similar to: AMI and pyst

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:

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

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

Hi I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP. However, we exclusively use the asterisk realtime architecture using the mysql connector. Looking at tutorials we have to set encryption=yes and transport=tls for any peer we want encrypted traffic for. Having a look at contrib/realtime/mysql/sippeers.sql from the source code shows that the encryption column is

Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer <peer-name> load. Has anyone got any experience of connecting to Lync using ARA?

Hi people I'm just mailing to see what people are using for CTI solutions with asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce? Thanks in Advance 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 Address: PACKNET LIMITED, Duplex 2,

HI there I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? Thanks in Advance 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 Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester,

Hi Is there any harm in using res_mysql for some things and res_odbc for others? We already use res_mysql for ARA but could do with having CEL logged to MySQL. Thanks in Advance 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 Address: PACKNET LIMITED, Duplex

Hi I'm using asterisk 1.8 but I'm sure this applies to other versions. If someone puts a call divert on a handset such as a Snom phone I get this type of SIP message on receipt of an inbound call: Got SIP response 302 "Moved Temporarily" back from xxx.xxx.xxx.xxx:xxxxx Which then triggers a local channel to make the call. Is there any way I can access that IP address inside

Hi We're using asterisk 1.8.23.1. Our inbound calls are routed into the default context with explicit number matching. If found they are passed on to a distinct context for the number being called using the Goto application. If the call is successful or even if it has no answer, the cdr dcontext field has the correct second context. However, if the call fails or is busy, and even though we

Hi I'm using asterisk 1.8.23.1 but I've seen this same issue in previous versions of 1.8. I have created some work arounds but the behaviour is incorrect. This is the scenario: Call comes in and goes to appropriate dialplan In the dialplan the call is forwarded to another number using a Local channel (and using /n ) e.g. Dial(Local/<my-number>@outbound-context/n,60) The number is

Hi I'm using asterisk 1.8 Does anyone know how to use the queue reload command. The built in help doesn't really help. queue reload {parameters|membe Reload queues, members, queue rules, or parameters Regards 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

Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? Thanks in Advance Ish -- Ishfaq Malik

Hi Is anyone using asterisk on CentOS 7? If so, is it working fine and as expected? Regards 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 Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552

Hi This is a bit of an exploratory question for groundwork before I start playing with asterisk 12. I've spotted the very useful looking file contrib/realtime/mysql/mysql_config.sql in the source. Are the table names starting ps_ all to do with PJSIP? Direct MySQL connection has been deprecated for quite a while, will I need to use ODBC for PJSIP or will it be supported by the old

Hi Is there any way to change the preferred audio playback format in asterisk (I'm using 1.8.25.0) i.e. first check for gsm, if doesn't exits then check for slin? Regards 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 Address: PACKNET LIMITED, Duplex

Hi I'm using asterisk 1.8. How are channel names constructed. I always thought they were <technology>/<peer>-<hex counter> but I've had a lot of instances where a channel name doesn't have the correct peer as part of it. Is it unwise to use channel names to extract the peers involved in a call? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited

Hi Using asterisk 1.8 NoOp and Verbose both put messages into the logs as VERBOSE, is there any way to put a message into the logs as NOTICE from within a dial plan? Thanks in advance 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 Address: PACKNET LIMITED,

Hello all If I look at the sip peers table definition as provided with the source of asterisk-1.8.23.0/ (looking at contrib/realtime/mysql/sippeers.sql) for the sendrpid column it's an enum with 2 possible values, yes and no. However, the sip.conf allows 4 values, no, yes, rpid and pai. Is this discrepancy an oversight? Is it possible to set the system default to pai but an individual peer

Hi queue reload(queue name) or queue reload all for example queue reload reception From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ishfaq Malik Sent: Thursday, January 8, 2015 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] queue reload command Hi I'm using