similar to: Asterisk+mISDN drops calls after 3-4 secs

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

http://dovecot.org/releases/dovecot-1.0.rc30.tar.gz http://dovecot.org/releases/dovecot-1.0.rc30.tar.gz.sig So, this is it. Unless you can find a new and important bug within a week, this release is the same as v1.0. I'll only update the version number and NEWS file. Changes since rc29: * PAM: Lowercase the PAM service name when calling with "args = *". Linux PAM did this

http://dovecot.org/releases/dovecot-1.0.rc30.tar.gz http://dovecot.org/releases/dovecot-1.0.rc30.tar.gz.sig So, this is it. Unless you can find a new and important bug within a week, this release is the same as v1.0. I'll only update the version number and NEWS file. Changes since rc29: * PAM: Lowercase the PAM service name when calling with "args = *". Linux PAM did this

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then continue. Asterisk console says: analog: -- Attempting call on Zap/2/3391818250 for 104@inbound_originate:1 (Retry 1) > Channel

Hello, For those interested, I have released the NHW v0.3.0-rc30 new version. I continue to fine-tune the nhw_kernel weights.This new version has then more precision and a better visual quality. More at: https://nhwcodec.blogspot.com/ Cheers, Raphael -------------- next part -------------- An HTML attachment was scrubbed... URL:

R Help: I have read a number of tables into R with identical headings and I would now like to make a single table that has all the data appended under this single heading line. for example: t1 <- read.csv("f1",header=TRUE) t2 <- read.csv("f2",header=TRUE) all <- c(t1,t2) #all is now twice as wide as t1 or t2 with the same number of row!!!! #I need to know how

Is there a configuration file in asterisk used for modifying Daylight savings time or is this strictly being dictated by linux? ________________________________ Mark Galley * Systems Administrator acgroup <http://www.bflo.com/> Aurora Consulting Group, Inc. * 7625 Seneca Street * East Aurora, NY 14052 Phone 716.655.9000 X118 * Cell 716.238.3535 * Fax 716.655.4957 * www.BFLO.com

Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip----[asterisk]----E&M----PSTN. As endpoint I had tested another asterisk box (with a FXS),

Please see the attached file and let me know if you need any more information. /Sam Sam Sexton <mailto:sam.sexton@reuters.com> Reuters Coventry Automated Dealing Technologies Phone: +44 24 7625 6562 Fax: +44 24 7655 5203 --------------------------------------------------------------- - Visit our Internet site at http://www.reuters.com Get closer to the

Is rsync a good way to make a valid copy of offline database files (TSM, in this case) from one xfs filesystem to another? If so, which options should I use? Once again, the database services would be shut down, so the database would be offline. Bill Dorrian Network Administrator Desk: 904-273-7625 Cell: 904-859-9471 " Follow the 2012 FedExCup season at PGATOUR.COM." "The

R-Help, I am trying to do simple plots of the characteristics of cellular phones. I am values that fit along an axis that has many data points around 800 and 1800. I am not interested in the "dead space' between the two clusters of data. I am not interested in a linear axis. I would like to "cut out" the white space between the two pockets of data when plotting. I

R-help, I would like to place nine (3X3) plots per page. I am not properly implement mfrow(3,3) in the script below: jpeg("xyplot.jpg") #names output file my_args <- commandArgs() #sets up to take args from dos batch command mfcol(3,3) #set page for 3X3 TEMPS <- c(-15,25,85)#list of temps VBATS <- c(3,3.6,4.7)#list of Bats BOARDS <-

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only