Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030819/9b0f11d4/attachment.htm
Seeing as no one else has replied, I figured I may give it a shot. At least it'll start something. Now, correct me if I'm wrong someone, but as far as I understand in this situation you can do both. Normally the RTP packets would be swtiched through *, but you can set in you sip.conf file the 'canreinvite=yes' option which will allow the RTP stream to be direct if a compatible codec is negotiated. I'll double check if I ever get my server up and running again. J On Tue, 19 Aug 2003 11:17:20 -0500 "Jorge Cisneros Flores" <jorge@redenlaces.com.mx> wrote:>Hi > > > Is posible to make a call from site A to Site C, and >my question is, the rtp data is from A to C or is from A >to B to C > > > > > Site A Site B > Site C > ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 > >ThanksRegards, Jamie Carl Jazz Inc. Email: me@jazz-inc.net Web: www.jazz-inc.net Phone: +61-414-365-466 Jabber: jazz@netmindz.net
On Tue, 19 Aug 2003, Jorge Cisneros Flores wrote:> Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C > > > Site A Site B Site C > ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186That depends. Initially, RTP data is sent via Asterisk (B). Asterisk will then try to reinvite A and C so RTP is sent directly between them. If it succeeds, RTP is stent only between A and C, if it doesn't, asterisk will act as a proxy. If the reinvite succeeds but a firewall is blocking the direct RTP stream, the call will get silent. You'll have to put "canreinvite=no" in the sip.conf for at least one of the ata186 devices in that case, so it doesn't even try... Cheers, Siggi
Is there a way for a natted client with a dynamic ip address to receive call from the asterisk box ? I can call from the natted phone using tasterisk but I can't receive call in the natted phone because * does not know the ip address of the phone I have enabled the registration but when I launch the show peers I have: 281/281 (Unspecified) (D) 255.255.255.255 0 Unmonitored instead in the local network phone I have specified the ip address. Someone can help me ? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040903/5bf3ec5b/attachment.htm
Hi, I have a provider sending me data through SIP, but with no registration. (there are constraints that forces us to work like this). And, as far as I am concerned, that's fine. Here is the relevant portion of my SIP.conf file. [5149075555] context=5149075555-inbound type=friend host=11.222.222.23 language=fr disallow=all allow=ulaw dtmf=rcf2833 Basically, I understand that I am saying everything coming in from 11.222.222.23 should be sent to the context 5149075555-inbound. Right? If it is, how do I ask this provider for another DID, let`s say 555-555-5555, and send those calls ina different context (let`s say 5555555555-inbound)??? There doesn't seem to be a way of differenciating between calls meant for 5149075555 and 5555555555, since they both come in from the same provider (hence same IP address). What am I missing to treat those calls differently? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060203/15c6f16f/attachment.htm
Benjamin, Thanks a lot for the answer. Sometimes the obvious escapes me, and this was the case here. Regards, Mike> I'd change your definition to something like > > [providerX] > context=providerX-inbound > host=11.222.222.23 > > in your providerX-inbound context you can match the different > extensions > > [providerX-inbound] > exten => 5149075555,1,NoOp(5149075555) > exten => 5555555555,1,NoOp(5555555555) > > Now a question I've always wondered, What if providerX uses > multiple IPs. Is > there any way to specify a range of IPs for the "host" in sip.conf ? > So far I've had to make a sip entry for each IP my provider uses.