similar to: AMI disconnects

Hi there, I need a Python interface to asterisk manager for my own project. The voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+manager+Examples) lists 4 python projects for this purpose: Fats, py-Asterisk, pyst and StarPy. Because my project is rather small and I don't want to involve twisted in, the options left for me are py-Asterisk and pyst. So I want to ask your opinion:

Hello Guys, In the case of a multiserver environment for outbound automatic calls, can you share you experience and preference between call files and ami originate ? thanks -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi people Just having a quick check to see if anyone is using any AMI proxies and which are the most popular. For our purposes it must be able to connect to multiple asterisk instances. Thanks for the help. Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

Hello, Is it possible to run Starpy and Asterisk on different machines ? A quick glance at http://www.vrplumber.com/programming/starpy/ seems to tell it is possible but Debian's python-starpy package installs Asterisk. What do you think ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks for replying. So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian setup the alternatives are either : - to install it from source - tto build my own custom package removing this asterisk dependency (is it easy or even possible ?) - to use another solution such as pyst. Regards 2014/1/16 Adolphe Cher-Aime <acheraime at gmail.com> > Yes you can. This what

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re <tab><tab> realtime reload shirley*CLI> core show version Asterisk

Where this revers IP comes from ? == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack -- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack -- Hungup 'IAX2/0.0.29.199:4569-5255' -- Executing [s at

Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show <queue_name> I get the following numbers: <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hi Everyone, I want to know each and every parameter's detail that can be included in the read= write= in manager.conf Where can I find this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110107/eacb9f1a/attachment.htm>

Hi All, I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend. My dialplan: exten => _XXXX,1,Dial(SIP/${EXTEN},60,rt) Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument In 1.6 there was no problem, I have got Channel is

Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I

Hi all, I want to upgared from asterisk-1.6.2.6 version to asterisk-1.8.0 version. When i execute "make" command for compilation i have seen below errors. In file included from /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31 /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:233: error: field ?AST_DATA_IPADDR? has incomplete type

Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)

Hi, I'm giving Queue-Tip a try, following installation instructions in http://queue-tip.rubyforge.org/install.html. My setup is : ruby 1.8.7 rubygems 1.3.7 rails 3.1.3 Adhearsion 1.2.3 I'm struck in step 7 in the above installation procedure : # rake --trace db:create (in /usr/local/src/queue-tip) rake aborted! no such file to load -- initializer

Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. I know there is a DBdeltree(<family>) function. Is there a context that only runs once (automatically) at Asterisk startup (so that I can call this function)? Also is AstDB lookup faster than a func_odbc lookup? Is there a

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton