similar to: Asterisk 1.2.5 Released

Displaying 20 results from an estimated 6000 matches similar to: "Asterisk 1.2.5 Released"

2005 Oct 14
5
sip accounts
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
2005 Sep 05
9
Asterisk Follow ME
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
2006 Mar 22
2
Pickupexten not working
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
2005 Oct 07
3
Digium G.729 codec modules updated
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
2006 Apr 07
2
Announcing Astmanproxy 1.20
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
2006 Mar 02
3
Native music on hold - Error
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
2006 Jan 20
5
iDEFISK (mac iax2 softphone) release
] 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.
2005 Oct 05
3
SIP Attended Transfer using REFER and Replaces: headers
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
2006 Feb 27
5
res_features pickupexten
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
2006 Apr 05
3
queue issue
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
2001 Oct 19
2
wine 20010824 and quake
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.
2006 Jan 31
5
Queue() with timeout=0
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=
2006 Mar 17
3
Numbered Voicemails when you still delete them.
Hello, I am using Asterisk 1.2.5 and the voice mail application. I don't actually want to keep the voicemails on the server so I am emailing them. However if I have the line in my voicemail.conf 1234 => 4242,Temp_User,voicemail,,delete=1 Then my voicemails are all sent with the same subject line and Message id eg: [PBX]: New message 1 in mailbox 1234 Is there anyway, to delete a
2005 Oct 10
2
DTMF Question (misunderstood '*' button)
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
2005 Jul 01
3
pattern matching based on callerid, not working
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
2004 Sep 19
1
RE: [Asterisk-Dev] Hardware details for the Digium TDM400P
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
2004 Sep 26
2
Asterisk <-> WellGate 3502a : ulaw/alaw only?
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
2004 Oct 01
1
How to configure the voicemail message playback sequence
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
2004 Nov 24
1
Re: Asterisk timer for Freebsd
Hello, I'm just wondering what is the situation today, 24 Nov 2004, regarding asterisk timer for freebsd. I would like to know if there is any way to run Meetme on Freebsd or if there is anybody currently working on it. Cheers, Victor. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Jan 10
1
IAX2 keep alive?
In a setup I've made i have a problem in the two way origination of the call. Asterisk 1 <==> Public internet <==> NAT <==> Asterisk 2 note the two way pointing arrows. When you call from the *2 to the *1, the call passes, but if you try to call from the *1 to the *2, it will not connect, unless you call from *1 to *2, and inmediately after hang-up, you call in the other