similar to: Manager API and astmanproxy

Is this still maintained ? Last update that I can find was nearly 18 months ago. I am currently using 1.22pre. I have had some problems with the astmanproxy process dying on me, and was wondering if anyone else has had this problem, or knows of any newer versions. Julian.

I've got around 45 people who need to place calls from our inhouse app. What is the considered "best practice" for placing these calls: 1) All clients connect to astmanproxy, and use AMI API Originate command 2) All clients connect directly to the astersik AMI and use the API Originate command 3) All clients create a db record, some process reads the record and writes out a call

Does any one know how to change astmanproxy to be able to a) compile without optimisations b) dump a core I've had it crash several times over the past couple of months, but there is no way to debug what's going on. I like the way a core is produced when (if!) * crashes, and would like to have the same thing on astmanproxy. Thanks Julian

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

I have concocted a system for my children's primary school where parents can dial in and subscribe to an "emergency broadcast" message so that they can be automatically contacted in case of a problem like the school being shut because of snow etc. I would like to provide an 0800 number service for this, so that there is no cost to the parents, but obviously I would like to get

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

Hello, I cannot find documnetation explaining how to access AstManProxy. I am working with Asterisk Java and accessing Asterisk Manager. I wonder if AStManProxy is using the same API as Manager? Can I access it with Asterisk java too ? Regards, Daniel

Hi folks -- For those of us trying to develop interactive web-based Asterisk applications, it can be a challenge getting Asterisk integrated in a cgi/mod_perl/php environment. The load associated with making multiple connections to asterisk via the manager port, having to teach our applications to parse "manager-ese", and a variety of other factors have all pointed to the need for

I am using 1.4, and have the above device, and it worked really well with monitoring 18 "hints" aka devices. Now, I've moved us to a hotdesking paradigm where the user is the "extension" not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ?

Anyone got any thoughts on this and how it compares to the chan_skype that's due soon ? "OpenSky is a free service provided by Gizmo5 which allows *any* mobile phone, web browser or IP aware phone network (SIP, asterisk, etc) to communicate with Skype users. OpenSky supports sending text messages and voice calls." http://www.gizmo5.com/pc/opensky/ Julian

We've just had the problem where our DNS server went down, and * started to act "funny". Is the best solution to install a local DNS server on the * box, and have no other DNS servers ? - this is an internal app, no need for any external DNS resolution at all. Julian. ______________________________________________________________________ This email has been scanned by the

Greetings -- Many of you have downloaded and tried out Astmanproxy, a multi-threaded C-based proxy for Asterisk's Manager Interface. It has been under development since April 2005 and was presented at the Madrid Astricon in June, and will also be presented at Astricon in Anaheim in October. There has been interest in setting up an Astmanproxy mailing list specifically devoted to

Greetings everyone, I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy allows you to communicate with multiple Asterisk boxes from a single point of contact using a variety of I/O formats, now including support for XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. Astmanproxy is

Does anyone have any idea on how to force cepstral to convert a number to speech ? I have noticed that sometimes it speaks the number correctly, and at others it doesn't. 1) 787 is pronounced 7-8-7 2) 123 is pronounced one-hundred and twenty-three. 1) is wrong for what i need, 2) is perfect. Is there anyway of forcing numbers to be pronounced as 2) ? I've tried looking at the ssml

Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

I have a GXP2010 phone, the one with 18 blinky lights ;) I currently provision the phone by writing out the conf file, encoding it and sending it to the tftp server. I was wondering if anyone had managed to automate the web side of provisioning ? TIA Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security

Can anyone give me any idea on where to start looking for this ? 1.4 svn (ish) It has appeared twice in the last hour on a system that gets numerous inbound calls to the same number TIA Julian [Mar 27 17:21:07] WARNING[3239]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 2 ^ [Mar 27 17:21:07] WARNING[3239]: