Re: [asterisk-users] How to deal with error messages passed as Early Media

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 > -- [image: Digium logo] Scott Griepentrog Digium, Inc ·

Re: [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11

ew 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/asteris

Re: [asterisk-users] Custom PHP for Call Files

s: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [image: Digium logo] Scott Grie

Re: [asterisk-users] same sip username with realms and chan_sip

> -- 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

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

Are you using this method of setting headers on PJSIP? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER On Thu, Aug 27, 2015 at 4:54 PM, Dan Cropp d...@amtelco.com wrote: Thanks Scott. I was able to get the basic concept to run. However, it seems PJSIP INVITE

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

-html-chunk/ACD_id289508.html Basically, read that book, and if you get stuck ask for help. On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp d...@amtelco.com wrote: Thanks Scott. I’m taking over for someone else’s code, so I must admit I’m still learning the Agent and Queue concepts. Local

Re: [asterisk-users] Windows Asterisk Help

recent release of Asterisk (version 13 at the moment).​ -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] Windows Asterisk Help

test the most recent release of Asterisk (version 13 at the moment).​ -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http

Re: [asterisk-users] asterisk segfault debian jessie asterisk 11.13

-- 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 -- [image: Digium logo] Scott Griepentrog

Re: [asterisk-users] Dell portability

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com

Re: [asterisk-users] chan_sip.c: Hanging up call

You mean sip set debug on ? ​Yes, that's correct for chan_sip. Sorry, I was vague -- there is now a different command for chan_pjsip​, didn't know which you were using. On Thu, May 28, 2015 at 12:49 PM, Ethy H. Brito ethy.br...@inexo.com.br wrote: On Thu, 28 May 2015 11:15:45 -0500 Scott

Re: [asterisk-users] chan_sip.c: Hanging up call

To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http

Re: [asterisk-users] ARI echo test

, and then the usual methods of moving channels in to bridges with ARI could be used.​ On Sat, May 23, 2015 at 1:33 AM, Nick Awesome jl...@me.com wrote: recreate Echo, if that is possible. trying to recode all dialplan to stasis application On 22 May 2015, at 19:29, Scott Griepentrog sgriepent

Re: [asterisk-users] ARI echo test

/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] Custom UUID in originate and AMI

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 -- [image: Digium logo] Scott Griepentrog Digium, Inc

Re: [asterisk-users] FXO advice

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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan

Re: [asterisk-users] Update peer IP address

PM, Daniel Heckl daniel.he...@gmail.com wrote: Okay, Scott, I think we are on the wrong path. Maybe I'm wrong though. I will summarize again briefly the problems together: - The peer ip address could be another than the ip address of incoming invites - After an re-register

Re: [asterisk-users] Update peer IP address

PM, Daniel Heckl daniel.he...@gmail.com wrote: Scott, I have changed the configuration as said it and will test it. I’m curious. Can you briefly explain what insecure=invite,port does? ;insecure=port ; Allow matching of peer by IP address without ; matching port number ;insecure=invite ; Do

Re: [asterisk-users] Update peer IP address

://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] Update peer IP address

: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] PJSIP configuration for AWS/EC2 based Asterisk 13.1.0

mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check

Re: [asterisk-users] connect call to queue to specified agent

-- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] asterisk -r spammy

://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

[asterisk-users] Dial Plan Issue

') [2015-02-10 15:01:50] VERBOSE[32567][C-000f] pbx.c: == Spawn extension (subMachine, xx, 4) exited non-zero on 'SIP/SMtrunk1-000f' I copied the context from the FreePbx box over to the new box so the code should be the same. Any help would be appreciated. Thanks, Scott If you

Re: [asterisk-users] Dial Plan Issue

One follow-up. At the end of the call, after it dis-connects I get the following error: [2015-02-10 15:33:42] NOTICE[4524]: pbx_spool.c:402 attempt_thread: Call completed to SIP/SMtrunk1/xx Thanks, Scott Haley 5-2244 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk

Re: [asterisk-users] constantly increasing load in Asterisk 11.14

options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http

Re: [asterisk-users] Problem with Cisco Phones

://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] MWI issue

Notify come in from Avaya for the extension (I did this with a tcpdump). My question is how do I configure Asterisk to act on that request and call an agi program to do what I want. Any help would be appreciated. Thanks, Scott Haley If you are not the intended recipient of this message

Re: [asterisk-users] Problem with Cisco Phones

-- 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 -- [image: Digium logo] Scott Griepentrog

Re: [asterisk-users] Problem with Cisco Phones

://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256

Re: [asterisk-users] Problem with Cisco Phones

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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW

Re: [asterisk-users] sip show channelstats reliable?

: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct

Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue

| |(VM) | | 192.168.1.239 | |---| On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing

Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue

list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us

Re: [asterisk-users] Smartphone Mobility App?

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256

Re: [asterisk-users] Register multiple phones to a single AOR with PJSIP

://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] Call forwarding from Phones and getting the referrer IP

/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] dialplan reload context

-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 -- [image: Digium logo] Scott

Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda itemDeprecate AMI/AGI(Ben Klang)

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com

Re: [asterisk-users] PJSIP and NAT behind a dynamic IP address

visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http

Re: [asterisk-users] Issue playing high quality white noise

: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US

Re: [asterisk-users] Sent ami event from AGI?

-- 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 -- [image: Digium logo] Scott Griepentrog

Re: [asterisk-users] Voice Mail Questions

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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW

Re: [asterisk-users] how can queue agents choose which call to answer?

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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan

[asterisk-users] compiling Asterisk

server CentOS 6.5. Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone: 314-515-2244 Email: scott.ha...@edwardjones.commailto:scott.ha...@edwardjones.com If you are not the intended recipient of this message (including attachments), or if you have received this message

Re: [asterisk-users] compiling Asterisk

Had to re-install and change selinux to disable. Works now. Thanks, Scott Haley 5-2244 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Friday, September 12, 2014 1:00 PM To: Asterisk Users Mailing List - Non

Re: [asterisk-users] Call Transfer Fails - Not a Valid Extension

To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out

Re: [asterisk-users] asterisk SugarCrm integration

via another app). I would recommend filing an issue on the yaai project for 7 support. There may also be some other resources I've missed. On Thu, Aug 28, 2014 at 3:18 PM, Marek Cervenka cerv...@fpf.slu.cz wrote: it's old. sugarcrm v7 is not supported Dne 28.8.2014 v 14:54 Scott Griepentrog

Re: [asterisk-users] asterisk SugarCrm integration

? 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software

Re: [asterisk-users] PRI timing settings

On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.netmailto:j...@jeff.net wrote: 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. I don’t

Re: [asterisk-users] Asterisk peer definition registration

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317

Re: [asterisk-users] [OT] Split a recording based on a presence of beep sound

/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] Asterisk seding 2 INVITEs all of a sudden

To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out

Re: [asterisk-users] enable features

mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check

Re: [asterisk-users] Internal timing under load is critical ?

list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out

Re: [asterisk-users] Directory app not working with realtime

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317

Re: [asterisk-users] Directory app not working with realtime

For clarification: I was speaking of the directory.php which didn't support realtime last I looked at the code. The app_directory built in to Asterisk should support realtime. Can you determine which one you're using? On Wed, Jul 30, 2014 at 9:46 AM, Scott Griepentrog sgriepent...@digium.com

Re: [asterisk-users] Directory app not working with realtime

describing why it isn't working. On Wed, Jul 30, 2014 at 10:32 AM, Tech Support aster...@voipbusiness.us wrote: Scott; I’m using Asterisk’s built-in application “Directory”, not the php script. Thanks; John *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users

Re: [asterisk-users] Limit Asterisk

:29 PM, Scott Griepentrog wrote: Your bottleneck is most likely your drive bandwidth. Even with SAS drives, you'll need to move to a raid 5+ solution with 6+ drives to continue to increase the concurrent calls, or use a storage appliance. To confirm this, install the tool nmon and use the v

Re: [asterisk-users] Asterisk 12.4.0 not able to install pjsip

: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http

Re: [asterisk-users] Limit Asterisk

logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] Simultaneous Ring

(${DIALGROUP1},20,t) ;Dial Group 2 exten = Group2-101,1,Verbose(2,Dialing Group2 set of phones) same = n,Verbose(2, Waiting 10 seconds before dialing) same = n,Wait(10) same = n,Dial(${DIALGROUP2},${TIMER2},t) Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun

Re: [asterisk-users] Simultaneous Ring

That worked. I had to use the *two* underscores in the agi script where I was setting the values. Thanks. Thanks, Scott Haley If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete

Re: [asterisk-users] Simultaneous Ring

Thanks AJ, this sounds like what I need. Thanks, Scott Haley If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any email

[asterisk-users] Simultaneous Ring

) Is there a way to do this without interrupting the first call? Thanks, Scott Haley If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive

Re: [asterisk-users] recording in mp3

-- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org

[asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card

Machine check exceptions MCP: 1 1 Machine check polls ERR: 1 MIS: 0 Any ideas on how I can further diagnose and pursue this? Google does not reveal much related to this issue that is useful. Thank you! -- Scott L. Lykens Keystone Medical Management Solutions, Inc

[asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card

0 99 0 0 0 0 0 7714712 42752 17632400 0 4 205 414 0 0 99 0 0 0 0 0 7714760 42804 17632400 023 216 430 0 0 98 2 0 0 0 0 7714756 42812 17632400 0 4 201 409 0 0 99 0 0 Thank you. -- Scott L. Lykens Keystone

Re: [asterisk-users] wct4xxp Excessive Interrupts Resulting in Unusable System or Card

the hardware independent of its present OS. Thank you. Scott -- _ -- 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] Trunk issue

/agi-bin/tbsdial.agi': File does not exist. The file is in that directory and is owned by the user asterisk. Why does it say the file does not exist? Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message

Re: [asterisk-users] Trunk issue

-set_variable($vmvariable, $vmvalue); $agi-set_variable($timer, $timervalue); $agi-set_variable($branch, $branchvalue); Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us

Re: [asterisk-users] Trunk issue

Here is the directory listing: [root@nxdasterisk-3 agi-bin]# ls -al total 12 drwxr-xr-x. 2 asterisk asterisk 4096 Apr 28 12:11 . drwxr-xr-x. 12 asterisk asterisk 4096 Apr 28 12:26 .. -rwxrwxr-x. 1 asterisk asterisk 590 Apr 28 11:55 tbsdial.agi Thanks, Scott Haley 5-2244 -Original

Re: [asterisk-users] Trunk issue

One more thing. I have this exact same script working on an Asterisk 1.8 box. This is a new Asterisk 11.7 box. Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent

Re: [asterisk-users] Trunk issue

AGI Script /var/lib/asterisk/agi-bin/tbsdial.agi tbsdial.agi: Failed to execute '/var/lib/asterisk/agi-bin/tbsdial.agi': Permission denied Thanks, Scott Haley 5-2244 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad Sent

Re: [asterisk-users] Trunk issue

That seemed to fix it. Thanks to everyone. Thanks, Scott Haley 5-2244 If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. If you do not wish to receive any

Re: [asterisk-users] Trunk issue

It is just plain Asterisk. I solved the original problem of it not being in the from-pstn context, now I am getting a rejected error I believe from the CM. Thanks, Scott Haley 5-2244 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun

Re: [asterisk-users] Asterisk -rx, how expensive is it? Should you avoid spamming it?

://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239

Re: [asterisk-users] Asterisk -rx, how expensive is it? Should you avoid spamming it?

/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org

Re: [asterisk-users] Asterisk -rx, how expensive is it? Should you avoid spamming it?

the same go for issuing a sip show peers through the AMI? And do you know where I could find information of what asterisk versions may use cached information instead? What would you suggest be better ways to monitor asterisk information? On 24 April 2014 17:58, Scott Griepentrog sgriepent

[asterisk-users] Trunk issue

, reason (1) Hangup [Apr 23 08:20:59] NOTICE[19157]: pbx_spool.c:392 attempt_thread: Queued call to SIP/SMtrunk/913145152244 expired without completion after 0 attempts Thanks, Scott Haley IS Voice Projects Team Edward Jones Investments Phone: 314-515-2244 Email: scott.ha

Re: [asterisk-users] Strange dropped calls

://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256

[asterisk-users] SIP 603 Declined error message

:172.17.184.46:31285 --- INVITE sip:51...@edj.devjones.com SIP/2.0 From: Haley, Scott sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00 To: sip:51...@edj.devjones.com Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 INVITE Max-Forwards: 71 Via: SIP/2.0/TCP 172.17.184.46;branch

[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Unknown)

2012/12/19 Scott Huang gyration.hu...@gmail.com Hi I've saw some similar case in the mail list, but seems no standard answers, so I decide ask here again. Is there anyone see the message below ? I use asterisk(1.8.11-cert 9) in my openbts2.8, and when I made a phone call

Re: [asterisk-users] No rtpmap codec info in 200 OK

Ok. Asterisk sends the rtpmap info for the codec. Is it possible to remove this from the 200 OK sent by Asterisk? Possible direction I should look. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] No rtpmap codec info in 200 OK

It seems quite unlikely that the presence of an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any problems. Thanks for the reply. I'll expand on the scenario... This particular ATA does not send 'a=rtpmap' for any codec. When talking to a Asterisk PBX everything

Re: [asterisk-users] No rtpmap codec info in 200 OK

On 20 December 2011 12:51, Bruce B bruceb...@gmail.com wrote: I could be wrong but this sounds like a NAT issue rather SIP related packet issue. I looked at this to start with. Spent sometime comparing addresses and ports between successful and failure packets. Couldn't see any ports that

Re: [asterisk-users] No rtpmap codec info in 200 OK

On 20 December 2011 14:22, Bruce B bruceb...@gmail.com wrote: Can you register with Eyebeam to VSP and have it work? Make sure you are on the exact same network as the ATA when making this test. This should isolate the NAT issue. Great tip. Eyebeam dosen't send a rtpmap for known codecs

[asterisk-users] No rtpmap codec info in 200 OK

Hi, My VSP uses Asterisk to which I'm connected with an ATA. When I receive an inbound call the invite includes the following... v=0 o=root 32218 32218 IN IP4 202.52.129.50 s=session c=IN IP4 202.52.129.50 t=0 0 m=audio 16864 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101

[asterisk-users] About X100P and TDM400P analog card in China

have to cut the phone line to force it hang up. So can TDM400X work with such a system without tone only with music and voice? Thanks. Regards. Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] About X100P and TDM400P analog card in China

provide ) answer supervision. This will certainly complicate what you want do do. John Novack Scott Zhang wrote: Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27

Re: [asterisk-users] About X100P and TDM400P analog card in China

Thanks. I see. Regards. Scott On Wed, May 11, 2011 at 3:43 AM, John Novack jnov...@stromberg-carlson.orgwrote: Assuming you have read the link you provided, and understand most of what it said, the link really doesn't address calling out over a POTS (copper) line. When Asterisk dials out

[asterisk-users] Registration problems - Vitelity

Hi All-   I have successfully routed calls into our asterisk system from several DID providers in the USA, but for some reason I'm having a problem getting Vitelity to work.   We are using the IAX protocol, and the symptom is that only about 50% of the calls terminate properly into my

Re: [asterisk-users] [Dahdi 2.3.0.1] Does it need all those modules?

I just went through a Dahdi rebuild, and I seem to recall a message that all modules will be loaded until you set up the dahdi configuration files. regards Scott On 7/9/2010 11:41 AM, Gilles wrote: Hello To use Dahdi + Asterisk with a PCI card with a single FXO port, I just... 1

[asterisk-users] Conditional includes in iax.conf

Hello- For maintenance purposes, if possible I'd like to use the same iax.conf file in several different asterisk systems. However, on one of the systems only, I would like to include an IAX register command to another external system. Within iax.conf or other configuration files (other

Re: [asterisk-users] Conditional includes in iax.conf

different from each other. cheers Scott -- _ -- 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

Re: [asterisk-users] Conditional includes in iax.conf

On 7/7/2010 11:52 AM, Kevin P. Fleming wrote: On 07/07/2010 01:46 PM, Scott Stingel wrote: On 7/7/2010 11:25 AM, Danny Nicholas wrote: -- Rather than trying to determine what system you are on, just make the included file be empty on all except the desired server. OK

Re: [asterisk-users] Warning spamming for any unsynchronized ISDN port with dahdi-2.3.0.1

(or chan_dahdi.conf depending on your setup) Scott -- _ -- 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

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

zapata files that you had CPE for two spans and NET for the other two, and your dahdi_chan setup is set up the same. But I'm thinking perhaps during testing you plugged a CPE on your new setup to a CPE on the other, which would produce the symptoms you see. -Scott

Re: [asterisk-users] DAHDI: Inbound BRI call, DDI not presented

On 6/22/2010 2:03 AM, Tzafrir Cohen wrote: On Mon, Jun 21, 2010 at 09:08:02AM -0700, Scott Stingel wrote: Hello- I have a system with one D410P and one B200P (both OpenVox). All is well with the D410P, inbound and outbound, and I can initiate calls on the B200P BRI span

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