Re: [asterisk-users] PRI timing settings
On 2014-08-19 23:56, Jeff LaCoursiere wrote: Hello, I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as "busy". The telco has been working with their switch manufacturer and took the output of "pri show span 1" from me and came back with this: quote--- Please check your timers below. How did you determine your settings? * Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T201: 1000 T202: 2000 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000Our switch: Telcordia National ISDN 2: Range 10 - 90 seconds, default 90 seconds. * T312: 6000 T313: 4000 T316: -1Our switch: Telcordia National ISDN 2: Range 10 - 120 seconds, default 30 seconds. N316: 2 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 ---end quote--- Now I have no idea what T309 or T316 represent, but it seems odd that timers and counters would cause such an odd result... and the failure is immediate, not after some amount of timeout. Can anyone shed light on these settings and tell me if they are configurable? I don't think that this has something to do with the timers. I recommend to enable debug of your pri connection to get a trace to a sprint phone number. There you should get the disconnect cause of the call. Then your provider should be able to help you out. Maybe the reason of the problem is an incorrect number setting. Regards Hans -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI timing settings
Doubtful that T309 or T316 are causing the problem, but you can always change them to correspond with their defaults. http://www.nmscommunications.com/manuals/6272-16/appendxe.htm --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, August 19, 2014 4:56 PM To: asterisk-users@lists.digium.com >> Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI timing settings Hello, I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as "busy". The telco has been working with their switch manufacturer and took the output of "pri show span 1" from me and came back with this: quote--- Please check your timers below. How did you determine your settings? * Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T201: 1000 T202: 2000 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000Our switch: Telcordia National ISDN 2: Range 10 - 90 seconds, default 90 seconds. * T312: 6000 T313: 4000 T316: -1Our switch: Telcordia National ISDN 2: Range 10 - 120 seconds, default 30 seconds. N316: 2 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 ---end quote--- Now I have no idea what T309 or T316 represent, but it seems odd that timers and counters would cause such an odd result... and the failure is immediate, not after some amount of timeout. Can anyone shed light on these settings and tell me if they are configurable? Cheers, j -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI timing settings
Hello, I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as "busy". The telco has been working with their switch manufacturer and took the output of "pri show span 1" from me and came back with this: quote--- Please check your timers below. How did you determine your settings? * Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T201: 1000 T202: 2000 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000Our switch: Telcordia National ISDN 2: Range 10 - 90 seconds, default 90 seconds. * T312: 6000 T313: 4000 T316: -1Our switch: Telcordia National ISDN 2: Range 10 - 120 seconds, default 30 seconds. N316: 2 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 ---end quote--- Now I have no idea what T309 or T316 represent, but it seems odd that timers and counters would cause such an odd result... and the failure is immediate, not after some amount of timeout. Can anyone shed light on these settings and tell me if they are configurable? Cheers, j -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error opening file for reading: Permission denied
Grepping the output of the strace revealed this:
stat("/root/.terminfo", 0x7fff8622ed50) = -1 EACCES (Permission denied)
open("/root/.asterisk_history", O_RDONLY) = -1 EACCES (Permission denied)
open("/root/.odbcinst.ini", O_RDONLY) = -1 EACCES (Permission denied)
[this one many times]
That must be because I'm starting asterisk as "root". When I su to
asterisk first, then start it, those above disappear. Problem solved!
Thanks Steve!
Mitch
On 08/19/2014 03:39 PM, Steve Edwards wrote:
On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn
No, that's not it. The wording is different.
Can you run Asterisk via strace? Something like:
sudo -u asterisk strace /usr/sbin/asterisk -c -p -U asterisk
--
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Re: [asterisk-users] Error opening file for reading: Permission denied
On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn No, that's not it. The wording is different. Can you run Asterisk via strace? Something like: sudo -u asterisk strace /usr/sbin/asterisk -c -p -U asterisk -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 12.5.0 Now Available
On Tue, Aug 19, 2014 at 3:38 PM, Asterisk Development Team wrote: > The Asterisk Development Team has announced the release of Asterisk 12.5.0. > This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk > > The release of Asterisk 12.5.0 resolves several issues reported by the > community and would have not been possible without your participation. > Thank you! Congratulations, team. -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HangupRequest Infinite Loop
Asterisk 11.8.1 Today our Asterisk server locked up. Looking through the asterisk full log, I see the following over and over again and the log just continues to grow until we reboot Asterisk. Has anyone seen anything like this before or have any idea what may cause it? [Aug 19 14:41:26] DEBUG[7362] manager.c: Examining event: Event: HangupRequest Privilege: call,all Channel: SIP/esivrproxy1-44ba Uniqueid: 1408472633.17594 Cause: 111 [Aug 19 14:41:26] DEBUG[25630] manager.c: Examining event: Event: HangupRequest Privilege: call,all Channel: SIP/esivrproxy1-44ba Uniqueid: 1408472633.17594 Cause: 111 [Aug 19 14:41:26] DEBUG[10337] manager.c: Examining event: Event: HangupRequest Privilege: call,all Channel: SIP/esivrproxy1-44ba Uniqueid: 1408472633.17594 Cause: 111 -- Deric Page ArcGIS, PhoneMaster, CallCapture and TapiToIvue Programmer E&O x2335 Building 3, 3rd Floor, Section C * "Everything starts as someone's daydream." -- Larry Niven -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error opening file for reading: Permission denied
I tried grep too. No 3rd party modules - this is an out-of-the box download and build. I'm guessing that some library function is being called to read a file and the error is happening there? Mitch On 08/19/2014 02:33 PM, Matthew Jordan wrote: On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn wrote: No, that's not it. The wording is different. grep doesn't turn up your phrase: ~/projects/12$ grep --include=*.c --include=*.h -r "Error opening file" . ~/projects/12$ Are you using any 3rd party modules that aren't delivered with Asterisk? -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 12.5.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 12.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Improvements made in this release: --- * ASTERISK-24036 - ARI: Recording resource should allow copying a recording (Reported by Samuel Galarneau) * ASTERISK-24037 - ARI: RecordingFinished event should return duration of recording (Reported by Samuel Galarneau) * ASTERISK-21178 - Improve documentation for manager command Getvar, Setvar (Reported by Rusty Newton) * ASTERISK-23692 - ARI: Add a Messaging Capability (Reported by Matt Jordan) Bugs fixed in this release: --- * ASTERISK-23852 - ARI mixing bridges should propagate linkedids. (Reported by Richard Mudgett) * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an empty string is a bit over zealous (Reported by Matt Jordan) * ASTERISK-23985 - PresenceState Action response does not contain ActionID; duplicates Message Header (Reported by Matt Jordan) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23987 - BridgeWait: channel entering into holding bridge that is being destroyed fails to successfully join the newly created holding bridge (Reported by Matt Jordan) * ASTERISK-23969 - SendMessage AMI action Cant Send Text Message Over PJSIP (Reported by Andrew Nagy) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-23847 - Alembic voicemail script - 'recording' column should be longblob on MySQL (Reported by Stephen More) * ASTERISK-23825 - Alembic scripts - table queue_members missing unique index on column uniqueid (Reported by Stephen More) * ASTERISK-23909 - Alembic scripts - table sippeers could use a longer useragent column (Reported by Stephen More) * ASTERISK-23941 - ARI: Attended transfers of channels into Stasis application lose information (Reported by Matt Jordan) * ASTERISK-18345 - [patch] sips connection dropped by asterisk with a large INVITE (Reported by Stephane Chazelas) * ASTERISK-23508 - Memory Corruption in __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) New Features made in this release: --- * ASTERISK-24000 - chan_pjsip: Add accountcode setting (Reported by Matt Jordan) * ASTERISK-24119 - HEP: Add module that exports RTCP information to a Homer Capture Server (Reported by Matt Jordan) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.5.0 Thank you for your continued support of Asterisk! -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.12.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.12.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.12.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: --- * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an empty string is a bit over zealous (Reported by Matt Jordan) * ASTERISK-23985 - PresenceState Action response does not contain ActionID; duplicates Message Header (Reported by Matt Jordan) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection dropped by asterisk with a large INVITE (Reported by Stephane Chazelas) * ASTERISK-23508 - Memory Corruption in __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) Improvements made in this release: --- * ASTERISK-21178 - Improve documentation for manager command Getvar, Setvar (Reported by Rusty Newton) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0 Thank you for your continued support of Asterisk! -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.30.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.30.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.30.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs fixed in this release: --- * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an empty string is a bit over zealous (Reported by Matt Jordan) * ASTERISK-23814 - No call started after peer dialed (Reported by Igor Goncharovsky) * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy (Reported by Corey Farrell) * ASTERISK-23818 - PBX_Lua: after asterisk startup module is loaded, but dialplan not available (Reported by Dennis Guse) * ASTERISK-18345 - [patch] sips connection dropped by asterisk with a large INVITE (Reported by Stephane Chazelas) * ASTERISK-23508 - Memory Corruption in __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) Improvements made in this release: --- * ASTERISK-21178 - Improve documentation for manager command Getvar, Setvar (Reported by Rusty Newton) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.30.0 Thank you for your continued support of Asterisk! -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error opening file for reading: Permission denied
On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn wrote: > No, that's not it. The wording is different. > grep doesn't turn up your phrase: ~/projects/12$ grep --include=*.c --include=*.h -r "Error opening file" . ~/projects/12$ Are you using any 3rd party modules that aren't delivered with Asterisk? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Way to dump PRI settings?
Perfect! Exactly what I was looking for. Cheers, j On 08/19/2014 12:23 PM, Eric Wieling wrote: I doubt PBX settings (other than CallerID) would break calling to only one specific carrier. Have you tried pri show span X? From one of our boxes: pbx*CLI> pri show span 1 Primary D-channel: 24 Status: Up, Active Switchtype: National ISDN Type: CPE Remote type: Unknown node type Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T201: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T312: 6000 T313: 4000 T316: -1 N316: 2 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Q931 RX: 6350 Q931 TX: 5828 Q921 RX: 42786 Q921 TX: 42716 Q921 Outstanding: 0 (TEI=0) Total active-calls:2 global:1 CC records: Overlap Recv: No -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, August 19, 2014 5:00 PM To: asterisk-users@lists.digium.com >> Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Way to dump PRI settings? Hello, I am having odd issues with a new PRI based installation. Outbound calls work for all numbers except those that terminate at Sprint! The telco is new to PRI (this is in the Caribbean) and say that Sprint is rejecting the calls, and asked for our PRI settings so they can work with their switch manufacturer. We are using the bare minimum default settings, but I can imagine that a whole ton of configurable settings exist that I have never played with. Is there a way to dump all of the configurable settings for a PRI? The card is a Digium TE122. Thanks, j -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Way to dump PRI settings?
I doubt PBX settings (other than CallerID) would break calling to only one specific carrier. Have you tried pri show span X? >From one of our boxes: pbx*CLI> pri show span 1 Primary D-channel: 24 Status: Up, Active Switchtype: National ISDN Type: CPE Remote type: Unknown node type Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T201: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T312: 6000 T313: 4000 T316: -1 N316: 2 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Q931 RX: 6350 Q931 TX: 5828 Q921 RX: 42786 Q921 TX: 42716 Q921 Outstanding: 0 (TEI=0) Total active-calls:2 global:1 CC records: Overlap Recv: No -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, August 19, 2014 5:00 PM To: asterisk-users@lists.digium.com >> Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Way to dump PRI settings? Hello, I am having odd issues with a new PRI based installation. Outbound calls work for all numbers except those that terminate at Sprint! The telco is new to PRI (this is in the Caribbean) and say that Sprint is rejecting the calls, and asked for our PRI settings so they can work with their switch manufacturer. We are using the bare minimum default settings, but I can imagine that a whole ton of configurable settings exist that I have never played with. Is there a way to dump all of the configurable settings for a PRI? The card is a Digium TE122. Thanks, j -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Way to dump PRI settings?
Hello, I am having odd issues with a new PRI based installation. Outbound calls work for all numbers except those that terminate at Sprint! The telco is new to PRI (this is in the Caribbean) and say that Sprint is rejecting the calls, and asked for our PRI settings so they can work with their switch manufacturer. We are using the bare minimum default settings, but I can imagine that a whole ton of configurable settings exist that I have never played with. Is there a way to dump all of the configurable settings for a PRI? The card is a Digium TE122. Thanks, j -- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error opening file for reading: Permission denied
No, that's not it. The wording is different.
Mitch
On 08/18/2014 02:28 PM, Paul Greenberg wrote:
Mitch,
Is it the below error?
if ((fd = open(filename, O_RDONLY)) < 0) {
ast_log(LOG_WARNING, "Cannot open file '%s' for reading:
%s\n", filename, strerror(errno));
return NULL;
}
Regards,
Paul
From: asterisk-users-boun...@lists.digium.com
on behalf of Mitch Claborn
Sent: Monday, August 18, 2014 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Error opening file for reading: Permission denied
Asterisk 12.4
I am seeing message "Error opening file for reading: Permission denied"
several times during the asterisk startup (asterisk -cv) but it
doesn't say which file. Is there a way to find out which file is having
trouble?
--
Mitch
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[asterisk-users] Alternative billing for A2Billing because of using Dial function with analogue lines
Hello All; After trying A2Billing and certainly when the trunk is analogue lines (FXO ports), I faced a problem that the channels were not hanged up properly from time to time which cause us to do restart for the dahdi. Without A2Billing, I was able to handle the Dial scenario properly and no hanging for the analogue channels and no need to restart dahdi from time to time. Really I would if there is alternative Billing software (open source) for A2Billing and its working mechanism differs than A2Billing (I always prefer to keep handle the Dial scenario from asterisk configuration file and not through the billing software to be sure that it is hanged up properly in all the cases), I would if there is a billing software that do only the billing part and send control the disconnection for the call by sending commands to asterisk while the dialing scenario is handled totally from the asterisk configuration. Or at least, is there billing software that work in better performance than A2Billing and does not cause for me problems if the trunk is analogue? Regards Bilal-- _ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
