similar to: packet2packet bridging

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

Hey guys Is it possible to record with mixmonitor straight into mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Sameer Rathod <sameer at hostnsoft.com> </div><div>Date:30/06/2014 9:23 PM (GMT+02:00) </div><div>To:

Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me in doing the same. Thanks in advance. -- Regards Sameer Rathod 8109413462 -- Regards Sameer

Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: binary <dreamer.binary at gmail.com> </div><div>Date:01/07/2014 6:09 PM

Hi, I had tried all the steps which I used to inatall Asterisk 12.3.2 Pjsip in Asterisk 12.3.2 is working but in new release Asterisk 12.4.0 it is not working I am getting XXX in make menuselect resource_module. I tried all trouble shooting steps along with ldconfig etc. I think its a bug can any one help me on this ? -- Regards Sameer Rathod 8109413462 -------------- next part

Currently using tikal crystal call recording Do you guys know of any better ones? Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: binary <dreamer.binary at gmail.com> </div><div>Date:01/07/2014 6:33 PM (GMT+02:00) </div><div>To: asterisk-users at lists.digium.com </div><div>Subject: Re:

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017

I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well as trying to get some of the RTP traffic offloaded from the network. I think I'm misunderstanding what the console messages mean when it says "Packet2Packet Bridding SIP/blah to SIP/blah". I though that meant that it had successfully (re)INVITED and the media was no longer going through my Asterisk

Hi, I have Asterisk set up on Fedora with a single SIP trunk, with a few handsets configured. The Asterisk box has both public and private addressing, so "canreinvite=no" is set on both the SIP trunk and handset configurations so I can get around the nasty NAT issues. One odd behaviour I am seeing is certain destinations are resulting in different SIP codes being sent back to Asterisk,

Hi everybody Hope I picked the right mailing list. If not, please tell me. We've got a problem with call forwardings. It's exactly the same problem as described in bug 12708, which is resolved by now. Situation: Caller -> asterisk -> call forward to mobile (packet2packet bridge) Quote from original bug reporter: 'One issue that we have noticed repeatedly is that there is a

I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for my problem Hello, During a call with canreinvite = no, at the beginning of the call I lose 2 seconds of audio. is obvious when I call autoattendant. schema: SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1) --> Operator SIP capture of voip1: - Executing [0825387205 at

Hi, I am using chrome version 36 and opera with asterisk 11.9.0 and cent os I am getting the below error if i do call on sipml5 from blink 1. Failed to set remote offer sdp: Called with SDP without DTLS fingerprint. tsk_utils.js?svn=224:128 1. tsk_utils_log_errortsk_utils.js?svn=224:128 2. tmedia_session_jsep01.onSetRemoteDescriptionError

Hi, I want to authenticate user with a random authentication key before registration in asterisk for a click2dial feature in my website. The goal is to not to display the password to the client. The client will be provided with a authentication key and when the request comes to the server form the web browser (via webrtc) it will fetch the relevant userId and password, register the sip and the

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

The password is related to a rsync server, when you communicate through the rsync port: grep rsync /etc/services rsync 873/tcp # rsync rsync 873/udp # rsync the authentication therefore is done against the rsync serverpassword. Yo wanted to "file in" the ssh password? Not possible in this way ... Rainer -----Urspr?ngliche

Hi all, i have installed rails 3 using RVM . the prob is now i have two parallel projects where one is in 2.3.5 and another is in rails3. i use Mac o x 10.6.3 mysql :Server version: 5.1.56 MySQL so can i switch in between rails3 and rails 2.3.8?? this is my gem list... actionmailer (3.0.5, 3.0.5.rc1, 3.0.4, 3.0.3, 2.3.8, 2.3.5) actionpack (3.0.5, 3.0.5.rc1, 3.0.4, 3.0.3, 2.3.8, 2.3.5)

Hello friends, Can you guys give me some idea about how to Create XML file using Nokogiri gem. -- Posted via http://www.ruby-forum.com/.

Hi, I have 2 ftp servers with 3 identical users. I want server B to be updated of server A. Whatever changes mmade by users on server A should be made on server B say every 5 hours. Is rsync the right tool to use here? Can rsync be used such that only changed files are downloaded? Thanks a lot and bye. With warm regards, -Payal -- For GNU/Linux Success Stories and Articles visit:

Hi, A majority of our work inolves ftp to my clients'' side over our slow connection. Now we need to allocate a greater b/w for this protocol. Is there anyway I can do it using lartc easily? Any suggestions on this please? With warm regards, -Payal _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc

Hi, we r using HTB algorithm,for traffic shaping, we are facing a problem. we are able to create multiple classes,filters. But when we delete 1 filter all filter gets deleted. how do we avoid that. waiting for you reply Regards Jayesh ------------------------------------------------- Shop & Save at Sifymall.com! Special Festive Offers - up to 60% off on DVD players, MP3 Players. Mobile