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...n commit access but haven't received a response (either positive or negative). What's the expected turnaround time for this (I contacted him five days ago)? In case he's currently unavailable, is there someone else I could contact? Thanks, Martin -- Martin Böhme Software Engineer mboehme at google.com Google Germany GmbH Erika-Mann-Str. 33 80636 München Geschäftsführer: Matthew Scott Sucherman, Paul Terence Manicle Registergericht und -nummer: Hamburg, HRB 86891 Sitz der Gesellschaft: Hamburg Diese E-Mail ist vertraulich. Wenn Sie nicht der richtige Adressat sind, leiten Sie...

...t. Is there anyone out there with news on VAD/Silence support in Asterisk? Thanks, Matthew -------- Original Message -------- Subject: [digium.com #25781] VAD in G729 Date: Wed, 20 Jul 2005 13:22:28 -0500 (CDT) From: Chris Hozian via RT <support@digium.com> Reply-To: support@digium.com To: mboehm@cytelcom.com Hello, We do not know of anyone else who is providing the G.729 codec for Asterisk. Regards, Chris Hozian > [mboehm@cytelcom.com - Wed Jul 20 13:04:21 2005]: > > Are there other ways to get G729 into asterisk other than using your > codec? I'd like to buy the code...

...o use a T1 strictly as bandwidth? Is there a HOWTO somewhere? Wiki has nothing I could find. I've got plently of public IPs I can assign to it but don't know how. Thanks, Matthew ----- Original Message ----- From: "Matthew A. Nicholson via RT" <support@digium.com> To: <mboehm@cytelcom.com> Sent: Thursday, December 23, 2004 10:38 AM Subject: [digium.com #12961] T100P as bandwidth > Yes the hardware, and our drivers suppport both voice and data modes, > but because of the nature of the configuration we do not support it. > Configuring one of our T1 cards for...

...and read on wiki all relate to "Outlook". Has anyone been successful in getting "Outlook Express" to do click to dial? -Matthew -- ------------------------------------------------------------------------ Matthew Boehm, IT Director Cypress Telecommunications mboehm@cytelcom.com 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. --------------------------------...

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

...ld see all 4 procs fall to 80% idle (20% used on each). Is this wrong? What about the g729 library from digium? Is that multi-proc aware? -Matthew -- ------------------------------------------------------------------------ Matthew Boehm, IT Director Cypress Telecommunications mboehm@cytelcom.com 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. --------------------------------...

I don't know...now I have a _X. in my CDR. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent: Saturday, March 12, 2005 8:05 PM To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Subject: Re: [Asterisk-Dev] SetVarCDR You must have some fux0red config 'cause using _X. works fine here. I haven't had an 's' in my CDRs for over sever...

Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar

Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have

...l does. How do I hangup her voicemail and resume the previous call? The example on the wiki assumes the transferee picks up the phone. :/ -Matthew -- ------------------------------------------------------------------------ Matthew Boehm, IT Director Cypress Telecommunications mboehm@cytelcom.com 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. --------------------------------...

.... No PRI lines to be plugged into the 410 card. MeetMe requires an external timing source. Right now, using the dummy driver. Is it possible to use the card solely for timing purposes? Any benefit to doing so? Or should I just sell the cards? Matthew Boehm, Senior MySQL DBA, The Planet E: mboehm at theplanet.com C: 832-253-8258 O: 281-714-4018 Certified MySQL 5.0 DBA, #226735534 Certified MySQL Cluster 5.1 DBA, #227094842 RedHat Certified Engineer, #805009324738628 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/ast...

Hey guys, I sent an email to the list at 2:57PM central. I just now see it on the list, and its 3:23PM. Anyone else experience this? I am sending this email at 3:24PM central. Lets see when it gets posted to the list. -Matthew

...ediatly, but how the hell am I suposed to find out wtf happened with no core file? Debug log doesn't say anything either. AGRHHHHHHHH -Matthew -- ------------------------------------------------------------------------ Matthew Boehm, IT Director Cypress Telecommunications mboehm@cytelcom.com 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. --------------------------------...

Hello: I found a reference to the application SetVarCDR in the following post but I don't seem to have this available to me in my version of *. HYPERLINK "http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.html"http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.html My version of * is CVS-HEAD-03/10/05-18:42:35 I would like to change the

Hey gang, We currently have a class 3 switch (CSX) that..well..it sucks. It does terrible CDR writes, doesn't support LCR, the list goes on and on. We want to replace this with several asterisk boxes each running one or two 4 port PRI cards. The problem is: I can plug in 20 PRI lines into the CSX (from PSTN) and have 1 come from CSX into asterisk. If 1 call comes in on each of the 20 pris,

Lets say that I have 9 phones with extensions 1 thru 9. (All SIP) Parking extension is set to 700. When I try to park a call to 700 as soon as I press the 7 key (to start typing 700) the person at extension 7's phone rings. So instead of transferring to extension 700, it went to 7. How can I slow down the response time of * to let me get to 700? Thanks, Matthew

Per the wiki: Logging off 1. call the extension for AgentCallbackLogin 2. enter your password followed by # 3. when asked for the extension number just press # But if your exten=> is this: exten => 2010,1,AgentCallbackLogin(3333|3044@mycontext) How do they logoff per the wiki's directions? If you use ACBL as above, it never asks you for the extension number because you have

...n to turn off this crap for lists? I don't care if you are on vacation and I'm sure that everyone else on this list doesn't care either. -Matthew ----- Original Message ----- From: "Dijkstra, Roelof" <Roelof.Dijkstra@nl.compuware.com> To: "Matthew Boehm" <mboehm@cytelcom.com> Sent: Thursday, January 06, 2005 9:24 AM Subject: Out of Office AutoReply: [Asterisk-Users] asterisk addson > Currently i'm on holiday. I'm back in the office on the 24th of Jan. > If you have any urgent questions , please contact the service desk. > > > --...

...iled the module that handles the transcoding it could take advantage of say a AMD64 bit cpu. Although I bet cache has a big impact on performance with this. Brian Greul Texas Shirt Company www.txshirts.com 713-802-0369 / 713-861-6261 (fax) -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent: Monday, January 10, 2005 4:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 64 Bit Support? I wouldn't mind seeing the g729 codec written to take advantage of 64bit processing. Might make a machine capable of handling more PRI...

I just started getting this error today (I've gotten this error befor) and its preventing me from having any incoming calls: chan_zap.c:7542 pri_dchannel: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. PRI has been working fine. I didn't know anything was wrong until someone came and said their DID wasn't working. You call their DID, asterisk shows the