[asterisk-users] Location with PRI / Analog lines

can use any channel and out going calls from any extension can use any channel. If somebody dials emergency then that specific extension dials specific channel which has its physical location in Telco database. I would highly appreciate your thoughts. Shahnawaz Mir http://www.aaanetworkx.com

[asterisk-users] Michael Wegner

http://www.villasantilles.com/home.php -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk

Xorcom XR005 is highly recommended. They work great. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=120550484883ssPageName=STRK:MESELX:IT Smir On Tue, Mar 30, 2010 at 7:34 AM, Juan Miguel jmigu...@gmail.com wrote: Hello Joseph I recommend that you use The Mediatrix 4100 Series are very

[asterisk-users] Priority based softhangup

Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will

Re: [asterisk-users] Priority based softhangup

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority based softhangup On Tue, 30 Mar 2010, mir shahnawaz wrote: Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels

Re: [asterisk-users] Priority based softhangup

Thanks Steve and Danny for your help. S Mir On Tue, Mar 30, 2010 at 3:16 PM, Steve Edwards asterisk@sedwards.com wrote: Un-top-posting... [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir shahnawaz Can you please give me idea about Softhang.agi On Tue, 30 Mar 2010

[asterisk-users] softhangup

Hi all, I am trying to drop a random channel in asterisk 1.6. The following line in extensions.conf works fine for the first channel exten = 911,4,SoftHangup(DAHDI/1-1) But I need to drop random channel for emergency not any specific one. Can someone show correct syntax for this Thanks smir

[asterisk-users] softhangup

Hi all, I am trying to drop a random channel in asterisk 1.6. The following line in extensions.conf works fine for the first channel exten = 911,4,SoftHangup(DAHDI/1-1) But I need to drop random channel for emergency not any specific one. Can someone show correct syntax for this Thanks smir

Re: [asterisk-users] Asterisk on MPLS VPN

Are you able to ping 172.16.0.1 from your router (which is connected to MPLS Network). If not then you have to tell your MPLS router where is 172.16.0.0 connected. Do you have access on this router? If you copy paste config here that would be helpful. I suspect that your ping request is reaching

[asterisk-users] 911, channel full

Hi, I am trying to implement 911 funtionality in my PBX. A call should drop if all lines are busy. Here is my context nineoneone from extensions.conf [nineoneone] exten = s,1,Set(SET_EMERG_FLAG=0) exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten = s,n,Set(EMERGENCY=1,g) exten =

Re: [asterisk-users] 911, channel full

fallthrough, channel 'SIP/501-0137' status is 'CONGESTION' Regards Shahnawaz On Wed, Mar 3, 2010 at 10:54 AM, Steve Howes steve-li...@geekinter.net wrote: On 3 Mar 2010, at 17:21, mir shahnawaz wrote: [nineoneone] exten = s,1,Set(SET_EMERG_FLAG=0) exten = s,n(checkavail),ChanIsAvail

[asterisk-users] dropping line (s) for 911

]?inprogress) exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1) exten = s,n,Wait(12) exten = s,n,Goto(checkavail) exten = s,s+2(inprogress),Congestion exten = s,checkavail+101(notavail),Goto(trunkbusy) exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3) exten = h,3,Set(EMERGENCY=0,g) Regards Shahnawaz Mir

Re: [asterisk-users] dropping line (s) for 911

I got it working now. I was not including context ninioneone in default context. On Fri, Feb 12, 2010 at 1:49 PM, mir shahnawaz shahnawaz...@gmail.com wrote: Hi all, I am trying to implement call dropping funtionality in asterisk for 911. I mean if all lines are busy and someone wants

Re: [asterisk-users] 911, location

Thanks very much everybody who contributed their thoughts. I would try to get some DID's so that each physical location can be identified by 911 call Center. Regards Shahnawaz On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: Leif Neland wrote: 2: Often callers are answered with an

[asterisk-users] 911, location

Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to

Re: [asterisk-users] 911, location

...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mir shahnawaz wrote: Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean

Re: [asterisk-users] 911, location

) - exten = _911,4,Background(emergencyin${IMAT}) Where you would record /var/lib/asterisk/sound/emergencyin100 for extension 100, etc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir shahnawaz Sent

[asterisk-users] chan_dahdi.conf for TDM404E

Hi there, I am trying to configure chan_dahdi.conf for TDM404E. Should I separate channels for dialing out and recieveing calls on this card or should I leave it random so that outgoing and incoming call get first available channels. ;FXO Modules group = 2 echocancel = yes signalling = fxs_ks

[asterisk-users] PSTN to SIP line ratio

Hi, I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me

Re: [asterisk-users] PSTN to SIP line ratio

Thanks Tim, Your response is really helpful. Its not going to be very busy. I was expecting 10:1 but I will start some where between 4-10. Thank you very much. Regards Shahnawaz Mir On 15-Oct-09, at 11:11 AM, Tim Nelson wrote: - Steve Edwards asterisk@sedwards.com wrote: On Thu

[asterisk-users] Dial without phone

Hi Is it possible to have Asterisk dial an external number without having a phone? I want to make a box that can generate calls into a normal PABX and just play MOH or similar. It's for stresstesting applications I'm developing. If it could be done via the Manager interface, it would be

[asterisk-users] Re: Calls on hold

Doesnt anyone know if this is possible? 2006/9/13, Mir [EMAIL PROTECTED]: Hello Is there a possibility for sending an event on the managerinterface (AMI) when a call is put on/off hold? Or is there any other way to detect when a call is placed on hold? Michael

[asterisk-users] Calls on hold

Hello Is there a possibility for sending an event on the managerinterface (AMI) when a call is put on/off hold? Or is there any other way to detect when a call is placed on hold? Michael ___ --Bandwidth and Colocation provided by Easynews.com --

[asterisk-users] Adding own info in AMI

Hello (again) When doing a STATUS request on the AMI, I get informations like these: Event: Status Privilege: Call Channel: SIP/311-08b97790 CallerID: 310 CallerIDName: Aastra Account: State: Ringing Uniqueid: 1157967815.19 My problem is that I want to tie my own information to a call,

[asterisk-users] Outgoing callerid in AMI

Hello I have a problem with callerid in the manager interface. I think that Asterisk has a strange way to handle callerid, until I found out to set the o-switch in the DIAL statement, it did not work the way I wanted, it still doesnt, but now it works ok in one direction. My extension is 311,

Re: [asterisk-users] Submenus

Mir wrote: Hello I'm doing an IVR-service, where pilot can check metar (airport weather information), they enter the 4 letter airport code on their phone, and get the metar read back by text-to-speech. [Metar] exten = 1,1,answer exten = 1,2,Background(Met_welcome) exten = 1,3,set

[asterisk-users] Submenus

Hello I'm doing an IVR-service, where pilot can check metar (airport weather information), they enter the 4 letter airport code on their phone, and get the metar read back by text-to-speech. [Metar] exten = 1,1,answer exten = 1,2,Background(Met_welcome) exten =

Re: [asterisk-users] Phone status

of that + more: http://micpc.com/eventmonitor/earlOn Thursday 24 August 2006 14:17, Mir wrote: Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query

Re: [asterisk-users] RE: [asterisk-dev] Phone status

[EMAIL PROTECTED]: So how about inventing a car? The auto industry is much more profitable.The point; there is no point in reinventing the wheel, why are you writing this from scratch?On 8/24/06, Mir [EMAIL PROTECTED] wrote: What do you mean? I'm not looking for someone elses work, I'm developing

Re: [asterisk-users] RE: [asterisk-dev] Phone status

What do you mean? I'm not looking for someone elses work, I'm developing an application from scratch. Michael 2006/8/24, Andrew Kirch [EMAIL PROTECTED]: Umm… Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of MirSent: Thursday, August 24,

[asterisk-users] Phone status

Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the

Re: [Asterisk-Users] IAX only speech one way

Thanks for your suggestion. Unfortunately, it didnt change anything, A can still not hear B, but B can hear A, strange.. Michael 2005/10/19, Rich Adamson [EMAIL PROTECTED]: I have two Asterisk's connected via IAX, they are sitting on the same network, via a VPN, so there should be no

Re: [Asterisk-Users] IAX only speech one way

Bingo, that was the answer. Originally, both ends had trunk=yes, and I had tried to comment it out, but it didnt help. But when I wrote trunk=no in end B, it worked. Thanks for your help, everybody. Michael 2005/10/19, David Uzzell [EMAIL PROTECTED]: Mir wrote: Thanks for your suggestion

[Asterisk-Users] IAX only speech one way

Hello I have two Asterisk's connected via IAX, they are sitting on the same network, via a VPN, so there should be no problems with firewalls. My problem is that when a person calls from A to B, A will not hear B speak. B hears A fine. I doesn't matter who initiates the call. One of the

Re: [Asterisk-Users] Quad BRI with Fedora, anyone?

Actually, the QuadBRI card is not from Digium, but manufactured by Junghanns.net Michael 2005/10/15, Lars Dybdahl [EMAIL PROTECTED]: We have a QuadBRI ISDN card from Digium. We would like to make it work with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of bristuff from the

[Asterisk-Users] Monitor DTMF problems

Hello We have discovered a problem with DTMF on Asterisk. We have a setup with a T1 from PSTN going into an Asterisk box, and then out again on T1 and into a normal PBX (EADS) We use it to record all calls going to/from the PBX. The problem is that when we record the calls (with MONITOR

[Asterisk-Users] Monitor in AGI

Hello Does anyone have an example of how to use the MONITOR command from an AGI-script ? I have tried different methods, but none of them worked :-( I'm using Python MIR ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users

[Asterisk-Users] Monitor in AGI

Hello Does anyone have an example of how to use the MONITOR command from an AGI-script ? I have tried different methods, but none of them worked :-( I'm using Python MIR ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] Play sound on connect

PROTECTED]: Hi Mir, You would need to put a Play command in before the Dial command. For example: Exten = 108,1,Play(108-greeting) Exten = 108,2,Dial(SIP/108) Etc. This however, will play on _every_ attempted call to 108. If 108 is offline or unreachable the caller will still hear the message

[Asterisk-Users] Play sound on connect

Hello A calls B, on connect I want B's greeting to be played to caller A. I can see it is possible to play a sound to B on connect (DIAL(SIP/123 ,A(hello)), but I cant se how to play a sound to A, is this possible? Thank you Michael ___ --Bandwidth