Re: [asterisk-users] configure my voip provider

nal Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Monday, March 05, 2012 4:51 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] configure my voip provider Le 05/03/2012 07:37, Baha @ S

[asterisk-users] configure my voip provider

Hello all, I am trying to configure my voip provider on asterisk box, I cannot get the right configuration after trying many possibilities, always getting circuits busy ,message, please let me know the meaning of the debug message. And by the way, when I use a normal sip phone, I can dial normall

Re: [asterisk-users] making announcements

: [asterisk-users] making announcements On 10/05/2011 03:36 PM, Baha @ SH wrote: the phone hardware has to support it, and be setup to allow it (answer without ring and go to speakerphone mode) if it does its pretty straightforward to get working - but I have had issues in the past where when you

[asterisk-users] making announcements

Hello everyone I would like to know how possibly to make an announcement with asterisk. i.e. I have a phone registered with extension 100 when I dial (for example) 500, then all the phones with extension 501,502,.510 will automatically answers my call, and I speak my announcement. Any he

[asterisk-users] changing sip port

Hello How can I run the sip service on asterisk on another port beside 5080? I mean asterisk will still take sip requests on port:5080 and another custom port, lets say port:6080 Thanks for any help -- _ -- Bandwidth an

[asterisk-users] all circuits busy now

I am getting all circuits busy now. But with my sip phone I can dial normally this provider... I cant see what's wrong here, any idea how to dig into this? My in/out: canreinvite=yes type=peer qualify=yes insecure=very host= x.x.x.x port=5080 username= xxx secret= xxx context=outbound-allroutes a

[asterisk-users] MGCP

Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user & password, I tried a custom trunk: MGCP/$outn...@user:passw...@66.152.163.106:4000 Not seems to help, Any suggestions plz? --

Re: [asterisk-users] a2billing muting "enter the phone number"

er 25, 2010 10:01 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] a2billing muting "enter the phone number" -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On

[asterisk-users] a2billing muting "enter the phone number"

How can I mute the message "please enter the number you wish to call and press the # key" in a2billing??? I tried use_dnid = YES but still I keep getting the message prompt... thanks -- _ -- Bandwidth and Colocation Provided b

Re: [asterisk-users] dials a trunk when off hook

-users] dials a trunk when off hook On Fri, Oct 22, 2010 at 7:10 AM, Baha @ SH wrote: > How can I let asterisk immediately dials a trunk when off hook? > For DAHDI: chan_dahdi.conf [channels] immediate=yes channel => 1 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: p

Re: [asterisk-users] dials a trunk when off hook

r AGI it is possible:) On Fri, Oct 22, 2010 at 1:42 PM, Gordon Henderson mailto:gordon%2baster...@drogon.net> > wrote: On Fri, 22 Oct 2010, Baha @ SH wrote: > How can I let asterisk immediately dials a trunk when off hook? If it's a SIP phone then read the manual for the

[asterisk-users] dials a trunk when off hook

How can I let asterisk immediately dials a trunk when off hook? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] a2billing

ot;Press 8 to fill up with a voucher". System Settings is the last in the menu. -Bruce On Tue, Oct 19, 2010 at 2:31 AM, Baha @ SH wrote: I am sorry , but where is System Settings??? And what is the parameter name? And also, id like to mention that the voucher is working, only when balance

Re: [asterisk-users] a2billing

authentication or CLID authentication that their balance is below threshold and they should pay. -Bruce On Mon, Oct 18, 2010 at 2:35 PM, Baha @ SH wrote: Not sure if a2billing can be shared here, but ill give a shot If the credit < min_credit the IVR play: sorry you have 0 credit and hangup

[asterisk-users] a2billing

Not sure if a2billing can be shared here, but ill give a shot If the credit < min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFI