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...re in conversation stop hearing each other. I can hear that they cant hear each other. We are using chanspy and passing the exact session-name to it as an argument. Asterisk 1.6.0.6 is being used a B2BUA. It receives calls from users connected via analog phones to a Quintum and forwards them to a PSTNGW. Analogphone--Quintim<------ SIP------->Asterisk<-----SIP------>PSTNGW(T1s) All media is passing thru asterisk (canreinvite = no). Thanks in advance for any help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asteris...

Search google with "sip pstn site:www.microsoft.com" You will find out how to configure LCS static routing to SIP Gateway, like Asterisk but you need patch Asterisk to support TCP. http://bugs.digium.com/view.php?id=4903 Step1: configure LCS 2005 to let sip uri: *@pstngw.domain to route to next hop: pstngw ip address Step2: patch your asterisk chan_sip.c to support TCP Step3: configure your Asterisk sip.conf, extensions.conf simple example :-) sip.conf context=sip_incoming extensions.conf [sip_incoming] exten => _XX.,1,Answer exten => _XX.,2,Noop(do trust...

hello, I would like to realize a VoIP Gateway, with some extra-features. The aim is to get the phone number of the caller, to make research in our database, and to put him automatically through the good employee. The company is equipped with a VoIP network : software : - PSTNgw - Ohphone - OpenGatekeeper hardware : - Quicknet phonejack - gateway : Voicetroniw OpenLine4 Is it possible to do it with Asterisk ? I don't know if Voicetronix cards are well supported, but it's ok to buy a new one if needed. Thanks, Fabrice Tereszkiewicz

Hi there, I?ve been out for some months now, haven?t been checking the list at all. Does anyone know if the problem with the Quicknet Linejack (FXO) card dial out to PSTN with asterisk was solved? Is anybody working on it? Cheers, -Z -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup

...thout needing a restart. What can this be? Surely Polycom is re-registering every 3600 before Asterisk times it out. But Asterisk is just refusing it. By the way, anyone know whether Asterisk is geared towards RFC3261 or RFC2543? I know Asterisk is not a fully SIP Proxy but lets say if a SIP PSTNGW or a SIP phone is designed under the spec 2543 as suppose to 3261, will it work better or the same with Asterisk? David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040204/662e3e56/attachment.htm

...ension rings. When we pick up to spy, parties involved in the call that we are spying on are suddenly unable to hear each other. This happens as soon as we start spying... why is this happening? We are using asterisk 1.6.0.5 the setup is: analog phone<->quintum<-->asterisk<--->PSTNGW This makes no sense since all we are doing is "listening' to the call which is in progress. Why would the people suddenly stop hearing each other. The options to chanspy we are using are "q". We also tried "qb" but that makes no difference. Thanks for any help ------...