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Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

Seems to be all morning. I have not been able to access for several hours now. W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 7:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi

Anyone know why zaptel would ignore a facility message from an ISDN PRI. I am trying to get Callerid name to work. The carrier says it on and I see it in the pri debug but asterisk never gets it. Any help would be appreciated. Thanks John Bittner Simlab.net Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 572/0x23C) (Terminator) > Message type: ALERTING (1) > [1e 02 81 88]I> > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1)...

When an agent logs into a queue using AgentCallBackLogin, he should be ready to take calls until he logs out right? For some reason the first time a customer calls the queue, it rings the agent just fine but after the agent hangs up the phone and the next caller calls the queue, no more calls will be transferred to the agent. He shows as logged in, but the calls wait in the queue forever and

...k but the sample config files are hard to find. My next step, if I cant find anything on this list is to purchase a Nortel Communication Server for testing. If anyone has a used NCS that works with these phone via SIP please email me off list. Any help on this is appreciated. Thanks John Bittner Simlab.net 9734333011

...ages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful ------------------------------------------------------- ----- Originalnachricht ----- Betreff: [Asterisk-Users] DTMF after answer Von: john@simlab.net An: asterisk-users@lists.digium.com Datum: 05-08-2004 19:57 > Hi, > I am trying to link up a comdial PBX to Asterisk using T1 tieline E&M. I > have it working for comdial to asterisk but not the other way. Comdial does > not listen for any DTMF before answering the ZAP channel...

[root@asterisk zaptel]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.5-1.315/misc/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy [root@asterisk

...ilt with > clang and libstdc++. > > Related to the first question: do the above flags also make clang use > libc++ by default, or is this configurable via a different set of flags? > > Thanks a lot in advance > > Michael > > -- > Michael Schlottke-Lakemper > > SimLab Highly Scalable Fluids & Solids Engineering > Jülich Aachen Research Alliance (JARA-HPC) > RWTH Aachen University > > Wüllnerstraße 5a > 52062 Aachen > Germany > > Phone: +49 (241) 80 95188 > Fax: +49 (241) 80 92257 > Mail: m.schlottke-lakemper at aia.rwth-aachen.d...

...+, whereas the used clang/LLVM compiler was originally built with clang and libstdc++. Related to the first question: do the above flags also make clang use libc++ by default, or is this configurable via a different set of flags? Thanks a lot in advance Michael -- Michael Schlottke-Lakemper SimLab Highly Scalable Fluids & Solids Engineering Jülich Aachen Research Alliance (JARA-HPC) RWTH Aachen University Wüllnerstraße 5a 52062 Aachen Germany Phone: +49 (241) 80 95188 Fax: +49 (241) 80 92257 Mail: m.schlottke-lakemper at aia.rwth-aachen.de Web: http://www.jara.org/jara-hpc

...d air for 15 to 30 seconds then a fast busy. I am working with the carrier to get this fixed but its not going easy. Is there anyway when asterisk sees the progress code to cancel the dial and playback a message mapped to the progress code type. Any help on this would be appreciated. John Bittner Simlab.net 9734333011

Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing

...or all capital tools placed in the technicians possession by the company. Benefits: *40 hour work week *Full medical and dental coverage after 90 days *Paid Holidays *Paid vacation after six months *Company fuel card *Company Van *Company supplied logo shirts John Bittner Please email me at john@simlab.net with responses. Thank You.

...setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net This is a what I have in my dialplan. exten => 207,1,SetVar(ALERT_INFO="Ring Answer") exten => 207,2,Dial(SIP/207) exten => 207,3,Hangup

Has anyone got Polycom Shared Call Appearance working with Asterisk ? If Asterisk doesn't support this, I am willing to put up a bounty of 1000 to get it to work. John Bittner Simlab.net Shared Call Appearance Signaling A shared line is an address of record managed by a server. The server allows multiple endpoints to register locations against the address of record. SoundPointR IP supports shared call appearances (SCA) using the SUBSCRIBENOTIFY method in the "SIP Specif...

...acro in exten-vm and replace it with a normal Dial(SIP/2000) the phones rings, everything works. It looks like the problem is with the dialparties.agi script. I see it exiting with 0 before going into voicemail. Anyone have any idea why? Any help would be much appreciated. Thanks John Bittner Simlab.net

...uded in asterisk distribution as of October. ? I tried patching my current 1.0 but seems the patches are for an older version. I posted a bounty of $250 to get this to work with the newest stable. Needs be able to monitor bridged sip calls with or without a monitoring beep. Thanks John Bittner Simlab.net

...> Related to the first question: do the above flags also make clang use >> libc++ by default, or is this configurable via a different set of flags? >> >> Thanks a lot in advance >> >> Michael >> >> -- >> Michael Schlottke-Lakemper >> >> SimLab Highly Scalable Fluids & Solids Engineering >> Jülich Aachen Research Alliance (JARA-HPC) >> RWTH Aachen University >> >> Wüllnerstraße 5a >> 52062 Aachen >> Germany >> >> Phone: +49 (241) 80 95188 >> Fax: +49 (241) 80 92257 >> Mail: m...

Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI> <-- SIP read from 216.86.35.24:63549: INVITE sip:9734333001 at 69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56 From: <sip:9733901090 at 216.86.35.24>;tag=4F9EF08-163B To: <sip:97...

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040331/c2abf19f/attachment.htm -------------- next part -------------- Hello, Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk. Regards, Tony

Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl