similar to: Dialplan variables

Hi, I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone. However, I cannot find the SIP tab. Would someone please give me a few pointers? The screen capture can be seen at URL below http://www.dslreports.com/forum/remark,12022987~mode=flat Regards, Norman Zhang

Finally I decided to rewrite my dialplans according to the right sintax, that is exten => someexten,priority,application(arg1,arg2,...) should be exten => someexten,priority,application,arg1|arg2... Isn't there anybody skilled enough in regular expressions that could write a quick Search 'n' Replace vi command, please? TIA, Alex

Hi, May I ask what ports are necessary for SIP communication through a firewall? I read somewhere that UDP/5060 alone is enough. Some recommends more ports to be opened for RTP. Regards, Norman Zhang

Hi, I'm running asterisk-1.0.2-2mdk. When I tried to start it with /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvgc, I get [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Ouch ... error while writing audio data: : Broken pipe # ps aux | grep mpg123 root 5237 0.1 0.4 5816 4444 pts/0 S 18:45 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

A trick I use is to give tab-delimited text files created by excel or R the .xls extension: this may apply, and it may be better, to CSV: R couldn't see any difference whichever the extension will be, and windows will be foolished by the .xls extension an will open the file with Excel that will handel it CORRECTLY. If you need to have the data always alligned in Excel and R this trick will

Hi, I'm using samba-*-3.0.6-4.3.100mdk and libkrb51-1.3-6.3.100mdk on LM10.0. A similar summary to what I'm seeing could be found here. http://lists.samba.org/archive/samba/2004-July/090210.html My relevant config info could be found below. May I ask how could I solve this in LM10.0? What packages do I need to update? The problem does not arise with NT. It happens to only W2K, XP,

Hi, I am thinking a network configuration where a SAMBA server is installed in a window machine so that the web server on that window machine can talk to linux devices. Is it possible, or it is a stupid idea? Thank you. Kind Regards, Jim

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

Hi, I'm using samba 3.0.6. I see computer_name folders created in homedir. This never happened before I switch from Domain to ADS. Could someone please give me a few pointers? Regards, Norman Zhang [global] workgroup = ARKONDOMAIN realm = HQ.ARKONNETWORKS.COM server string = Samba Server %v security = ADS obey pam restrictions = Yes

Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and

Dear R-users, I have a factor variable within my data frame which I derive week after week from a POSIXct variable using the cut(var,"weeks") command I have found in the chron package. The levels() command gives me: [1] "2009-03-30 00:00:00" "2009-04-06 00:00:00" "2009-04-13 00:00:00" "2009-04-20 00:00:00" "2009-04-27 00:00:00"

I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial("IAX2/firefly@89280250/3",

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

Hi List; My Asterisk version is 1.4 and I am trying to use the ResponseTimeOut() application to control the response time of the Background function, but when the running arrive for the ResponseTimeOut() then the call drop and my debuging says: No Application 'ResponseTimeout' for extension (Test_Bilal,s,3) Spawn extension (Test_Bilal,s,3) exited non-zero on 'Zap/1-1' Hangup

In data venerdì 26 dicembre 2014 16:17:49, Hu Tao ha scritto: > Signed-off-by: Hu Tao <hutao@cn.fujitsu.com> > --- > daemon/btrfs.c | 33 +++++++++++++++++++++++++++++++++ > generator/actions.ml | 16 ++++++++++++++++ > src/MAX_PROC_NR | 2 +- > 3 files changed, 50 insertions(+), 1 deletion(-) > > diff --git a/daemon/btrfs.c b/daemon/btrfs.c > index

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in