similar to: pattern matching based on callerid, not working

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

Has anyone used the H option for Queue() with Callback queues? I want customers in my queues to be able to jump out to voicemail when they get tired of waiting, but in my setup when I pretend to be a customer and press '*' [when I am waiting in the queue] I see the message 'User hit * to disconnect call.' but then just jump out to the outer loop where queued callers wait to

Hello, I am trying to alter the outbound callerID for extensions within a context I have created. I wrote the following: exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$ {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560) exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$

How can I dial *67 on a Sipura 3000 if I dial from a SIP phone connected on an asterisk server. I always get a message saying that authentication failed for INVITE for sip221@192.168.1.6. If I dial a number without doing *67 it's working fine... sip 221 being the extension of my Cisco phone and 192.168.1.6 being the IP of my asterisk server... I have my outgoing context configure

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

Hi, I need to create a simple number checking for authorizing the calls. if a person dial 91800555121212345 where 12345 is the authorization code. If the authorization code is correct the call will go through if not it will play something saying wrong authorization code or just hangup. This my dialplan to get the authorization code AUTH=12345 exten => _9.,1,Answer() exten =>

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

I'm getting a strange error on one of the two controllers on an AVM C2 card under chan_capi-cm-0.6.3. I have two ISDN controllers defined, both in the same group, both connections are UK ISDN2e Point to Point: On the third outbound call (both of the first two calls are handled by the second controller "ISDN2",) I get this error : chan_capi.c conf_error 0x2001 PLCI=0x301

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment:

Dear all, I have the following lines in my extentions.conf file; ;All US Calls exten => _9001XXXXXXXXXX,1,Dial(IAX2/dornoch:xxxx@10.xx.xx.xx/${EXTEN:1}@outbound) ;Dial 9 for outgoing numbers exten =>_9.,1,Dial(Zap/g1/${EXTEN:1}) ;include Brunswick switch => IAX2/dornoch:xxxx@xx.xx.xx.xx/sip What I'm trying to do is to send any calls starting with 9001 out through

Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good).

My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

Hi there, I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO line. The reason being is that Voicetronix sends out the DTMF too fast even before the line is fully established with the carrier. Usually when dialing an 8 digit number, only 7 digits are actually successfully heard by the carrier. Currently, my dial plan is: exten => _9.,1,Dial(vpb/1-1/${EXTEN:1}) Daniel

Hi, I have two small companies that are going to be sharing a * box. I have 2 TDM400's with 4 fxo ports each. Each company has its own sales person and they would like the sales people to always show their own caller id and have their own lines ring directly to their phones. Company 1 sales person uses the 1 port on the tdm400 and company 2 sales person uses the 2nd port of the tdm400.

hi, I am new to * and setting up a test system. here my setup : - debian (from knoppix 3.3) - Asterisk 0.7.1 (from the debian package) - AVM Fritz card used with i4l - softphone I use for testing SJphone on windows - I can make great softphone - softphone calls - I can call from an outside line * and get connected to a softphone here my problem: I can not make outbound calls. I place a call

Hi list , I have a question with account codes, all my outgoing calls are authenticated, but now the boss wants to monitor these calls with the codes. example: maria has an extension "110", but peter was in place and use the phone maria , maria then says that she did not make that call to that number of cell. like to know who made it?, I think the pin code is my friendo , my users have

Hello all, Recently I purchased an QuadBRI card from junghanns.net after some playing around, reconfiguring dialplans etc with the exception of 1 thing everything seems to work: I seem to be unable to set the outbound callerid. The dutch telecom operator (KPN) provided me with 4 MSN's on 1 BRI interface. In the past years I'm more then used to setting the MSN without the leading 0, this