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...org/svn/asterisk/tags/1.4.25/ChangeLog The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help! * Allow H.323 Plus library to be used in addition to the OpenH323 library. - Closes issue #11261. Reported by vhatz. Patched by jthurman. * Delay signalling progress until a PRI channel really signals progress. - Closes issue #13034. Reported, patched, and tested by klaus3000. * Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. - Closes issue #14373. Reported, and pa...

Hi, A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

....com/svn/asterisk/tags/1.6.0.7/ChangeLog The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help! * Allow H.323 Plus library to be used in addition to the OpenH323 library - Closes issue #11261. Submitted by vhatz. Patched by jthurman. * Make the sip_standard_port function more granular by allowing separate type and port arguments. - Closes issue #12761. Reported and patched by asbestoshead. * Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary. - Closes issue #14538. Report...

...org/svn/asterisk/tags/1.4.25/ChangeLog The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help! * Allow H.323 Plus library to be used in addition to the OpenH323 library. - Closes issue #11261. Reported by vhatz. Patched by jthurman. * Delay signalling progress until a PRI channel really signals progress. - Closes issue #13034. Reported, patched, and tested by klaus3000. * Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. - Closes issue #14373. Reported, and pa...

...oses issue #11583. Submitted by sobomax. Tested and additional coding by sobomax, dvossel, murf. * Update app_fax to work with spandsp-0.0.6 - Closes issue #13688. Reported by and patched by irroot. * chan_h323 with H323Plus for TRUNK (SVN rev. 89183) - Closes issue #11261. Reported by vhatz. Patched by jthurman. * Wrong usage of sscanf with use of uninitialized variable caused accidental parsing of RTP/SAVP - Closes issue #14000. Reported and patched by folke. * Realtime peers are never qualified after 'sip reload' - Closes issue #14196. Reported, tested, and patch...

...oses issue #11583. Submitted by sobomax. Tested and additional coding by sobomax, dvossel, murf. * Update app_fax to work with spandsp-0.0.6 - Closes issue #13688. Reported by and patched by irroot. * chan_h323 with H323Plus for TRUNK (SVN rev. 89183) - Closes issue #11261. Reported by vhatz. Patched by jthurman. * Wrong usage of sscanf with use of uninitialized variable caused accidental parsing of RTP/SAVP - Closes issue #14000. Reported and patched by folke. * Realtime peers are never qualified after 'sip reload' - Closes issue #14196. Reported, tested, and patch...

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

Hello, Does anyone know whether chan_sip in Asterisk supports DTMF in clock rates other than 8000? I looked for telephone-event/16000 in the changelog and in Jira but no luck. Any help would be appreciated. -- Best regards, Vlasis Chatzistavrou.

Hello all, Sorry for the slightly off-topic issue, I need to have a capture from a network sniffer (like Ethereal for example) from a call setup with the MGCP protocol. I thought that since Asterisk now supports MGCP some of the people who develop the MGCP channel driver may have such a capture available. I need this for my MSc thesis and unfortunately, I don't have any MGCP compliant

Hello, I have been wrestling with installing the CAPI drivers for AVM Fritz in order to use chan_capi with Asterisk. I use an SMP machine, RH 9. I have found rpm's for CAPI and AVM drivers (namely: capi4k-utils-2003.06.16-08.mungo.RH9.i686.rpm and kernel-2.4.20-8- avmfcpci-03.11.02-08.mungo.RH9.i686.rpm), but I believe that they support only single processor machines. I've already

Hello, I would like to ask if anyone has solved the problem with Asterisk+ISDN4Linux cards, where there are no call progress tones or announcements from the PSTN when we dial ouot through the i4l card. For the moment, if we don't inject the r option in the Dial command, there is only silence during the call negotiation... Using Asterisk RC2 with Eicon passive PCI 2.01 card... Thanks for

Hello, I use Asterisk CVS-v1-0-12/21/04-11:05:29 and I noticed that when we receive calls from a partner's IP address (who has a static host entry in the sip.conf file) but the RTP comes from a different address than the signaling, our * sends a 403 forbidden message and drops the call. This problem does not llow us to receive calls from SIP proxies. Was this fixed in newer versions of

Hello, I have used the Dial option 'r' before in older Asterisk versions and it behaved as expected, that is, Asterisk would always give ringback audio before the call was answered no matter what. It has been a while that I have used version 1.4.33.1 without any the 'r' option. Now that I had to use it for a while, I noticed that 'r' would not give ANY audio until the

Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2

Hello, I'd like to ask if there are any news about CE certification of the E1 boards. I know that the T1 boards are FCC certified but I'd also like to know what is the status for CE certification. Thanks for any input, Vlasis Hatzistavrou.

Hello, I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 installed but failed. I applied the patch to the required OpenH323 library according to the instructions, and set the proper directories in the Makefile. Here is what I receive after I issue make: ******************************* g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -