similar to: asterisk 13 webrtc

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 seem to have stumbled on a bit of a problem. When trying to send a fax with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the current svn version, with FFA 1.2 I get a core dump each time. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use)

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Trying to make the fax detection work. My current setup (with no fax) is done without Answer(), so the call is answered only when someone actually picks-up the phone. But when the incoming call is fax, I can her the tone and call is never forwarded to "Fax" extension. But... Strange thing happens when I (mistakenly) put a call on hold: -- Executing [youngandson-test at

Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number.

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

I solved the issue by not answering the call as I assume others have done. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect) Hello Le

Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

Hi everyone! Situation: when I call from cell phone to a asterisk-connected phone, all works fine. When I call from the asterisk-connected phone (a Cisco 7960 SIP) to the cell, the connection gets made, but there is no audio going in either way... Asterisk reports the following: Sep 16 08:27:41 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c =

i can register to asterisk with msn, place calls, etc. However, if I = unplug the ethernet cable and get a dialup connection I get the = following error when i try to place a call with MSN (i can register = withoug problems); chan_sip.c (process_sdp) No compatible codec! I have disallow gsm and allow ulaw on my sip.conf Any help would be greatly appreciated. Dan

If I understand correctly, setting encryption=no means that Asterisk will make outgoing calls without encryption, but will be happy to accept incoming calls regardless of whether the caller wants encryption or not If encryption=yes, then Asterisk not only uses encryption for the outgoing calls but it will refuse to accept incoming calls unless they use encryption too If I have encryption=no

Hello Le 17/02/2015 17:00, Administrator TOOTAI a ?crit : > Hi, > > as stated in the documentation, it's allowed to set > FAXOPT(faxdetect)=yes/no to allow fax detection. > > It's done (see below) but still fax detection :-( Extension 300 is > hylafax with iaxmodem. > > On the upper Asterisk gw it's the same, despite the faxdetect set to no > we also

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

Hi Everyone. I am getting this error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer >From the sdp can anyone suggest why secure audio cannot be provided ????v=0 ????o=- 6611325078116277019 2 IN IP4 127.0.0.1 ????s=- ????t=0 0 ????a=group:BUNDLE audio ????a=msid-semantic: WMS YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l ????m=audio

I've seen the following appear in some tests with Asterisk 11.11: WARNING[3938][C-00000003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 Specifically, it always happens from a Firefox 24 host but it works without this error from another host running Firefox 26 I did a diff on the SDP and couldn't see