similar to: Visual Dial Plan application: Recommendations?

My international calls are not connecting. [general] pridialplan=dynamic ;prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 localprefix= I have the above in my zapta.conf - yet when I dial an international number, I get a ring, then I get the message "the person you are calling, is currently unavailable" This is an ubuntu machine, with a sangoma card, with

Hi, all I want to add more than 200 sip accounts into sip.conf, username from 6000 to 6199, password is the same, i remember there is a better way to do this case, however, i have not searched the method yet. Anybody can tell me this method, TIA. BR Mike Li -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much.

Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

Hi Can anybody help me on this ? I am using Asterisknow 1.5.0-Beta(Freepbx) I am having a problem getting the sip trunks to register. It makes no different which provider one is using. Trunk name: callcentric Peer Details: context=from-pstn fromdomain=callcentric.com fromuser=1777xxxxxxx host=callcentric.com insecure=very secret=pasword type=peer username=1777xxxxxxx Register String:

Hello, I'm trying to setup a very simple queue with 5 SIP phones. I do NOT want the agents to have to be on the phone to get calls, but I want those 5 SIP phones to ring (according to the strategy chosen in queue.conf) to dispatch calls. Is this a call back queue, or is a callback queue a queue that calls back the customer? There is conflicting info when searching for "callback

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am googling for a while but google seems to be broken today, no solution yet :D I have a PHP script which needs to be started after Dial() has ended. If there is no answer, busy, etc., it is not a problem, because dialplan continues after the Dial() application, but when the call is established (i call macro in Dial() with AGI execution, working ok) and after the call ends, dialplan

Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via

hi i am using asterisk 1.4.22 ubuntu 8.4 i have two Ethernet one for ssh and other one only for voip calls when i start a call using originate in the manager or the cli in the voip Ethernet i get something like 4Mbits/sec of traffic only 1 G711 call. if i start the call using a soft phone everything is normal. any idea? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny

I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail.

Dear, because of using dial(local/...) each incoming calls (_12X.) makes 4 ports on asterisk. I can not use goto , because of some limitations. is any way to decrease it? Best, [MAIN] exten => _12X.,Dial(LOCAL/${EXTEN}@TEST/n,60) .... [TEST] exten _X.,1,Dial(${EXTEN}@next_gateway,60)

I have attached my PRI debug out put when making an international call - hopefully it can shed some light on the situation. I am sorry if this attachment gets to the list twice, I sent one early this morning, but it has yet to appear - i may have sent that one in error. Kind Regards: Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am looking for someone that could share their code for this function: Outgoing call -> macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones -> if human connect to agent else hangup and write status to cdr. Need help with this! Regards / Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password

hi i want to ask for your opinion what is better for a call center 100 current calls and other 200 current calls make the server step by step or use a auto install cd like asterisk now, druid elastix ....? and why? Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part