similar to: Asterisk 12.4.0 not able to install pjsip

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:

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

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, 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

Mates, Still can't get the latest samba binaries to inatall on SuSE 9.3. I've downloaded them from the samba site and from ftp://ftp.primastasys.com/pub/Samba-Packages/ still no joy. The problem is library related: nemesis:/home/david/Documents/linux/rpms/samba320 # rpm -Uvh * warning: cifs-mount-3.0.20-0.1.i586.rpm: V3 DSA signature: NOKEY, key ID 414a57c3 error: Failed

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, after update on 12.4.0 asterisk crashes on MeetMe ending on 12.3.2 it worked well. Is some one else have this issues? should someone open a ticket?

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:

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

Cross-posting scares me, but... Have you looked at the latest RPMs for SuSE, from ftp://ftp.suse.com/pub/projects/samba/ ? They've worked like a charm for me, thus far (revision after revision, a simple rpm -U * seems to work...) -----Original Message----- From: david rankin [mailto:drankin@cox-internet.com] Sent: Mon 9/12/2005 5:30 PM To: samba; Suse Linux Subject: [SLE] SuSE 9.3 and

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

Hi, I have 2 interfaces - one for adsl and other for LAN on my Linux gateway machine. The IP addresses are 10.10.10.3 & 192.168.10.101 respectively. Now my routing tables show this particular entry. What exactly is this? 169.254.0.0/16 dev eth0 scope link Or by traditional route -n, 169.254.0.0 0.0.0.0 255.255.0.0 U 0 0 0 eth0 What exactly is this 169.254.0.0/16

Hi, My branch office as got a 256Kbps b/w from their service provider at a very very high rate per Mb. They don''t require 256Kbps at all but the ISP does not offer anything low. Can we restrict the bandwith to say 64Kbps nothing fancy? How do I go about it? With warm regards, -Payal

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

Hi, A friend of mine has 2 lines of 512kbps terminated in two Linux boxes. He now want to remove those 2 boxes and have some device which will loadbalance the two ISPs and also have a failover arrangement. But he has agreed to give me a chance to do it on Linux for my own satisfication. Is this easy to do with lartc? How do I go about it exactly? I have very less time to do it since his

Hi, I want to use rsync from a script. Before that I am trying it from command line. I use it as, $ rsync --password-file=pass -e ssh -av legal.txt accounts@127.0.0.1:/home/accounts accounts@127.0.0.1's password: I don't want to be prompted for password. $ ls -l pass -rwx------ 1 payal payal 9 Sep 30 19:11 pass* $ cat pass accpass123 [Don't worry it is not on real