similar to: Manager API - howto

Hello, Is there a way to trace into a log file, incoming AMI requests ? For instance, I've got several apps accross the LAN, sending AMI requests such as : Action: originate Channel: Local/7539 at internal Exten: 00123456789 Priority: 1 ... Some of them might be sometimes producing some erroneous requests. I would like to check from server side, that each received request is formatted and

Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in

Hi All, I've just upgraded our CRM and it has an Asterisk Integration Module that I would like to test out. The CRM is running on one of our hosted servers in the cloud. The Asterisk server is running in my office. I am running Asterisk 1.4.21.2~dfsg-1ubuntu3. Reading the page http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf got me a little concerned

Hi anybody who would like to test drive a pre-release of the first OSX Assistant please visit the Wiki ... http://www.voip-info.org/tiki-index.php?page=Asterisk+MacOSX+Support if you find any bugs please let me know by email: benjamin (at) sunrise-tel (dot) com. Please take into account that this assistant is in the tradition of Apple's setup assistants which are always trading ease of use

Hi All, I am new to Asterisk and VOIP. I managed to get it working with sip(X- Pro) and skinny(Cisco 7940,7960). I have a call manager to which all the phones are connected. I would like some assistance integrating CCM with Asterisk. I was trying to understand the H323.conf file, but got nothing in it. Any steps, any config, any help would be highly appreciated. Thanks & Regards, Gurdeep

Thank you for all of the beta testing. New and improved graphics in this release along with drag and drop transfers and hold for all technologies. There's a screenshot on the link below. Also improved documentation so read the included README. There's also a sample xml configuration included. http://www.voip-info.org/tiki-index.php?page=Asterisk+WAMI Christian Hoffmeyer YottaDot

hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/8888888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten => s,1,Answer() exten => s,n,Wait(10) exten => s,n,Hangup() when the phone 8888888 pick up , it will come to callout context, after hangup, one cdr generate, but the

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hi all, How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command : Channel = SIP/0041435215301 Context = default Exten = 00982166501553 Priority = 1 CallerID = 0041435215301 this works with out any problems I initiate a call from one of my

Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Can someone please show me what the proper syntax is for Variable: in an Originate. Basically, I'm looking to pass a string into my dialplan as ${MYSTRING} so it's available from whatever Exten => I originate to. Thanks - -jwb -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux)

Asterisk 1.0.3 / TE410 / ISDN/PRI Zap channels As I understand it using the "Async: True" in an originate action is supposed do a "Fast Originate" "originate a call from a channel to an extension without waiting for call to complete". I'm finding no difference using Async or not, calls always wait for completion before connecting to extensions. Don't know

Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read in places that you use "originate" command and wait for an event back, does that mean you cannot place another "originate" until the event comes back ? Is it true that multiple API connections to Asterisk

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

Is there any built in functionality when using Originate to retry a call based on the DIALSTATUS? Similar to the .call file where you can set max retries and time between them? I've tried putting the logic in an outbound context/macro, but it just times out if the time between retries is too long...I suppose your not supposed to leave calls in limbo too long. Is there any other method

I have written an asterisk manager client which creates an outbound call using Asterisk manager API's Originate action. when the call is connected I run 3 applications on it. 1)read a dtmf digit from user 2)A customized application which I have written,(It plays something to user) 3)Hangup If user hangs up while app 2(see above) is executing I get a 'Event Hangup' from asterisk in my

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,

Hello, After battling with the Asterisk Manager interface(and getting it to pretty much do everything I want to do with it) I thought I'd share my experiences with those who are developing or are thinking of developing applications using it. First here's a list of some of the things the manager interface will let you do: - Dial a call from any extension/resource to any other

I am originating a call using the "Originate" action in the Manager API. It calls one party, then when they answer does the "Dial" application and calls another party and connects the two. Is there a way using the Manager API to check back later on the status of this call (is it still up, etc.)? I have found the "Status" API action, but I don't know how to get

Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does