similar to: Monitoring a call in an Call Center Environment

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

With the deafening silence from my previous questions, I feel seriously alone in the desire to have ChanSpy available. I want to be able to perform a "ZapBarge" on an Agents conversation, and ChanSpy was the answer to my prayers. Bug #2379 (http://bugs.digium.com/bug_view_page.php?bug_id=0002379) was closed "bkw918 10-27-04 17:06 Closed pending new changes in cvs-head."

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.

Does anyone know of any companies where I can interconnect with for SMS? ? .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office

Can someone explain what this error is? -- Got SIP response 500 "Server Internal Error - Invalid CSEQ number" back from 209.xxx.xxx.xxx How do I fix this? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

I created some scripts to logrotate. I am having a problem. After I do it, I am sending kill -HUP to the process its not using the newly created messages file again. Could someone help me out with how I can rotate asterisk's log's without killing the process? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

I don't completely understand this.. Lemme try it out.. [default] exten => 1112223333,1,Macro(happy-did) [macro-happy-did] exten => s,1,Goto(${MACRO_EXTEN},1) exten => _XXXXXXXXXX,1,NoOp(Normal "s" exten stuff here) So when this is ran it will cut the cdr and the s will show the actual DID not the s correct? But then the NoOp would be something like: ....

I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried looking on VOIP-info.org's ChanSpy page (http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to the link regarding bug 3836 (http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I downloaded the attachments and tried to use the patch and compile the source. However, it seems that

I have packet8 and I have spent many hours on the phone with them. If someone has found away around there DTA configuration I would like to know so I can bring it in house to my * box. But as far as your question is concerned. No. Not that I know of. They wouldn't give me any information about the configs. .o-------------------------------------------------------o. Brian Fertig Network

I have delete=yes and attach=yes. But my messages are not getting deleted after they're sent. I'm running asterisk as root so it can't be a permission issue. Any ideas?

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

Except that mixmonitor still has a bug in it. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -sound quality-critical! Matt Roth

When I setup Queuing I wasn't to give the user the ability to drop out and leave a voicemail. ok to accomplish this I understand I have to set the context in the queues.conf file. Now I have done this but when I go to invoke the voicemail function so they don't have to wait in queue it doesn't work. It only seems to work when it tried to dial one of the agents. Can someone give

Thanks for that. I just got rid of packet 8 and went with 100% asterisk in my house. But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But would like to have an extra FXS laying around just in case.. .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From:

Can someone tell me how to clear hung SIP channels in asterisk without restarting? Currently I have 62 channels and only show 10 in use.. this is some of the sip show channels output.. xxx.xxx.xxx.xxx 00xxx24xxx 04240xxxxxx 00103/00001 UNKN (d) ?How can I remove these? from * without rebooting? ? .o-------------------------------------------------------o. Brian Fertig Network

Does anyone know if this has been implemented? I have been around the sites and haven't really found much. I know there was an old patch that would make it work but it doesn't do anything but break the application now. ? ? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office