search for: arnarson

...l command is not suitable unless I am able to tell somehow that the call in question is being forwarded (which is of course not the case, as the Dial command is called befer the REFER is sent). Can anyone think of a way to get the call back to the transferrer after this timeout? Best regards, ?rn Arnarson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100126/589ebb2c/attachment.htm

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

...erisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when S...

Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some

...and couldn't find anything indicating different. Haven't started looking at what the output looks like, but it would be nice if someone could point me to a document going through the changes so I don't have to re-invent the wheel. Anyone have any info on either one? Best regards, ?rn Arnarson

Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to

Hi, Not sure where to submit this to so I'll try here. Below is the toneset for Iceland. Hopefully this can be added into the asterisk package. [is] description = Iceland ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/5000 congestion = 425+250/250,0/250 callwaiting = 600/100,0/100,600/100,0/9000 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0

...5555555? There's nothing in the invite indicating that as a host. Furthermore, the verbosity level was at the highest level, and I never saw the INVITE above come into the Dialplan. All I saw was the INVITE, and then the "No such host" error. Any ideas whatsoever? Best regards, ?rn Arnarson

Hello, What are the current methods for playing digits on different languages? I presume the big ones like German have been dealt with, saying 2 and 20 to announce 22. How is this currently decided? What about languages that say 20 and 2? Is there a way of configuring via config files or recordings somehow? Obviously you could record the sound file for 20 as "twenty and", but that

Hi, I've noticed that the UNIQUEID for a call is not the same in the Dialplan (when executed e.g. exten => s,n,NoOp(${UNIQUEID}) as it is when passed via STDIN to an AGI script. Is this normal, and is this supposed to behave this way? The UNIQUEID received in the AGI is usually .001 higher than the one in the dial plan -- but sometimes it is also a second behind. Here's an example

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

...ISK-18995 <https://issues.asterisk.org/jira/browse/ASTERISK-18995>] - Support for OGG/Speex file format (Reported by Timo Teräs) - [ASTERISK-26087 <https://issues.asterisk.org/jira/browse/ASTERISK-26087>] - Icelandic grammar support for voicemail and numbers (Reported by Örn Arnarson) - [ASTERISK-26058 <https://issues.asterisk.org/jira/browse/ASTERISK-26058>] - [Patch] Add uptime and last reloaded to FullyBooted AMI event (Reported by Niklas Larsson) - [ASTERISK-25925 <https://issues.asterisk.org/jira/browse/ASTERISK-25925>] - Allow Early Bridges on...

Hi, I am running Debian etch on dom0. I want to run debian unstable on one DomU, but I keep getting 4gb seg fixup errors, presumably because the libc6-xen installed there is much newer than on Dom0 (or perhaps there is some bug with Xen and libc6-xen version 2.7-16). I have two questions; 1. Why does the libc version on Dom0 matter at all for DomU (if that is the case)? As far as I've read,