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...okups for incoming media frames are a very fast hash lookup instead of an absolutely insane array traversal. In a quick test, I was able to get more than 3600% more IAX2 channels on my machine with these changes. On Thu, Apr 24, 2008 at 6:51 AM, Mike Clark <mike at infoanywhere.com> wrote: > I upgraded one of our servers to 1.4.19.1 last evening, but ended up > having to drop back because of IAX calls failing at a near 50 % rate. > Here is the message that we would receive on the console (multiple > times), and then it would hangup the call. > >...

I submiited to the list last night, but it never showed up. Here we go again. I've tried building Asterisk 1.4.15 on Solaris based on instuctions here, http://forums.digium.com/viewtopic.php?t=5888. However, this is the message I get. This is Solaris on X86. Any ideas? [CC] stdtime/localtime.c -> stdtime/localtime.o stdtime/localtime.c: In function `localsub':

...ity found on the server, and the client sees absolutely nothing. What's strange is I personally run CVS-head at my house, dated 11/10/04, it has no problems at all. If anyone has info on this please help, it's killing us :D Matt -----Original Message----- From: Mike Clark [mailto:mike@infoanywhere.com] Sent: Thursday, March 17, 2005 11:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Netlogic inbound DID issue Anyone out there using NetLogic DIDs? And have inbound working? I got outbound working, but no joy so far with inbound. Here are the relevant parts from my conf...

Anyone out there using NetLogic DIDs? And have inbound working? I got outbound working, but no joy so far with inbound. Here are the relevant parts from my conf files: iax.conf [general] tos=lowdelay jitterbuffer=no register => username:secret@zoot.netlogic.net [netlogic] type=friend host=dynamic context=sourcekit-main auth=plaintext username= secret= disallow=all allow=ulaw allow=all

Can anyone point me to a sample or information on using MeetMe like this? Conference room is set up with 2 PINs, one for the moderator and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad).

Can realtime extensions be used within AEL contexts? I've tried without success. Also did some Googling and looking around on the Wiki, but couldn't find much. Thanks, Mike Clark

I have noticed other parts of the U.S. and the world posting about user groups, and wondered if we have enough Asterisk users and interest in the Carolinas to start a group. I am willing to help with the organizational effort if there is interest and maybe one or two other folks who would be willing to help with organization. I tried to post this message last week, but accidentaly used the

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

We have gotten stuck trying to get a highly available Asterisk cluster fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's behind the virtual public IP. I got as far as getting phones registered and being able to place calls that rang and you could answer, but there was no audio. So, I enabled RTP debugging and discovered Asterisk was still attempting to send the audio

I have installed Asterisk 1.4.15 on Solaris and got it all running seemingly fine. However, when I record a message or voicemail, it will not recognize the '#' key to stop recording. Hanging up is the only way to end the recording. DTMF seems to work fine elsewhere. Is there a common problem or issue that I am missing? I've tried Google, but have had no success. Thanks, Mike

I upgraded one of our servers to 1.4.19.1 last evening, but ended up having to drop back because of IAX calls failing at a near 50 % rate. Here is the message that we would receive on the console (multiple times), and then it would hangup the call. Avoiding IAX destroy deadlock Anyone else having similar problems? Thanks, Mike Clark

When agents are on the phone, and the CLI queue show command shows their status as busy, the queue still tries to send them calls. Running Asterisk 1.6.0.17 and using AddQueueMember to dynamically add agents. ringinuse is set to no for queue. Agents are using Polycom 430s. dialplan: exten => s,n,Queue(orders,itT,,,80) queue definition in queues.conf: [orders] maxlen=20 queue-thankyou=

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

I would like to do something like: exten => s,1,playback(http://my.server.com/file.wav) I tested and it does not work. It seems highly likely that someone would already have done this one way or another. I know I could do a system wget and then play the local file, but wanted something a bit more elegant. Thanks, Mike Clark

We are having an occasional one w-way audio problem that occurs about every 25 - 30 calls on a system configured as follows: Asterisk 1.2.12.1 Sangoma A101 w/wanpipe beta9 Polycom 500s w 1.5.3 This happens only on inbound calls from the PRI. The external caller can hear our customer answer and say hello, however, our customer cannot here their caller. Typically, the caller calls right back