search for: kchua

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

hi guys! i can't seem to find where in the code the system puts the "system alias" i have a multiple protocol voip network wherein i run asterisk along with ser and cisco callmanager. when i call through h.323, the callerid name always shows up as "asterisk XXXX" (i wanna change "asterisk") where XXXX is a 4-digit extension defined in extensions.conf and when i

hi guys! i'm going to share a workaround for authentication from msn messenger, you have to change two lines in chan_sip.c msn messenger is known to look for the correct realm in authentication, therefore, change the realm in chan_sip.c, line 2061 and line 2910 (release 0.4.0) i hope the realm can be parsed from extensions.conf in the next release... ~kelvin =) -------------- next part

hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030725/ae4b2f25/attachment.htm

there's this one way audio problem using h323 (CVS) with cisco callmanager? has anybody encountered this problem? oh323 works ok though... or is there any workaround for this? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030731/ea855e43/attachment.htm

hi guys, i'm in need of several 24port or higher fxs device which supports sip, aside from mediatrix and audiocodes (cisco's vg248 doesn't support sip), do you have any idea who else manufactures such device? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030801/67eb12dd/attachment.htm

hi guys, i'm encountering one way audio on cvs using netmeeting and chan_h323.so is there a quick fix or workaround for this? compiled using openh323 1.12 pwlib 1.5 i also saw this in earlier version of openh323 and pwlib.... thanks for any info ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL:

guys, what firmware version of audiocodes mp104 fxs are you using with asterisk? i'm having sip stack problems. ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/50d5f5de/attachment.htm

hi guys, i'm using sept 30 cvs and oh323 5.5 i'm having 5 second latecy(on only 1 audio path) when a call is transferred.... the scenario is this: sip--------->asterisk----->h323:operator (who then transfers the call) ---------------->h323:destination ------------------audio path 5-second latency---------------->

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

* compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled --