search for: t38passthrough

(This is a shameless copy-paste from the note I posted on http://bugs.digium.com/view.php?id=5090) I have again backported the whole T.38 shebang to the stable branch. The port was based on two versions of the t38passthrough branch: r19125, the latest unconflicted automerge, and r13623, the latest version without the new chan_sip flag structure. Basically, the port contains everything that matters, EXCEPT: * chan_sip.c: r15282, new flag structure (not the point of this backport) * chan_sip.c: r15329, handle_common_...

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =&g...

...4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)} Codec=${CHANNEL(audioreadformat)}/${CHANNEL(audiowriteformat)}/${CHANNEL(audionativeformat)}/${SIPCHANINFO(t38passthrough)} QOS=${RTPAUDIOQOS}) The problems I have so far: *1. CODEC *Codec is reported only for A-Leg. When transcoding asterisk logs the above line as: slin for read / slin for write / the codec of A-Leg / 0 for t.38. Is there a way to get the codec for both legs of a call? *2. RTP Qos is reported on...

Trying out SVN-oej-t38passthrough-r12677 on a server that also needs to pass some calls to another using IAX and attempts to use the Dial command results in multiple messages "Out of idle IAX2 threads for I/O, pausing!". Since this server needs to support IAX I'll have to back out this version and find another id...

...ch suggests no problem with the NAT router the SPA-2100 is behind). Here are the various SVN checkouts I did (all have the problem -- if you can't guess I'm no SVN guru -- SCCS was the last version control system I mastered). svn checkout http://svn.digium.com/svn/asterisk/team/oej/t38passthrough asterisk svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 Here is the sip.conf entry: [t38.nvc.a] type=friend context=t38-inbound callerid=John Doe <1234> host=dynamic secret=xxyyzz qualify=2000 nat=...

We are testing the new T38 passthrough code (SVN-oej-t38passthrough-r13347): - we are using a Sipura SPA-2100 as the T.38 user device - we are using a Patton SmartNode 2400 as the T.38/PRI gateway - we are using Asterisk in the middle We have the following in the [general] section of our sip.conf: t38pt_udptl = yes t38pt_rtp = yes When a fax call comes i...

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: