[asterisk-users] Question regarding call queue and penalty's
Hey All - I've got an interesting problem, here is what I'm trying to accomplish: Six agents, two queues, three skill levels Queue A (queue B is the same) - Level 1 -- Agent 1 -- Agent 2 - Level 2 -- Agent 3 -- Agent 4 - Level 3 -- Agent 5 -- Agent 6 I'd like a call to come in to Queue A, ring agent 1 and agent 2 for X seconds, if agent 1 and agent 2 ignore the call, try level 2, ring level 2 (agents 3 and 4) for X seconds, if agent 3 and agent 4 ignore the call, move to level 3. I'm using dynamic agents via the Add Queue Member method (SIP/EXTENSION). I can accomplish this by adding autopause=yes, but my agents who ignore the call are then paused and won't receive the next call (level 1). I need to accomplish this using a queue to get reports. Using 1.4 Any thoughts? -Chris ___ -- 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] Question regarding call queue and penalty's
Ahh - so use three queues and not one queue with three penalties? On Thu, Mar 19, 2009 at 4:04 PM, Danny Nicholas da...@debsinc.com wrote: Wouldn’t this work? Exten = s,1,Queue(level1,20) Exten = s,n,Queue(level2,20) Exten = s,n,Queue(level3,20) Exten = s,n,voicemail ; nobody answered -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Christopher Aloi *Sent:* Thursday, March 19, 2009 2:56 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Question regarding call queue and penalty's Hey All - I've got an interesting problem, here is what I'm trying to accomplish: Six agents, two queues, three skill levels Queue A (queue B is the same) - Level 1 -- Agent 1 -- Agent 2 - Level 2 -- Agent 3 -- Agent 4 - Level 3 -- Agent 5 -- Agent 6 I'd like a call to come in to Queue A, ring agent 1 and agent 2 for X seconds, if agent 1 and agent 2 ignore the call, try level 2, ring level 2 (agents 3 and 4) for X seconds, if agent 3 and agent 4 ignore the call, move to level 3. I'm using dynamic agents via the Add Queue Member method (SIP/EXTENSION). I can accomplish this by adding autopause=yes, but my agents who ignore the call are then paused and won't receive the next call (level 1). I need to accomplish this using a queue to get reports. Using 1.4 Any thoughts? -Chris ___ -- 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
[asterisk-users] Question regarding custom announcements in queues.conf
Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! ___ -- 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] Question regarding custom announcements in queues.conf
Here's the version - Asterisk SVN-branch-1.4-r143404 Just static queues. Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir for these queue announce files? So my custom file should live in that dir right? Thanks for the help :) On Tue, Feb 17, 2009 at 1:05 PM, Mark Michelson mmichel...@digium.comwrote: Mark Michelson wrote: Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto:7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson Hmm, my realtime question is a bit silly since you provided config for a static queue with a static member in it. My question about the version is still relevant, though. Mark Michelson ___ -- 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] Question regarding custom announcements in queues.conf
Yah - Found my problem, I can't spell - periodic-*annouce* = SD-PLS-HOLD periodic-announce-frequency=10 : ) On Tue, Feb 17, 2009 at 1:19 PM, Christopher Aloi chris.a...@gmail.comwrote: Here's the version - Asterisk SVN-branch-1.4-r143404 Just static queues. Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir for these queue announce files? So my custom file should live in that dir right? Thanks for the help :) On Tue, Feb 17, 2009 at 1:05 PM, Mark Michelson mmichel...@digium.comwrote: Mark Michelson wrote: Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto: 7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson Hmm, my realtime question is a bit silly since you provided config for a static queue with a static member in it. My question about the version is still relevant, though. Mark Michelson ___ -- 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] Debug meetme
Hello, I'm not 100% sure if it's the same on 1.2 as I'm on 1.4 now, but when I need to debug DTMF I add the following: full = notice,warning,error,debug,verbose,dtmf Then do a logger reload from the console You should then see the following if you do a logger show channels in the console: [HCCAPP1 0.01]--logger show channels Channel Type StatusConfiguration --- --- /var/log/asterisk/full.HCCAPP1.usad File Enabled- Debug DTMF Verbose Warning Notice Error Console Enabled- Notice Error [HCCAPP1 0.01]-- Make sure you see DTMF listed in your logger channel On 6/5/07, Adrian Marsh [EMAIL PROTECTED] wrote: Thanks Chris, I tried: debug = debug,dtmf console = notice,warning,error ;console = notice,warning,error,debug messages = notice,warning,error ;full = notice,warning,error,debug,verbose And restarted the logger, but I don't see any DTMF output in the debug log file when I call into meetme Version 1.2.18 of Asterisk. Adrian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Aloi Sent: 05 June 2007 02:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Debug meetme Add a comma dtmf to your debug line in logger.conf then do a logger reload. That will get you DTMF. -Chris On 6/4/07, Adrian Marsh [EMAIL PROTECTED] wrote: Hi, I'm having complaints from some users about calls into dynamic meetme sessions failing. I'm guessing that they are dialling the wrong DTMF keys, OR that DTMF is hearing the digits entered wrong (or not hearing some). I've put debug = debug into logging.conf, and searched through the file, but I'm not sure how to debug. EG, Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'USER ABC 2060' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2060' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2098' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'from-sip' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'SIP/460-b7310bf0' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'MeetMe' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '|DsM' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:31:51' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:31:51' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:32:33' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '42' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '42' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'ANSWERED' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'DOCUMENTATION' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '1180704711.1969' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Where would it show what DTMF they entered? Cheers, Adrian ___ --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 -- @Christopher Aloi ___ --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 -- @Christopher Aloi ___ --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] Debug meetme
Add a comma dtmf to your debug line in logger.conf then do a logger reload. That will get you DTMF. -Chris On 6/4/07, Adrian Marsh [EMAIL PROTECTED] wrote: Hi, I'm having complaints from some users about calls into dynamic meetme sessions failing. I'm guessing that they are dialling the wrong DTMF keys, OR that DTMF is hearing the digits entered wrong (or not hearing some). I've put debug = debug into logging.conf, and searched through the file, but I'm not sure how to debug. EG, Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'USER ABC 2060' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2060' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2098' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'from-sip' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'SIP/460-b7310bf0' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'MeetMe' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '|DsM' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:31:51' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:31:51' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:32:33' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '42' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '42' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'ANSWERED' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'DOCUMENTATION' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '1180704711.1969' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Where would it show what DTMF they entered? Cheers, Adrian ___ --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 -- @Christopher Aloi ___ --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] Asterisk and iBasis
It should work the way you outlined; if your running through a firewall just make sure you allow traffic to/from both of the provider's IP addresses. On 5/19/07, Yossi Ben Hagai [EMAIL PROTECTED] wrote: iBasis, like many providers uses a softswitch in which separate elements handle the signaling (SIP/H.323) and media gateways handle the media (RTP). when you send a call with the Dial command you state iBasis signaling address and the Asterisk sets it's own media IP/Port in the SDP. when iBasis send back a response it states it's own media IP/Port in the SDP (which can be different from the signaling IP) so the asterisk will know where to send the RTP packets. so in terms of asterisk configuration you don't need to do anything different from what you would usually do.. On 5/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks for the prompt response, but would you care to explain this a bit further? It could be due to my ignorance, and if so, I apologize. But, how can we send the INVITE to one IP and then the media to a different one? Do we just simply send the call to the INVITE IP using the Dial command and that's it? Thanks On Sat, May 19, 2007 12:29 pm, Yossi Ben Hagai [EMAIL PROTECTED] said: Asterisk supports it and the good news is that you don't have to do anything for it to work. On 5/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, We are currently trying to setup Asterisk with iBasis. One question/problem we have is that Ibasis has told us to send the INVITEs to one IP address and all media to a different IP address. How can we do that in Asterisk? Thanks ___ --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 -- -- Christopher T Aloi -- ___ --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] getting call status using Manager API
Hello - On 5/16/07, mitcheloc [EMAIL PROTECTED] wrote: Matthew, If you hold open the connection to the manager api after initiating your call you should see packets returning from Asterisk which you can parse to get the call state. I'm not sure if there is a way to do it later after the fact. You can execute commands like Show Channels through manager though. If these are two SIP calls; knowing the channel name to parse for is going to be a bit of a challenge. Is it possible to do something like the following? - manager sets channel to SIP/foo-UID - manager queries status of channel SIP/foo-UID Since there is a SIP trunk on both sides of this call; no real extensions are involved. If this was a Zap bridge I think getting the channel would be easier. I'm answering a question with more questions -Chris On 5/16/07, Matthew M. Boedicker [EMAIL PROTECTED] wrote: I am originating a call using the Originate action in the Manager API. It calls one party, then when they answer does the Dial application and calls another party and connects the two. Is there a way using the Manager API to check back later on the status of this call (is it still up, etc.)? I have found the Status API action, but I don't know how to get what to pass in for the channel parameter. Thanks, Matthew Boedicker ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --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 -- -- Christopher T Aloi -- ___ --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] Aastra phones?
Hello - I have a few 480i's and just yesterday received two 57iCT's. Thus far they are pretty nice, the large display is clean, and sound quality is good. The setup / configs are just like the 480's -Chris On 5/8/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Sorry for being a little off topic, but I'mconsidering a few new phones for my Asterisk installation. I have a mix of Polycom 500/600s and an Aastra 480i CT. I'm considering adding a couple of Aastra 57i or 57i CT. Does anyone here have experience with the 480i CT and the newer 57i CT? I'm curious as to the real differences. Thanks, Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] [EMAIL PROTECTED] FWD 54245 ___ --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 -- -- Christopher T Aloi -- ___ --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] Re: Virtual IP Adresses and SIP requests failing...
Hello - Well I've been able to find a bit more about my problem. Again - I am not bound to a specific interface (0.0.0.0) When a SIP invite addressed to the .36 address, Asterisk replies FROM the .38 address. Is this the expected behavior? Wouldn't it make sense for Asterisk to reply on FROM the IP it rec'd on? Any thoughts??? Is this a bug? On 5/3/07, Christopher Aloi [EMAIL PROTECTED] wrote: Hey All: Question; when using a virtual IP on an Asterisk server, I am having trouble getting sip user to register to the ViP. They are able to register with the true IP, just not the virtual. It seems Asterisk is rejecting the SIP invite, register, etc (like it's not destined for this server) I've added all the IP's to the domain listing in sip.conf and in the Asterisk console a sip show domains shows both the virtual and the physical IP. Am I missing something? I have Asterisk bound to 0.0.0.0 which should tell it to listen on all IP's, right?? Some Details: ## ifconfig eth1 - inet addr:69.67.250.38 eth1:0 - inet addr: 69.67.250.36 (ViP) ## sip.conf [general] domain=69.67.250.36 domain=69.67.250.38 bindport=5060 port=5060 bindaddr=0.0.0.0 ## sip show domains Our local SIP domains: Context Set by 69.67.250.36 (default) [Configured] 69.67.250.38 (default) [Configured] ## tshark -i eth1 -R sip ## Call to .38 10.818719 66.218.1.47 - 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.818903 69.67.250.38 - 66.218.1.47 SIP Status: 200 OK 10.820676 192.168.0.102 - 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.821626 69.67.250.38 - 192.168.0.102 SIP Status: 200 OK 10.829019 66.218.1.47 - 69.67.250.38 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.830792 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 10.835473 66.218.1.47 - 69.67.250.38 SIP Request: ACK sip:[EMAIL PROTECTED] 10.841651 66.218.1.47 - 69.67.250.38 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED] , with session description 10.841880 69.67.250.38 - 66.218.1.47 SIP Status: 100 Trying 10.847744 69.67.250.38 - 69.67.248.83 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.847874 69.67.250.38 - 66.218.1.47 SIP/SDP Status: 183 Session Progress, with session description 10.848852 69.67.248.83 - 69.67.250.38 SIP Status: 100 Trying 16.724167 69.67.248.83 - 69.67.250.38 SIP/SDP Status: 183 Session Progress, with session description 16.725928 69.67.248.83 - 69.67.250.38 SIP/SDP Status: 200 OK, with session description 16.726053 69.67.250.38 - 69.67.248.83 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 16.726373 69.67.250.38 - 66.218.1.47 SIP/SDP Status: 200 OK, with session description 16.731913 66.218.1.47 - 69.67.250.38 SIP Request: ACK sip:[EMAIL PROTECTED] 19.561514 69.67.248.83 - 69.67.250.38 SIP Request: BYE sip:[EMAIL PROTECTED] 19.561617 69.67.250.38 - 69.67.248.83 SIP Status: 200 OK 19.562158 69.67.250.38 - 66.218.1.47 SIP Request: BYE sip:[EMAIL PROTECTED] :5004;transport=udp 19.565798 66.218.1.47 - 69.67.250.38 SIP Status: 200 OK ## Call to .36 90.821676 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 90.821873 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 91.321664 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 91.822061 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 92.322452 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 92.821931 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 94.323765 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 94.452850 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 94.453240 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 94.822695 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 98.324204 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 98.453399 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 98.822235 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 102.325048 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 102.821775 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 106.325130 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 106.822293 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 110.326101 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 110.587025 66.218.1.47 - 69.67.250.36 SIP Request: CANCEL sip:[EMAIL PROTECTED] 110.587101 69.67.250.38
[asterisk-users] Virtual IP Adresses and SIP requests failing...
Hey All: Question; when using a virtual IP on an Asterisk server, I am having trouble getting sip user to register to the ViP. They are able to register with the true IP, just not the virtual. It seems Asterisk is rejecting the SIP invite, register, etc (like it's not destined for this server) I've added all the IP's to the domain listing in sip.conf and in the Asterisk console a sip show domains shows both the virtual and the physical IP. Am I missing something? I have Asterisk bound to 0.0.0.0 which should tell it to listen on all IP's, right?? Some Details: ## ifconfig eth1 - inet addr:69.67.250.38 eth1:0 - inet addr:69.67.250.36 (ViP) ## sip.conf [general] domain=69.67.250.36 domain=69.67.250.38 bindport=5060 port=5060 bindaddr=0.0.0.0 ## sip show domains Our local SIP domains: Context Set by 69.67.250.36 (default) [Configured] 69.67.250.38 (default) [Configured] ## tshark -i eth1 -R sip ## Call to .38 10.818719 66.218.1.47 - 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.818903 69.67.250.38 - 66.218.1.47 SIP Status: 200 OK 10.820676 192.168.0.102 - 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.821626 69.67.250.38 - 192.168.0.102 SIP Status: 200 OK 10.829019 66.218.1.47 - 69.67.250.38 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.830792 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 10.835473 66.218.1.47 - 69.67.250.38 SIP Request: ACK sip:[EMAIL PROTECTED] 10.841651 66.218.1.47 - 69.67.250.38 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.841880 69.67.250.38 - 66.218.1.47 SIP Status: 100 Trying 10.847744 69.67.250.38 - 69.67.248.83 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.847874 69.67.250.38 - 66.218.1.47 SIP/SDP Status: 183 Session Progress, with session description 10.848852 69.67.248.83 - 69.67.250.38 SIP Status: 100 Trying 16.724167 69.67.248.83 - 69.67.250.38 SIP/SDP Status: 183 Session Progress, with session description 16.725928 69.67.248.83 - 69.67.250.38 SIP/SDP Status: 200 OK, with session description 16.726053 69.67.250.38 - 69.67.248.83 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 16.726373 69.67.250.38 - 66.218.1.47 SIP/SDP Status: 200 OK, with session description 16.731913 66.218.1.47 - 69.67.250.38 SIP Request: ACK sip:[EMAIL PROTECTED] 19.561514 69.67.248.83 - 69.67.250.38 SIP Request: BYE sip:[EMAIL PROTECTED] 19.561617 69.67.250.38 - 69.67.248.83 SIP Status: 200 OK 19.562158 69.67.250.38 - 66.218.1.47 SIP Request: BYE sip:[EMAIL PROTECTED]:5004;transport=udp 19.565798 66.218.1.47 - 69.67.250.38 SIP Status: 200 OK ## Call to .36 90.821676 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 90.821873 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 91.321664 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 91.822061 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 92.322452 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 92.821931 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 94.323765 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 94.452850 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 94.453240 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 94.822695 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 98.324204 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 98.453399 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 98.822235 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 102.325048 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 102.821775 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 106.325130 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 106.822293 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 110.326101 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 110.587025 66.218.1.47 - 69.67.250.36 SIP Request: CANCEL sip:[EMAIL PROTECTED] 110.587101 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 110.587133 69.67.250.38 - 66.218.1.47 SIP Status: 200 OK 111.087270 66.218.1.47 - 69.67.250.36 SIP Request: CANCEL sip:[EMAIL PROTECTED] 111.087332 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 111.087359 69.67.250.38 - 66.218.1.47 SIP Status: 200 OK 111.587718 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 112.087661 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request
Re: [asterisk-users] internal sounds of asterisk / freePBX
Try getting rid of all those macros etc.. so you can see what's going on, something simple like: exten = 500,1,Answer() exten = 500,n,Playback(beep) exten = 500,n,Hangup() Then dial 500 from your soft phone and see what happens. On 4/17/07, EWV2 [EMAIL PROTECTED] wrote: The codecs are correct, so you are having other type of problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --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 -- Carlos Jerónimo ___ --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 -- -- Christopher T Aloi -- ___ --Bandwidth and
[asterisk-users] 1.4.1 - T38 Pass Through - Seeing some odd errors but the fax works.....
Hello List - Here's the setup: Mediatrix 1102 ATA (t38enabled) -- Asterisk 1.4.1 -- IP -- SIP GW -- TDM The T38 call comes up perfect - I see the initial invite, followed by G711, Re-Invite, T38 establishes, Fax Completes, T38 Stops, Call Down. here's the problem - I see the following in my console: [Mar 19 05:09:38] WARNING[4745] chan_sip.c: Can't send 0 type frames with SIP write And don't know what it means :) I also see timing errors: [Mar 19 05:08:46] DEBUG[4745] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=25) sip.conf has t38pt_udptl=yes for all peers and globally. A copy of the debug file during this call is located http://chrisaloi.com/static/DEBUG.txt Anyone have any thoughts? Christopher Aloi ___ --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] How to best manage my dial plans as the continue to grow, and grow, and grow....
Hello List - I've been slowing growing my extensions.conf file and have been wondering how everyone manages their systems. I currently have my main extensions.conf where I reference my sub extensions (for tenants or customers) files using the include statements and define my global variables. Today while watching the asterisk console I noticed a call from a voicemail user bounced into another tenants extensions file using the # key. What i'd like to accomplish is true separation for tenants on a multi-tenant system. I'd like to remove the chance of context hopping etc... How does everyone manage their systems as they continue to grow? Thanks for reading, -- -- Christopher T Aloi -- ___ --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] Thoughts on CPE server...
Hey List - I've been looking into the various options for small form factor customer premise gear, and am wondering what your using and what your reccomendations are. I'd like to drop a unit at the customer premise to handle their internal routing and trunk their outgoing calls back into my datacenter. The CPE unit would be useful if their WAN link drops as internal calls will stay up and if they have a PSTN line the Asterisk server could failover to the FXO trunks. Some of the ports I think would be useful on this piece of gear... - 1 WAN port - 1+ LAN ports - 1+ FXO ports (optional) - Would be nice for faxing - 1 Console - 1 VGA / USB / PS2 It wouldn't need to be a powerhouse as i'm only serving between 10-15 seats. Anyone have any ideas? -- -- Christopher T Aloi -- ___ --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] Question Regarding Visual Park Functionality - Hardware/Software
Hello List! I am hoping someone may be able to assist with the following feature I am looking to implement: I would like to use a visual park function if possible, here's how I see it working - Call comes into caller A - Caller A places the call in orbit (or park) by dialing 700 - A message is then sent to illuminate a line on each of the phones in this office indicating a call is parked - Anyone can retrieve this call by pressing the illuminated line key on their phone I assume this must be accomplished with specific phones, I know I can do the park/retrieve - but not sure how to handle the visual indication of a park. My first thought was to send a ringing to each SIP phone, this won't work because they only want a light to illuminate, no audio. Maybe a phone that supports a quiet mode? I could then send ringing, and a custom ring in the ALER_INFO SIP parameter?? Anyone have any suggestions on the best phone to accomplish this goal? Anyone have any advice on getting this to work? Anyone do this now? Thanks!! -Chris Aloi ___ --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] POS Terminals
Hello List - I've got a question regarding POS terminal transactions (credit card machines, ATM, etc...). Currently we setup customers in the following manner: Customer Location -- Data T1 -- DataCenter - PSTN Termination We are currently using Mediatrix gear for fax transmissions from the customer location, but they don't seem to handle POS modem sales very well. Does anyone have any experience using POS terminals? Is something like an IAXy at the customer prem a good idea? -Thanks for any advice, -- -- Christopher T Aloi -- ___ --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] POS Terminals
Hello - Because this is the equipment some end users have purchased. -Thanks, Chris On 11/16/06, Al Bochter [EMAIL PROTECTED] wrote: Why are you using VOIP for credit cards? You have the Internet look into a bank with a credit card gateway. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=email For new and used security itemshttp://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICEShttp://www.bochterservices.com/?j=platingt=email Christopher Aloi wrote: Hello List - I've got a question regarding POS terminal transactions (credit card machines, ATM, etc...). Currently we setup customers in the following manner: Customer Location -- Data T1 -- DataCenter - PSTN Termination We are currently using Mediatrix gear for fax transmissions from the customer location, but they don't seem to handle POS modem sales very well. Does anyone have any experience using POS terminals? Is something like an IAXy at the customer prem a good idea? -Thanks for any advice, -- -- Christopher T Aloi -- -- ___ --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 -- Inbound (clean). Database: 0649-0, 11/15/2006 - 11/16/2006 11:30:46 AM ___ --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 -- -- Christopher T Aloi -- ___ --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] Mitel 5224 Asterisk Distinctive Ring -- Anyone have it working?
Hello List,
So I have a few MiTel 5224 IP phones running in SIP mode. Per the
phones documentation they honor SIP distinctive ring tones. I am able
to send the correct ALERT_INFO message in an invite from Asterisk to
the phone, but I don't know what ring tone to call. From the reading
I've done the syntax is:
[3155791234]
exten = s,1,Set(_ALERT_INFO=Ring 8)
exten = s,2,Answer()
exten = s,3,Set(CALLERID(name)=FOOBAR ${CALLERIDNUM})
exten = s,4,Dial(SIP/3155791234,,r)
When I place a call to 3155791234 I see the following from Asterisk:
-- Executing Answer(SIP/69.67.248.00-b7b1eb08, ) in new stack
-- Executing Goto(SIP/69.67.248.00-b7b1eb08, 3155791234|s|1)
in new stack
-- Goto (3155791234,s,1)
-- Executing Set(SIP/69.67.248.00-b7b1eb08, _ALERT_INFO=Ring
8) in new stack
-- Executing Answer(SIP/69.67.248.00-b7b1eb08, ) in new stack
-- Executing Set(SIP/69.67.248.00-b7b1eb08,
CALLERID(name)=FOOBAR 311234) in new stack
-- Executing Dial(SIP/69.67.248.00-b7b1eb08,
SIP/3155791234||r) in new stack
And the SIP invite looks like this:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 69.67.250.00:5060;branch=z9hG4bK184f04fd;rport
From: FOOBAR 311234 sip:[EMAIL PROTECTED];tag=as26559e24
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Thu, 09 Nov 2006 23:19:28 GMT
Alert-Info: Ring 8 --- HERE IS THE ALERT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 242
So it seems I have everything setup correctly.
The problem is I don't know how the phone refers to it's ring tones, I
used Ring 8 because the phone uses Ring 1-16 in the user web
interface.
Anyone have any thoughts?
--
--
Christopher T Aloi
--
___
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Re: [asterisk-users] Powering SNOM 200 phones?
Hey Brad - I have a Snom 200 at the office, If I remember correctly the power is about the size of an RJ11 jack; it's a weird connector. I haven't used PoE with the 200 though. Not sure if that helps or not :) -chris On 11/9/06, Brad Templeton [EMAIL PROTECTED] wrote: Ok, not exactly an Asterisk problem, but... I picked up some SNOM 200 phones because SNOM's have been recommended for use with Asterisk and they have line buttons that can subscribe to presence. However, they don't appear to power up when connected to my Negear FS108P, which is an 802.3af Power-over-ethernet capable hub. I am pretty sure these are the SNOM 200b, in that the ethernet connectors are at the back rather than on the bottom, and there doesn't even seem to be a jack for plugging in any other kind of power adapter (and I don't have another one.) Anybody had experience with these phones and powering them? Is it just an icompatability with the Netgear, or do I have 2 dead phones? Would getting a different PoE box be a good idea? (Frys has the airlink for $29 from time to time, which is a great price. Otherwise many older PoE boxes tend to cost more than the modern cheaper phones they might power.) ___ --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 -- -- Christopher T Aloi -- -- -- Christopher T Aloi -- ___ --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] Multiple queue_log files based on queue - is it possible??
Hello List, Question: Has anyone been able to create multiple queue_log files in /var/log/asterisk for multiple queues? We are designing a multi-tenant system and separating the log files would be useful, instead of dropping all queue actions into one file. Is it possible this is a user configurable option I am missing? Cheers, -- -- Christopher T Aloi -- ___ --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] Possible to set max voicemail mesasge limit per user
Hello - Setting up a voicemail application and would like to have a ceiling for the amount of voicemail messages per user. Example - 20 users x 10 messages each = total of 200 messages = XXX disk space. Is it possible to define this? I didn't see it here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf Thanks! -- -- Christopher T Aloi -- ___ --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] Shared NFS or Shared MySQL for redundant secondary server?
Hey List!What are your thoughts on redundancy?? Is it best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one falls out of service the other takes over or is it best to have each node pull from a shared DB?? Cheers!-- --Christopher T Aloi-- ___ --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] Help compiling asterisk-addons on Debian?
Hello All -Running the following:Debian StableAsterisk SVN-branch-1.2-r41069Checked out the following from SVN:asterisk-addons/branches/1.2 When I attempt to compile asterisk-addons I get the following: /usr/src/asterisk-addons$ make/clipasterdev1:/usr/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or directoryres_config_mysql.c:53:19: mysql.h: No such file or directoryres_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directorymake -C format_mp3 allmake[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'/clipIs this looking for MySQL? I do have MySQL installed and running, a bit confused here anyone have any thouhts? -- --Christopher T Aloi-- ___ --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] Help compiling asterisk-addons on Debian?
Thanks for the tip!libmysqlclient12-devGot it doneOn 8/25/06, Rushowr [EMAIL PROTECTED] wrote: Do you have the development libraries installed too? I believe on Debian it's something like libmysqlclient From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Christopher AloiSent: Friday, August 25, 2006 8:36 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Help compiling asterisk-addons on Debian? Hello All -Running the following:Debian StableAsterisk SVN-branch-1.2-r41069Checked out the following from SVN:asterisk-addons/branches/1.2 When I attempt to compile asterisk-addons I get the following: /usr/src/asterisk-addons$ make/clipasterdev1:/usr/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or directoryres_config_mysql.c:53:19: mysql.h: No such file or directoryres_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directorymake -C format_mp3 allmake[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'/clipIs this looking for MySQL? I do have MySQL installed and running, a bit confused here anyone have any thouhts? -- --Christopher T Aloi-- ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- --Christopher T Aloi-- ___ --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] Asterisk Real Time Engine - Fails to Connect to MySQL
Debian StableAsterisk SVN-branch-1.2-r41069Hello List.Okay, tonight I dove into Asterisk Real Time.I have the module compiled and installed and have the following setup:MySQL running with a DB created asterisk *how original right? Username 'astuser' password 'foo'I can locally login to MySQL (as user astuser with pw foo) and select the following:mysql show tables;++| Tables_in_asterisk |++ | sip_friends |++1 row in set (0.00 sec)I have a sip_friend 800 setup in the table.My extconfig looks like this:[settings]sippeers = mysql,asterisk,sip_friends My res_config_mysql.conf looks like this:[general]dbhost = 127.0.0.1dbname = asteriskdbuser = astuserdbpass = foodbport = 3306dbsock = /var/run/mysqld/mysqld.sock And here is what I see in the console; i wasn't able to get any more than this out of the debug file either.Anyone have any thoughts? What is error 2013???Thanks :) !Aug 26 01:13:02 ERROR[6256]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1 (err 2013). Check debug for more info.Aug 26 01:13:02 DEBUG[6256]: res_config_mysql.c:652 mysql_reconnect: MySQL RealTime: Cannot Connect (2013): Lost connection to MySQL server during query -- --Christopher T Aloi-- ___ --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] Re: Asterisk Real Time Engine - Fails to Connect to MySQL
Scratch that email.problem was using 127.0.0.1 - I changed that to localhost and all is well.Can I have the last three hours back now :) On 8/25/06, Christopher Aloi [EMAIL PROTECTED] wrote: Debian StableAsterisk SVN-branch-1.2-r41069Hello List.Okay, tonight I dove into Asterisk Real Time.I have the module compiled and installed and have the following setup:MySQL running with a DB created asterisk *how original right? Username 'astuser' password 'foo'I can locally login to MySQL (as user astuser with pw foo) and select the following:mysql show tables;++| Tables_in_asterisk |++ | sip_friends |++1 row in set (0.00 sec)I have a sip_friend 800 setup in the table.My extconfig looks like this:[settings]sippeers = mysql,asterisk,sip_friends My res_config_mysql.conf looks like this:[general]dbhost = 127.0.0.1dbname = asterisk dbuser = astuserdbpass = foodbport = 3306dbsock = /var/run/mysqld/mysqld.sock And here is what I see in the console; i wasn't able to get any more than this out of the debug file either.Anyone have any thoughts? What is error 2013???Thanks :) !Aug 26 01:13:02 ERROR[6256]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1 (err 2013). Check debug for more info.Aug 26 01:13:02 DEBUG[6256]: res_config_mysql.c:652 mysql_reconnect: MySQL RealTime: Cannot Connect (2013): Lost connection to MySQL server during query -- --Christopher T Aloi-- -- --Christopher T Aloi-- ___ --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] Simple CDR parser to print to webpage
Hello -I'm searching for a simple php or perl script to parse Asterisk's CDR csv into a formatted webpage - anyone have any suggestions?-- --Christopher T Aloi-- ___ --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] Re: SIP Debug to file - Is it possible?
Thanks for the ideas!I've been looking at sip_scenario for parsing SIP signalling into nice ladder diagrams etc.. If I was to call eithet ngrep or tcpdump from my dial plan and pipe it into sip_scenario.pl; anyone have any thoughts on how to kill the ngrep process once the call is complete? This would be for a 'debug' context, something like this[debug]= system(ngrep -q sip | sip_scenario.pl /var/www/foo.trace)= Answer()= Blah = system(killall ngrep) The killall would work if there was only one call right? But if there were two users in my debug context, one at step one and one at step kill - all the ngrep processes would die. Anyone have any thoughts??On 8/22/06, Jan Fousek [EMAIL PROTECTED] wrote: tcpdump -i eth0 -s 1500 -w ./dump #8216;udp port 5060#8242;creates dump of communication on port 5060 (default sip)__ Od: [EMAIL PROTECTED] Komu: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Datum: 22.08.2006 05:18 Předmět: RE: [asterisk-users] Re: SIP Debug to file - Is it possible? ngrep is also good if you only want to see SIP traffic and filter all the lower level stuff. -Original Message- From: Brandon Galbraith [mailto: [EMAIL PROTECTED]] Sent: Mon 8/21/2006 8:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Re: SIP Debug to file - Is it possible? Try Ethereal (I think it's called WireShark now). Does nice decoding of the packet stream to show you what's going on. Supports SIP for sure, notso sure about IAX though. -brandon On 8/21/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Christopher Aloi wrote: Hello List -I'm a big fan of call traces to diagnose a problem; I often use pri set debug file X to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to afile? Anyone have any ngrep scripts that do the trick? tcpdump :) ? ___ --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 -- Brandon Galbraith Email: [EMAIL PROTECTED] AIM: brandong00 Voice: 630.400.6992 A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost___--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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- --Christopher T Aloi-- -- --Christopher T Aloi-- ___ --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] Re: SIP Debug to file - Is it possible?
Hello List -I'm a big fan of call traces to diagnose a problem; I often use pri set debug file X to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to a file? Anyone have any ngrep scripts that do the trick?Thanks!-- --Christopher T Aloi-- -- --Christopher T Aloi-- ___ --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] Anyone use TACI.pl for a click to call app? - Doesn't seem to want to work for me
Hello List -Asterisk Version: Asterisk SVN-branch-1.2-r38420MI'd like to create web GUI for basic internal outbound dialing.I came across TACI which, if I could get it to work, would fit the need. My goal is to provide the user a form with the following:OriginateUsers# _Number they wish to terminate to CallerID they wish to pass Context they wish to terminate through ___I'm sure there are many ways to accomplish this goal, I found this: http://www.azxws.com/asterisk/.The script seems to fail due to a missing priority, I added the extra space as outlined on voip-info, but the call still fails (doesn't even start) Debugging hasn't gotten me too far, Asterisk shows the app authenticate against the manager.conf file, then logoff.Anyone have this working? Anyone have another solution I may have missed?Thanks in advance, -- --Christopher T Aloi-- ___ --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] Looking for carrier grade redundant solution
Thanks for the informative replies.While Enswitch sounds like a a nice solution, I don't believe it's for our shop. I really would like to develop this solution internally.My thoughts thus far are pointing towards a scalable, redundant solution based on stock hardware.
I am thinking of running a matched server pair (1 or 2U) as the application servers (both running a matched Asterisk build) most likely on RHL. The servers will contain redundant hardware (power supplies etc...)
The application servers will be backed up by a matched pair of MySQL servers which will use MySQL replication between each other and will use a floating IP; a heartbeat will beat between the pair and nominate one a master should another fail.
The disk storage will be local or offloaded to an existing netApp filer.From the reading I've done, running SER as a proxy to the two application servers should serve it's purpose well, providing load balancing and monitoring of the application server behind it (should one Asterisk box return a 3XX-5XX I would then route advance to the second application server and take the first offline.
Question - Is running SER by best bet here? Would I be better off running a heartbeat between the two app servers?I am not too worried about SIP registration/expiration's of the SIP users or peers:1) Most of the call center traffic will be terminated out another platform; the calls are being sent to a DID (not a SIP URL).
2) My SIP peers are all within the trusted network core and will not need to maintain status of each other.Here's where it gets tricky:As agents login to a queue or become 'available' Asterisk is maintaining their state; I believe an agents state is maintained across an Asterisk reload. Could an agents state be maintained should the second app server take over for the first?
Anyone attending ClueCon next week??Thanks,_Chris_On 7/28/06, Douglas Garstang
[EMAIL PROTECTED] wrote:
What
about sip registration replication?
What
about SIP subscription replication?
What
about BLF replication?
What
about using DUNDi to replicate applications for redundancy?
How
would you handle different phones ability to failover if they don't do it so
well?
How
would handle the fact that the config files have a hard coded database
IP?
And so
on...
I
don't think anyone has a great solution to date.
-Original Message-From: Stephen Wingfield
[mailto:[EMAIL PROTECTED]]Sent: Friday, July 28, 2006 4:14
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] Looking for carrier grade
redundant solution
Chris,
Heartbeat failover will usually be your best
mixed approach.
As always there is a cost benefit to be
considered.
Where the call absolutely has to stay up then
Fault-Tolerant software and hardware is the only option that works with
Asterisk to date.
If however you wish to keep costs to a mimimum
then possibly an onsite / hosted model where the back up is available
remotely. This modeldepends on set up however.
In all cases I would suggest you take a peruse of
PBXware : http://www.bicomsystems.com/products/online_demo/
which
is our SMB Edition. We will next week launch our Call Center Edition that is
packed with features and functions to assist the running of a dedicated to
running a Call Center efficiently.
Feel free to contact me offline steve {at]
bicomsystems {dot] com and can make more precise suggestions according to
requirement.
Regards
Steve
- Original Message -
From:
Christopher
Aloi
To:
Asterisk Users Mailing List -
Non-Commercial Discussion ;
asterisk-biz@lists.digium.com
Sent: Friday, July 28, 2006 3:44
AM
Subject: [asterisk-users] Looking for
carrier grade redundant solution
Hello List -We are looking add Asterisk to the core
of our voice/data network. Our first application will provide a hosted
call center application for a number of tenants (customers) who will have
between 5-20 agents (seats) answering ingress calls. The calls will
ingress and egress the Asterisk server SIP (all TDM is handled by Sonus
switches). My goal is to design a redundant solution using a
multiple Asterisk servers with an NFS mounted filesystem.I've done
some reading regarding Asterisk redundany, and so far it seems the best
approach is running redundant hardware (power supplies etc), matching
servers (with a heart beat ping between them) and a NFS filer for storage
(hot swapable) connected to each box via gigE. Am I on the right
track? Any other suggestions or resources I might have missed
regarding developing a redundant solution?Thanks for your
time,_Chris_
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Re: [asterisk-users] registration process
What about limiting the simultaneous SIP sessions for each subscriber to one or two?On 7/28/06, Joshua Colp [EMAIL PROTECTED] wrote:- Original Message -From: unplug[mailto: [EMAIL PROTECTED]]To: Asterisk Users Mailing List -Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com]Sent:Sat, 29 Jul 2006 00:51:38 -0300 Subject: Re: [asterisk-users] registrationprocess Thanks! I think I can't restrict the access by IP in my system.As you said, there is no way to prevent it, right? I am using ARA in the system.Can I detect it if prevention is not possible?When it comes down to it - unless you want to go to great extremes, it is not possible. Sorry.Joshua ColpDigium___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Looking for carrier grade redundant solution
Hello List -We are looking add Asterisk to the core of our voice/data network. Our first application will provide a hosted call center application for a number of tenants (customers) who will have between 5-20 agents (seats) answering ingress calls. The calls will ingress and egress the Asterisk server SIP (all TDM is handled by Sonus switches). My goal is to design a redundant solution using a multiple Asterisk servers with an NFS mounted filesystem.I've done some reading regarding Asterisk redundany, and so far it seems the best approach is running redundant hardware (power supplies etc), matching servers (with a heart beat ping between them) and a NFS filer for storage (hot swapable) connected to each box via gigE. Am I on the right track? Any other suggestions or resources I might have missed regarding developing a redundant solution?Thanks for your time,_Chris_ ___ --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] Recommendations for best Voicemail application manager?
Hello List.I am looking to build an Asterisk Voicemail application to serve approx. 100 users.I will be building the Voicemail system using a standard Asterisk install on a stable Debian system.The system will house 100x20mb/each voicemail boxes. On to my question:The Voicemail system will most likely be maintained by a single person at the customer location, most likely an office admin. I wouuld like the office admin to be able to conduct standard moves/add/changes/resets etc.. Any thoughts on the best WWW UI to provide these moves adds and changes?Thanks!!_Chris_ ___ --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] Help with incoming SIP routing
Hello -I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone on the list can assist me.Here's the relevant info: Ingress SIP trunk:IP: 123.45.45.3456DID's XXX-XXX-XX00-XX10sip.conf:[general]useragent=Asteriskport=5060context=defaulttos=lowdelaydisallow=allallow=ulawallow=alaw allow=gsmrtptimeout=300rtpholdtimeout=600// // My thought in this context is I will grab any incoming SIP call from the IP address of my SIP trunk and pass it to my sip-defaul-in context // // in extensions.conf [sip-default-in]type=frienddefaultip=123.45.3456host=123.45.3456nat=noinsecure=verycontext=sip-default-incanreinvite=nodtmfmode=rfc2833// // My thought here is I will grab any incoming SIP call form the IP address of my SIP trunk that matches XXX-XXX-XX00 and pass it to my XXX-XXX-XX00 context in extensions.conf[00]type=frienddefaultip=69.67.248.51host=69.67.248.51fromuser=00nat=nocontext=00 insecure=veryAnd a look at extesions.conf:// // My thought is here I will route my incoming calls to a DID i haven't specifically routed to my default context (GoTo(XXX))[sip-default-in]exten = s,1,Answer() exten = s,2,Playback(beep)exten = s,2,Ringingexten = s,3,Wait,1exten = s,4,GoTo(XXX)// // My thought here is I will handle my incoming calls to XXX-XXX-XX00 and pass it to a specific context, say a queue [00]exten = _00,1,Answer()exten = _00,2,Playback(beep)exten = _00,3,GoTo(queue-test,s,1)What am I doing wrong??I can receive calls fine, but they aren't routing properlyI think I overlooked something. Thanks list!!/Chris ___ --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] Help with incoming SIP routing
Hello - Thanks for the suggestions, I actually learned since my post that the problem is related to the formation of the initial SIP invite sent from our Sonus gateway through our NexTone SBC. Using your idea regarding the variable I think i've found a work-around.
When the Sonus sends the inital invite the format is:INVITE sip:315579;npdi=[EMAIL PROTECTED] SIP/2.0It appears the Asterisk server is able to parse this message if I am only using one context, it appears to fail when I use multiple contexts to route my ingress calls.
From what I can tell Asterisk parses the invite and looks to send the call to s in the default context (domain 315579).Looking for s in default (domain 315579)If 's' doesn't exist in [315579] the call chokes.
If I create a variable based on the ingress 'domain' using:exten = s,1,Goto(${SIPDOMAIN},s,1)The call is sent to the correct context.I think it's a work-around, but it seems to do the trick.
-- Executing Goto(SIP/15241-08198868, 315579|s|1) in new stack -- Goto (315579,s,1) -- Executing Answer(SIP/15241-08198868, ) in new stack
Notes on Digium:http://bugs.digium.com/view.php?id=7208nbn=24
On 6/28/06, El Flynn [EMAIL PROTECTED] wrote:
Christopher Aloi wrote: Hello - I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone
on the list can assist me.snipUnless I'm misunderstanding you, how about trying this:1. In your sip.conf:[general]useragent=Asteriskport=5060context=default
tos=lowdelaydisallow=allallow=ulawallow=alawallow=gsmrtptimeout=300rtpholdtimeout=6002. In your extensions.conf:[default]exten = s,1,Goto(${CALLERIDNUM},s,1)[123456789]
exten = s,1,Answer()exten = s,2,Playback(beep)exten = s,3,GoTo(queue-test,s,1)So if you get an incoming SIP call from 123456789, it enters the defaultcontext and is then routed to the 123456789 context.
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[Asterisk-Users] Thoughts on building a Voicemail only Asterisk server?
Hello List -I've done some reading on voip-info regarding hardware requirements for an Asterisk server; but I haven't been able to find anyone doing what we plan to, so I am hoping you can assist.We are looking to provide a voice mail only Asterisk solution for approx. 100 homeless people, a customer of ours is planning to provide the service. The Asterisk service will reside in our data center which will provide the TDM-SIP GW so the Asterisk will receive all it's calls via SIP.A rough overview of what I think we will need:- A non-redundant server running Asterisk -- -- The Asterisk build will have a very simple dial plan-- -- -- Two inbound DID's (one for checking vmail and one for leaving voice mail for an extension)-- -- -- A management interface for the voice mail boxes, so I will need to run Apache - A disk array (either local RAID or external NAS) to house the voice mail storage.-- -- The voice mail system will allow 30MB of storage per user, so 30MBx100users=3GB-- -- I'd like the 3GB of storage to be in either in the RAID or dumped onto an NFS or NAS Does anyone have any recommendations on a server that might fit the bill above? Or experience running a similar application?Just looking for some thoughts on RAM, Processor speed, Disc etc...Thanks in advance. -Chris ___ --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] Asterisk queue log solution?
Do you have multiple tenants? I was under the impression that queue-metrics did not offer a hierarchy that would allow a tenant to view only his own queues etc...On 6/21/06, Matt [EMAIL PROTECTED] wrote: Does a solution exist that I am overlooking that may provide the functionality I am after?I don't understand why Queue-Metrics will not do what you need? Werun it and it does everything you just said you wanted to do. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Creating Queues on Asterisk server - Sending ingress calls off-net to either PSTN or another VoIP application - thoughts?
Hello,Long time subscriber/reader of this list - thank you for all the great ideas.Scenario:We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality:- Agents having the ability to login to a specific queue- Call distributed to that queue based on criteria- Basic reporting (ASA, AHT etc..) Solutions:- Flip the Mitel phones to load a SIP firmware and speak to AST (althought i'd love it, the powers that be probably won't)- Use the Asterisk queueing ability to send calls off network (AST) to the NCI platform (the Asterisk box can send these calls via SIP or TDM through a gateway). Goals:I'd like to create an Asterisk server running multiple queues for multiple tenants (or customers) that can provide the ability for agents to login remotely (either via an ingress call to AST or a www gui). The call flow would be similar to this: Agent#1 - logs into Mitel phoneAgent#1 - Dials XXX XXX into AstersikAgent#1 - Hears a prompt on Asterisk to login to a specifc queueAgent#1 - Passes DTMF and becomes 'available' in the eyes of Asterisk Agent#1 - Is now in queue*repeat for three agents*Now, all three agents are in an available state to Asterisk, and logged into our one queue. If Asterisk receives a call on a specific DID it will attempt to send the goal to agent#1, if agent#1 rings three times or returns a 'busy here' the call will pass to agent#2 etc. The challenge I see will be configuring an off-network queue, is anyone working with a similar setup?Does anyone have any thoughts on how to better accomplish my goals?Thanks in advance./Chris ___ --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
Fwd: FW: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts?
-- Forwarded message --From: Christopher Aloi [EMAIL PROTECTED]Date: Jun 18, 2006 9:52 PM Subject: Re: FW: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts?To: Alexander Lopez [EMAIL PROTECTED]Alexander,Thanks for your reply, may I ask a few questions?- Does the Asterisk server maintain any type or presence for the agents? (i'm assuming this wouldn't be possible since your shooting the call out POTS) - How do your off-network callback agents identify their location to the Asterisk server?- Are you able to describer your dialplan configuration in detail?Thanks again,/Chris On 6/18/06, Alexander Lopez [EMAIL PROTECTED] wrote: I do this type of thing right now, with both agents that are logged in and callback agents, All off site and via PSTN From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Christopher Aloi Sent: Sunday, June 18, 2006 8:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts? Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to login to a specific queue - Call distributed to that queue based on criteria - Basic reporting (ASA, AHT etc..) Solutions: - Flip the Mitel phones to load a SIP firmware and speak to AST (althought i'd love it, the powers that be probably won't) - Use the Asterisk queueing ability to send calls off network (AST) to the NCI platform (the Asterisk box can send these calls via SIP or TDM through a gateway). Goals: I'd like to create an Asterisk server running multiple queues for multiple tenants (or customers) that can provide the ability for agents to login remotely (either via an ingress call to AST or a www gui). The call flow would be similar to this: Agent#1 - logs into Mitel phone Agent#1 - Dials XXX XXX into Astersik Agent#1 - Hears a prompt on Asterisk to login to a specifc queue Agent#1 - Passes DTMF and becomes 'available' in the eyes of Asterisk Agent#1 - Is now in queue *repeat for three agents* Now, all three agents are in an available state to Asterisk, and logged into our one queue. If Asterisk receives a call on a specific DID it will attempt to send the goal to agent#1, if agent#1 rings three times or returns a 'busy here' the call will pass to agent#2 etc. The challenge I see will be configuring an off-network queue, is anyone working with a similar setup? Does anyone have any thoughts on how to better accomplish my goals? Thanks in advance. /Chris ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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
