Thanks Liviu and Bill, Sorry for a late reply...had to bike across the cityLiviu - I've got canreinvite=no on the [general] as well as for each peer createdBill - handset, or speakerphone? handset connected to the Dlink ATA and x-lite have the same behaviour --> the audio is not being transmitted at all. Asterisk CLI as well as the devices show the call is connected between both legs HTHThanks again for your input and have a great week end! Claudius
Date: Fri, 1 Jun 2007 21:04:55 -0400From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [on-asterisk] What am I doing wrong? No Audio on remote SIP clientsHave you triedcanreinvite=noin all your peers ?LiviuOn 6/1/07, Bill Sandiford <[EMAIL PROTECTED]> wrote: > > This may sound like a stupid question but.> > Is this a call with the handset, or speakerphone? If the handset, are you> actually hanging up after the other side hangs up?> > Bill> ----- Original Message ----- > From: Claudius Fortis > To: [email protected] > Sent: Friday, June 01, 2007 1:49 AM> Subject: [on-asterisk] What am I doing wrong? No Audio on remote SIP clients > > Hello everyone,> > Thought I'd shoot out this question which may appear basic for most of the> guys/gals here.......hey, I'm no expert myself, so pardon my ignorance!> > I'm having trouble getting audio from a particular remote location.> > - I've got my Asterisk (1.2.18) running without no problem (been running for> about a year)> - Asterisk is running behind my linksys NAT box > - Asterisk is on a DMZ> - "I'm using a 3web DSL connection with dynamic IPs (which changes not too> often)"> - SIP clients are internal and external >> register without any problem > > - But I've noticed that a friend of mine would register but receives no> audio in both ways (inbound or outbound); the only thing that tells me his> device (X-Lite and/or DLINK Vonage Adaptor) registers and makes SIP activity > (call in/out progress) is my Asterisk CLI.> > - For the sake of it, I installed a local Asterisk (1.4.0) at his location> and did set up a sample ECHO-test app which I pointed out a DID directly > from my Provider to his box: This box needs to register w/ DID provider via> SIP> > - When I dial 647.555.1111, the echo-test app executes successfully, I can> hear it, so can it (no problem communicating ) > - This second Asterisk box is also behind NAT device and DMZ on a Cable> connection> > > The problem is:> - When his SIP devices connects to my box > registration successfully but no > AUDIO in/out> - When his SIP devices connects internally to his box >> same problem> - When I (from my network) SIP-connects to his >> same problem> > > My Question: > - Am I missing something here?> > You may ask if I tried a SIP device from a different network and see how> that works? Not yet, but I'm going to try tomorrow. But I know for a fact> that my internal SIP devices work fine (via internal or public IP of > Asterisk box)> > Thanks again for your input and have a great one!> > Claudius> > > > > ________________________________> Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE
