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hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

Hello, I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring but I don't hear the caller and the caller doesn't hear me (all IP Phones have the same problem). This problem appear also if the call is directly send to the second E1 of the digium card who is connected to an IVR. It does not depand on the charge of the server (I have the problem with only one call).

My SIP seems to be configured correctly as I can dial out and my minutes show up on my broadvoice bill, but whenever anyone calls my broadvoice # inbound they just get a busy signal. I dont get anything in the logs saying anything came in from broadvoice at all. Has anyone had this/simmilar problem with inbound from Broadvoice? Any suggestions? Thanks Neri -------------- next part

Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal.

Hello We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since. We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: It does show up in asterisk a few seconds after the UIP200 reboot: -- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200 but

Hi, People here often work on 2-3 places (office 1, office 2 and home). I would like to give them 1 extension (XXX) and to ask them to 'register' the phone they use at a certain moment. The idea is that, when you need someone, just dial XXX and the phone near him (in Office 1, Office 2 or at Home), will ring. This will keep my queue system and other tricks intact, where I always use the

Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could possibly be wrong? If it were interrupt overload, I'd hear lots of cracks in my echo, right? I'm not hearing that. Besides, a telephony card is not involved.

I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context: [callback] exten=> 501,1,Congestion() exten=> 501,2,Hangup() exten =>h,1,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =>h,2,Hangup() With the above, the call comes into the trigger number, then the call