[Asterisk-Users] MusicOnHold Native Mode, Please Clarify

Hi Guys, Ive attempted to get this moh-native thing to work with no success. Ive reviewed wiki, mantis and e-mail postings and Im confused. The latest Ive read is native moh should be in asterisk-addons in format_mp3, but what version will it work with? Ive tried asterisk 1.0.1,

[Asterisk-Users] re: MusicOnHold Native Mode, Please Clarify

Hey guys, I got moh-native working with todays CVS of asterisk and asterisk-addons so Im guessing there were some code problems with versions 1.0.1, 1.0.4 and current CVS stable. Following the wiki instructions worked fine. Also the mp3s that come with Asterisk sound perfect, whereas my

[Asterisk-Users] Asterisk on KNOPPIX, I have it working, somewhat.

Asterisk Brethren, This has been a fantasy morphed into reality; albeit not quite how I had it planned. Learning how to re-master KNOPPIX into my own customization WAS a challenge but very educational. Once I learned the process of re-mastering, installing Asterisk was a whole new

[Asterisk-Users] ParkAndAnnounce Problem

Hi Guys, I've been trying to get this application to work but I'm getting inconsistent results. My music on hold works fine by itself, I can park calls all day, works every time. I can dial the Console/dsp all day by itself and it works every time, sounds great, even play announcement gsm

[Asterisk-Users] Linux basics and Asterisk basics

On Fri, 2004-12-10 at 16:29, Jim Guy wrote: Hello, I am just starting to research Asterisk and I would like to install it on a PC to try out. I have looked around quite a bit but I haven't found much information on the Linux part. I know you need to put Linux on the PC first but what

[Asterisk-Users] Very Cool.........Asterisk Made Wired Magazine

Hi Guys, The article They've Got Your number in the Dec 2004 issue of WIRED magazine mentions Asterisk PBX (on p.100). The article is about phone phreaks hijacking cell phones with Bluetooth technology along with spoofing CID to pull some clandestine hacks on the PSTN. Anyhow, Asterisk is

[Asterisk-Users] Carrier Access CMG/FXS MGCP to Asterisk, Works Fine

FYI, Follow the Quick Start Guide from Carrier Access to setup the CMG (Customer Media Gateway) Router card. Follow the Asterisk mgcp.conf wiki page setup. The only issue I had was with the CAC CMG card, it defaults to strict policy message exchange and dial-tone will not come across

[Asterisk-Users] Carrier Access CMG/FXS MGCP to Asterisk, Works Fine

FYI, Follow the Quick Start Guide from Carrier Access to setup the CMG (Customer Media Gateway) Router card. Follow the Asterisk mgcp.conf wiki page setup. The only issue I had was with the CAC CMG card, it defaults to strict policy message exchange and dial-tone will not come across

[Asterisk-Users] RE: Problems with ADIT 600 - latency, loss, etc

Ralph, My experience with the Adit (going on 3 years), it's a solid platform. I've also had one backed into * using MGCP to CMG card for a bit with great success. The thing you said about the bandwidth usage and call quality sucks for both 300KB and 1.5Meg is the clue to this problem being a

[Asterisk-Users] test e-mail, please disregard

Test. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

* Brethren, It's a sad day in our community. Please join me in a moment of silence for the death of responsible journalism. Silence.good enough. This article goes on to tell about Pingtel's announcement of forming the first open source community aimed at creating SIP based

[Asterisk-Users] FW: Network Magazine 04/04/04 Article pg 19 (Free IP Telephony PBXs?)

The [EMAIL PROTECTED] appears to be broken. I dug around the magazine contacts and found Doug Allen, senior editor, you can send comments to [EMAIL PROTECTED] . I didn't get a bounce back from that e-mail so I assume it made it to the editor. JR

[Asterisk-Users] Embedded Asterisk System

I have had an embedded * server for a while, a one-off project I've been working on in some spare time. I want to write a white paper about it but haven't started yet due to other priorities. A CF used in an IDE adapter is the way to go. The development environment is a bit tricky but here is a

[Asterisk-Users] Embedded Asterisk Paper Complete

Hi all, The journey is complete, at least for this project. http://lists.digium.com/pipermail/asterisk-users/2004-October/067289.html I spent the better part of Halloween putting this together, I hope its useful, enjoy. My ftp server is on the fritz so feel free to post on any

[asterisk-users] Voicemail .lock- files voicemail box not accessible

someone point me in the right direction to resolve this? I'm runnning 1.2.9 on Debian Sarge. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list

[asterisk-users] perl script to generate new sip.conf users

Hi All, I remember some folks had put together a web page or perl script to generate many sip.conf entries from a file defining the users, vmbox, secret, CID and other variables. Can someone please point me in the right direction. Thanks. JR -- JR Richardson Engineering for the Masses

[asterisk-users] Time Limit on Call or Conference Room?

calls within the dial plan, per call. I have another customer who wants to offer free calls, for 5-10 minutes with auto disconnect. Can anyone point me int he right direction? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth

Re: [asterisk-users] Time Limit on Call or Conference Room? NEW ASTERISK PROVERB

On Fri, 3 Aug 2007, JR Richardson wrote: Can anyone point me int he right direction? At the risk of coming off in a gratuitiously self-aggrandising manner quoting myself: http://lists.digium.com/pipermail/asterisk-users/2007-May/188438.html -- Alex Balashov Thank you, Alex

[asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback

to follow for new users. Your feedback is appreciated. 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

[asterisk-users] DUNDi, So Easy A Caveman Could Do It!

Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- JR Richardson Engineering for the Masses

Re: [asterisk-users] Overhead paging over IP

then bam, your passing audio to the paging system. You can pick these up for ~$200. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Build your own appliance concept

tips in here ofr trimming down debian and having a re-producible build environment. Good luck. JR -- JR Richardson Engineering for the Masses ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation

[asterisk-users] SIPAddHeader cmd from Realtime MySQL, not getting all the 'appdata' field

) I'm wondering if the colons or the back slash is affecting this coming into asterisk? Thanks. JR -- JR Richardson Engineering for the Masses ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth

[asterisk-users] Astricon Ride From Airport to Conf Hotel

. If anyone is coming in around the same time and needs a ride, contact me off-list. Thanks. JR -- JR Richardson Engineering for the Masses ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided

Re: [asterisk-users] Anyone use the Linksys phones?

time, up and running, which is really great. Lots of good features, solid mid to low end cost business phone, customers seem to like it and not many support calls once the users get used to using it. JR -- JR Richardson Engineering for the Masses

Re: [asterisk-users] DUNDi, regcontext, softphones.. fail.

I'm having an issue deploying softphones into my DUNDi/regcontext setup. My current design is that all SIP users get registered into a sipregistration context in the sip.conf. I then have a dialplan function that includes that and does the dial: include = sipregistration exten =

[asterisk-users] Ultrastmonkey? Ultramonkeyast? Astrimonkey? High Availability and Asterisk

would like to learn or implement this type of solutions. I'm happy to facilitate and document these solutions and share my successes and failures. Please contact me if you would like to participate or just be in the loop. Thanks. JR -- JR Richardson Engineering for the Masses

Re: [asterisk-users] Asterisk Realtime woes

Good luck. 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] Asterisk under VMWare; Great Topic

more light on why this is and is anyone trying to improve this? 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] Backport Func_ODBC question

Hi All, Ingnorant question, how do you apply the backport func_odbc to 1.2 branch? 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] Realtime on Asterisk 1.2.24

and is working fine for me. Please clarify if you have a moment. 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

Re: [asterisk-users] Realtime context

and you can catch up on the progress with the patch, try it, it may work for you. 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

Re: [asterisk-users] DUNDI setup help

that as well. 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] * crash when forward voicemail message

Hi all, When I attempt to forward a voicemail message to another voice-mailbox (option 8), asterisk crashes. I can restart is immediately and all seems fine. If I attempt to forward the voicemail message to a non-existing mailbox, I get a message saying that mailbox does not exist. I've been

[Asterisk-Users] * crash when forward voicemail message [problem solved]

plan on using the older kernel for my implementations for now. Hope this helps. JR -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 1:57 PM To: JR Richardson Subject: RE: Re: [Asterisk-Users] * crash when forward voicemail message I don't

RE: [Asterisk-Users] New to asterisk? RUN... don't walk.

Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness.

[Asterisk-Users] * crash when forward voicemail --Nicolas Gudino

Hey Nicolas, That did it. I ran that export command you suggested, then launched *, everything worked fine. I'm still looking for info on what that command actually does. Can you shed some light please? Thanks. JR -Original Message- From: JR Richardson [mailto:[EMAIL PROTECTED

RE: [Asterisk-Users] Static Noise coming from Wildcard FXS: Wildcard TDM400P

I had the same problem, I reseated the daughter card on the board and that helped but just for a short time. Eventually the port wouldn't even break dial-tone. All other ports 2, 34 on the card were fine, it was only port 1 giving a problem. I swapped daughter boards around between ports but the

[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] 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

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

[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

[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

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

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] 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] 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

[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] 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

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,

[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

[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] 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

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] 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] 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

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

[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

[asterisk-users] Re: Dial outbound trunk numbers in a round-robin sequence?

Hi All, Customer is requesting 1 incoming toll free #, that dial out to 4 different terminating numbers, not ring all at once but ring #1, then #2, then #3, then #4, then back to #1 consecutively on inbound calls, regardless if someone is on #1. So this is not like a hunt group,

[asterisk-users] SER/OpenSER, I Finally Get It.............General Observation

be a true Astriholic; and second, I can't seem to break OpenSER and if you can't break-em, join-em. Can I use OpenSER as a voicemail server, blah, blah, blah??? JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided

[asterisk-users] agi timeout

Hi All, Is there a way to specify a time-out option when you call an AGI command from the dialplan? If my AGI fails or doesn't get a response, the call drops, not good. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth

[asterisk-users] Re: agi timeout......clarification

n 4/24/07, JR Richardson [EMAIL PROTECTED] wrote: Hi All, Is there a way to specify a time-out option when you call an AGI command from the dialplan? If my AGI fails or doesn't get a response, the call drops, not good. I'm running asterisk 1.2 and calling a fast agi script exten = s,1,Agi

[asterisk-users] Re: Voicemail on Different Server, Voicemail with NFS

-Original Message- From: JR Richardson [mailto:[EMAIL PROTECTED] Sent: Saturday, June 17, 2006 2:30 PM To: asterisk-users@lists.digium.com; Douglas Garstang Subject: Voicemail with NFS (working, I think) I'm using a stand-alone VM server and exporting the VM files ro for MWI

[asterisk-users] Re: Voicemail on Different Server, Voicemail with NFS

for remote vocemail/mwi. Using NFS works great in a Cluster arrangement, all servers on the same subnet, location. I'll add to the wiki as well. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com

[asterisk-users] Re: Voicemail on Different Server (MySQL Replication split thread)

replication or data corruption when replication is not implemented properly or setup in an environment not particularly suited well for replication. Just wanted to add my own experience. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth

[asterisk-users] RE: Voicemail on Different Server (MySQL Replication split thread)

Having master and slave servers in the same switch fabric is the only situation in which I would consider replication. The cases that I described were with machines in separate subnets. Replication simply doesn't work that well when there is significant latency. Did they mention that in

[asterisk-users] Re: is dundi worth pursuing in this situation?

an ever changing environment and truly need dynamic extension DID routing between remote locations/servers. Hope this helps. Sorry if it's clear as mud. Good Luck JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided

[asterisk-users] Re: headsets for linksys/sipura phones?

. These work fine with the SPA-942's http://brandcell.stores.yahoo.net/planm1headha.html Plantronics M175, we get them for $23, but this site is cheaper. Regards, JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided

[asterisk-users] RE: Mr. Spencer Written

Mr. Spencer written the article Using DUNDi with a Cluster of Asterisk Servers http://www.voip-magazine.com/content/view/3644/0/1/0/ in the VoIP Magazine and the piece follow: [lookupdundi] exten = _X,1,Goto(${ARG1},1) switch = DUNDi/priv exten = i,1,Goto(lookupmysql,${INVALID_EXTEN},1)

[asterisk-users] Re: FastAGI hangs up channel if server is not available

arquments $number = $ARGV[0]; $AGI-exec(agi,agi://192.168.1.175/calldirect?checknumber=$number); end of file--- Also you need the asterisk perl agi modules at http://asterisk.gnuinter.net/ Good luck. JR -- JR

[asterisk-users] Re: DUNDi configuration problem

-priv-canonical,0,SIP,${NUMBER}@the real IP Address,nopartial The rest looked ok I think. Good luck. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] RE: DUNDi configuration problem

Thank you for the quick response. Do I need to create a route to the other machine? like a trunk? On the SIP side of things, yes, you can create a SIP trunk for each server-to-server relationship, or you can just send the sip call to the default context and use a goto statement to get the call

[asterisk-users] RE: Ser vs. DUNDi

With all of the recent talk on the list about DUNDi, I have a question. From the outset it appears that SER is often used for high availability solutions and as a tool for almost clustering Asterisk boxes behind it. It appears to me that DUNDi is providing a lot of this as well. Now I know

[asterisk-users] Polycom or Linksys phones bootp tftp config setup

for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] Re: Polycom or Linksys phones bootp tftp config setup

if TFTP and the Polycom came right up and acted as expected. I'm still poking around with the Linksys. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

[asterisk-users] CDR not recording accountcode on SIP Response 302 Call Forward From Phone

be billed to. I have a feeling this is normal behavior for Asterisk as no real channel gets invoked with an accountcode parameter, but there has got to be something that accounts for this situation. Does anyone have a work around or remedy? I'm running 1.2.9. Thanks. JR -- JR Richardson

[asterisk-users] RE: Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

servers will usually dominate singe servers in relation to cost. All, Nice discussion, and thanks for posting your benchmark results and feedback. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com

[asterisk-users] RE: Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

try adding the command ulimit -n 8192 to the script that starts Asterisk. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided

[asterisk-users] custom cdr fields and cdr_mysql, howto?

1.2.3. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: custom cdr fields and cdr_mysql, howto?

On 6/7/07, JR Richardson [EMAIL PROTECTED] wrote: Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten = s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten = s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specified

[asterisk-users] Re: Write to multiple databases as redundancy scheme

to setup. Asterisk writes to the Master database and the Master replicates changes to slave databases for backup. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

[asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working

dtmf settings in the 2600 and have tried all the dtmf settings in Asterisk. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk

Re: [asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working, Working now

On 6/26/07, JR Richardson [EMAIL PROTECTED] wrote: Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600

[asterisk-users] FW: Realtime Voicemail Password Change Not Working

I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been

[asterisk-users] Re: Realtime Voicemail Password Change Not Working

On 1/17/07, JR Richardson [EMAIL PROTECTED] wrote: I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new

[asterisk-users] RE: Realtime Voicemail Password Change Not Working

Interesting, well if you're seeing the other selects in the mysql.log then this update not showing up is bizarre. It would also mean that permissions are irrelevant if doesn't even attempt to change the password, as you'd rightly pointed out as well. I just tested it again and this is what I

[asterisk-users] Re: Realtime Voicemail Password Change WORKING NOW

across table names, I did change the voicemail table column 'uniqueid' to 'id', like the sip, iax and exten tables. I changed it back to 'uniqueid' and password update is working fine now. Thanks to all who replied for your help. -- JR Richardson Engineering for the Masses

[asterisk-users] iax2 prun realtime peer only can't prune user

Hi All, I'm running 1.2.9.1. I can prune sip realtime peers and users and iax realtime peers but no command to prune iax realtime users. Was this implemented in a later version? Thanks. JR -- JR Richardson Engineering for the Masses

[asterisk-users] iax.conf setvar= like sip.conf setvar=?

Hi All, I'm running 1.2.9.1, is setvar= implemented in iax.conf in a later version of asterisk? If so, which one? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

[asterisk-users] max tnt pri voice channels 56k or 64k, does it matter, selection parameter?

to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-users] requesting real world meetme capacity numbers

load, perfect audio. I'm working on a conf bridge for 150+ users, could use some advice, if anyone has accomplished such a feat or has any ideas on how. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided

[asterisk-users] Asterisk to Cisco's Rescue...again...Authenticate LD Calls

for the Cisco CM and the Cisco Gateway to play nice together. The real hero here is Asterisk, Digium, and the Community that supports it! Thank you All JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation

[asterisk-users] RE: Asterisk to Cisco's Rescue...again...AuthenticateLD Calls

From: Jason Aarons \(US\) [EMAIL PROTECTED] Glad to hear you had a workaround. I would suggest re-queing your TAC case, perhaps you got a outsourced or less experienced engineer at Cisco. Their support has varied depending on which city/group you get. Some have more experience then

[asterisk-users] Re: Authentication Command

example, * version, hardware, more info about your setup. Make sure you answer the call first, before you invoke the authenticate cmd. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

[asterisk-users] Re: Polycom Firmware

, had to shelf the phone and go with a 601 with 1.6.6. That's the only thing I'm aware of is presence seemed to break with the latest firmware. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com

[asterisk-users] RE: Polycom reject button

I have users in my dialplan that go from SIP to Cell When they are at their desk and they hit reject call, it goes to the next thing in the dialplan, thus transferring to their cell. Not what they want. Is it possible to change the reject button to make it go to voice mail or a new ext? I

[asterisk-users] Asterisk SIP to MAX TNT Gateway, Sporadic Echo

really having trouble isolating anything. I'm wondering if this could be a bad DSP on the TNT, and how would I isolate. We have 600+ DSP's in this chassis. Any experience or ideas with this type of issue would greatly be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

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