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We have recently started making more frequent use of the meetme conference of our * system. We are using v1.0.8 with a 2.6.11 kernel on our system. We generally have 4 callers in it: two with the gsm codec and 2 with g729. Initially, the conference works fine and there is little latency. After about 15min., though, the latency is very noticable and by 25min it's unbearable. If we all leave

I have a * box running our house and on one extension we are getting spurious DMTF tones during conversations. It only happens on one of the 3 FXS ports and it's the one w/ a cordless phone on it. At first I thought someone was being careless and just hitting a button on the other end of the line, but it's happening too much for that... Has anyone run into this before? -- -M There

Every so often we need to send the # dtmf tones but * interprets that as the initiation of a transfer. The best solution I've found so far is outlined at: http://lists.digium.com/pipermail/asterisk-users/2004-March/039501.html This disabled transfer for a call. I take this to mean that there is no way to send dtmf for # with transfer enabled? Thanks! -- -M There are 10 kinds of people

I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in

I may be way behind here, but I see that digium redesigned their site. I cannot find the mailing list search screen. I have found the mailman list page, but that doesn't have have a nice search ability. Do I need to just rely on google and other generic search engines or is there a search on the digium site? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in

I just deployed * on my home system last Sunday. 2x since then the Zap hardware seems to have malfunctioned on some way. One time it would just screech out one FXS, even though it would ring. The other time * would bridge to my FXO but it never got out on the line. I have a new TDM400 with 3 FXS and 1 FXO. Both times I tried unloading the zaptel drivers (which worked) and reloading them,

Hello, all! I'm looking at the wiki page and info on the mailing list and I'm getting conflicting info... I am using the manager API from the telnet CLI and I am testing creating calls with it. I login with events: on and I can originate calls just fine. However, when I set ActionID on an Originate, I cannot see anywhere where that actionid carries into the Event output. But I found

I haven't seen this appear on the list, so I thought I would resend it... Sorry for the repost if it did appear before... ----- Forwarded message from Michael George <george> ----- Date: Wed, 3 May 2006 21:48:09 -0400 From: Michael George <george> Subject: meetme conference latency degrades... To: asterisk-users@lists.digium.com We have recently started making more frequent use

iH have a TDM400 w/ 1 FXS & 1FXO card. after about a day or so the FXO stops working. after a reboot i get the following error. to get the card working again i have to shutdown, physically unplug/replug the card (reboot only, power off/on does not work) any ideas what may be going wrong here? thanks - hcir Zaptel Configuration ====================== Channel map: Channel 01: FXO

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk

I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above,

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now

I have a Digium TDM400 in my system and I'm using my main system as my asterisk box at home (very light load). When I start up *, though, it grabs my sound card and I cannot play other music through it (e.g. x/ XMMS). I have moved the alsa.conf and oss.conf files so that there is no configuration for them (though those files seemed to do little), but still the sound card is grabbed. How can

I'm using * at home and I planned on having * let the answering machine in my kitchen to the "general" voicemail getting. However, about 6s into the call * will hang up the line. I found a post about OHT somethingorother, so I can probably work around it, but I'd like to know what's happening and if there's a better way around. Thanks! -- -M There are 10 kinds of

I just got a Budgetone 101 and I have it hooked to my * box. I thought I'd read somewhere that we can program the buttons on these phones to send DTMF tones, thereby effectively programming them. However, according to the user's manual, they have predefined SIP functionality. My dialplan implements the festures I want (transfer, message, stuff like that), so for uniformity, I'd just

I just put together a SuSE 9.1 box and got the * and zaptel drivers today. Everything built fine but now I get: FATAL: Error inserting zaptel (/lib/modules/2.6.5-7.104-smp/misc/zaptel.ko): Invalid module format Can't find too much in the list on it for this, just ztdummy. First thing I'll try is rebuilding the kernel with the same copiler as the drivers. Any other suggestions? -- -M

On the extensions.conf explanation page is a mention of the "switch" statement and it refers one to the "connecting two * servers" page. The only mention of the switch statement there is brief and in the example. However, the example seems to have some errors in it. It shows a sample of what's in the extensions.conf file, but it clearly has sections which would be in the

I've a question about the bandwidth consumed by IAX2/GSM. According to the wiki page, the GSM codec should run about 13 kilo-bits/sec for a voice encoding. However, watching gkrellm when I initiate a call to Digium, it looks like the channel is taking a consistent 5-6 kilo-bytes/sec. That's a lot more bandwidth than it should take. Is there perhaps a setting I have wrong somethere in

I'm trying to set up queueing on my system and for the most part it works fine. However, I'm trying to give the user the ability to break out of the queue. Putting the "H" option into queue() doesn't seem to work. That seems to hang up from the extension that was being dialed and then after the delay start ringing them again. Looking at the mailing list archives, others