: Hi. A cross-over cable won't work, the isdn network provides signalling and adressing functions. When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg, around $1k used from ebay. They appear to have come down: http://search.ebay.com/isdn-bri-simulator_W0QQfrppZ50QQfsopZ1QQmaxrecordsreturnedZ300 Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk on 2.4 kernel ... scheduler problem?
inline ... On 9/28/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 28, 2006 at 03:24:27PM +0200, Cesc wrote: Hello people! I have an inquiry (not a doubt ;D ). Actually, two. I am trying to run asterisk on an embedded Power PC platform on which we have a linux with a 2.4.2x kernel. Still uses 2.4 today? Not a very good sign. yeah, i know ... but multiple issues here: management and money; backwards compatibility (not all sectors are concerned about cutting edge) and reduced size of the 2.4 kernel. Anyway ... it is just the way it is :) In there, the linux scheduler runs at 100Hz. On a 2.6 kernel, the scheduler is at 1KHz. I just take this from a colleague ... hope it is true :) I only need to run the VOIP part, thus no POTS or external hardware. Actually, I just need SIP and H323 (channels/h323). Is there any problem to be expected from the scheduler difference? Or any other from running on a 2.4 kernel? Some colleague said that asterisk needs the 1KHz scheduler, but i cannot believe that it won't run on a 2.4 kernel ... Anyway, that is why i am asking. If you really want a 1kHz timing source for 2.4, build zaptel. But you'll need a USB UHCI chip. my question is, do i really need the 1Khz scheduler? remember i just want to operate SIP and H323 ... If i do really need it ... there is NO chance to add any extra software or the like. It has to be all software based, otherwise it is a nono. No one else uses asterisk with a 2.4 kernel? forget the Power PC story .. but just 2.4? The other inquiry ... as the system is embedded we have not so much disk space available. So, i need a minimal asterisk installation. When compiled and stripped, the biggest amount of space is taken by the modules. My question is, can asterisk work with just the chan_sip.so, chan_h323.so and the codec_*.so? is there any other module needed? I need only be able to bridge sip to h323, no extra fancy stuff needed (parking, echo, blah, blah, ... ) Don't autoload modules in modules.conf . Load only the modules you need. Use 'load' from the CLI to manually load modules to see if you need them. One shortcut you may take is to use DeStar. It is an Asterisk configuration generator that generates a configuration with explicit load in modules.conf, rather than loading everything... ok, i will do that ... i will try to cut to a bare minimum. Tks! REgards, Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk on 2.4 kernel ... scheduler problem?
Hello people! I have an inquiry (not a doubt ;D ). Actually, two. I am trying to run asterisk on an embedded Power PC platform on which we have a linux with a 2.4.2x kernel. In there, the linux scheduler runs at 100Hz. On a 2.6 kernel, the scheduler is at 1KHz. I just take this from a colleague ... hope it is true :) I only need to run the VOIP part, thus no POTS or external hardware. Actually, I just need SIP and H323 (channels/h323). Is there any problem to be expected from the scheduler difference? Or any other from running on a 2.4 kernel? Some colleague said that asterisk needs the 1KHz scheduler, but i cannot believe that it won't run on a 2.4 kernel ... Anyway, that is why i am asking. The other inquiry ... as the system is embedded we have not so much disk space available. So, i need a minimal asterisk installation. When compiled and stripped, the biggest amount of space is taken by the modules. My question is, can asterisk work with just the chan_sip.so, chan_h323.so and the codec_*.so? is there any other module needed? I need only be able to bridge sip to h323, no extra fancy stuff needed (parking, echo, blah, blah, ... ) Tks a lot! Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960 part numbers ...
Hi,I requested a quote from a cisco reseller (or something like this) for 2 cisco 7960 phones, ideally preloaded with SIP firmware ... and i got the quote back with: 1x CP-7960-CH1 and 1x CP-7960-CCME. My question is, what is the difference between the two? If these are not the part number for the pre-loaded SIP phones, what part number is the correct? and what about the service contract ... when is it needed? Thank you very much ...Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] alcatel ip touch 4068 ... sip?
Hi, Quickie ... is the alcatel ip touch 4068 (or any other in that series) sip enabled? If not, does alcatel have a sip-enabled phone? Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] Asterisk and H.323
On 7/24/06, Aaron Anderson [EMAIL PROTECTED] wrote: I have been messing with both all day. I think what might be tripping me up is the extensions.conf. i do think so too :) I was able to receive an incoming connection from the client, but all the system returned was a busy signal. This call was to a known good number (my phone) so I'm not sure what's wrong. if you got the incoming connection but then got a busy tone, it is because the peer info in your asterisk conf is not good ... try the standard demo setup in asterisk ... there you can call extension 600 (i think) and you get an echo test ... later, extend this to call whoever you want ... but you'll need to enter the location info somehow (i never did it ... ). I used sjphone for testing ... try it ... you can add an extension to dial directly to the IP address of the sjphone. So, you dial the MMM extension on your phone, get to asterisk and then asterisk dials the IP address of sjphone ... nice and easy for testing. Will gnugk do a translation from h.323 to sip so I don't have to make any major modifications? Is there an example gnugk.ini and h323.conf file I can look at to get this all running? gnugk has no clue of sip. gnugk is a gatekeeper for h323 ... you can do stuff with the dialled numbers ... forwarding the call to various [asterisk] h323 gateways and the like ... but the conversion of h323 to sip is done in asterisk. Cesc Thank you in advance Aaron [EMAIL PROTECTED] wrote: Hello, Try both chan_oh323 and gnugk . Harry --- Aaron Anderson [EMAIL PROTECTED] a écrit : I have been scouring the net the last couple of days looking for some kind of tutorial or walkthrough on setting up a h.323 channel in asterisk. What I need to do is basically this: I have a client who wants to be able to connect to me via h.323 and make a local phone call (local to me, he is in a different country). The call is an automated process and no callee interaction is required. My client simply wants to be able to call a user and give them a verification number and then hang up. He's using some in-house software so unfortunately, h.323 is his only option. Can someone point me to a doc or perhaps give me a simple breakdown of what I need to add to asterisk in order to be able to do this? I am on a tight deadline and my searches have not revealed the information I am looking for. I have built chan_h323 and it is loaded but I'm not sure how to set it up beyond that. Any help would be much appreciated. Thank you Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This message was scanned by Sunny-Net SpamVirus Detector Gateway If this is not spam but was Maked with SPAM in the subject line please forward this e-mail to [EMAIL PROTECTED] if this is spam but was not marked spam please forward the e-mail to [EMAIL PROTECTED] Thankyou ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] Asterisk and H.323
On 7/24/06, Aaron Anderson [EMAIL PROTECTED] wrote: Thanks for the response. I have been able to now receive calls over h.323 using sjphone through the built in ooh323 channel driver. It seems to work ok for a bit but then asterisk seems to stop accepting connections and the server needs to be rebooted. huh? i didn't do extensive testing ... but on my 1.2.9.1 it seemed to work in a stable way ... i would reboot every now and then, but it would easily resist more than a few calls ... Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip to h323 ... direct RTP?
On 6/21/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - Johansson Olle E [EMAIL PROTECTED] wrote: No. It's certainly possible but at this time there's no interaction between the RTP clients, the various channel drivers. I believe this is incorrect; all the RTP-using channel drivers supply 'ast_rtp_bridge' as their native bridge method, so assuming they also implement the 'set_rtp_peer' method, then an RTP native bridge between dissimilar channels should work fine. If the channel driver(s) also support sending the RTP peer address to the endpoint (as chan_sip does with reinvite), then a direct media path should also be possible. Sorry, but I am confused. What does this mean for the bare-bones user like me? That technically it would be possible but that it is not implemented? (i use the sip and h323 channels shipped with the latest sources tarball) Or that it is possible to configure via some obscure setup file? Thanks! Cesc -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing with multiple servers
In general, you are talking of distributed conferencing, which in SIP it was tried once to standardize but never reached anything. It is just not commercially popular, i guess. Now, this doesn't mean that it cannot be done or that it has not been done ... but it is propietary implementations. And in my particular knowledge, i know asterisk was not the choice. Just my 2 cents. Cesc On 6/20/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Patrick [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 12:05 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Conferencing with multiple servers On Tue, 2006-06-20 at 15:22 +0100, Wildheart wrote: Hi, I am trying to join 2 asterisk servers together using a sip channel. This is so, if a user joins a conference on box A and another user joins a conference on box B, providing they are in the same conference room, the two conferences are joined via the sip channel. We only want to join the conferences together if they have users in them and we don't want to point all the conferences to one server as we would like to try to balance the load a bit. This is a general problem with the 'enterprise grade' aspects of Asterisk. As far as I know, there is no way to distribute applications (eg: Queue, Meetme etc) between multiple Asterisk systems. You really need to run the applications that will serve a common set of phones on the same Asterisk system, and then fail over to a secondary if necessary. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk h323
Hi, I am also testing asterisk with H323, with the channel included in the latest sources. It works ( i had some problems with media configuration when calling from an SJPhone ... but it seems more an SJPhone problem than asterisk). I also bridged from SIP to H323 ... it works fine. I have a question for those out there ... i compiled with pwlib 1.9.2 and openh323 1.17.3. These are the versions mentioned in the README file for the h323 channel. Now, has anyone tried any newer version? I would be more comfortable using the latest stable release for these required libraries with asterisk ... just to make sure i get the best. Any experiences?? Cesc On 6/20/06, Alberto Sagredo [EMAIL PROTECTED] wrote: Im using several Asterisk Box with chanh323 from asterisk, and it works fine. Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability. A fail (crash) last month with about 600 calls per day. Regards Alberto Sagredo hakem voip escribió: You can do this by installing a h323 module. Conversion works simetimes good, sometimes not good. H323 behaviour on asterosk with my experience with kind of unpredictable. 2006/6/20, Khaled Chehab [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi Can asterisk work as sip and h323 protocol in the same time ,and how is the conversion protocol works . Please if u know send me how to active h323 protocol or the conversion protocol Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Hakem Voip ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 soft phone known to be run with asterisk.
I had problems with sjphone ... same version as yours. Finally, i managed to solve it by: - in sjphone, media channels settings: untick Use remote codec preferences and Open audio streams after remote opened ... it was trial-error ... now it works (to Echo and Sip-H323 call). - in asterisk, h323.conf ... the codec configuration ... i commented all lines related to it ... ;disallow=all ;allow=all ;allow=gsm ;disallow=g723.1 (just in case) (again, trial-error) Cesc On 6/21/06, Pawel [EMAIL PROTECTED] wrote: Hallo group members Could You tell me a h.323 soft phone that runs well with asterisk. I tried the following so far, but in general I cannot compile them (fc.3) or I cannot configure them to run with asterisk: http://www.pacphone.com/downloads/PacPhoneSetup117.exe - cannot configure http://www.sjlabs.com/softphone/SJphone-289a.exe - cannot configure http://cphone.sourceforge.net/ - cannot compile http://www.ekiga.org/ - cannot compile http://www.openh323.org/ - cannot compile Greetings. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip to h323 ... direct RTP?
Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk just as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone to h323 phone). It should be possible ... but is it possible with asterisk? Thanks! Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip to h323 gateway ...
Hi, I am familiar with asterisk, though never actually tinkered with one myself ... so i don't know the full extent of its capabilities. I am facing a request to bridge a sip network and an h323 network. I would like to operate the sip with ser as the proxy and some gatekeeper on the h323 side (not required though). Actually, i have a few more points that may make it simpler - i do not need codec negotiation: both sides are configured use the same (g711 alaw) by default. - I have just a few phones on each side, so even static routing can work, if that is of any help. - it is not a production environment, for now. It is a demo/lab The question is ... can asterisk do the job? Ideally, the bridge would be only signalling-wise (rtp to be direct end-to-end). But, if someone had bad experience with this and would recommend to use a B2BUA approach, please, tell me. I don't know if it makes a difference, but most of the calls would go from the H323 side to the SIP side ... but i don't really want to restrict SIP-H323. Thanks a lot! Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] open source sip softphone (Window OS version )
minisip (http://www.minisip.org) - it is LGPL, GPL ... windows support is in testing status On 6/15/06, Kerry Garrison [EMAIL PROTECTED] wrote: None of those are open source that I recall. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: Thursday, June 15, 2006 6:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] open source sip softphone (Window OS version ) Asterisk guy wrote: are there any open source sip softphone (Window OS version )? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users xlite sjphone firefly (3rd party version) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip to h323 gateway ...
So, asterisk does the bridging ... I asked on another list and the answer was that asterisk could not do the job :O The truth is that my setup should be fairly simple ... i do not need any cool feature (voicemail and the like). I just need to call from one side to the other, for a reduced amount of users (so name mapping could even be manual ... no problem). Cesc On 6/15/06, Gary Richardson [EMAIL PROTECTED] wrote: I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an asterisk SIP setup. It works. There are issues, but that has more to do with Unity voicemail than the h323 implementations. On 6/15/06, Cesc [EMAIL PROTECTED] wrote: Hi, I am familiar with asterisk, though never actually tinkered with one myself ... so i don't know the full extent of its capabilities. I am facing a request to bridge a sip network and an h323 network. I would like to operate the sip with ser as the proxy and some gatekeeper on the h323 side (not required though). Actually, i have a few more points that may make it simpler - i do not need codec negotiation: both sides are configured use the same (g711 alaw) by default. - I have just a few phones on each side, so even static routing can work, if that is of any help. - it is not a production environment, for now. It is a demo/lab The question is ... can asterisk do the job? Ideally, the bridge would be only signalling-wise (rtp to be direct end-to-end). But, if someone had bad experience with this and would recommend to use a B2BUA approach, please, tell me. I don't know if it makes a difference, but most of the calls would go from the H323 side to the SIP side ... but i don't really want to restrict SIP-H323. Thanks a lot! Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip to h323 gateway ...
Ok,
I will trust you guys :) But it is puzzling that on another list (SER)
someone (actually 2 people) told me that asterisk would not do the
job. One suggested Yate ... any opinion? (I know ... not nice to ask
in the asterisk list ... but it is good to compare, i think).
Now, just to make sure that if it works for you it does for me ...
what version of asterisk you are running? :D
Cesc
On 6/16/06, Tigran Kocharyan [EMAIL PROTECTED] wrote:
It should do the job!
In my setup, I call from an IAX phone to an h323 Gateway, and all is
fine. The opposite direction also works fine.
Though this is an IAX setup, SIP should perform likewise.
REgards,
Hohenzolern
Gary Richardson wrote:
Nope, asterisk does the bridging. Asterisk can talk to SIP phones and
H323 gateways/phones. It can also cross connect them.
Since I have SIP users plugged into asterisk, I have a dial plan that
looks something like:
exten = 100,1,Macro(local_sip_user,SIP/bill)
exten = 101,1,Macro(local_sip_user,SIP/bob)
exten = 102,1,Macro(local_sip_user,SIP/steve)
exten = _XXX,1,Macro(call_ccm,${EXTEN})
exten = _8XXX,1,Macro(call_ccm,${EXTEN:1})
So, if you dial 100-102, you get a sip call, but if you dial 103, it
would try to dial my CCM. If you dial 8100, it would call CCM anyway.
From the cisco side, I have some similar logic. That's pretty much it.
On 6/15/06, *Cesc* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
So, asterisk does the bridging ... I asked on another list and the
answer was that asterisk could not do the job :O
The truth is that my setup should be fairly simple ... i do not need
any cool feature (voicemail and the like). I just need to call from
one side to the other, for a reduced amount of users (so name mapping
could even be manual ... no problem).
Cesc
On 6/15/06, Gary Richardson [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an
asterisk SIP
setup. It works. There are issues, but that has more to do with
Unity
voicemail than the h323 implementations.
On 6/15/06, Cesc [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I am familiar with asterisk, though never actually tinkered with
one
myself ... so i don't know the full extent of its capabilities.
I am facing a request to bridge a sip network and an h323 network.
I would like to operate the sip with ser as the proxy and some
gatekeeper on the h323 side (not required though).
Actually, i have a few more points that may make it simpler
- i do not need codec negotiation: both sides are configured use
the same (g711 alaw) by default.
- I have just a few phones on each side, so even static routing
can work, if that is of any help.
- it is not a production environment, for now. It is a demo/lab
The question is ... can asterisk do the job?
Ideally, the bridge would be only signalling-wise (rtp to be direct
end-to-end). But, if someone had bad experience with this and would
recommend to use a B2BUA approach, please, tell me.
I don't know if it makes a difference, but most of the calls
would go
from the H323 side to the SIP side ... but i don't really want to
restrict SIP-H323.
Thanks a lot!
Cesc
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