similar to: T.38 - By reinvitation only?

Hello, We're trying to receive faxes on the Asterisk server, but for the time being T.38 negotiation fails. The SDP that the Asterisk reINVITE sends contains these lines: ---------------------- m=image 4968 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400

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

Hello everyone. I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID provider here in Brazil (GVT - Vox IP service). Here's my scenario: When faxes arrive by a specific DID, they are routed thru this simple macro: [macro-recebefax] exten => s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1]) exten => s,n,Set(FAXCOUNT=${DB(fax/count)}) exten =>

Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need for app_rxfax then asterisk crashes with segfaults on startup asterisk[2624]: segfault at 30353466 ip b7eb538b sp bffda26c error 4 in

Has anybody else noticed cygwin.com and sources.redhat.com being down? The machine that answers to those addresses (and sourceware.org) is refusing HTTP connections, and has been for at least 48 hours now. Anybody here know anything about it? I haven't been able to find out diddly-squat about it on the web... -J

On 24/04/19 19:51, L.P.H. van Belle wrote: > Hai, > >> -----Oorspronkelijk bericht----- >> Van: samba [mailto:samba-bounces at lists.samba.org] Namens >> Rowland Penny via samba >> Verzonden: woensdag 24 april 2019 12:13 >> Aan: samba at lists.samba.org >> Onderwerp: Re: [Samba] User mapping/login issue >> >> On Wed, 24 Apr 2019 11:38:58 +0200

is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't

Hi everyone, I've got the same problem described in http://bugs.digium.com/view.php?id=10647 (unfortunately, the bug is closed and I could not find the way to reopen it). Wiki says, " When options t, T", "h", "H", "w", "W" or "L" (with multiple arguments) are applied, Asterisk will remain in the media path, even if

Is there a way in iax to have to endpoints talk to each other directly (after the call is setup by *) without going through *. In sip, with * you can do it by configuring sip.conf with canreinvite = yes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030728/0c711d05/attachment.htm

Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once

Hello, I've just set up DISA on my * server. I'm using it to avoid cellular overseas calling charges from support staff in the field at our customer sites. Support staff often spend hours on the phone to our UK factory. However, I'm not sure about the implications of reinvite in this arrangement. A support engineer calls in to a DID that I have from VoicePulse Connect. They match

Hey Folks, Is it possible to transfer an incoming call back out without a "trombone" effect. For instance; Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the caller selects an option --> asterisk transfers the call to my cell phone via broadvoice and removes itself from the equation so I end up with... Caller --> Broadvoice --> Cell Phone Vs.

Hi, We're routing SIP calls through Asterisk and we want to be able to reinvite calls without Asterisk performing codec conversion. We've performed the following test: Asterisk has license for G.729 installed sip.conf [general] context=default autocreatepeer=yes disallow=all allow=alaw allow=g729 canreinvite=yes nat=no We have configured two endpoints: EP1, preferred codec order

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

It is not clear how exactly g729 pass-through can be enabled. I have a SIP call off a gateway come into an Asterisk menu, and then I send the SIP call to another SIP gateway using Dial(). Even though codec preferences have g729 listed first, it never gets used. Both gateways have separate peer entries in sip.conf, and both have canreinvite=yes set. Can Asterisk change the media type during

I am wondering if the SIP protocol and its implementation in * allows for changing codecs mid-connection. I've seen some questions regarding this on the list, but I've not found any clear answers. I've also seen the SIP_CODEC variable, but it's not clear that it will change the codec on an existing call. Also, there are mentions of needing a reinvite to make the change, but most

I've been wondering for a long time why my reinvite option is not working with Broadvoice anymore. It happend during the time Broadvoice was having a lot of issues, so I decided to wait. Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and

Hi, just a question: Let say I have 2 phones with G729 onboard, but no 729 licence for Asterisk. Preferred codec set up in phones is G729, followed by ULAW, in Asterisk I have allow=ULAW deny=ALL. When call is reinvited by Asterisk will the two phones use G729 between each other or they will stick to ULAW they used for first part of the call ? A quick test showed that they will use ULAW ...

Hi Although canreinvite option is yes, the asterix doesn't send reinvite and the media is going through the asterix instead of between the two sip phones. Both sip phones (handytone 486) don't use NAT and are configure with canreinvite option yes and use the same codec G.729. And Dial() command don't contains t or T. Any suggestion on what could be the problem ?

I've an Asterisk server running in my office, which forwards all long distance calls to a third party SIP service using an extension rule: exten => _1XX0.,1,Dial(SIP/{EXTEN:4}@external_sip_server.com) (1XX0 is the international calls rule for Chile) Also, in my sip.conf, I've defined canreinvite=yes to decrease the network load to the server caused by the RTP. However, the external