similar to: Using RedirectAction with queues

Hi, I have a problem dialing a SIP phone which is logged in as different Astesrik machine from the one I am working with. I want to call a phone in Another astersik machine in , if it answers, calling a SiP phone registered in my ASterisk: My dialplan is: [mariaSIP] exten => _1.,1,Wait(1) exten => _1.,2,Dial(SIP/6021@192.168.0.51:5038,20) exten => _1.,3,HangUp() exten =>

I anyone has any hesitations in upgrading their 405P (or 410P) to V2 of the firmware, read below; I installed one today (turnaround time around 2 weeks to Australia, inc. economy freight in both directions... impressive!) and have noticed immediate, significant improvements. Audio levels are better (have set tx and rx gains back to 0.0) and missed frames have gone (popping, clicking,

Hello We have discovered a problem with DTMF on Asterisk. We have a setup with a T1 from PSTN going into an Asterisk box, and then out again on T1 and into a normal PBX (EADS) We use it to record all calls going to/from the PBX. The problem is that when we record the calls (with MONITOR command), DTMF tones gets obscured, and is not understood in the other end, if we dont Monitor, there are no

Hi! I am in the process of upgrading our 1.2 servers to 1.6. We have a lot of realtime extensions with app=Macro and appdata=stdexten|080512|SIP/080512 But this does not work in 1.6. It is expecting , and not | as the argument seperator. If I change the | to , then it does not work in 1.2. Is there any backwards compatible switch you can enable in 1.6, so it accepts | as a argument seperator

Hi! We have some problems with calls that get disconnected in the middle of a call. We are using Asterisk 1.2.1 with a TE410P (2.gen firmware). When the call is disconnected Asterisk writes this to the log: Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy, avgtone: 300, avgsilence 2090 Jan 9 14:56:17 DEBUG[4404] dsp.c: Requesting Hangup because the busy tone was detected on

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hello! If I have more than a hundred analog telephones (analog lines) that need to be connected to Asterisk PBX, what kind of hardware do I need, and where can I buy it? Thanks in advance!

Hi, I have some interesting problem with htb , I set up root class and sub-classess: $TC qdisc add dev eth0 root handle 1: htb $TC class add dev eth0 parent 1: classid 1:1 htb rate 1990kbit ceil 2000kbit $TC class add dev eth0 parent 1:1 classid 1:10 htb rate 190kbit ceil 200kbit $TC class add dev eth0 parent 1:1 classid 1:11 htb rate 1400kbit ceil 1600kbit $TC class add dev eth0 parent 1:1

Hello! Does TE110P (a 32-bit PCI) fit into PCI Express x8 slot? I'm thinking of buying a Sun X2100 and it has a PCI Express x8 slot. Or perhaps, does Digium produce PCI Express E1 cards? Thanks in advance!

Hello, We have a webrick server running our nice app, and an apache server being used to serve the rest of the site and act as a proxy for the webrick app. <code> <IfModule mod_proxy.c> ProxyRequests Off <Proxy *> Order deny,allow Allow from all </Proxy> ProxyPass /appname http://server.com:3000 ProxyPassReverse /appname

The problem with the extra ptime descriptions in the SDP has been fixed in Asterisk (see http://lists.digium.com/pipermail/svn-commits/2006-October/017694.html). I've got the latest version of the 1.4 branch from SVN and have verified that the codec negotiation is working again. If you don't want to try the latest SVN version, then you'll have to restrict the phones to a single codec

Hi! We have a SIP server with a TE410P card with asterisk version Asterisk CVS-D2005.02.12.14.37.11-04/13/05-16:14:03. Sometime the calls get disconnected with now reason and the users get a busy signal. The log file show this for one of the calls that got disconnected: Aug 31 22:51:53 VERBOSE[3911]: -- Accepting call from '46362302' to '36917474' on channel 0/5, span 1 Aug

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

Adding gluster-users. On Wed, Jan 31, 2018 at 3:55 PM, Misak Khachatryan <kmisak at gmail.com> wrote: > Hi, > > here is the output from virt3 - problematic host: > > [root at virt3 ~]# gluster volume status > Status of volume: data > Gluster process TCP Port RDMA Port Online > Pid >

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming

I would like to setup my Asterisk server to process an incoming SIP URI and redirect all requests to a specific context. Example: (1) using a sip phone I'd like to be able to call: sip:somedomain.com *or* sip:someone@somedomain.com (2) i'd like my asterisk server to answer the call and route it to the context=in-from-sipclient which would play thru some DP actions Can anyone give