similar to: Cisco router and "488 Not acceptable here" messages

> James Harper wrote: > > >Additionally, just to satisfy myself that I wasn't going mad I changed > >the port from 5060 to 5070 and now things are working, so something is > >definitely playing up on port 5060. > > > >James > > > > > > > You probably have are behind NAT and your NAT device has a SIP ALG. > Changing the port disables

> On Jun 11, 2006, at 8:15 AM, James Harper wrote: > > > Additionally, just to satisfy myself that I wasn't going mad I changed > > the port from 5060 to 5070 and now things are working, so something is > > definitely playing up on port 5060. > > > If you are behind a NAT perhaps two SIP devices are both trying to use > 5060? > Packets are getting out,

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

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

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, 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

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

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

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 all, A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line with errors as follows: -- Executing Dial("SIP/ata186-c1cf", "SIP/29086988@110.100.231.2:5060|30|r") in new stack -- Called 29086988@110.100.231.2:5060 Apr 15 16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp:

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

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!

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.

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

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,

Hi all, Thought I should say something - it''s been a little while. :) Anyway, I''m still futzing with how I want the Ruby API to look. The more I look at the Perl version, the more I think it was too faithful to the C API. Right now I''m looking at the trigger() method. For example, I don''t see any reason to make the "type" key take a hash as an

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

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

Hi, I'm having real difficulty in getting calls to go through with Asterisk. I've managed to check that my SIP connection is made to my provider. Below is an email I received from them: ----------------snip--------------------------------snip--------------------------------snip---------------- I am not certain of the reason for rejection but it has to do with the SDP, it does not