similar to: Does Asterisk just pass thru RTP if the codec is the same between two extensions?

Hi all, I'd like to know how Asterisk process a RTP data flow. Is there any clue to find out about this? The rtp.c? Thanks. Regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/1572210f/attachment.htm

Hi all, I'd like to know how Asterisk process a RTP data flow. Is there any clue to find out about this? The rtp.c? Thanks. Regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/6386cacd/attachment.htm

Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456 at default:1] AGI("SIP/100-00000000", "call.php") in new stack -- Launched AGI Script

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

I want to make sure that RTP is not going thru my asterisk. I read you should avoid in the dial commands: m music while ringing t,T transfer calls from caller and called party What else do I need to take care? remote phone ===> registered to local asterisk ===> calling remote gateway should have the RTP remote phone ===(RTP)==> calling remote gateway bye Ronald

Hello All, Currently my setup uses Xlite and Asterisk and i found that all the RTP voice packets are transfered via the asterisk server from one xlite to another. Is there any possibility that we can make all the RTP Packets to be transfered directly between the two clients once the connection is established?. Any one please help me. Thanks and Regards, Senthil Murugan.V

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used "canreinvite=yes" but it didn't work. Description from asterisk conf. File; (canreinvite=yes ; allow RTP voice traffic to bypass Asterisk) Thanks Erdem HAKI - erdemh@tesas.com -------------- next part

This is a problem that is completely stumping me, and my understanding of Asterisk dialplans tells me this should never be a problem. Moreover, this scenario works on Asterisk 1.4 but not 1.6. We have a customer with several Aastra 6731 phones. They want incoming calls from the PSTN to work and they also want to be able to call each other "internally" on a special non-DID number (like

After just having update to the latest CVS I am getting the following message when I call VoicemailMain(): -- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing

Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? cheers, Aby Azid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080424/b442d5af/attachment.htm

What happens if asterisk receives a T.38 call? Will asterisk pass it thru? I've seen a few ATA devices that support T.38 and I'm wondering what happens if a fax is sent thru one of these ATAs into asterisk. Maby I have the terminology wrong. Is T.38 a protocol like SIP or is T.38 a compression like G729 using SIP? Thanks, Matthew

I've configured asterisk for pass-thru mode according the following two URLS: http://voip-info.org/wiki-Asterisk+G.729+pass-thru http://lists.digium.com/pipermail/asterisk-users/2004-March/039663.html I don't believe I have it working, as show channels reports that my calls are bridged. I'm assuming that bridged is the opposite of passed-thru...is this correct?

Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension

Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in "pass-thru" mode. I mean setup like this: Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same

Hello, I'd like to use g729 pass-thru when I dial out to a sip provider from my IP phone but because I have no license for g729 I'd like to use g711 ulaw for asterisk voicemail, conference bridge and other services. When I set in [general] section of sip.conf the following: disalow=all allow=g729 allow=ulaw the g279 pass-thru works fine with my SIP provider but when I call the

Hello, I have just updated my Asterisk installation from 1.2x to 1.4 (on FreeBSD) - mostly everything seem to work fine. However, I use G.729 pass-thru - and I have before successfully used the following setup: http://www.voip-info.org/wiki/index.php?page=Asterisk%20G.729%20pass-thru However, it is not working with 1.4 - I see the following errors: [Apr 16 15:59:24] WARNING[10139]:

Hello I am having a weird issue with a PCI-X Intel dual gigabit ethernet nic. While I am partially successful in pass thru with kvm, I have a weird issue with my pfsense freebsd vm. It see the two nics but gives them the same Mac address. I'm not sure if this is a pfsense bug or what. The nic works for everything but this Mac address issue. Long story short, I guess what I'm asking is

Dear all, Did anyone successfully test T38 fax pass thru to Cisco as53xx? We've tried 1.2.4 with latest patch and latest svn trunk and T38 patch but still not work. Reinvites from Cisco are correctly passed back to the originating gateway, but fax never able to connect. Cisco IOS 12.3.x configuration voice service voip fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback