search for: infopact

List, I've got 2 queues with 10 agents in both queues. One of the agents is mainly responsible for queue_1, and the others mainly for queue_2 so i've defined the following in my queues.conf [queue_1] strategy=ringall member=>Agent/1,2 member=>Agent/2,1 member=>Agent/3,1 member=>Agent/4,1 [queue_2] strategy=ringall member=>Agent/1,1 member=>Agent/2,2

Hello, I've got the following queue.conf: [testQ] music=jr_80 ;Bore the caller with some 80's music announce=queue-testQ ;Announcement to play to the Agent answering strategy=ringall ;Let all hell break lose timeout=60 ;We should answer within 60s retry=5 ; announce-frequenty=15 ;Tell them where the are every 15 seconds announce-holdtime=yes ; Give them

I just recompiled my version from this morning's CVS Head. My systems voice files (voicemail, time etc) were playing nicely. Until that is I added an extension and now the files won't play. Worse than that, * thinks the files have played and goes to the next step in the dial plan. What gives? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads

Hello, I'm using asterisk to relay sip to another SIP provider, I've setup a friend in sip.conf for my softphone and a user & peer section for the SIP provider, when my softphone calls out to the sip provider and the sip provider returns an error (404 Not Found for example) the sip message is not relayed back to my sip phone, it just sits and waits for a timeout. Is it possible to

Hello, I would like the dialing party to know what happened to the call, since asterisk doesn't relay a sip error back to the originating sip channel (would be nice, a if (org_channel = sip && dst_channel = sip, relay error to sip client) I want to set up audio feedback on the call status. I've changed the county setting to NL in indications.conf and created this test

I'm having some troubles with my * machine, when i place a call on hold the callee doesn't hear any MOH and the call is dropped because of lack of RTP. I also don't see * starting MOH on the SIP channel the callee is on (moh class is defined, there are MP3 files and mpg123 is active). I'm using * 1.0.6 right now with Cisco 7940's, i can see * recieving a SIP invite with

Anyone got the screen xml function to work yet? i've setup an URL in my line 1 (the only line I use) but i don't even see a GET request to my webserver. Kind regards, Erik

Hello list, I've got a problem provisioning my snom 360's in the office (about 20 of them). I'm using DHCP option 66/67 to set the provisioning URL but the phone won't connect to it to retrieve it's configuration. We are using a Cisco Catalyst Epress 500 to power the phones (poe), however if i power hem via the adapter and hook them up to a hub instead of the switch the phone

Hi list, I know * generates it's outgoing RTP stream based on the incomming RTP stream, i'm having some audio problems after i recieve an rtp reinvite from my carrier. Situation: Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN Asterisk A: reinvite = no Asterisk B: reinvite = no If i dial out on phone via asterisk A, Asterisk B relay's the INVITE to the carrier, after the

Hello list, I'm currently looking into building a new Asterisk server, due to some codec problems i've got to transcode most of my channels between Alaw -- G729. Is there any indication on how many channels you would be able to transcode on a certain platform? I'm looking into dual Xeon or dual Opteron configurations, which of these platforms would perform better? And how much power

Hello, I'm trying to implement a function available on the PSTN net here, if you dial a number which is busy and you press 5, you will be called back when the busy party hangs up. Figuring out if a SIP user is busy isn't to hard, ${DIALSTATUS} produces a BUSY message, however, how can I implement the call back? IE, I dial to extension 712, but that extension is busy, I dial 5 and

Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? IE, the following setup: SIP Phones -------------- Asterisk ------------------------ SIP carrier | | ------- Asterisk (standby) ------ Asterisk is used as a PABX in this setup, so the