similar to: Codec problems only with AGI ?

Hi all, I realise that asterisk's codec negotiation has been discussed in the past multiple times. What I haven't been able to understand is how asterisk decides which video codecs to advertise to the other end when canreinvite=no in sip.conf and the initial caller doesn't support video. My tests are quite simple, I use an asterisk with 4 peers all on the same LAN. My sip.conf

Hello I got the similar error while trying a call. -- Executing Answer("SIP/8000104-86ef", "") in new stack -- Executing Wait("SIP/8000104-86ef", "2") in new stack -- Executing AGI("SIP/8000104-86ef", "areskicc.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php areskicc.php:

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as

Dear All, Here the version 2.2 a new version of your dear CallingCard Software !!! http://www.areski.net/areskicc-doc-v2/ Many new features have been added and several enhancements made! Newest features : - A new re-build rate-engine - LCR & LCD management (OOOOHHH YESSSSS) - Billing Increment - Progressive Rate - Scheduled Rates (days of the weeks) - Expiration rates - Buy rates

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Hi All, I am trying to get the opus codec working with linphone. I followed the instructions... This shows me its loaded core show translation paths opus --- Translation paths SRC Codec "opus" sample rate 48000 --- opus:48000 To g723:8000 : No Translation Path opus:48000 To ulaw:8000 : (opus at 48000)->(slin at 48000 )->(slin at

Today's "frustrated programmer" award goes to Linphone, which has the following debug output: > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that

Hello I would like some help or advice on how to implement or could insert the opus codec to softphone Linphone, thank you very much. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20140424/ef366db5/attachment.htm

Jean-Marc Valin wrote: >Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, >Linphone just sends raw packets, as specified in the RTP draft. > > Asterisk expects speex frames to have a terminator. The phone I was referring to was the X-Ten/X-Lite phones, which seemed to be adding something _before_ the speex data to indicate the length of the frames.

On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards <asterisk.org at sedwards.com>wrote: > On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards <asterisk.org at sedwards.com> >> wrote: >> >> I strongly suggest using an existing library for the language of your >> choice. >> > > On Mon, 6 Jun 2011, A E [Gmail] wrote: > > Copy that. Not planning to

Excellent work! Thanks a lot ------------------------------------- Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF

Hi all, I have encountered this problem: if the caller is connected to the callee using Dial() command called from extensions in extensions.conf, there is no problem. But if the same caller and callee are connected using an AGI->exec('Dial'...), the line is disconnected when asnwer. There's a problem bridging. If the codecs are the same on both ends then there is no problem.

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

I just downloaded the latest CVS. RTP streams from a X-Lite on a PC, to the Linux box running Asterisk and Linphone, seem to connect: > -- Called m4 > -- SIP/m4-dd8f is ringing > -- SIP/m4-dd8f answered SIP/m12-3649 > -- Attempting native bridge of SIP/m12-3649 and SIP/m4-dd8f But then Linphone sends an endless stream of complaints: > (linphone:20223):

Hi, I am facing a (for me) strange problem. When placing a SIP-Call I normally get connected and the dialplan is executed. The Call-Flow is controlled by a PHP-Agi-Script. The script answers the call (via AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get disconnected immediately after the Answer - without any reason. This happens about all fifth call. Later on you will find

Requirements: 1. http://sourceforge.net/projects/odbcsock 2. The SQL server must be in the same subnet as your * server Howto: 1. Install ODBCSocketServer on your SQL server and verify connection with the included VB COM app from a Windows box. 2. Decide how you want to transform the Caller ID. In my case, I want to do a lookup of the CallerID number in SQL server and prepend the Caller ID

Hi, I've compiled the speex codec and installed as per docs, speex works perfectly between Asterisk gateways using IAX protocol. My problem is between X-Lite softphone and a Asterisk Gateway users cant hear anything when the speex codec gets selected. Has anyone else experienced this with X-Lite or any other softphone client? Is their any other Softphone for Windows that makes use of the

I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call