[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

Re: [asterisk-users] DUNDI with minimal features

ent/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 communit

Re: [asterisk-users] Unicast RTP Paging

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

[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

[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

calls. So now I'm thinking it is a NAT issue, but only when using outbound proxy, doesn't make sense, now I'm really confused. Any feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

[asterisk-users] Voicemail Prepend Message Forwarding Not Working

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 for the Masses

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

and now prepending voicemail works. 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] Use Allworx Phones With Vanila Asterisk PBX?

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 http

[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 Masses

[asterisk-users] Asterisk SIP Refer Transfers

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 every Thurs

Re: [asterisk-users] Auto ban IP addresses

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 -- _ -- Bandwidth

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 suggestion but commenting out 309-312 refused to compile: cdr_csv.c

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.digium.com Subject: [asterisk

[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 Engineering

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

is 'rtcachefriends=yes', that should do it. 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

[asterisk-users] Asterisk Configuration GUI Question

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. Thanks. JR -- JR Richardson Engineering

[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

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 for years

[asterisk-users] Asterisk 1.8 Manager Perl Script Problem

response. I've searched through all the upgrade docs but nothing mentions command syntax changes. Any help is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http

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

. 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] asterisk and fail2ban

this or has resolved it, thank you. I have F2B set to ban after 1 attempt. The most I have seen in the logs is 4-5 attemps before ban is applied. I am calling scripts that apply the ban to a cisco access-list, so there is script/telnet/config delay but it is very minimal and works very well. JR -- JR

Re: [asterisk-users] asterisk and fail2ban

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 JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth

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

are going to run an automated blocking mechanism, you should get proficient with un-blocking as well for accidental blocking. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api

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

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 every Thurs: http

[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. JR -- JR

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

in 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 where it

[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 winsl...@pa.net 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

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

for the parties involved. Please contact me if you have time to work on this and are interested. I'm sure the Project Honeypot guys will be willing to pick this project back up and work on it. Thanks. JR -- JR Richardson Engineering for the Masses

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

this is not exactly a round robin distribution but works for what I need. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

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

),.,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 -- _ -- Bandwidth and Colocation Provided by http

[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

couldn't be of any help, but I feel your frustration. 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

[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

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

Faxes on one server? Could the Attrafax software handle that volume? Thanks in advanced for any feedback. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

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

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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

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

=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 Provided by http://www.api

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

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. JR -- JR Richardson Engineering for the Masses

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

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson jmr.richard...@gmail.com 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 like to use the function

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

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 Provided by http

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

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 http://www.api-digital.com -- asterisk-users mailing

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

variables according to supplied arguments $number = $ARGV[0]; $AGI-exec(agi,agi://agi.server.com/script.agi?user=usernamenumber=$number); *** Any assistance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

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

On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson jmr.richard...@gmail.com 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 would like a timeout of 1 second, then return

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 repeated 3 or more times prior

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

: 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 mailing list

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

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.com/asterisk-users@lists.digium.com/msg234015.html Good luck. JR -- JR Richardson

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

in this 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 Engineering for the Masses

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

,,) Something 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

[asterisk-users] Asterisk SIP to Cisco IAD2430 Series?

. 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 Colocation Provided

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

? 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

[asterisk-users] strange cisco nat issue

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 for the Masses

Re: [asterisk-users] DUNDi + SIP Realtime

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 what is it. :) Vin?cius Fontes www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP [JR Richardson] I

[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 Masses

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

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: http

[asterisk-users] Asterisk SIP trunk to Cisco IAD2400

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.digium.com/mailman/listinfo

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

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 UNSUBSCRIBE or update options visit

[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

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

so the Economic Downturn has affected them enough to reposition their margin strategies. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

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

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 for the Masses

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

Is 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 options

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

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 Provided by http://www.api-digital.com

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

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 Provided by http://www.api-digital.com -- asterisk-users mailing list

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

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 Richardson Engineering for the Masses

Re: [asterisk-users] fax / t38 gateway

- 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] How to add contexts in asterisk realtime?

. 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

[asterisk-users] Asterisk T38 and Dialogic DMG 2000

=image? If I disable udptl in Asterisk, call 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

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. Cust has there own Asterisk PBX with SIP phones

[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 Richardson

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

and I've adjusted them all with no change in the results. Pretty much the same results when testing t38 pass through to a Cisco pri gateway as well. So my question is: Does anyone else have this solution working and wouldn't not mind sharing configs? Thanks. JR -- - JR Richardson

[asterisk-users] Intermittent T.38 pass through

pass through from these ATA's, without the need to use the #99 in every dial string from the fax machine? Thanks. JR -- - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

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,

Re: [asterisk-users] multiple asterisk approach

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.pdf ftp://208.81.55.228/Using_DUNDi_with_a_Cluster_of_Asterisk_Servers.pdf Good Luck. JR -- - JR Richardson Engineering for the Masses

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

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 Richardson

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 tnt

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

on the local machine. # mysql -u **user** -p In /etc/mysql/my.cnf ensure: bind-address = 0.0.0.0 or bind-address = 127.0.0.1 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

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

TNT shouldn't fax work with T.38... Does anyone have any experience with this configuration ? Thanks, I have been wanting to do this for months, but just can't find the time to work on it. If you do get it going, I would really appriciate knowing how. Thanks. JR -- JR Richardson

[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 for the Masses

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

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 help also. JR -- JR Richardson Engineering for the Masses

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 [general] and well as the [peer]. I get an error

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

-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 or update options visit: http://lists.digium.com

[asterisk-users] T38 Passthrough Verification

]: 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 ___ -- Bandwidth

[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

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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

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 Subject: asterisk-users Digest

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

, 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 -- asterisk-users mailing

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 things still blow up. The DUNDi

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 database. JR -- JR Richardson

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

(use strict;) in CFG file (10.10.14.102.cfg) does not make sense I named the script file the IP address of the server.cfg instead of asterisk-mrtg. I call 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

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

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 channels during script execution. Any ideas? Thanks. JR -- JR Richardson Engineering for the Masses

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

=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 Provided by http://www.api-digital.com -- asterisk-users mailing list

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 10 10 You should get 4 lines of numbers

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

-- 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

Re: [asterisk-users] Realtime: Should I say or should I go (now) ?

will be 1001. To make this change, you need a central provisioning server, update the config file and reboot the phone to update the display name. Hope this helps and doesn't confuse things. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth

[asterisk-users] resync linksys SPA9XX config file from Asterisk

Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list

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