similar to: speex in asterisk 1.0.10

Hi guys, I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at the third days I activated setting jitterbuffer=yes and suddenly there is no voice when the call is picked up. It's really weird as if asterisk stops sending rtp packet. I've checked asterisk log and found nothing suspicious. Just weird :S. I tried it in 3 asterisk server and all of them are having

Hi, Does anyone know how to make this thing (7940) work with asterisk (chan_sccp module) ? I've set the configuration according to the wiki and now the phone just keep asking for CTLSEP<xxx>.tlv from my tftp server. In the cisco's web interface, I found this in the device logs : 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 ...

Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : - when I called from 7910 to another sip phone in the same asterisk server, the call took place normally. - when I called from 7910 to another sip phone in different asterisk server, the call is answered but I cannot hear nor say anything. The phone just immediately lose its tone. - when I got

Hi, I've tried to upgrade my asterisk to 1.2.3 again after disastrous bug incident yesterday but when I called and the phone was picked up, there was simply busy tone... Weird, is this another bug in asterisk 1.2.3? Currently, I rollback again to asterisk 1.0.10...:( Is there any configuration change issue in 1.2.3 cause I've just used my configuration that worked in asterisk1.2.2 ?

Dear all. I have install samba 2.2.7a in mandrake 9.1 and success access by client, but there is a problem about it. Server samba is giving an ip address 192.11.11.xxx and client is having ip address 192.11.11.xxx and 192.11.12.xxx. in computer client (windows NT, Win2k, and win98) which have an ip address 192.11.11.xxx server can access straightly from network neighborhood (without searching

Hi, Recently, I've just installed asterisk along with AMP.. Everything seems to work fine, just when I tried to record my voice via ivr, asterisk won't play the file if I call it. When I test by dialing *99, the record is played, but when I call straight to the digital receptionist, it just stand there about 7 seconds, playing no sound at all and then hung up.. I use AMP version

Hi, After days tinkering with this digium card (TDM04B), I notice that this card has a slow response in detecting ring signal from pstn and hanging up when the call is over. The delay can consume up to several seconds... Is this normal? Best regards, Stevanus

Hi, Sorry for keep hammering the list with this annoying question. Can we use Intel 536 ep (not 537ep that is in wiki) as x100p clone? I know I've asked it in this list a couple days ago but none responded so far and I'm getting frustrated pairing it with asterisk as the zaptel driver could not detect it. I just need more information before I throw this intel 536 EP to the garbage can

Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29

hi, recently I installed zeroconf for asterisk... I've already followed the asterisk+zeroconf how to (which is too short), but it came with an error message... asterisk: relocation error: /usr/lib/asterisk/modules/res_zeroconf.so: undefined symbol: DNSServiceRegister Ouch ... error while writing audio data: : Broken pipe it's weird since I've double checked the library and header

Hi, My asterisk is frequently dead by itself. It leaves messages: /usr/sbin/safe_asterisk: line 40: 24890 Segmentation fault (core dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 139 Asterisk exited on signal 11. Automatically restarting Asterisk. Anyone has any idea of the cause? Thanks.. Best Regards, Stevanus

Hi, Why does my asterisk keep forking instances at random times everyday? When I do ps aux, I got this: asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk -vvvg -c asterisk 23558 0.0 5.1 26040 12248 ? S 09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S 11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S

Description of what I am trying to do: I am using the xyplot code below to plot the variable ?MeanBxg? against the variable ?PercentVarExplained? for all 9 possible combinations of variables ?bdg? and ?bdx?. Within each of these 9 scenarios I am plotting a separate line for each of up to 9 different methods that I used to estimate the variable MeanBxg. These methods are identified by the numeric

Hey all, I''m not sure if this is a ZFS bug or a hardware issue I''m having - any pointers would be great! Following contents include: - high-level info about my system - my first thought to debugging this - stack trace - format output - zpool status output - dmesg output High-Level Info About My System --------------------------------------------- - fresh

Canuck15, No, I hadn't played with the gains. But I've now done so and no difference unfortunately. Thanks for the suggestion though. I have discovered that after Asterisk has answered the call and the remote caller has hung up, if I lift the receiver on a phone connected to the line (in parallel with Asterisk), Asterisk then DOES instantly hang up. Would it be reasonable to assume the

I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxxxxxxxxxx type=peer username=0406082250 Regards Anders Svensson -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sorry for the repetition, unless I've got bad deja vu this questions been asked before but I couldn't turn up an answer on CRAN. Is there already any code in existence for local dependence measures such as Moran's I or Getis and Ord G? Also, S-Plus has a number of interstingly named Cluster Algorithms based on some previous stand-alone fortran algorithms (agnes, daisy etc.) which

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

I am completely new to this.. I am trying to make simple traffic control.. I have read quite some manuals and posts that i found, but i don''t understand much, i think.. My situation is: i have linux server which i am using for firewall for local network.. also on same linux server i have torrentflux for downloading torrents.. What i want to do is to give priority to local clients(2-3

I'm guessing that I need more info entered into the 'message centre' section. What did you key in? Paul Hales IT Support Adairs -----Original Message----- From: Jeff Pyle [mailto:jpyle490@gmail.com] Sent: Friday, 17 September 2004 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 I have two IP 500's on my Asterisk