Re: [asterisk-users] VoIP monitoring tools

Homer for voip / flow capture Smoke ping has a sip based server test feature in it as well Sent from my iPhone > On 27 Sep. 2016, at 7:17 pm, "sysad...@reed-media.com" > wrote: > > Hello, > > you can have a look on Homer > > http://sipcapture.org/ > > regards >

Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP

How about running a second asterisk instance on the same box with different IP/Port combo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP

And if we were the trunk provider in this case - I would probably trunk it to a separate machine, or run a second instance of asterisk on the same machine and bind it to the secondary IP address / sip port combo, peer the two instances and then send it out via the second instance. Sent from

[asterisk-users] Hints realtime table structure Ast 11

Hi All, Has anyone used hints in realtime ? (As in storing and loading hints from odbc) I cannot find a table structure for this anywhere...? Thanks Neeraj -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Fwd: Unable to place outbound calls

Could potentially be a NAT problem. Could you do a sip debug and see whether there is any traffic to / from the address? Sent from my iPhone > On 15 Mar 2016, at 9:52 PM, Administrator TOOTAI wrote: > > Le 15/03/2016 11:20, Feroz Ahmed a écrit : >> Hi I need help > > Hello

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

Hi Travis, Have a look at this: http://www.ipcom.at/en/telephony/siptapi/ I have used this in the past to do something similar, unless you have an Exchange Enterprise setup in which case I would suggest exploring unified messaging Thanks, Neeraj On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis

[asterisk-users] Cisco BLF c7975 notifications not working with asterisk realtime

Hi All, I've set up asterisk 11.20.0 with the blf patch and registered cisco phones to the server. I can see the subscriptions and the last state as idle, type = pidf+xml In hints I also have: Location Hints DeviceState PresenceState Watchers XXX@YYY

Re: [asterisk-users] Pops clicks at the end of sound files

Hi all, We recently decided to get a professionally recorded set of prompts for our asterisk based IVRs and received these as the following: Bit Rate: 1536Kbps Sample Size: 16bit Channels: Stereo Sample Rate: 48kHz Format: PCM I use Wavepad to convert it to: Bit Rate:64Kbps Sample Size: 8bit

Re: [asterisk-users] Features.conf - Blind Transfer

Hi guys, I'm trying to get blind transfer to work and automatically transfer call to another number on key sequence press. Extensions.conf_snippet [from-pstn] exten = _0399377744,1,Set(__DYNAMIC_FEATURES=blindxfer) exten = _0399377744,n,Set(__GOTO_ON_BLINDXFR=to-pstn ^0388924326^1) exten =

Re: [asterisk-users] record individual callers in confbridge

I'd suggest try recording in ulaw first and then convert all to wav after. It may have something to do with timing since you're using iax, what are you using as a timing source? Hardware or software? Sent from my iPhone-- _

[asterisk-users] record individual callers in confbridge

While you're testing, capture iax2 debug info as well as that may point to other factors/issues. Possible fixes: File conversion: Yes asterisk can do the conversion File convert xyz.ulaw xyz.wav As long as you selected wav format in the initial build. Timing module for iax: Check your

[asterisk-users] Ast 1.8_CentOS5.5 with timerfd as timing source

Hi All Just finished setting up a vm with centos 5.5 and asterisk 1.8.3 Using timerfd as a timing source. Has anyone got a similar setup in production ? How's performance? Thanks, Neeraj  -- _ -- Bandwidth and

Re: [asterisk-users] Email from Dialplan

I use the following: Exten = s,n(status-NOTIFY),System(echo '${DIALSTATUS} on ${CALLERID(num)}' at ${STRFTIME(${EPOCH},,%H%M%S)} | mail -s Call Unsuccessful on DNIS '${ARG10}' neeraj.ch...@ocis.com.au) -- _ -- Bandwidth and

Re: [asterisk-users] MS-SQL / Freetds -- func_odbc

Hi folks, How would I go about running a stored procedure call from asterisk via func_odbc. I'm after an example entry in func_odbc if possible for ast 1.4 Also, if someone could post an insert statement that actually works, would be nice. Thanks, :) --

Re: [asterisk-users] Asterisk ODBC Insert issue

Hi guys, Having issues with doing an insert statement using ast 1.4.24: [START] dsn=mssql-asterisk write=INSERT INTO testdb (callarrival,callerid) VALUES ('${VAL1}','${VAL2}') SET(ODBC_START()${TIMESTAMP},${CALLERID(num)}) No errors pop up on execute, but nothing gets inserted. Read and

Re: [asterisk-users] SKYPE - Authenticate incoming call

Hi All, After getting licences for Skype for asterisk a while ago I finally got around to setting up a server with two channels and setting up a bcp on the skype end. The idea behind this is the following: Users can dial into the PBX, get authenticated and only after

Re: [asterisk-users] SKYPE - Authenticate incoming call automatically

Hi All, After getting licences for Skype for asterisk a while ago I finally got around to setting up a server with two channels and setting up a bcp on the skype end. The idea behind this is the following: Users can dial into the PBX, get authenticated and only after authentication get

Re: [asterisk-users] asterisk and cisco 2800

Hi Giorgio, Why don't you terminate calls on the cisco router via SIP? -- Message: 11 Date: Fri, 02 Jul 2010 18:54:31 +0200 From: Giorgio Incantalupo gincantal...@fgasoftware.com Subject: [asterisk-users] asterisk and cisco 2800 To: Asterisk Users Mailing List -

Re: [asterisk-users] CDR to MS-SQL via ODBC issue

On Wednesday 05 May 2010 18:29:26 Neeraj Chand wrote: --- Message: 10 Date: Wed, 5 May 2010 10:26:34 -0500 From: Tilghman Lesher tles...@digium.com Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue

Re: [asterisk-users] CDR to MS-SQL via ODBC issue

Hi guys, Having issue with getting CDR to write to MS-SQL via ODBC. cdr_odbc: Connected to freetds-connector cdr_odbc: Error in PREPARE -1 cdr_odbc: Query FAILED Call not logged! == Spawn extension (cisco, ##, 2) exited non-zero on 'IAX2/ast-507 Isql

Re: [asterisk-users] CDR to MS-SQL via ODBC issue

asterisk-users@lists.digium.com Message-ID: 201005051026.34929.tles...@digium.com Content-Type: text/plain; charset=iso-8859-1 On Wednesday 05 May 2010 06:51:48 Neeraj Chand wrote: I can connect to the database and run via isql, and also use func_odbc, etc with res_odbc configured with the same

Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install

Use kickstart to configure your default packages, and then set up a shell script to install the additional stuff you need. :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

[asterisk-users] Asterisk realtime chat

Hi, Do a google search for openfire + asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] MYSQL queries from dial plan

Hi all, I currently run small scale mysql queries from the dialplan exten = s,n,MYSQL(Connect exten = s,n,MYSQL(Query resultid ${connid} exten = s,n,MYSQL(FETCH fetchid exten = s,n,MYSQL(Clear ${resultid}) exten = s,n,MYSQL(Disconnect ${connid}) This currently takes about 4 seconds to

Re: [asterisk-users] FAX for Asterisk

Just finished with the instructions from digium website/ net on how to compile FFA: After restart, modules did not get loaded so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26]

Re: [asterisk-users] asterisk-users Digest, Vol 65, Issue 38

Did you check the jitter settings on asterisk the phones as well? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Extension in use

There are a couple of ways you could see that, One would be by having a service .NET connected to the manager interface and watching for activity on the phone, this way you could tell if the phone is busy or not. [If phone has more than one line then set call-limit=1] Is this for routing

Re: [asterisk-users] CDR userfield -

Set(CDR(userfield|r)=blah) This works for me on 1.4.24 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] odbc to ms-sql server

Hi all, I'm trying to set up an odbc connection to a ms-sql server from an asterisk 1.6.1 install My problem is that I cannot get asterisk to build func_odbc res_odbc.so I installed yum -y install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel And then went on to reconfigure /

Re: [asterisk-users] odbc to ms-sql server

Gotcha! Missed libtool! :) -Original Message- From: Neeraj Chand Sent: Friday, 6 November 2009 6:43 PM To: 'asterisk-users@lists.digium.com' Subject: RE: odbc to ms-sql server Hi all, I'm trying to set up an odbc connection to a ms-sql server from an asterisk 1.6.1 install My

Re: [asterisk-users] Astreicon presentations

Hi Folks, Are all the astricon presentations up? I'm especially after the one that tilghman did. I caught the tail end of the prez when I decided to skip the session I was attending and go for that one. :) ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Asterisk/Cisco AS5300 = Two problems in

Please post your dial peer configurations. We have as5400 (5) working with asterisk servers also. The cisco routers are at the edge of the network (connected to PSTN via E1) and send calls to asterisk over SIP ___ -- Bandwidth and Colocation

Re: [asterisk-users] SIP debugging enabled : written to log

There was a presentation at astricon by Clod, that covers just this CLI Filters What this does is show only the filters that you set on asterisk cli, and your /var/log/asterisk/full log file also only contains the filtered output. I believe it would have been handier to have filtering, but

Re: [asterisk-users] DUNDi + SIP Realtime

JR - couldn't find your whitepaper from astricon06 online, links are broken would it be possible for you to email it to me? I have not tried setting up DUNDi yet, but from the sound of it, seems like it would be pretty handy. I have sip phones registering to two asterisk servers [primary and

Re: [asterisk-users] Custom auto-install asterisk using ks.cfg

Hi guys, Anyone done this with CentOS and asterisk 1.4? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users

Re: [asterisk-users] dCAP Exam

Hi All, Thanks for all the wonderful contributions, from cell phones right up to proxies, etc... Many thanks also to Tony Turner for the great advice. As for Jared, what can I say...simply legend... :) I believe this is what I was after. :) For all those attending AstriconSee you there!

Re: [asterisk-users] dCAP Exam

To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 200909150838.05001.tles...@digium.com Content-Type: text/plain; charset=iso-8859-1 On Monday 14 September 2009 22:31:26 Neeraj Chand wrote: Is there anywhere I can possibly get a model

[asterisk-users] dCAP Exam

Hi folks, Is there anywhere I can possibly get a model of the exam itself, maybe possible scenarios for the prac, etc? To people who have done the examany helpful hints ? Thanks, ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Time of Day Routing

Asterisk version 1.4 From: Neeraj Chand Sent: Friday, 14 August 2009 8:17 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] Time of Day Routing Hi David, With this: ifTime(00:00-12:00|*|*|*) Whatever time you specify at the end

[asterisk-users] Time of Day Routing

Hi David, With this: ifTime(00:00-12:00|*|*|*) Whatever time you specify at the end, I believe asterisk continues to evaluate this condition as true for 2 more minutes. So in this case, it will be valid for 00:00-12:02, even though you've specified 12:00 Cheers! Neeraj

[asterisk-users] Astricon 2009 - dCAP

Hi folks, Going to astricon this year? Feeling a bit nervous as planning to take the exam this time. Any one else doing the same? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15

[asterisk-users] FW: re: Asterisk Outbound with Failover, alarm notification, dial status and hangupcause capturing to CDR from Dialplan

Managed to implement this on asterisk v1.4.24.1, Also, Hangupcause updating to user field. However, this only works on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort

Re: [asterisk-users] Writing Hangup causes to CDR record

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? Neeraj Chand Support Analyst Fiji Islands Australia T: +6793342526 T: +61388924326

[asterisk-users] FW: Writing Hangup causes to CDR record

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] Fwd: Asterisk With Cisco Voice Router

Hi, We have AS5400's set up with asterisk boxes. Initially we had similar issues, but as described, you need to have dial peers to handle both incoming and outgoing peers. Please post your dial peer configs as well as the serial interface configs. I also found that until I add [isdn

Re: [asterisk-users] asterisk-users Digest, Vol 58, Issue 9

--- SIP read from 192.168.32.245:5060 --- SIP/2.0 481 CallLeg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport From: asterisksip:aster...@192.168.32.16;tag=as2ff08179 To: sip:5...@192.168.32.245:5060;user=phone;tag=c0a80101-2ce1bc03 Call-ID: