similar to: How to stop the update of astdb?

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with

Hello everybody, I have been googling for hours and also searched on http://www.voip-info.org/wiki-Asterisk, but I still can not find any information for the problem I have. So I hope one of you could help me out. I have actually very little experience in Asterisk and also Linux. But by following installation guide, luckily I could get asterisk working. That is only with SIP and IAX channels

Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756]

Hello everybody, I have set dtmfmode=auto in my sip.conf, but that does not work and I still got the following message: WARNING[4980]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 According to http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode: auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- Tomislav Parcina tparcina#lama.hr

I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a

All the 'unknown device' means is that your 'lspci' doesn't know what the card is. That's all. Nothing more. --Rob ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aryanto Rachmad Sent: Tuesday, 4 October 2005 7:43 AM To: asterisk-users@lists.digium.com Subject:

Hello, I have Asterisk version: Asterisk SVN-branch-11-r420435 I have the following code: exten => 303,1,NoOp(Dialing ${EXTEN}) ? ? ? ? same => n,NoOp(DBKey = ${DBKey}) ? ? ? ? same => n,DB_DELETE(office/${DBKey}) ? ? ? ? ? same => n,Playback(auth-thankyou) ? ? ? ? same => n,Hangup() And I get the following error: [2014-08-09 18:00:30] WARNING[4338][C-00000067]: pbx.c:4869

Hello I upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10) And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one. In the UPGRADING.txt in Asterisk it says: * The applications DBGet and DBPut have been deprecated in favor of functions. Here is a table of their replacements: DBGet(foo=family/key)

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

Hi all, since a couple of days i have been wanting to upgrade Asterisk, but my system complains. (Se errors below) Since im ripping the cvs version i guessed that it was something temporary. But it seems persistant. Is this only me or is it happening to other people? Anyone who has a workaround? Or naturally even better, a solution... Oh, i almost forgot im running: # uname -a Linux

I have built Asterisk 1.4.4 on my Solaris 10 x86 box: LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib' CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw --without-oss --without-vpb --prefix=/opt/asterisk-1.4 The build and install go fine but the asterisk executable reproducibly dumps core with a segmentation violation. If I start it as: asterisk -gc and

hello, I'm getting a lot of messages in the log file like : May 2 17:12:07 NOTICE[15877] cdr.c: CDR on channel ' Local/1913@from-internal-a363,2' not posted May 2 17:12:07 NOTICE[15877] cdr.c: CDR on channel ' Local/1913@from-internal-a363,2' lacks end May 2 17:12:07 WARNING[15901] func_db.c: DB requires an argument, DB(<family>/<key>) May 2 17:12:07

> Message: 1 > Date: Tue, 15 May 2007 23:01:24 -0400 > From: Frank Tarczynski <ftarz at mindspring.com> > Subject: [asterisk-users] Asterisk 1.4.4 reproducibly dumps core on > Solaris 10 > To: asterisk-users at lists.digium.com > Message-ID: <464A7404.5000706 at mindspring.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I have

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.

Problem: *Call* *waiting* comes in, I press flash to answer it, and the first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP. System: Analog stations and trunks running on a pair of TDM400's. It does NOT have * call* *waiting* from the phone company, and I have enabled it in all my conf files. The trunks are set to FXSKS and the stations are FXOKS. I am not using *call*

Hi, On Lenny, I typed "apt-get install db4.5-util " then (as root) : # db4.5_dump /var/lib/asterisk/astdb db4.5_dump: /var/lib/asterisk/astdb: unexpected file type or format db4.5_dump: open: /var/lib/asterisk/astdb: Invalid argument # file /var/lib/asterisk/astdb /var/lib/asterisk/astdb: Berkeley DB 1.85/1.86 (Btree, version 3, native byte-order) Is db4.5_dump appropriate to dump an

Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100127/b13d6a62/attachment.htm