similar to: Alternatives to Digium 729

Even though they make the cards and advertise that they support data modes, digium won't support data mode on the $500 card they sold to me, so I must turn to the list. Has anyone configured a T100P to 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 -----

All of the stuff I've googled 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

Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last

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

We have asterisk running on a quad processor dell. The kernel has been compiled with SMP. However, asterisk seems to only use 1 processor. 3 of the 4 always stay at 100% idle. Is it pointless to have a multi-proc machine? I was going to buy a new dual 3.6Ghz Xeon server but if nothing will take advantage of the other proc... Perhaps my conception of multi-proc/threaded is warped. If asterisk is

This is frustrating. Asterisk has crashed now twice today and neither crash has produced a core file. My ulimit is unlimited. I'm using safe_asterisk so asterisk is restarting immediatly, 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 --

I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the

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 =

Hi all, supposing to have speex working at 3.95, when VAD/CNG is employed, which is the size of those "silence" samples respect to voice samples? I am wondering if it could be possible to lower even the transmission of VAD/CNG samples by specifying "outbound" in the communication channel only 1 bit instead of sending the whole VAD/CNG packet. I am doing the same for AMR

Hi List, i've set up a cisco 7912 for my asterisk box. I've had problems with VAD and CNG. After googling a bit, i've found an article about asterisk not supporting these two protocols, therefore it's better to turn them off. Since then i did not found answer to my two questions, maybe somebody here could help me: a) am i even able to turn off vad/cng on cisco 7912? SIP

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:

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

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

Has it been a whole week since a question about VAD? :-) I am curious why the VAD algorithm described in G.729 Annex B is not used by Speex. Complexity, Patent, Quality, or what? -mjc

Hi guys, I got 2 licenses of g.729 and while running the asterisk with Monitor (for recording a channel) and using one channel for the call... I receive this error: WARNING[23826]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! many times.... it starts only when the call through the Zap channel takes place. while this error is being running on my screen I ran the cli command:

Anyone used the combination above? We are and it sounds like crap. The audio drops out in regular intervals which suggests that someone's g729 isn't doing its job correctly. I'm blameing XTen cause when I make a ulaw call that gets converted to 729 using digium's 729, calls sound fine. Anyone else? Similar experiences? -Matthew --

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

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

Jean-Marc Valin wrote: >>In particular, (I'm not really sure, because I don't thorougly >>understand it yet) I don't think your jitterbuffer handles: >> >>DTX: discontinuous transmission. >> >> > >That is dealt with by the codec, at least for Speex. When it stops >receiving packets, it already knows whether it's in DTX/CNG mode.