similar to: priorityjumping not working, Dial goes to n+1 not n+101

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",

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.

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

I've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten =>

Hello list, I have successfully set up Asterisk, but girls from our office complain to me that when they hit Flash to transfer a call and pick the number, they need to wait until the call is answered, and only then they could hangup. On the analog PBX we had before the transfer was in "background", and if called party did not answer the call, then the call returned to the girl in the

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

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

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

Dear all, I know this is not imq mailing list. But many of the users over here have done exactly what i want. Requirement:- I want to tight bound eth1 for 100 kbps but after i want to create many classes of 64 kbps or 50 kbps and others. But the total sum of classes is more than 100 kbps so my eth1 is not restrciting total bandwidth at 100kbps. According to search on google imq is the solution.

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/

Hello everybody. i have the folowing problem: i have this in the top of PREROUTING chain in mangle table iptables -t mangle -A PREROUTING -j CONNMARK --set-mark 0 # rule 1 iptables -t mangle -A PREROUTING -m connmark --mark 5 # rule 2 iptables -t mangle -A PREROUTING -m connmark --mark 6 # rule 3 i think when packet is passing trough my POSTROUTING in mangle table

Hi folks. Let''s say I would like to make some bandwidth control on my network using HTB. I have 2 clients: PC1: 192.168.100.2 PC2: 192.168.100.3 Server: 192.168.100.1 This has 2 NIC''s eth1 is local and eth2 is connected to the internet. It could be nice to have a script, where you could specify, how much bandwidth you want for a specific host on a network, like, PC1 has

Hi All, Does anyone know of a way to limit the speed of *individual* TCP sessions, but without placing any overall bandwidth limits, and without requiring an explicit QoS entry for every ip address the machine is communicating with ? The scenario is a mailserver - say you want to limit individual TCP sessions (pop3, smtp etc) to no more than 512Kbit so that an individual session

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the

Hi, I made a simple menu using the Background application and some wav files. I converted the wav files using for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm"; done (from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk) The first two files "01/bemvindo" and "01/menu_top" are good.

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why?