Re: [Asterisk-Users] SIP -h.323
Hi Ryan! Interesting what experience you have made in this issue. We have setup the alternative channel for H.323 (the * built in chan_h323), and we are now in a testing phase. I was wondering (in case no transcoding is needed), how your setup treats the RTP streams. Do the RTP streams go end-to-end or always via Asterisk? Another question I'd be interested in: Have you also gained some experience with bridging _video_ calls between H.323 and SIP? cheers, Bernie PS: I'd be glad, if I also could get the relevant config files from you. On Fri, 13 Aug 2004, Ryan Wilkins wrote: Yes, it can.. I'm doing it at my home. My current setup is Asterisk-1.0-RC2 using the oh323 driver. I have a SIP connection to Broadvoice talking to Asterisk. I have a e-tel (now Qtelnet) H.323 VoIP telephone adapter as my end point talking to Asterisk. For processing sake, you may want to keep your codec the same all the way through. Originally I ran G.711u on the SIP connection and G.711a on the H.323 connection. It worked just fine but the logs always said something about transcoding between u-law and a-law. I reset the H.323 link to G.711u and now it says nothing about transcoding. In theory you would lose a bit of audio quality in the translation process. In reality I don't really know. email me privately if you want a sample config. Ryan Wilkins On Fri, 13 Aug 2004, Yiannis Costopoulos, Web2Net Solutions Ltd. wrote: is there a definite answer if asterisk can pass calls between SIP and h.323 protocols? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video Calls between SIP and H.323
Hi! Does anybody have experience with * as a video gateway between SIP and H.323? We managed successfully to make voice calls between the two worlds). However, we have noticed, that * stays in between the RTP streams (as an RTP proxy). Is this the mormal behavior or can this be avoided somehow, in the sense, that the RTP streams flow end-2-end? We did not manage, to get any video streams to work, e.g. making a video call from SIP to H.323, * simply does not offer the video capabilities to the H.323 side. In sip.conf and h323.conf I have set the codecs h261 and h263 to allowed and the option videosupport=yes ? Did I forget anything in the config? Looking forward to your answers! cheers, Bernie PS: I am using * version 1.0-1 0, which is in the current default in the Debian Testing distribution. I am using chan_h323 (the * built in), (but not chan_oh323). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hwo to get CallerID: SIP - ISDN
Hi! I trying to configure * in a way, that it uses a different CLIP (Caller-Id in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far always the main (1st) number of the number-block is sent to the ISDN. I have a E100P from Digium and use the zapata stuff (chan_zap). All SIP calls are coming through an SER. One idea I had in mind is to assign userid's in SIP, that match the extension of the number block, e.g. 854. * could then take the user part of the From header field of the incoming SIP INVITE and relay this numeric user part (e.g. 854) to the chan_zap, so that the CLIP in the ISDN appears as the number assigned to SIP user. Another idea I had was ENUM. But as in ENUM one can only resolve one way, i.e. E.164-number - SIP address, * would have to lookup the whole number block (every entry) from time to time and cache it in a mapping table. No so nice solution, I guess. Does anybody have some experience in this? Any hints, instructions and HowTo's are warmly welcome. cheers, Bernie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault, exit status 139, ...
Hi!
I am running an * 0.7.2 on an X86 debian stable 2.4.25 (with
backports.org). The HW I am using is Digium's E100P on an HP DL 380.
Quite often it crashes, e.g. after a call has finished. Below some logs
form the * Console as well as from the /var/log/asterisk/messages
(Replaced some stuff with XXX).
Any idea what there could be the reason for this segmentaion fault?
What other indormation (e.g. configs) would be required to analyse
this problem further?
Thanx for you help!
cheers,
Bernie
* Console:
Apr 5 18:01:18 WARNING[24594]: app_dial.c:331 wait_for_answer:
Unable to forward voice
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Dial(SIP/xxx.switch.ch-0894ea58,
Zap/g1/04176XXX) in new stack
-- Called g1/04176
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
== Spawn extension (SIP, +4176XXX, 1) exited non-zero on
'SIP/xxx.switch.ch-0894ea58'
astra*CLI
/usr/sbin/safe_asterisk: line 6: 20873 Segmentation fault
asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Disconnected from Asterisk server
[EMAIL PROTECTED]:/etc/asterisk$ /usr/sbin/safe_asterisk: line 6: 20905
Segmentation fault asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 6: 20925 Segmentation fault asterisk
${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: /dev/tty9: Input/output error
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
* /var/log/asterisk/messages:
Apr 5 18:01:18 WARNING[24594]: Unable to forward voice
Apr 5 18:01:34 ERROR[1024]: Unable to load config iax1.conf
Apr 5 18:01:34 WARNING[1024]: Ignoring port for now
Apr 5 18:01:47 WARNING[16401]: Timeout, but no rule 't' in context 'SIP'
Apr 5 18:01:57 ERROR[1024]: Unable to load config iax1.conf
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
