similar to: Background transfers with callback

Me again, another problem. As I said before, I have 2 lines going into "incoming" context. When client calls, I press Flash, client hears music on hold (only on voip line as said in previous post), when I get back and press Flash again to get back to my client I cannon hear him, but he hears me without problems. I have just tested in on the LAN, same situations, happens everytime.

Asterisk 1.4.6 at FreeBSD6.2-RELEASE Client hears pure silence when waiting for call answer. Music on hold stops when transferer pics a number and client doesn't even hear ringing. Is this normal behaviour? How to change this? Log says everything, MOH should stop after call pickup, not before Dial. -- Executing [113 at firma:1] Dial("SIP/zytek-08737000",

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack -- Called zytek

Hello all, this is my first post here. Sorry for my english. Gentoo LAN router, 2.4.26-hardened-r2 There are 2 WAN links, one LAN link. I am doing some iptables/routing/tc magic in my scripts. What''s interesting is marking packets traveling from all IP''s in LAN. Interesting commands are: ------------- for ip in `seq 50`; do $IPTABLES -t mangle -A FORWARD -o eth2 -d

Correct me if I''m wrong, I just want to help my friend who needs a tc solution with fairness to hosts on a 512K/s DSL line, but few of them should be restricted to 64K/s I thought about htb + esfq (sfq with ip based fairness, not connection) parent class with CEIL=500Kbit (no RULE? see *1) and attached esfq to this parent class, now child class with CEIL=64Kbit and RULE=10.0.0.1

What I need to do: shape every user on my LAN to 256Kbit -- class for web trafiic with rate X ceil 256Kbit -- class for other(p2p) traffic with rate 1Kbit ceil 200Kbit This is good because even if they have p2p programs running they will always have fast web surfing. I can do it with bash scripts - one class per ip with 2 children. But I wonder if something like this would work: # class

Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer

asterisk-1.4.7, Fedora 7, intel emt64 - nocona: == Parsing '/etc/asterisk/alsa.conf': Found ALSA lib pcm_dsnoop.c:558:(snd_pcm_dsnoop_open) unable to open slave [Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:365 alsa_card_init: snd_pcm_open failed: No such file or directory [Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:481 soundcard_init: Problem opening alsa I/O devices == No sound

Hi, I've recently upgraded Asterisk to the latest version 1.4.9 on a PBX that manages several queues, but at least on one queue strategy (leastrecent) it doesn't seem to be distributing the calls has it should. I think this strategy should work like this: a) Make a list of available agents and their idle time (time since last call) and

Hi, All, I have a question about agents and queues. Right now we have about 4 queues in our system. Some agents are in multiple queues. Our main queue is for technical support and it's by far our busiest queue as well. We have the autologoff feature set to 14 sec right now in the agents.conf file. The problem I'm running into is we don't want people in our sales queue (who are

Is DTLS available for Centos? Either Centos 4 or 5. DTLS is TLS over UDP. Highly valued to protect SIP traffic.....

Hi there, I'm experiencing some problems accessing some of the services on llvm.org: $ svn co https://llvm.org/svn/llvm-project/llvm/trunk svn: E200029: Couldn't perform atomic initialization https://llvm.org/bugs/show_bug.cgi?id=24734: "undef error - DBD::mysql::st execute failed: Got error 28 from storage engine" On Linux, errno 28 corresponds with ENOSPC, so my suspicion

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

Hi All, Currently i am integrating webRTC demo. I have issue using firefox,someone suggest me to enable DTLS for webRTC working in firefox using Asterisk. I am using Asterisk 11.9.0. https://groups.google.com/forum/#!searchin/doubango/bhavik/doubango/Mv9u0YkNb90/55VElJ1TdY8J Can any one tell me how to enable DTLS ? -- Thanks, Bhavik Patel -------------- next part -------------- An HTML

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people

>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing with chan_sip.c:4083 retrans_pkt: Hanging up call 7238b48c11581d4166b899bf747a05f7 at 130.211.62.184:0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). is there any way to configure to have the previous behaviour? Im trying to set

Hi Hans, 2016-07-30 0:57 GMT+02:00 Hans Wennborg via llvm-dev <llvm-dev at lists.llvm.org>: > There are still open merge requests and bugs, but I'd like to get the > real testing started to see where we're at. Just to double-check, is this bug also on people's radar? It causes Clang to crash when trying to build a fair number of Open Source packages/libraries. In my

Hi All We've been reading this in the CLI a lot lately: Received SSL traffic on RTP instance '0x7fe7481faad8' without an SSL session How can we find details about this particular RTP instance? "rtp set debug" needs an IP which is precisely what I want to know (and I don't)! Cheers Ethy

I was trying to install iproute2-2.2.4. I get an error when i run the makefile. I get a parse error in /usr/include/arpa/inet.h. Can someone help me? Thanks. _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/