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...ent settings...)? I personnaly think we can rule out network config, since both successful and unsuccessful users work in the same lan. Is there anybody with experience in using channel gtalk? Should we start debugging? What can we learn from jabber debug logs? Any help is very much appreciated! koenvi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070621/bcb4b581/attachment.htm

I have a TE110P connected in euroisdn as pri-cpe. When I dial out from a sip phone to a number over the pri, I get an error Unable to set channel 1 (index 0) to linear mode On the destination phone, I only get a terrible noise when answering the call. There doesn't seem to be a speech path... Config: libpri 1.2.2 - zaptel 1.2.5 - asterisk 1.2.6 on Fedora Core 4 (kernel 2.6.11) Anybody

If I edit the value of a global variable in my dialplan, could there be a risk of collision between calls? More in details: could a global var be used to build a counter that will be incremented by every call that passes. I think when 2 calls come in almost sumiltaneously, they could both be incrementing and saving the same value... which is bad! Anybody knows how asterisk handles this? K

Is there a way to use 'application mapping' from features.conf without the built in features (pickup, blind transfer, etc.) nor call parking? I have been trying to comment out everything in features.conf, but my asterisk stills shows the defaults... Koen -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have the most stupid problem in my dialplan. I need to do something as trivial as splitting a string, with a semicolon as separator. I was thinking the 'CUT' function would be perfect for this. But the problem is the semicolon. In the dialplan it is always understood as a separator for parameters. What I have tried so far: [macro-eva-on-sip] exten => s,1,NoOp(${CALLERID(name)})

I'm trying to connect my asterisk 1.4.6 to a system that provides video content (through SIP). Problem is my video system only speaks H263-2000 version (aka H263++). As far as I can see, * only understands H263 and H263+ in and sdp. Can anybody tell me how to extend asterisk so it'll support H263++?

Hi, I wonder if anyone is using Digium's TE110P card on an E1 connection. I have been try to, but so far it wasn't much of a success. It only works more or less in EuroISDN as PRI CPE. And even that config gives me some trouble with channel negotiation. My current config: *zaptel.conf:* span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=be defaultzone=be *zapata.conf:* [trunkgroups]

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: