similar to: Help T.38

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

Hi all.... I'm using asterisk for making PSTN calls from extensions registered on OpenSIPS...In peak hours ,number of calls Increase dramatically to a non logic number..When checking the calls using asterisk CLI I saw a lot of calls in ringing status and after 300s(rtphold timeout), asterisk release all calls...I checked the log file and found.. [Feb 28 11:34:14] NOTICE[19197] chan_sip.c:

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP

Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other

Dera All, I have the following scenario, A customer dial a DID number...The call is routed to a PSTN GW that send the call to asterisk... On asterisk I created an AGI Script that send the call to an extension registered on OpenSIPS server... The extension is ringing successfully, but as soon as I accept the call on OpenSIPS side the call is hangd up... I checked rhe SIP debug and it seems that I

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

Hello List, I have been trying to configure a sip profile ( peer / friend ) for each of my customers behind a sip proxy for some time, but I have had no success, so I would appreciate your help. Customer -> OpenSIPS -> Asterisk -> PSTN The opensips is working as a sip proxy with record route, for billing, load balancing and authentication purposes. I would like to be able to define

Hi gurus, In need of a little help here. I?m trying to do the Asterisk media release by using canreinvite=yes. But I found weird behaviour when comes to BYE. Below are my current setup: Client A is registered to Opensips Client B is registered to Asterisk A ? Opensips ? Asterisk ? B On hangup below are the SIP flow which I?ve notice from the Asterisk server itself: 1. Opensips forward the BYE

Hello, We're trying to receive G.711 (aLaw) faxes on the asterisk and convert them to tif. With T.38, we have several issues, so we are trying to use G.711, since the gateway 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

Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just

I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax (.tiff) from the first asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an INVITE with audio G.711. Asterisk2 accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to Asterisk1. But, Asterisk1 responds with "488 not acceptable here". I double check t38pt_udptl = yes in

Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request response saying I have a SIP syntax error. Flowroute support is recommending that I try again after

Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. When I try to send a fax in T.38 mode I receive this error ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel 'SIP/eutelia-sirio-out-00000000' is in an unsupported T.38 negotiation state, cannot continue. In my sip.config general section I have added this lines t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no If I comment this lines,

Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->

Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog

Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 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

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

Hello guys. When I am trying to send fax through T38 to linksys SPA (properly configured etc. - I have tried it with other systems), I'm getting error and fax is not delivered. I'm getting this errors in asterisk.log: WARNING[687] udptl.c: No UDPTL ports remaining ERROR[687] chan_sip.c: UDPTL creation failed WARNING[687] udptl.c: No UDPTL ports remaining then, couple lines down:

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