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...Here is the call flow: AST-Sanset*CLI> -- Accepting AUTHENTICATED call from 195.27.242.8: > requested format = g729, > requested prefs = (), > actual format = g729, > host prefs = (g729), > priority = mine -- Executing Goto("IAX2/comcentrixs-uplink-6", "sanset|944|1") in new stack -- Goto (sanset,944,1) -- Executing Macro("IAX2/comcentrixs-uplink-6", "stdexten") in new stack -- Executing Set("IAX2/comcentrixs-uplink-6", "LANGUAGE()=de") in new stack -- Executing Set...

currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet. here si the sip debug msg, we got a Message type: DISCONNECT (69) and unable to dial any numbers. Jul 12 12:56:26 WARNING[1523]: chan_zap.c:1931 pri_find_dchan: No

Currently I am creating .wav files and then converting them via SOX to .au file format, then running them through a gsm codec convertor which all works fine except that it sounds like the recording was made with a sock in my mouth !! Could someone in * land help me to get a good sound quality with gsm format. Thanks in advance. -------------- next part -------------- An HTML attachment

Grretings to all, I am having a problem with a customer's queue setup that I don't really understand. Background: Customer has 5+ queues and is using dynamic login to the queues based on SIP/XXX for example. There is a litle script that runs that allows agents to log into particular queues via the keypad. The user can log in to any queue that he wants, including multiple queues. The

I have setup several queues for a customer. Their periodic announcement says please wait for the next available agent, or press * to leave a voicemail. This does not work when the message is playing. The message stops, but the user is left in the queue. Q-exit with * works the rest of the time fine. Has anyone seen this or know if it shoudl actually work differently? Regards to all, Joe

On 4/8/06, Joe <jrothstein@comcentrixs.com> wrote: > Remove the SIP /400 entry from the Asterisk DB. > > Database del .... At asterisk prompt. > > Or look at the wiki for info on how to remove it. > > Or make sure the SIP/500 uses a different IP address than the old SIP/400. > > Joe > > > > than...

I was just looking through the Wiki for some info on how to retain the original caller's callerid when make transfers to internal extensions, and I came across the parameter below: useincomingcalleridonzaptransfer=yes There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering if this is still a valid parameter. If not, does anyone know how I can do this? Thanks, Joe

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

Anybody know the proceedure to factory reset the a 7960 phone running 6.3 SIP software? I've tried holding # when booting the phone and nothing, i can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. Also **# doesnt work either.. -- ~Shaun

Greetings to all, I ma having a problem with channel variables on a couple of our Asterisk boxes. Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN. On the External GW, we also have an IAX trunk to a VOIP provider if for some reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be

Geetings to all. I am having a hell of a time getting incoming SIP connections to work properly, and am hoping that someone can help me. Here is what I am using as a guide (from the wiki): "Incoming SIP Connections When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer]

Greetings to all. I am hoping someone can help me out with a problem I am having getting my Cisco phones, 7960s and 7940s, to download the appropriate files from our TFTP server. The TFTP server is running on Fedora Core 4. The TFTP server appears to be setup properly: service tftp { socket_type = dgram protocol = udp wait =

Greetings to all. I am currently implementing call queues for a customer and have come across several "problems". The customer is an airline representative, and will be using call queues for different airline reservations. The customer requires that any agent be able to login to any number of queues. This means that queue members have to be dynamic, not using "member =>

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

Hi, I also have the same problem, from memory is this not progress indicator=8 that deals with the dialing indicator ? Anyway I am also stuck with not having any dial indication, has anybody got an idea. MO <mailto:asterisk-users@lists.digium.com> -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have the following problem with our asterisk system: We have a very long no. 089xxxx3168205 (14 digits) or perhaps minus the 1st 3 digits as that is the area code. Some people are having issues when they phone from certain PABX?s or have to try many times to get through, which is really bad. I have noticed from my mobile this does not occur. pbx*CLI> -- Extension '2' in

Greetings to all, I'm try to compile * (1.2) on a Sunfire 210 with Solaris 10, but do not get past line 29 in the Makefile. Some innoccuous line with 'uname -s' as a variable. Would love to hear from anyone who has gotten Asterisk to compile on Solaris 10 specifically. Thanskt to all, Joe -------------- next part -------------- An HTML attachment was scrubbed...

Greetings to all, I am getting the following error on my PRI-connected Asterisk box, and am just wondering if anyone else has seen this, and if so how they solved it. Jan 18 10:10:23 NOTICE[6070]: chan_zap.c:7428 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jan 18 10:10:24 NOTICE[6070]: chan_zap.c:7428 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary

Greetings to All, I hope that someone can give me some guidelines with regards to CPU utilization, and at what level CPU utilization begins to effect call quality. Thank you in advance, Joe