similar to: How to check what codec translations are in use in a call?

hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info mailbox=1234@default disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus

This evening I posted a new set of Digium G.729 codec modules to our FTP server and web site, for Linux x86 and x86-64 processors. They were built using GCC 4.0.1, and they now report the processor they were optimized for when they are loaded. The previous x86-64 module required a non-standard Asterisk binary configuration, so this was corrected. In addition, there was only a generic version

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

Greetings everyone, I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy allows you to communicate with multiple Asterisk boxes from a single point of contact using a variety of I/O formats, now including support for XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. Astmanproxy is

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

] Hey ho, A few days ago we released the linux version of the phone, today we are very happy to have the mac version ready for a little field test. Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php At the same time, we also put a newer version of the windows and linux versions online. Let us know how you feel about it, a more mac look (brushed metal) is coming.

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind

Hi group. I have huge problem. My pickup exten #8 isn't working. This is what I have configured. pbx*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 #8 In sip.conf I have callgroup=2 pickupgroup=2 For called party and same for person that is trying to pick up the call. The person that is trying

Hi, I have several queues configured at my call center for different support levels. Today, something weird happened: - A client called queue 1 and was answered by an agent - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf - The user transferred the client to another Queue, by using the second channel and the XFer key of her

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=

i have quake v1.06 installed and running fine under windows. however, running it in wine gives a bunch of errors. see below: prophet% wine --winver win98 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD. Try --winver nt40 or win31 ! prophet% wine --winver nt40 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD.

Hi all! I'm experimenting a strange problem in my Asterisk PBX: I've got an Asterisk pbx in the office: I dial an external number; the dialled number answers me correctly, but as soon as I press the '*' button (i.e. to navigate through the menus or to enter a password) my Asterisk box put me on hold. (CLI transcription follows: -- Executing

according to the wiki, I should be able to do this: exten => _9./3003,1,Set(CALLERID(number)=2814444443) exten => _9./3004,n,Set(CALLERID(number)=2814444444) exten => _9./3005,n,Set(CALLERID(number)=2814444445) exten => _9./3006,n,Set(CALLERID(number)=2814444446) exten => _9.,n,Dial(SIP/${EXTEN:1}@mycarrier,30,wt) and have the correct calleridnum's set for each extension based

Google has given me too many responses, so I'll ask the list: Where in the Asterisk rtp source code can I find the default millisecond frame size? I've looked around for obvious pointers, but it's not clear. I'd like to "force" my Asterisk server to use a certain frame size all the time. (Of course, ideally I'd like to prefer or even force that frame size in a

Asterisk 1.2.5 is now available for download on the ftp. See the ChangeLog for details about what has changed. ftp://ftp.digium.com/pub/telephony/asterisk/ As mentioned in the release announcement for Zaptel 1.2.4, our releases now contain some extra files. The Asterisk release is available as asterisk-1.2.5.tar.gz. However, there is also a patch against the previous release as an option for a

Asterisk 1.2.5 is now available for download on the ftp. See the ChangeLog for details about what has changed. ftp://ftp.digium.com/pub/telephony/asterisk/ As mentioned in the release announcement for Zaptel 1.2.4, our releases now contain some extra files. The Asterisk release is available as asterisk-1.2.5.tar.gz. However, there is also a patch against the previous release as an option for a

asterisk-dev-bounces@lists.digium.com wrote: > I have a DSP based system that is working on a four port FXS system > using a 200MHz arm processor. Well.. since we are talking about this topic I owe you guys notes of my experience with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch etc.). We made a Linux distro and compacted it into 32MB flash. Installed asterisk and

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

Hi, I would like to change the sequence of messages that was playbacked when I call to the voicemail extension, VoicemailMain. Where can I changed it, can't find the definition for VoicemailMain in extension.conf and voicemail.conf __________________________________________________ Do You Yahoo!? Download the latest ringtones, games, and more! http://sg.mobile.yahoo.com