Re: [asterisk-users] how to set to make a call through a fixed ip on a 2 ips server?

Hi, I don't think there is a way for it inside asterisk but you achieve it by adding static route in Linux routing table and make interface having that IP as default route for the interested IPs traffic. Regards, Faisal From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] SIP Peer Name Variable

When you make a call asterisk always create a channel named as below, CheannelType/PeerName-uniquecode Like SIP/jon-312abf So here jon is the peer name. This can help you to identify a peer as long as A-Leg is active. Regards, Faisal -Original Message- From:

[asterisk-users] Mixmonitor concept's question

[RecordPrompts] exten = ,1,Answer() exten = ,n,NoOp(WelCome to conference section) exten = ,n,Playback(ConfDemoWC) exten = ,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.wav,ab) exten = ,n,Konference(${EXTEN},ADRSV) Hi My basic doubt is that if 1 or

Re: [asterisk-users] how to set to make a call through a fixed ipon a 2 ips server?

thanks, Faisal Hanif! i will try it! 2011-07-04 cnasterisk 发件人： Faisal Hanif 发送时间： 2011-07-04 15:59:34 收件人： 'Asterisk Users Mailing List - Non-Commercial Discussion' 抄送： 主题： Re: [asterisk-users] how to set to make a call through a fixed ipon a 2 ips server? Hi, I don’t think

Re: [asterisk-users] Distributing the incoming calls and the huntgroup

Just to be sure that I am working in the right direction. To do ACD, then I have to configure queue.conf and agent.conf? One more question: if the agent needs to be in the NotReady state, then how this can be acheived? Regards Bilal - FreeBPX calls them Ring Groups,

[asterisk-users] stream rtp from asterisk

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus -- _ -- Bandwidth and

Re: [asterisk-users] stream rtp from asterisk

On 07/04/2011 06:58 AM, Marcus Kvarsell wrote: Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Can you clarify what you mean by streaming? -- Alex Balashov

Re: [asterisk-users] stream rtp from asterisk

Sending the rtp-data to external server. One example which I have not gotten to work is this below: http://oreka.sourceforge.net/ September 02, 2009: Asterisk interception via Xorcom Asterisk patch Added support for recording of Asterisk voice calls (TDM and IP) using Xorcoms Asterisk patch.

Re: [asterisk-users] Mixmonitor concept's question

On Monday, July 04, 2011 05:10:43 AM virendra bhati wrote: [RecordPrompts] exten = ,1,Answer() exten = ,n,NoOp(WelCome to conference section) exten = ,n,Playback(ConfDemoWC) exten = ,n,MixMonitor(tmp/00Record/-${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}.w av,ab) exten =

Re: [asterisk-users] Mixmonitor concept's question

Hi Your suggestion is right if we want different recording for all channels. But my problem is that I want to know if more user call the same conference at different time gape(difference) then mixmonitor will take single asterisk thread for recording or multiple thread for recoding. On Mon,

Re: [asterisk-users] stream rtp from asterisk

On 2011-07-04 15:07, Marcus Kvarsell wrote: Sending the rtp-data to external server. One example which I have not gotten to work is this below: http://oreka.sourceforge.net/ September 02, 2009: Asterisk interception via Xorcom Asterisk patch Added support for recording of Asterisk voice

[asterisk-users] RINGNOANSWER events in queue log

Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595|1309550570.399965|2253|Local/05@from-internal/n|CONNECT|2|1309550593.399966|0

[asterisk-users] Testing Asterisk with media - sipp

I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help: This is the command I send at SIPp server: ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err This is the result I see: Last Error: Aborting call on

Re: [asterisk-users] Testing Asterisk with media - sipp

488 means no mutually acceptable codecs were negotiated between the endpoints. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 4, 2011, at 3:29 PM, Daniel -

[asterisk-users] Agent Login, Logout, Ready, Not Ready from the CTI application

Hi All; We know that agents can login and logout from the phone handset. But if we need the login, logout, ready and not ready to be from an application and to be integrated with the CRM, how to acheive this? Normally in Cisco and AVAYA, they use CTI integration and the CTI client (which is

Re: [asterisk-users] Agent Login, Logout, Ready, Not Ready from the CTI application

You need to use the AMI interface an deal with the events that are give to you. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 4, 2011, at 4:36 PM, bilal ghayyad wrote: Hi All; We know that agents can login and logout from the phone handset. But if we need

Re: [asterisk-users] Testing Asterisk with media - sipp

Thank you Alex, It's running without errors now and I can see the media flowing with 'rtp set debug on' but I can't still hear anything on the Asterisk's peers, any advice? Elder 2011/7/4, Alex Balashov abalas...@evaristesys.com: 488 means no mutually acceptable codecs were negotiated between

[asterisk-users] DTMF between sip trunks and PRIs

Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008.

[asterisk-users] Blind Transfer Connected

Hi all In asterisk if blind transfer failed ,call is not connecting back . For Eg: A make call to B through asterisk,then B transfer the call to C. If C did not answer the call ,A and B Call should connect back.But this is not happening with asterisk(A and B call is disconnecting).