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

[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

[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

[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