[asterisk-users] AGI Script for thinQ CNAM lookup

Hi All, Does anyone have and can share with me an AGI script to dip thinQ for cnam? oR perhaps dialplan curl using curlopts? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and

Re: [asterisk-users] DUNDI with minimal features

et/wp-content/uploads/2007/08/DUNDi_So_Easy.pdf Good luck!. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk c

Re: [asterisk-users] Unicast RTP Paging

it master branch, I'm not running git master in production but could really use this functionality. Any ideas on how I could backport/patch UnicastRTP to another branch? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- __

Re: [asterisk-users] Unicast RTP Paging

> On 15-10-20 07:18 PM, JR Richardson wrote: >> Hi All, >> >> I playing around with multicast paging, I saw a post from Josh Colp >> about adding unicast support into chan_multicast_rtp but not finding >> details if this is incorporated in dialplan functions or not

[asterisk-users] Unicast RTP Paging

-director cisco router. Can anyone point me in the right direction? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

[asterisk-users] Asterisk Tech/Eng Positions Open In Dallas TX

We have a couple of positions open, please contact me off-list if interested. http://www.ntegratedsolutions.com/voice-engineer-dallas/ These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

[asterisk-users] Cisco 7940 SIP 8.12 no audio when using Outbound Proxy

Hi All, Simple scenario: 7940 SIP>http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aster

Re: [asterisk-users] Voicemail Prepend Message Forwarding Not Working [SOLVED]

voicemail.c manually from the patch file (revision 233691), recompiled and now prepending voicemail works. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digita

[asterisk-users] Voicemail Prepend Message Forwarding Not Working

y running for a customer, 1.6.0.28. Does anyone have a patch file that will apply to this version or an app_voicemail.c file that is already patched and will compile with this versions to fix this particular bug? Thanks. JR -- JR Richardson Engineering

[asterisk-users] Use Allworx Phones With Vanila Asterisk PBX?

g if these will work with vanilla Asterisk system or are they hard wired for Allwork systems only? Any feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Job Posting

Ntegrated Solutions in Dallas, TX is still looking for voice guy. This position is for US hire only, will not sponsor H1B work visa. http://www.ntegrated.net/careers/ Thanks. JR -- JR Richardson Engineering for the Masses

[asterisk-users] Asterisk Tech Job Posting Dallas Texas

Hi All, Ntegrated Solutions is looking for a full-time Asterisk/Telecom Tech and a .net/php developer. http://www.ntegratedsolutions.com/careers/ Forward resume' to j...@ntegrated.com Thanks. JR -- JR Richardson Engineering for the M

[asterisk-users] Asterisk SIP Refer Transfers

well? Any guidance is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar e

Re: [asterisk-users] Auto ban IP addresses

; You may want to check out this presentation form the last Astricon, it may be relevant: http://www.astricon.net/2012/videos/Automated-Hacker-Mitigation.html Cheers. JR -- JR Richardson Engineering for the Masses -- _

Re: [asterisk-users] Asterisk 1.6.0 disable cdr account logs?

> JR Richardson wrote: >>> My bad. I sent Igor to the boneyard to fetch 1.6.0.28 and it appears to me >>> that by commenting out lines 309-312 and doing a fresh make you eliminate >>> the extra files (or make them empty). >>> >> Appriciate the sug

Re: [asterisk-users] Asterisk 1.6.0 disable cdr account logs?

> Just add noload=cdr_csv.so to modules.conf > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson > Sent: Friday, October 19, 2012 5:09 PM > To: asterisk-users@lists

[asterisk-users] Asterisk 1.6.0 disable cdr account logs?

Hi All, I would like to disable the cdr account logs but in 1.6.0 but the 'accountlogs=no' switch is not available till 1.8 as far as I can tell. Is the any switch I can turn off int he Mkae file for the cdr_csv.so module to disable accountcode logs? Thanks. JR -- JR Richardson E

Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug

       nat: NULL >            allow: ulaw >         disallow: g729 >         insecure: invite >         callerid: NULL > rfc2833compensate: NULL >          mailbox: NULL >   session-timers: NULL >  session-expires: NULL >    session-minse: NULL > session-refresher: NULL >

[asterisk-users] Asterisk Configuration GUI Question

terface, brand with my business logos, add or remove configuration elements. I kind of like the Digium Asterisk GUI but I'm just not real familiar with it, just test driving it a bit. What I do like about it is the flat file manipulation, no database needed. Any guidance is much appreciated.

Re: [asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks [WORKING]

to the new 1.8 call servers and on to the carrier. I don't know why this seemed to fix the issue, I'm not 100% convinced it really did. I reverted the change and could not reproduce the issue, so I also suspect the upstream carrier started working or may have changed something co

[asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks

Hi All, I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in routing calls to upstream carrier via SIP trunks out. I spent a lot of time in the lab testing 1.8 which included heavily testing DTMF with no issues that came up. It all just seemed to work fine. But then a

Re: [asterisk-users] Can we use MySQL native connector for ARA?

> Hello Everyone, > > The documentation suggests using unixodbc for asterisk realtime. Is > there any way > we can just use native database clients such as libmysqlclient from > MySQL? The native > clients tend to be more up-to-date. > > Thanks in Advance, > > Nick. I've used the MySQL addon fo

Re: [asterisk-users] Asterisk 1.8 Manager Perl Script Problem [SOLVED]

On Mon, Oct 3, 2011 at 5:01 PM, JR Richardson wrote: > Hi All, > > Trying to upgrade some call servers, in the lab making sure all my > applications work, ran into an issue with some manager perl scripts > that pull and reset database info, it seems the command and result > resp

[asterisk-users] Asterisk 1.8 Manager Perl Script Problem

nd(Event => 'DBGetResponse'); my $cnamreset2 = $result102[1]; ##disconnect the manager connections## $astman1->disconnect; $astman2->disconnect; print "Total CNAM Count for last month is $cnamtotal\n\n

[asterisk-users] Asterisk SIP Trunk with CUCM Express, Disable Comfort Noise?

on what else to look for? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every T

Re: [asterisk-users] asterisk-users Digest, Vol 80, Issue 73

Some of these bots are real malicious with attempts coming in the hundreds/sec so my philosophy is to block soon and block often. If you are going to run an automated blocking mechanism, you should get proficient with un-blocking as well for accidental blocking. JR -- JR Richardson Engineering

Re: [asterisk-users] asterisk and fail2ban

. > > So I forge one SIP packet and I get you to block the IP address of your > SIP trunk (or your IAX trunk)? > > Cool! > > -- >               Tzafrir Cohen Good thing I ignore my own IP blocks

Re: [asterisk-users] asterisk and fail2ban

lnet/config delay but it is very minimal and works very well. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a liv

[asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny

] section of asterisk.conf with no effect. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar e

[asterisk-users] sip attended transfer beep

Hi All, I see some patches about adding atxfer beep sound in the sip channel, but I'm not clear on when this was implemented in what version? I don't see the added function in chan_sip in 1.2.24 or 1.4.21 or 1.6.0.28? Where is this code implemented, what stable release? Thanks.

[asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician

Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

[asterisk-users] SIP and ANI

for any clarification. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Cisco SIP 8.5 and 9.0 Issues

> Hi list, > I was wondering if anyone had any solution to either one of two issues > I'm having: > I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware, > it works very well for the most part, but after less then a week of > heavy usage, eventually the phone gets into a state wher

[asterisk-users] OT: HUD3 and NON-Trixbox Asterisk?

? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT

> Date: Thu, 24 Jun 2010 15:32:39 -0400 > From: Ben Winslow > Subject: [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT > To: asterisk-users@lists.digium.com > Message-ID: <4c23b2d7.9090...@pa.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello folks, > > I've been tryin

Re: [asterisk-users] Being attacked by an Amazon EC2

usy with other matters. If anyone here would like to pick up the torch and move this along, I can certainly provide info on how far along we got and contact info for the parties involved. Please contact me if you have time to work on this and are interested. I'm sure the Proj

Re: [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution

k2,${EXTEN},1:siptrunk1,${EXTEN},1) [siptrunk1] exten => _X,1,Set(GROUP()=siptrunk1calls) exten => _X,n,Dial(SIP/${ext...@siptrunk1,60,) [siptrunk2] exten => _X,1,Set(GROUP()=siptrunk2calls) exten => _X,n,

Re: [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution

Thanks Steve, works great: exten => _X.,1,Set(uniqueidcut=${CUT(CDR(uniqueid),.,2)}) exten => _X.,n,Set(result=${MATH(${uniqueidcut}%2)}) exten => _X.,n,GotoIf($[${result} > 0 ]?siptrunk1,1:siptrunk2,1) Thanks. JR -- JR Richardson Engineering for the Masses --

[asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution

Hi All, I know I can do this pretty easily with one of the SIP Proxy/Routers, I already do this using OpenSER as a load balancer. I have a special requirement that insist an Asterisk server, 1.6.1.x, is used. I will have 2 SIP trunks coming into the server and I will have to send calls to

Re: [asterisk-users] Dropped Calls

lution was to downgrade to another version or switch to 1.2 or 1.6 depending on what features I need for the system. Sorry I couldn't be of any help, but I feel your frustration. JR -- JR Richardson Engineering for the Masses -- ___

[asterisk-users] Asterisk 1.6.1.12 with Grandstream HT502 T38 Fax

Hi All, I'm in the lab with Asterisk 1.6.1.12 and several ATA's testing T38. I hit a snag with the Grandstream HT502. It only seems to nail up a session at 9600bps. The Grandstream GXW4104 nails up consistently at 14400bps. I'm using the same equipment in the same configuration, just switch

Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license

c Speed, Disc I/O? Would there be a problem running 3 to 4 PRI's full of T38 to SIP Faxes on one server? Could the Attrafax software handle that volume? Thanks in advanced for any feedback. JR -- JR Richardson Engineering for the Masses --

Re: [asterisk-users] 1.6.x SIP allow incoming calls based on from ip address?

'host=ip address' is not being looked up. It works in 1.4 but not in this version. I'll do some more debugging and try to figure out what is going on. Thanks. JR -- JR Richardson Engineering for the Masses -- _

[asterisk-users] 1.6.x SIP allow incoming calls based on from ip address?

t=ip address context=provider_1_incoming or something like this: [from ip address] type=trunk context=provider_1_incoming authentication=none Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] Using SIPPEER status with CUT function? SOLVED

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)".  Seems to work fine. > > Now I would l

[asterisk-users] Using SIPPEER status with CUT function?

"OK (48 ms)" and then do some follow on stuff if the status is OK. I'm running into syntax errors in the Set command, I think due to the spaces in the SIPPEER status. Any suggestions on how to deal with the 'spaces' in the status? Thanks

[asterisk-users] Asterisk 1.4.28 intermittent one way audio?

hat can be identified and resolved. Or maybe suggest another version of 1.4 that does not have an issue like this at these volumes? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] AGI perl script set timeout within script?

f the AGI server in your dialplan? > Ok, I went with #4 for a bit, then resolved to #5 (pardon the pun), works fine. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by htt

Re: [asterisk-users] AGI perl script set timeout within script?

> >> On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson > > > wrote: > >>> problem I'm running into is if the DNS server is not responding, the > >>> script hangs and waits for 30 seconds before returning to the Asterisk > >>> dialplan. ?I

[asterisk-users] AGI perl script set timeout within script?

arse(); # Set variables according to supplied arguments $number = $ARGV[0]; $AGI->exec("agi","agi://agi.server.com/script.agi?user=username&number=$number"); *** Any assistance will be appreciated. Thanks. JR --

Re: [asterisk-users] Realtime mysql extensions mutiple queries for each priority?

> On Monday 28 December 2009 23:49:13 Tilghman Lesher wrote: >> On Monday 28 December 2009 18:09:15 JR Richardson wrote: >> > I turned on console debug to see the actual mysql queries and to my >> > surprise and concern, I see every query for an extension priority &g

[asterisk-users] Realtime mysql extensions mutiple queries for each priority?

ySQL RealTime: Everything is fine. test1-6*CLI> Any guidance on trouble shooting this will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9

>> for IFPs, and does not allow successful FAXing at any possible bit rate >> (except for 2400 bits per second using 10 millisecond IFPs, but no FAX >> stack would do that). >> I was having similar issues, trying Asterisk 1.6.1.12-rc1 resolved it. http://www.mail-archive.co

[asterisk-users] T38 Passthrough 1.6.1.12-rc1 Good Results

particular area and also thank the dev team for responding to the bug tracker, taking suggestions for improvements and doing the coding to make Asterisk the best it can be. I can't wait for T38 gateway. Keep up the good work. Thanks. JR -- JR Richardson Engine

[asterisk-users] SIP Change canreinvite=yes/no from dialplan?

ng like that. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Asterisk SIP to Cisco IAD2430 Series?

rking. I'm wondering if anyone has tried the new Cisco 2430 series IAD's and have been successful and reliable, care to share your experience and sample configs? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colo

[asterisk-users] Looking for the asterisk 'off' sound file

can send it to me, gsm or ulaw? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium

[asterisk-users] strange cisco nat issue

s an Cisco IOS bug or maybe a router overload? I've searched for a cisco nat bug with no luck. So my question is, has anyone else experienced this type of issue and if so, is there a solution to resolve? Thanks. JR -- JR Richardson Engineering

Re: [asterisk-users] DUNDi + SIP Realtime

g that I'm doing terribly wrong, > that would break everything and make the universe collapse into itself > when I apply the same principle on production? > > I'll be happy to provide more details in case there are any doubts. I > really appreciate your feedback, no matter

[asterisk-users] Using the PBX Directory from a Blackberry

, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. That is how folks can use a Blackberry effectively with the PBX Directory application. Hope this helps. JR -- JR Richardson Engineering for the M

Re: [asterisk-users] MeetMe and setting conference timeout

m-goodbye incomingconf136 6 Hangup JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] Asterisk SIP trunk to Cisco IAD2400

the POTS line config for the IAD. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.di

[asterisk-users] Strange voicemail problem when call forwarding off local PBX

y anyway so no disruption of local function when the user is in the office. Hope this helps. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSC

[asterisk-users] Realtime dialplan application versus REALTIME dialplan function

Hi All, I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with converting the Realtime application to the REALTIME function. I have the method down and understand simplistically what is going on, at least enough to get my old 1.2 apps to run in 1.4 functions. I do not under

Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories

but it make a good statement about why their prices where high for the past few years. The technology and how it is delivered has not changed much over the past year so the "Economic Downturn" has affected them enough to reposition their margin strategies. JR -- JR Richardson

[asterisk-users] Asterisk 1.4.21.1 intermittent presence working with Polycom

ain. We are using the same firmware on the phone that worked fine with the Asterisk 1.2 code, Polycom 650 with 2.1.1. So I'm guessing there is something particular with this version of Asterisk. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering f

[asterisk-users] cdr_addon_mysql 'Failed to insert into database' stops * call processing

there possibly a patch to addons that would relieve this issue? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update op

[asterisk-users] Asternic Call Center and Asterisk 1.4 Queues

the web page does not change, always shows 'not in use'. The page does update with 'Last In Call' info after hangup of a call. Any ideas? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation

Re: [asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz

,0) kernel /boot/vmlinuz-2.6.18--686 root=/dev/sda1 ro acpi=off initrd /boot/initrd.img-2.6.18-686 savedefault Reboot, and that should do it. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provide

[asterisk-users] What syntax to send user:pass in SIP Dial string?

to do is embed the username:password in the Dial string, something like this: exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]:[EMAIL PROTECTED]|30|) doesn't work though, can't create sip channel. I'm not sure if this can be done? Any guidance will be appreciated. JR -- - JR R

Re: [asterisk-users] fax / t38 gateway

fax over HTTPS. I will be testing the FaxBack products to see how they stack up. JR - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSU

Re: [asterisk-users] How to add contexts in asterisk realtime?

y there is no impact to a running system. JR -- ----- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] Asterisk T38 and Dialogic DMG 2000

ll setup fine with audio. Thanks. JR -- ----- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Regis

Re: [asterisk-users] Need application, CID number match list to call cell phone

> Is this a one VIP to one cell number match? Or is it on VIP to multiple > cells? > > On Tue, Aug 26, 2008 at 7:28 PM, JR Richardson <[EMAIL PROTECTED]> > wrote: > > Hi All, > > > > I received a request for a special application and need some guidance. &g

[asterisk-users] Need application, CID number match list to call cell phone

and deleted, through a web page on the PBX. So I'm thinking I need a dialplan app that has to interface with a MySQL database that holds the list of numbers, so I can build a webpage to add/delete the numbers. Any ideas would be much appreciated. Thanks. JR ----- JR Richa

[asterisk-users] Asterisk 1.4 T38 UDPTL Pass Through MAX TNT and Linksys 2102

Hi All, I finally got the time to test t38 pass through with a TNT, * 1.4.21.1 and Linksys 2102: PRI><2102> ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: h

[asterisk-users] Intermittent T.38 pass through

Hi All, I've been testing reliability with t.38 faxing pass through with * 1.4.21.1, Linksys ATA's 2102 x 2, Sharp UX-B800SE and Cannon ImageClass D880. cannon> <2102 #1> <*> <2102 #2> ___ -- Bandwidth and Colocation Provided by http://www.api-digital

Re: [asterisk-users] FAX t.38 on Asterisk 1.6?

> Asterisk 1.6 currently has T.38 origination and termination support. > It does not yet have fax gateway support. > > -- > Russell Bryant Russell, Can you please clarify what you mean. I think there is still a bit of confusion as to what termination and gateway and Asterisk 1.6 is all about, ca

Re: [asterisk-users] multiple asterisk approach

t; Use DUNDi, perfect for this. The protocol is very light, no load on the servers to run it, can handle hundreds of queries a second with no load. You want to use regcontext and a few other things to make it all work together. Here are some papers to guide you: ftp://208.81.55.228/DUNDi_So_Easy.pd

[asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation

100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Thanks. JR ----- JR R

Re: [asterisk-users] 911 via MAX TNT

> When I send a call out the MAX I get the following > > -- Got SIP response 484 "Address Incomplete" back from 172.16.10.230 > > Any ideas on how to make 911 appear as a ten digit number to the device so > that it will pass the number out to the PSTN ? This is not a max tnt problem, the

Re: [asterisk-users] asterisk-addons 1.6.0 "Command 'realtime mysql status'

My test is connecting fine to local and remote databases, I'm use Asterisk 1.6-current and addon-1.6-current from digium ftp, not trunk. Hope this helps. JR -- JR Richardson Engineering for the Masses ___ -- 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

Re: [asterisk-users] addons-1.6 not seeing installed MySQL packages

ge and the addon package did find mysql installed. So I'm guessing debian etch is putting mysql_client in some other place that /usr/sbin/. What I did notice is the addon sample config file for res_mysql.conf doesn't specify how to setup the read/write entries, clarification on that would hel

[asterisk-users] addons-1.6 not seeing installed MySQL packages

Hi All, I'm poking around with 1.6, tried to compile the addon package, but it doesn't see mysql_config installed. I have mysql-client, mysql-common and mysql-server installed. I'm running debian etch. Any suggestions? Thanks. JR -- JR Richardson Engineering

Re: [asterisk-users] T.38 w/ MAX TNT & ASTERISK

it. If you do get it going, I would really appriciate knowing how. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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

Re: [asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA)

= 3db-loss set line-interface voip-gain-control output-pad = 3db-loss Hope this helps. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE o

Re: [asterisk-users] T38 Passthrough Verification

> JR Richardson wrote: > > I have 1.4.9.1 setup, with the compiler flags enabled for T38, and > > have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes > > between devices but can't seem to invoke T38 pt UDPTL. It's enabled > > in sip.conf [gener

[asterisk-users] T38 Passthrough Verification

RNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet ! sip show channels shows the call setup with ulaw. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --

[asterisk-users] Anyone have a method of keeping an incremental tally of calls?

that gets called, so after a week or month, I can see how many times a specific dilaplan action has been used. Thanks for any advice. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW

ndom reboots are not uncommon on the 601 with sidecars if you're > running it on PoE. That makes sense but in my case the 601 w/3 sidecars did not reboot at all and it is run from POE. The 650 just seems to perform much better. JR --- JR Richardson Engineering for the Masses _

Re: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW

JR Richardson Engineering for the Masses> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of asterisk-users- > [EMAIL PROTECTED] > Sent: Saturday, March 01, 2008 12:00 PM > To: asterisk-users@lists.digium.com > Su

[asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue

processor, will this eliminate the issue? Has anyone experienced this or have ideas for resolution or further troubleshooting? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- as

Re: [asterisk-users] DUNDi with two servers

JR Richardson > Hi, > > I'm having difficulties with using DUNDi between two servers. If it were > three I think I could control looping by limiting TTL, but with two I'm > not > sure how to prevent a loop causing bad things to happen. I've tried ttl=1 > but

Re: [asterisk-users] Include in asterisk realtime

> I am trying asterisk realtime with mysql database. But i don't know how to > put the "include" entry. > Have you some ideas? You have to put the include statements in the static extensions.conf file in the proper [context]. You can't use include=context in the da

Re: [asterisk-users] Asterisk and MRTG, a little help please...WORKING

On 1/28/08, JR Richardson <[EMAIL PROTECTED]> wrote: > > You need to take a step back and first test the script without using > > MRTG. Execute it like this: > > # /opt/bin/asterisk-mrtg -h localhost -u XXX -p -1 SIP -2 Zap > > 10 > > 10 > > 1

Re: [asterisk-users] Asterisk and MRTG, a little help please...

d = yes port = 5038 bindaddr = 0.0.0.0 [user] secret = pass deny=0.0.0.0/0.0.0.0 permit=[subnet of mrtg server] read = system,call,log,verbose,command,agent,user Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Prov

Re: [asterisk-users] Asterisk and MRTG, a little help please...

-01-28 11:16:01: ERROR: Target[asterisklab2][_OUT_] ' $target->[0]{$mode} ' did not eval into defined data ntcp-mrtg:/var/www/mrtg# I can see the script log into the manager interface on the asterisk server at 10.10.14.102, and there are active SIP c

Re: [asterisk-users] Asterisk and MRTG, a little help please...

l the script from the command line: # env LANG=C /usr/bin/mrtg 10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2 JR -- JR Richardson Engineering for the Masses ___ -- 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] Asterisk and MRTG, a little help please...

p? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Asterisk High Availability and Clustering

Hi All, There is a new list available for collaboration in this subject. http://lists.digium.com/mailman/listinfo/asterisk-ha-clustering JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api

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