search for: t38pt_udptl

.... 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : "Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP provider account as well as the fax account. " But above, you can read "[general] t38pt_udptl = yes " Has this parameter name changed between 1.4 to 1.6 from t38_udptl to t38pt_udptl ? A asterisk remains silent when I add an unknown parameter "foo=bar", it would perfect if someone could point the right name (t38_udptl or t38pt_udptl). Regards -------------- next part ------...

...AT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38 on 0197673581, changing t38pt_udptl=no to t38pt_udptl=yes,fec and try to send from 0197673581 to 0851711201, it is not working, switches to T38 sendimg a lot of UDPTL packages but it looks like (at least for me) that addresses are "wrong". UDPTL (SIP/0197673581): packet from 90.230.92.67:33408 (type 0, seq 0, len 6) UDPT...

...aboxe line ? MMR and JBIG transcoding ? 2. A softphone like Zoiper is able to receive voice or fax calls. When setting those values (see bellow) in sip.conf, does it mean : "t38/udptl will be chosen if a fax signal is detected somewhere in the path. If none, then alaw would be picked" t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no disallow=all allow=alaw 3. How do you reload updtl.conf ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090209/4d551116/attachment.htm

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

...Direct777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12 File: sip_custom.conf [VoipDirect777821] type=friend host=141.122.139 username=VoipDirect777821 secret=wsPiOov8830 accountcode=5260477782 amaflags=billing context=Incomming disallow=all allow=g729 ;allow=alaw ;allow=...

I am running Trixbox 2.4 which has Asterisk 1.4.18-1 I have kind of followed: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 I added to sip_general_custom.conf ;NEEDED!!! t38pt_udptl = yes I did not add this to the actual SIP extension, as I assumed this being general it applies to all sip extensions, and doing a sip show peer ext# did indeed come up with t38pt_udptl = yes The fax is attached to a Grandstream 488, so I set it for fax mode: T.38 I did leave DTMF as inband (...

Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal success..Any suggestion here? Thanks -------------- next part --...

Dear All, I have created an inbound context in sip.conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes under General context...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with OpenSIPS and cal success..Any suggestion here? Thanks -----...

Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-----[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:...

...decs allow=g729 allow=gsm allow=alaw ; Allow codecs in order of preference dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes I have the definition of the phone in DB. voip-test-01*CLI> sip show peer 0625037998 voip-test-01*CLI> * Name : 0625037998 Realtime peer: No Secret : <Set> MD5Secret : <Not set> Context : sipresidential Subscr.Cont. : <Not set&...

Hello, Using 1.6.2-rc5, my settings include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5 subscribecontext=subs disallow=all allow=alaw t38pt_udptl=no setvar=accountcode=foo [168](local-phone) defaultuser=168 secret=pass168 callerid=John Doe<168> mailbox=168 setvar=longcid=015555555 setvar=accountcode=bar CLI> sip show peer 168 ... Variables : accountcode = bar longcid = 015555555...

...R[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down: WARNING[3514] chan_sip.c: Unsupported SDP media type in offer: image 16400 udptl t38 WARNING[3514] chan_sip.c: Failing due to no acceptable offer found sip_general_custom.conf contains t38pt_udptl=yes udptl.conf contains: [general] udptlstart=4000 udptlend=4999 T38FaxUdpEC = t38UDPRedundancy Asterisk version is 1.8.5.0 When I restart asterisk, everything is working good. Then, after some time, fax stop working. Do you have any idea what it could be? Thanks in advance.

...) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASS...

...ave been able to access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid="blah" <XXXXXXXXXX> host=dynamic nat=y...

...and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just after sending the OK with the proper SDP to the remote Proxy, the Asterisk initiates a new INVITE to the local extension and remote Proxy, with the normal audio codecs again. We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the local extension and remote Proxy, but it still forces the call to go back to a voice call. Any idea why it happens and how to debug it? We set verbose and debug to 20, but no "internal" info is provided to get a clear understandi...

...#39;t see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl = yes,fec,maxdatagram=400 faxdetect = t38 And in udptl.conf, I just uncommented 'use_even_ports = yes ;' and rest of it set as default. Here is the error I'm usually seeing in Asterisk side; [Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-00000ad6): Transmission error...

...I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is "t38pt_udptl = yes" which I have set under [general] & the peer.

...ay is located in the same LAN, so there's no bandwidth/packet-lose issue. We also use on the same Asterisk Real-Time process for the extensions.conf My question: Is the following syntax for disabling T.38 support correct? vm*CLI> -- Executing Set("SIP/Proxy-00000000", "t38pt_udptl=no") vm*CLI> -- Executing Set("SIP/Proxy-00000000", "SIP_CODEC=aLaw") vm*CLI> -- Executing Answer("SIP/Proxy-00000000", "") The aLaw Set command is taken into consideration, because the SDP of the OK that follows these lines includes only...

Hello using asterisk 1.8, compiled res_fax.so and res_fax_spandsp.so - both loaded successfuly in sip.conf set t38pt_udptl=yes but faxes still don't work even in passthru mode. if i do a 'sip show channel' on the channel via which i am sending fax it shows: T.38 support Yes however if i do sip show channel of my channel (from other server) it shows T.38 support No Advices appre...

All; I'm running Asterisk 1.8.15-cert3 with the newest version of spandsp. I've even tried unloading that and using Digium's FFA module but I receive the same error on an outbound transmission: [2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL (SIP/XXXXXXXXXXX_outbound-00000000): Buffer overflow detected (59 + 127 > 175) I only get this with one