similar to: Conversations cuts: "didn't get a frame from Channel: SIP/..."

I'm using EyeBeam from xten, and whenever I call another user, the callee phone rings but my SIP phone immediately hangs up. The other end keeps on ringing but when the callee answers, there is no sounds. I have found the "Didn't get frame from channel" error occurring in each such call. What does this mean? How can I fix it? -Mike- May 15 22:31:10 DEBUG[4792]:

I have just finished implementing an Asterisk system for my place of business (first one), and after three days of flawless usage, Asterisk seems to have crashed. I wasn't running with '-g', so I don't have a core dump. Here's the sequence of events leading up to the crash: 1. call comes in on our TDM2400P 2. all of our phones (about 26 Polycoms) ring. (it's after

Porque con "less" estás hacienddo un test de una cola. En tu ejemplo estás haciendo un test de dos colas: > 2*pt(-5.4349,df=21.982) [1] 1.855473e-05 El 21/12/2013 21:33, HERNANDEZ CORONADO JORGE escribió: > t.test(1:10,y=c(7:20)) > > Welch Two Sample t-test > > data: 1:10 and c(7:20) > t = -5.4349, df = 21.982, p-value = 1.855e-05 > alternative hypothesis:

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Si miras el código de t.test() verás que para calcular el p-valor llama a pt(). En la página de ayuda de pt() se cuenta que: "For the central case of pt, a normal approximation in the tails, otherwise via pbeta. For the non-central case of pt based on a C translation of Lenth, R. V. (1989). Algorithm AS 243 ? Cumulative distribution function of the non-central t distribution, Applied

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

Hello, To do some testing with Asterisk installed the latest Asterisk @ Home in a Vmware system. All worked fine, I can access the web interface (AMP). I have setup the extention and X-Lite softphone according to the description in the Wike (http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite). I can dial 200 (the softphone extention) and 1234 and they connect (the softphone shows this, as

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Hello, I have a TDM400P with 2x2 configuration of FXOs and FXSs. I set a test extension of '444' to dial out a specific zap trunk and call a local #. First time I call out to '444' everything works fine. If I hang up the call, and within 10 seconds dial the same number again, I get a fast busy. Seems it isn't letting go of the trunk or something, and I don't have a

hello, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call:

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

Hi Everybody, I am trying to use SIP (Sipura 2000) to connect to Asterisk which then dials out a local number using the Digium E100P. We have purchased the G729 codec licenses from Digium and loaded them into Asterisk successfully. However, the call drops immediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a

Hi all :) If I set the burst/cburst parameter to, let''s say, 1500, the command "tc -s -d class show dev eth0" says that the value is 1499b/8 instead of the (correct?) 1500b/8. Is this right or am I doing anything wrong? Many thanks in advance :) Raúl Núñez de Arenas Coronado -- Linux Registered User 88736 | http://www.dervishd.net http://www.pleyades.net

Hello all, I recently noticed the realTime effort and must say it is a nice idea! I would appreciate any help to get it running .. I downloaded the code & patches and succefully patched my asterisk (CVS-HEAD-11/29/04-12). - created a DB called asterisk, and a table sip using the schema supplied at http://bugs.digium.com/bug_view_page.php?bug_id=0002613. - entered an entry: insert into

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

Hi all :) I''ve been using a tc setup for almost two years, but at some point (probably when I switched to kernel 2.6.x, but I''m not sure) it has started making something very weird. For a certain class, the rate is 125000bit and the ceil is 270000bit, but the fastest rate I get is about 75-80000bit, instead of the "promised" 125000, *with no other traffic in

Hi all :) I think that this issue has already been discussed on this list, but google didn''t find anything interesting, so I''m bringing the subject again. The output of "tc" uses "b" meaning "byte" and "bit" for "bit". The "official" suffixes for those units are "B" and "b", respectively,

Hello, I'm using Asterisk@Home. I'm still new to Asterisk, and trying to grasp it all. I'm wanting to do a simple setup of One SIP provider (Broadvoice) and 3 SIP Software Phones. I'm able to call the VoIP provided line fine and get dropped to the digital receptionist (or mailbox). However, when I try to send outbound calls I get "Error 503 Service Unavailable" and

I am hoping some of you can help me troubleshoot this problem I am having with my home asterisk machine. I have incoming POTS service using a SPA-3000 (extension 119). Calls on that line go to an attendant recording that offers a menu choice: press 1 for Nancy, press 2 for the rest of us. In reality, pressing anything other than 1 sends the call to the rest of us by dialing both extensions 101