similar to: SV: Queues and call waiting indication

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r afoc@interconnessioni.it Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Queues

Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a "beep beep" (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a "single line" user agent, like firefly, still

The message waiting options seem to be geared for a normal office environment where there is a one-to-one relationship between a user and a phone/mailbox. I want to have a mailbox that is shared amoung a number of people and want the message waiting light to turn on on multiple phones. I can direct the voice mail to the common mail box. I can direct the individual users to the common mail

Hello, I have a partially working Realtime Queue using unixODBC and Firebird. I followed the steps on asterisk-guru and the voip-info asterisk message boards but have been having no success. I say that my queue is partially working because callers can join the queue. However the calls do not go to the agents even if the agent is not tied up and is available for calls. So the caller is stuck

Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout |

Hi there, this is my queues.conf, I'm using todays CVS: [599] joinempty = yes musiconhold = default strategy = roundrobin servicelevel = 60 wrapuptime = 0 maxlen = 0 timeout=15 announce-frequency = 15 member => SIP/381 member => SIP/300 At first call 381 rings, if you hang up and call again you get the 300 ringing ... this looks more rrmemory than roundrobin, there is something wrong

Hi, I would like to enable mysql handling of voicemail boxes ... following that tutorial http://www.voip-info.org/wiki-Asterisk+voicemail+database so I modified the makefile of /apps directory to include USE_MYSQL_VM_INTERFACE=1 and copied mysql-vm-routines.h in the /apps dir, set up the db and some boxes in the table, also edited the voicemail.conf file. Everything compiles just fine, then

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

Hi, I'm testing realtime right now, it does not seem to me that realtime contexts can be included in normal context, like this [sip] include=>sip-dial exten=>i,1,Hangup [sip-dial] switch=>Realtime/sip-dial Am I getting it wrong ? Tnx ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Hi, I get this message after password request in voicemail app: Unable to create lock file: No such file or directory Anyone got a clue about fixing that problem ? I can't understand what directory or file we are talking about .. Tnx for any help! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Hi, I'm having some problem with realtime: let's say I have a dialplan like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer A call from ip phone 300 comes in, and it's been answered. Then I "switch" the sip context to realtime, putting the exten in the db and using the variable name for this as in the file version. [globals] IPPHONES=_3XX [sip]

Hi Alessio Yes, the problems you report do seem similar to the issues I had. I found on the Dells that the audio prompts were very choppy and played slower than normal. Occasionally there would be 'bursts' oav a second or so of 'good' audio. I also suspected IRQ issues but the Dell Mobos had no way of adjusting them. Best thing is to try and get the card on its own unshared

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Hello all, First of all I''d like to thank everyone for making Xen so user friendly. I thought I''d be sweating through this but it''s been pretty straight forward (thanks to all the different tutorials on the net). I''ve got a domU that I created with dd if=/dev/zero of=vm_base.img bs=1024k count=4000 mkfs.ext3 vm_base.img dd if=/dev/zero of=vm_base-swap.img

Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled =

On Sat, Mar 23, 2013 at 04:59:49PM +0100, Tobias Grosser wrote: > With Chris's reply you also got the general OK to upstream the > patch, _AFTER_ the actual patches have been reviewed. One thing I was asking on IRC for is whether it makes sense for new backends to be committable in incremental steps. Especially for a complete backend like this, e.g. the asm parser would make a

Hello Asterisk Users, I guess the subject says the most of it; here goes some more detail: - Running Asterisk 1.2.14 - Objective: record all calls managed by a specific queue - Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID} Facts: - If the UNIQUEID chan var is used in the MONITOR_FILENAME, before calling the Queue() application, the two legs of the call are not

Hi! By iptables I set marking (-j MARK) just before -j IMQ. There are only three marks. Exectly all of them are ditected to 1:20 Why I see 3 pkts in default 1:30? How explain it? ### imq0: queueing disciplines qdisc htb 1: r2q 10 default 30 direct_packets_stat 0 Sent 631429 bytes 1812 pkts (dropped 0, overlimits 20) ### imq0: traffic classes class htb 1:20 root prio 0 rate 100Kbit ceil

Please can you explain in more details as to what your problem is? I have 2 cards working in one PC, but have had problems with Dell motherboards. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Alessio Focardi Sent: 17 June 2004 11:41 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] HFC ISDN card

Sorry for the stupid question: What's the purpose of defining a peer as trunk in iax.conf ? The question is also valid generally speaking (for other channel types), for instance: why define a Zap group as trunk in extension.conf ? Tnx for any help ! -- Best regards, Alessio mailto:afoc@interconnessioni.it