search for: cytelcom

Hey gang, I'm trying to setup the ability to dial a SIP user via their email address. I'm using SJPhone as my tester UA, but most clients will be using XTen Pro. I added an SRV DNS entry into our zone, and it returns: ; <<>> DiG 9.2.1 <<>> SRV _sip._udp.cytelcom.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 47390 ;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 2, ADDITIONAL: 3 ;; QUESTION SECTION: ;_sip._udp.cytelcom.com. IN SRV ;; ANSWER SECTION: _sip._udp.cytelcom...

...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 codec from so...

...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 data mod...

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

...d 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 =

..., 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. -----------------------------------------...

...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 several months...

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 be sent out SIP to my Cisco Gateway. Im running Asterisk on a Dell Dual P3 1.2ghz running Fedora. Is there a calculator or a spreadsheet which could tell me about how many calls I will be able to make through * ? ----------------------- Harold Workman CCNA, CCNP Cytel Communications hworkman@cytelcom.com Ph. 281-449-4000 x3098

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

...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. -----------------------------------------...

OK. I just downloaded asterisk-1.0.3.tar.gz and did a 'cvs co -r v1-0 asterisk' into 2 seperate directories. I then did 'diff -ur asterisk-cvs/ asterisk-1.0.3/' and there were source code line differences between the two. Some code that was in asterisk-cvs wasn't in asterisk-1.0.3 and vice versa. Which of those is "the most recent"? If someone wants to use cvs to

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

...nd it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AgentCallbackLogin - invalid extension Hi, I have setup agent 1000 in agents.conf, and the queue cytelcs in queues.conf, and my extensions.conf file looks like exten => 28,1,AgentCallbackLogin exten =&...

Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

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

...n W a l s h > _/ _/ _/ _/ _/ _/ _/ _/_/ kevin@cursor.biz > _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/ > > > > ------------------------------ > > Message: 5 > Date: Tue, 30 Nov 2004 11:57:07 -0600 > From: "Matthew Boehm" <mboehm@cytelcom.com> > Subject: Re: [Asterisk-Users] Really Get 96 Simul Calls? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: <016d01c4d706$142262f0$2700000a@cytelcom.com> > Content-Type: text/plain; charse...