Re: [asterisk-users] Queue - agent auto-answer
We do something similar to this by logging a Local channel (eg: Local/1234@AgentContext) into the queue that passes each call through a few lines of dialplan code before going to the SIP extension. Jordan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: 27 January 2011 16:46 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Queue - agent auto-answer Hi, Is there any way to have queue member interface answer automatically? Basically when agentA is called, his phone picks up with no intervention from his part? (assuming of course he's available and not on the phone, and not paused). I already manage this with the Page application (using exten => s,n,SIPAddHeader(Alert-Info: Ring Answer)) and Polycom phones. But how do I do this for calls that are handled by the Queue application? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic agent showing as "Invalid"
Hi, I have written some very simple dialplan logic for our call centre agent system so that when we log an agent into the queue they login as something like: Local/4...@roamingagent/n We have the occasional problem whereby Asterisk sees an agent as Invalid: Local/4...@roamingagent/n with penalty 10 (dynamic) (Invalid) has taken no calls yet Can anyone shed any light on what asterisk would consider Invalid? If I restart asterisk the problem goes away for some time and when it reoccurs it isn't always the same agent. All other agents using the methods to login/out are working fine. Just logging this agent out and back in again doesn't correct the problem. The RoamingAgent code just looks up the SIP extension for any given agent from the asterisk database and sends the call there. We're using Asterisk 1.6.2.0. Thanks Jordan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] State of 64 bits applications in Asterisk
I've used FFA briefly but successfully on Asterisk 1.6.2 x64. Jordan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: 05 March 2010 17:00 To: Asterisk-Users Subject: [asterisk-users] State of 64 bits applications in Asterisk Hi, what is the state at this time for 64bits applications and compatibility with 1.6.2 Mainly speaking about FFA, SFA, G729. Thanks for any information -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom VVX1500 video working yet?
We use Asterisk 1.6.2 with the latest Polycom firmwares on the VVXs with no problem. They key is the new "bootblock" polycom released a little while back. If you download the new BootBlock, BootROM and SIP Firmware from http://www.polycom.eu/support/voice/business_media_phones/vvx1500.html it works well with 1.6.2. I've not tried with anything lower than 1.6.2 though. Jordan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies Sent: 17 February 2010 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom VVX1500 video working yet? On 17 February 2010 16:56, asterisk wrote: > Can anyone tell if asterisk and Polycom VVX1500 work with video yet? > If so what version? Is there a patch? > > Thank you! > > Doug > According to my experimentation, Polycom VVX1500 phones work with "all" versions of Asterisk as far back as 1.2.30, and possibly older. This is with the earliest of the Polycom firmware to support this device. The problem (cause not known) is that Polycom VVX1500 phones only talk to other Polycom VVX1500 phones, making them essentially useless. Perhaps Polycom have fixed this in newer firmware versions - I've not seen such a fix in their changelogs. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and outlook
I've never used it but... http://www.snapanumber.com/ Looks ok feature-wise - plus there's a free version to take for a test drive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Soderblom Sent: 18 December 2006 14:46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and outlook Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi list. Has anyone used any commercial or open source application to integrate Asterisk into MS Outlook 2003 which can be used to place calls directly to contacts from Outlook? And if so how well does it work? Thanks, Richard Best Regards Richard Soderblom Network Configurations Cell: E-Mail: [EMAIL PROTECTED] Number of Attachments: 0 This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dual Wan Router with Failover
Dean, A small Linux box will make a very effective router (and firewall if required) and give load balancing/failover capabilities. I've done it in the past (many moons ago!) A link from my bookmarks: http://lartc.org/ - can be a little scary depending on your knowledge of ip routing and linux but there are plenty of examples to help! Jordan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: 14 November 2006 15:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Dual Wan Router with Failover Hi Jason, I was looking for an external solution outside of my asterisk box so that I can load balance my other website/email traffic as well. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jason > Sent: Tuesday, 14 November 2006 11:00 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Dual Wan Router with Failover > > If you don't mind using linux, linux can do some fairly intense load > balancing all built in. Check out the Linux Virtual Server project. As > for WAN failover, if you again don't mind using linux, you can script a > simple ping to the internet (I would ping at least 3 hosts) and if that > fails, fail to your second ISP. You can also do some crazy fun stuff > with linux advance routing and bonding. > > Jason > The place where you made your stand never mattered, only that you were > there... and still on your feet > > > > Dean Collins wrote: > > > > Are you looking for load balancing or failover. > > > > > > > > Also is there a cheaper way of implementing load balancing than $845 > > appliance? > > > > > > > > > > > > Cheers, > > > > > > > > Dean > > > > > > > > > > > > *From:* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *On Behalf Of *Todd- > > Asterisk > > *Sent:* Tuesday, 14 November 2006 9:26 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* Re: [asterisk-users] Dual Wan Router with Failover > > > > > > > > I've been looking for this as well.. I need to support up to 20 VOIP > > phones over Internet as the Asterisk server is off-site. We'll have > > multiple cable modems or DSL routers. > > > > > > > > I found this device which looks promising - does anyone have any > > experience with this? > > > > http://www.peplink.com/productsLoader.php?productName=balance > > > > > > > > Todd > > > > > > > > On Nov 13, 2006, at 8:49 PM, Dovid B wrote: > > > > > > > > Hi List, > > > > Does anyone know of a good dual wan router that can handle SIP well > > and can failover between connections if there is a SIP issue on one of > > the lines (meaning there still is a connection however there isnt > > enough bandwith or sip packets arent going thru etc.) ? > > > > > > > > Thanks. > > > > > > > > Dovid > > > > > > > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Problems
Hi,
I've got a setup whereby calls come into the asterisk server (1.2.7.1)
over a IAX2 trunk and into a dialplan that launches a php AGI script:
[live-full]
exten => _X.,1,Set(TIMEOUT(absolute)=0)
exten => _X.,2,NoOp(${EXTEN})
exten => _X.,3,DEADAGI(live-full.php)
exten => _X.,4,Wait,2
exten => _X.,5,Hangup
The script is using phpagi-2 from http://phpagi.sourceforge.net/ and
works flawlessly in all but one aspect which I believe is related to
asterisk rather than the script itself.
As the script is launched using DEADAGI I expect it to carry on after
the channel has been hungup (to save the results of the user input in
this case) which works unless the users are leaving a voice message at
the time. The script uses "record_file" and records ok if the user ends
the call with a keypress (#) but if the user hangs up once they have
finished leaving their message the script exits immediately rather than
carrying on:
Nov 2 11:45:57 VERBOSE[24262] logger.c: AGI Rx << RECORD FILE
/ivr/recordedtemp/1162467957 wav "#" 12 0 BEEP s=5 Nov 2 11:45:57
DEBUG[24262] channel.c: Set channel IAX2/AQL IAX-1 to read format slin
Nov 2 11:45:57 DEBUG[24262] channel.c: Set channel IAX2/AQL IAX-1 to
write format ulaw
Nov 2 11:45:57 VERBOSE[24262] logger.c: -- Playing 'beep' (language
'en')
Nov 2 11:45:58 DEBUG[24262] channel.c: Set channel IAX2/AQL IAX-1 to
write format ulaw Nov 2 11:46:01 DEBUG[3658] chan_iax2.c: Immediately
destroying 1, having received hangup Nov 2 11:46:01 VERBOSE[24262]
logger.c: AGI Tx >> 200 result=0 (hangup) endpos=22560 Nov 2 11:46:01
DEBUG[24262] pbx.c: Spawn extension (live-full,70,3) exited non-zero
on 'IAX2/AQL IAX-1'
Does anyone know why the channel is closing down immediately rather than
waiting for the script to exit?
Thanks
Jordan
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RE: [asterisk-users] Asterisk web interface is not parsing the PHPpages
Possibly a silly question, but do you have php installed and configured in apache? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alok MohapatraSent: 31 October 2006 15:45To: asterisk-users@lists.digium.comSubject: [asterisk-users] Asterisk web interface is not parsing the PHPpages Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 2.2. Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) . Thanks and Regards Alok Mohapatra ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sound (or lack of it) problems
I've just installed Asterisk using TrixBox 1.1 (previously has 1.0 installed and working). All my sip trunks and iax trunks connect and can receive calls (there are no phones connected to Asterisk - it's just used for incoming automated services), but the problem is that the line is silent. The Asterisk logs go into the dialplan and into the agi script but I get no sound (I know the scripts are ok). I can even trigger events in the scripts using the relevant buttons on my phone. The problem effects both the sip and iax trunks and I've opened the firewall right up to eliminate that. I'm at a loss as to what can be causing it - Asterisk doesn't seem to error anywhere and I've run "alsaunmute" but still nothing. Any suggestions would be very welcome! I'm getting fairly desperate at this point. Thanks Jordan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
