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...(.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

...t. 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 <cesc.santa at gmail.com> Date: Fri, Nov 28, 2008 at 3:26 PM Subject: Asterisk RTP pager To: Andreas.Brodmann at gmail.com Hi, I came across your "RTPpage" application and just made me very happy. If I may, some questions....

...9;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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090406/66d58337/attachment.htm

...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

...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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/a7bcdda1/attachment.htm

It seems renouveau doesn't cope well with missing GLX extension, unlike e.g. glxgears: rmh at cesc:~/renouveau$ glxgears Xlib: extension "GLX" missing on display ":0.0". Error: couldn't get an RGB, Double-buffered visual rmh at cesc:~/renouveau$ ./renouveau detect_devices: Creating probe window failed. We tried to create a window by using SDL. Our OpenGL tests require a...

...require(multcomp) summary(glht(a1,linfct=mcp(factortmnt="Tukey"))) The fact is that once I get both results, there are some occasions in which I get lower p values with Tukeys correction than in paired t-tests. How is it possible? Isn't Tukey more restrictive than paired t-tests? Cesc [[alternative HTML version deleted]]

...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

...hutdown=1, restart=0) at asterisk.c:945 #18 0x080be019 in handle_shutdown_now (fd=1, argc=2, argv=0xbffff830) 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=0xbffffd84) at asterisk.c:1019 (gdb) Regards, Cesc

...323/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

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

We are trying to install Samba 4 on a RHEL4 update 4 machine and are facing problems. We have downloaded the samba4 tar ball from http://repo.or.cz/w/Samba.git/snapshot/master.tar.gz After untarring it we have done cd source4 ./autogen.sh ./configure But at this stage itself we are getting the following error: /root/Samba/source4/wscript: error: Traceback (most recent call last): File