[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

track or is there something else I should be looking at? Thanks. JR -- _ -- 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] Can we use MySQL native connector for ARA?

use the cdr_mysql as well. I wrote a couple of papers on asterisk_clustering_with_mysql_replication. They are a bit dated but still relevant. I'll send over if you like. Good luck. JR -- _ -- Bandwidth and Colocation Provided

[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

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 Speaking blindly as someone who has yet to fool with F2B, I'd rather ban somebody after 5-20 attempts than have the overhead needed to ban them quicker.  Guess

[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

than there servers so I don't really expect it to work properly if at all with Asterisk. 7.5 is the only firmware version that I deploy on a few hundred units and works fine. Good luck. JR -- _ -- Bandwidth and Colocation

[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

to T38 and I haven't had time to investigate too deeply. Fax handling in TOAS was greatly improved in 14.0 version, I would suggest you upgrade to that if you can and start there. JR -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Dual Atom mobo - call capacity

Ditto I'm running a Supermicro Atom based dual core server and it's rock solid!!! These make excellent servers for Asterisk installation IMHO. On Fri, Jun 11, 2010 at 05:42, --[ UxBoD ]-- ux...@splatnix.net wrote: - Original Message - On Thu, 10 Jun 2010, Michelle Dupuis

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

or guidance. Thanks. JR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

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

switching out the ATA. I have the latest firmware on each unit. Any ideas on what could cause this? The configuration is pretty simple so I don't think I'm missing anything there. I'm guessing there is a built in speed limit on the HT502? Thanks. JR

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

[asterisk-users] DAHDI disconnect supervision timing

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for disconnect supervision to be provisioned on

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

Re: [asterisk-users] DAHDI hangup detection

Ok on the workaround, how would I implement it? I'd like to give that a shot. On Tue, Sep 15, 2009 at 12:23, Danny Nicholas da...@debsinc.com wrote: The issue is that POTS as a technology does not have Answer/Hangup Supervision control (This is per the good folks at Digium). Your local Telco

[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

the function is a bit more versatile than the old app, but I don't really see it just yet. Can anyone shed some light on this and expound on the benefits or reasoning behind the switch in the application usage? Thanks. JR ___ -- Bandwidth

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] Problems with ztdummy

I'm having trouble with ztdummy and I can't seem to figure it out. I am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates applied and I have compiled Zaptel from source along with a new kernel from Debian sources to include 1khz timer support. The modules build fine, yet when I load

Re: [asterisk-users] Problems with ztdummy

this with the etchandahalf (2.6.24) kernel? That was going to be my next step if I couldn't get it resolved Thanks, Stephen On Thu, Dec 18, 2008 at 11:57, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Thu, Dec 18, 2008 at 10:49:03AM -0500, Stephen Brown Jr wrote: I'm having trouble with ztdummy

[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] Intergrating vicidial with trixbox

I noticed that the vicidial site has documentation available which probably covers the topics required. However, I also see that they want $50-$100 to download the docs. Seems harsh. Ron Byer Jr. NetWeave Integrated Solutions, Inc. +1.732.786.8830 x120 -Original Message- From

Re: [asterisk-users] fax / t38 gateway

that Audiocodes had with zero improvement. So I have to say, my confidence in T38 is very low, at least where open Internet connections are being used. I'm now going to look at some other technology, fax over HTTPS. I will be testing the FaxBack products to see how they stack up. JR

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

[asterisk-users] Redundant PSTN PRI Gateways using Asterisk

I currently have two T1 PRI lines feeding our company's legacy PBX. All our numbers are DIDs and can pass over both PRI. Currently, if one PRI line (or T1 interface card in the PBX) is down, communication continues to function as normal (with the exception of the reduction of channels available

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?

, capability, functionality, call flow to what application, library requirements, spandsp versioning. And when do you think we can expect to see stable solutions for each. Thanks. JR ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

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

will pass anything you send to it, 911/411/7 digit/10digit/011 international, the question is, does your PSTN provider accept 911 call on the trunk your passing the call to? JR ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

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] Mail Server

Mike Hammett wrote: I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't

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

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