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Hi, I have been using asterisk 10.0.1 for 8 months and now I have updated it to 12.6.0. I have not made much changes in conf files. I am seeing continues warnings saying "Can't send 10 type frames with SIP/Gtalk write" on console. Which I had never seen in the previous version. I have seen compared chan_sip.c from both versions and It seems 12.6.0 is also made to support all the

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

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: asterisk-users at lists.digium.com </div><div>Subject: [asterisk-users] Fwd: Regarding packet2packet bridging </div><div> </div> Dear concern, I want to configure packet2p...

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

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

Hi, I am on asterisk 12.6.0. Previously I was using 10.0.1 and for Gtalk I was using chan_gtalk and jabber configuration. But on 12.6.0 I tried to use chan_motif, asterisk starts consuming 100% cpu. From pstack trace I got it is because of ICE and to run motif ICE is necessary. Has anyone else seen this issue or any solution for this? I am using neither STUN nor TURN. I have just enabled ICE in

Any help will be appreciated. Regards Mayank -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141205/f4841527/attachment.html>

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

Hi, I am new to asterisk I want to configure my asterisk server such that it only establishes the call rest the audio must bypass the server and transmitted directly to the peer In my config file I did changes which are below canreinvite=yes nat=force_rtp dirtectmedia=yes directsetup=yes I am using asterisk version 12.3 -- Regards Sameer Rathod 8109413462 -------------- next part

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