[asterisk-users] GUI for Asterisk 1.2 Source
Hi, Is there a GUI for Asterisk 1.2 compiled from source or would I need to upgrade to the 1.4 version to get the GUI that can be installed on servers complied from source? Any help is appreciated. Otis ___ --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] GUI for Asterisk 1.2 Source
Thanks... Tzafrir Cohen wrote: On Thu, Oct 25, 2007 at 01:46:53PM -0500, OCOSA ListAcct wrote: Hi, Is there a GUI for Asterisk 1.2 compiled from source or would I need to upgrade to the 1.4 version to get the GUI that can be installed on servers complied from source? Any help is appreciated. asterisk-gui[tm] requires asterisk 1.4 . Ther are a number of other graphical user interfaces for Asterisk which work well with Asterisk 1.2. /me recommends gvim and runs. ___ --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] 20min waiting time
Did not work either...Thank you! Otis Michiel van Baak wrote: On 15:02, Sun 12 Aug 07, OCOSA ListAcct wrote: exten=5,2,Dial(SIP/supportSIP/support2,2,tr) Make this line read: exten=5,2,Dial(SIP/supportSIP/support2,,tr) That should do the trick ___ --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] 20min waiting time
I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis ___ --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] 20min waiting time
Steve do you have an example that works for you. I am reading the queue literature nowThank you! Otis Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Thanks, 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] 20min waiting time
Eric so I should do this exten=5,1,Answer exten=5,2,StartMusicOnHold exten=5,3,Dial(SIP/supportSIP/support2,2,tr) exten=5,4,VoiceMail([EMAIL PROTECTED]) exten=5,5,PlayBack(vm-goodbye) exten=5,6,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Otis Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. ___ --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] 20min waiting time
Steve do you have an example of this... Otis Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, 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] 20min waiting time
Ok thanks. I will finish reading and see if I have any questions I will post and wait until you answer thank you! Otis Steve Totaro wrote: Yes, but I have to be up very early in the morning and it is getting late. The answer priority will work for you in the meantime. If you want to investigate using real queues, let me know and I will help you set it up. Most of the stuff is on the Wiki but I will give you exact settings that should work on your setup. If you plan on growing or ever want to collect data on queues, then this is the way to go. Thanks, Steve OCOSA ListAcct wrote: Steve do you have an example of this... Otis Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, 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 ___ --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] 20min waiting time
Steve / Eric When configuring the queue I tested works fine but one issue. My agent auto logs off after I am done with the call. I tried ignoring that option in agents.conf no luckAlso the below with the Answer line does not work either...still stays on and ring about 1:15 secs then goes to voicemail Otis Eric ManxPower Wieling wrote: Steve Totaro wrote: Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: OCOSA ListAcct wrote: I apologize if this question has already been answered / asked. I was searching on Google and nothing I do will work. All that happens is that the phones ring for 00:01:15 then voicemail kicks in. My goal here is to let the phones ring and ring until someone is not busy. I think 2 secs is long enough. Here is how the dial plan is setup exten=5,1,StartMusicOnHold exten=5,2,Dial(SIP/supportSIP/support2,2,tr) exten=5,3,VoiceMail([EMAIL PROTECTED]) exten=5,4,PlayBack(vm-goodbye) exten=5,5,HangUp() exten=1222,1,VoiceMailMain([EMAIL PROTECTED]) Any help is appreciated Otis Easiest way to solve your problem is to implement a support queue. Queues in Asterisk are horrid little creatures. Many SIP phones and ITSPs will disconnect the call if the destination rings for a long time. Put an Answer as your first priority, this should fix your problem. That is an odd statement about queues. I ran a call center handling over 15,000 calls a day using Asterisk and queues. No real problems. Please qualify your completely abstract statement, Queues in Asterisk are horrid little creatures. Statements like this are completely non productive to anyone. Thanks, Steve Sorry to reply to my own post but for clarification, we had four queues. English sales, English support, Spanish sales, Spanish Support. At peek times, we would have 200-300 agents logged in and 600 or so callers. This was usually when our ads were running during Jerry Springer or Judge Judy. I think his two agent single queue would work just fine. Add Queuemetrics which is free (I believe) for five or less agents and then you can actually get some reporting on how your support role is handled. In your situation it seems that queues work well for you. When you have dedicated agents answering calls full time queues work well. In non-call shops people forget to log out of the queue, are away from their desk often, and otherwise just screw up many of the assumptions that the Asterisk queue system makes. This is in addition to the learning curve. For a low number of calls and/or non-dedicated agents, a little bit of dialplan logic can do everything someone needs with something that is massively more flexible. ___ --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] Asterisk Freeze Problem
Does asterisk 1.2.23 solve the problem did not say in the release notes. Also Could this be a CentOS 5 problem maybe? I am running CentOS 5 -Asterisk 1.2.22 and Zaptel 1.2.19 Otis ___ --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 Freeze
I too am having problems with freezing the FXO lines drop and the whole system dies. I am running Asterisk 1.2.22 and Zaptel 1.2.19 what do you suggest? I am thinking downgrade... Also asterisk will not start on boot I notice when reviewing the details says Asterisk OK then dies and traced backed to Zaptel the modules can not load. At boot asterisk just restarts and restarts...Any help is appreciated. otis Noah Miller wrote: You really need to update to a later version of asterisk (and zaptel). There have probably been somewhere close to a thousand bug fixes since 1.2.10. If you still have this issue with the latest version, please collect as much information as possible (exact cli messages, turn on logging, your config files, etc) and post that information to this list. I am very wary of upgrading -- some versions of Asterisk do not work well in my environment. Thursday night I upgraded one of my 6 production system to Asterisk 1.2.22 and Zaptel 1.2.19. So far I have not had any reported problems. That's one of the reasons I always do upgrades on a test environment before deploying them onto my live servers. Of course, I've still been bitten by a bug or two, but I've generally found that things get better after upgrades rather than worse. Judging by some of your posts (and other people's posts), I guess the mileage varies a lot, though. - Noah ___ --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 Freeze
I have just some 1 port FXO cards otis Tzafrir Cohen wrote: On Mon, Jul 23, 2007 at 04:42:52PM -0500, OCOSA ListAcct wrote: I too am having problems with freezing the FXO lines drop and the whole system dies. I am running Asterisk 1.2.22 and Zaptel 1.2.19 what do you suggest? I am thinking downgrade... Which FXO device do you have, exactly? Also asterisk will not start on boot I notice when reviewing the details says Asterisk OK then dies and traced backed to Zaptel the modules can not load. At boot asterisk just restarts and restarts...Any help is appreciated. Asterisk starts and freezes later, or fails to start at all? Did it work before? At which versions? ___ --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 Freeze
Tzafrir Cohen wrote: On Mon, Jul 23, 2007 at 04:42:52PM -0500, OCOSA ListAcct wrote: I too am having problems with freezing the FXO lines drop and the whole system dies. I am running Asterisk 1.2.22 and Zaptel 1.2.19 what do you suggest? I am thinking downgrade... Which FXO device do you have, exactly? Single Port FXO Cards Also asterisk will not start on boot I notice when reviewing the details says Asterisk OK then dies and traced backed to Zaptel the modules can not load. At boot asterisk just restarts and restarts...Any help is appreciated. Asterisk starts and freezes later, or fails to start at all? Did it work before? At which versions? Worked at Asterisk 1.2.18 and Zaptel 1.2.19 then stopped then worked again once upgraded to 1.2.21 ___ --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] Music on Hold and Announcements
Does anyone know how to have an ad or announcement playing but in the background play a MP3 file? I think this would be done with the s extension and background application but not sure how? Any help would be appreciated!! -- Otis ___ --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] Music on Hold and Announcements
Wow seems a bit much? I use 1.2.22.yeah if you make it generic it would be nice and I would probably upgrade. I guess. The only other way to do this is to just drop the announcements and record a message on hold for a specific group with music in the background at the recording time. So for team 1 context [team1] and team 2 context [team2] and play various messages specific for the groups. Russell your a genius.nice setup. Otis Russell Bryant wrote: OCOSA ListAcct wrote: Does anyone know how to have an ad or announcement playing but in the background play a MP3 file? I think this would be done with the s extension and background application but not sure how? Any help would be appreciated!! Interesting question. I actually have some code that will almost do this sitting in a branch. The code actually started out as a joke, but I think I could make it more generic to where it could be useful. Right now, I have two modules - res_monkeys and app_monkeys. If you load res_monkeys on a system, it will pick a random active channel on the system once per minute and play the tt-weasels file to them. This would be a nice module to load on April 1st. :) app_monkeys gives you a dialplan application called Monkeys(). Once you run this on a channel, it will hear the tt-weasels file once a minute for the rest of its lifetime in the system, while executing other applications. I could probably make app_monkeys more generic so that you can specify a frequency and which sound file to play. The one thing you can't do with it is turn this periodic announcement back off. I think I could add it, though ... Anyway, this would only be for 1.6 unless enough people think its useful. Then, I might maintain an unofficial backport to 1.4. ___ --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] 's' extension Asterisk 1.2.18
Yeah thats what I thought I found everything running so I just upgraded and fixed the problem. Otis Anthony Francis wrote: Maybe a hardware problem? What does zttool and ztcfg -vvv say? Is Zaptel running? -- Original Message -- From: OCOSA ListAcct [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Sat, 14 Jul 2007 14:56:33 -0500 how can I fix this just started .. Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Otis ___ --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 Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com ___ --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] 's' extension Asterisk 1.2.18
how can I fix this just started .. Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Otis ___ --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] 's' extension Asterisk 1.2.18
Never mind the 1.2.18 messed and did not recognize the s extension any more so I just upgrade to 1.2.21.1 and fixed the problem,.weird. otis OCOSA ListAcct wrote: how can I fix this just started .. Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Otis ___ --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] Transfer Call to Cell Phone
Thanks work perfect,,
Otis Surratt Jr. / [EMAIL PROTECTED]
OCOSA Communications, LLC
PH: 918.585.9882 x 205 Fax: 918.585.5857
Visit Tulsa's Best Internet Data Center:
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Ryan Goldberg wrote:
OCOSA ListAcc wrote:
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example. I have the
system setup so it just dials out which ever line is not busy. Thanks!
I'm quite new to *, but I've got this in place in my first rendition, and
I'm pretty sure it does what you want:
exten = 101,1,Dial(SIP/${EXTEN},15,t)
exten = 101,n,Dial(Zap/4/12185551212,30,tpm)
exten = 101,n,VoiceMail([EMAIL PROTECTED])
exten = 101,n,Playback(vm-goodbye)
exten = 101,n,Hangup
caller dials extension 101. It first tries his desk for 15 seconds, then
it tries his cell over a zap channel (the 'p' turns on call screening),
then it finally hits voicemail. In our actual dialplan, the cell phone
call goes out over sip, so the line looks like this:
exten = 101,1,Dial(SIP/lesnet/12185551212,30,tpm)
Alternatively, the first line could be:
exten = 101,1,Dial(SIP/${EXTEN}Zap/4/12185551212,30,tpm)
which would dial both the desk and the cell at the same time...
See http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Hope that helps.
Ryan
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Re: [asterisk-users] Transfer Call to Cell Phone
works perfectThanks
--
Otis
John Faubion wrote:
We do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
The issue really isn't whether you have the ability to make toll calls on
your line. The concern here is in what the regulatory agencies call toll
bridging which is using a system to relay a call from one local calling are
to another local calling area to avoid a toll charge. This is one of those
gray areas that can become a problem if your not careful. The problem comes
up if you have customers that can call you as a local call and you are
forwarding them on to another party that is a local call for you but would
be a toll call for the customer. This is essentially what toll bridging is
about. Now your not likely to have to worry about the legal ramifications of
this since your merely connecting the customer with an extension of your
company, namely your salesman. Where this could become a problem for you
would be in transferring the customer using the same pots line. The reason
is that ATT is handling the transfer. When you transfer the call, it
essentially becomes a new call. The main difference is that you have
provided the called number. So the software in the Class 5 (End office)
switch, takes the number you provide and runs the call through its routing
translations (similar to the Asterisk dialing plan) and if it determines
that the destination number is outside the originators Local Area Transport
Area or LATA, then it will either drop the originator to a message that
says, You must first dial a 0 or 1 before calling this number or it may
deny the transfer allowing you to stay connected to the customer. Neither
one looks very professional. The only way around this would be to provide
another line or trunk to pass the call down. Now if your not in an
overlapping LATA this probably isn't an issue.
The only way I can get it to work is by have the call on the 1st
line then transfer it out on the 2nd line. After that is complete both
lines are free.
Are you saying that you are able to route a call from line 1 to line 2 and
have the call transfer, thus freeing the lines or that once the call
completes the lines are freed? I've never seen the first scenario. The
second scenario is the normal behavior.
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example.
In extensions.conf use something like this.
[global]
SIP-PROV = sip.urprovider.com
; Now set the call forward numbers
CFN21 = 551234 ; These are normally set in an external file
[internal]
exten = 21,1,Macro(stdext,${SIP/21},${CFN${EXTEN}})
[macro-stdext];
; Standard extension macro:
; ${ARG1} - Device(s) to ring
; ${ARG2} - Our call forward number
exten = s,1,Dial(${ARG1},10)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,GotoIf($[${LEN(${ARG2})}0]?s-CFWD,1)
exten = s-NOANSWER,2,Voicemail(${MACRO_EXTEN},u)
exten = s-BUSY,1,Voicemail(${MACRO_EXTEN},b)
exten = s-CFWD,1,Dial(SIP/[EMAIL PROTECTED],20)
exten = s-CFWD,2,Goto(s-NOANSWER,2)
exten = _s-.,1,Goto(s-NOANSWER,2)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
There is more to this but this should show the basics of what we use. I
store my Call Forward Numbers (CFN) in an external file. This allow me to
update the file externally (currently with a web interface but as soon as I
get the prompts recorded it will be done with an IVR) and then just reload
the extensions to activate the new numbers. Also I using SIP for pretty much
everything. Our TDM400 doesn't even have modules, it's just there for
timing. However you should be able to convert the SIP calls to ZAP calls for
you use. The internal context is included in our default context. Dialing
extension 21 calls the stdext macro. This dials the local extension first.
If not answered after 10 seconds, we check to make sure we have a phone
number to send the call out with. If not we send it on to voice mail.
Otherwise we send it to the s-CFWD. The check listed here is a very
rudimentary check but again I hope you get the idea. Next we try the call to
the CFN. If not answered in 20 seconds, then we send it to voice mail.
Finally if the user presses the star button during the attempt, we send them
on to Voicemail mail so they can check their messages.
Hopefully this helps.
John
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Re: [asterisk-users] Transfer Call to Cell Phone
John Faubion wrote: by the way selling does not depend on the amount of lines you have and we are very productive trust me True, very true. There are lots of very productive sales people that don't need a phone at all. From the paper boy to car dealers, lots of sales don't require many phone lines. Of course at the same time, a typical call center wouldn't be very productive with only two lines. I have seen a million dollar corp work off four lines so your statement is quite vague... We have a few agents that have million dollar months and even a couple that have had million dollar weeks! But that isn't the point, is it Otis? The problem is you got your feelings hurt because instead of reading my reply, you assumed that I was putting your company down. My first paragraph was kind of a open thought process so that you and others might comprehend the basis of my reply. What I was trying to wrap my head around, was just how productive a system with only two lines could be if a single call came in and was then routed back out the other line to the outside sales guy. Now if you were using a digital line, perhaps we could consider using the signaling to redirect the call from the originating source directly to the salesman's phone and thus free up the lines for the next call. But no, you said pots lines as in Plain Old Telephone Service (POTS) which means we don't have the option of using some fancy, out of band signaling to redirect the call. So my thinking was, as I said before, Surely you have more than two lines. In my twenty-two years of telephony experience, dealing with everything from single line phones to key systems to PBX systems to Nortel DMS-500 switches, I only remember one sales office that only had two lines and that office was literally an 8 foot by 8 foot closet with two phones and all calls were outgoing. You are right but finish reading thisTo be honest I did not get my feelings hurt, so assumptions are not needed I was simply stating one scenario where a local company here was very product to better your understanding of how some companies work off 2 to 10 lines and still produce. If you have read my first statement you would have understood that I did read and I did appreciate your reply. As the other methods has no interest to me at this time. I do agree maybe I should have sent a paragraph with details but I felt like knowing only two lines for the sales office was plenty. Now do not confuse us with a call center. Trust me by no means could we be as productive as others but then again we are not a giant but a small business and do not sign up thousands a day so we do not have a need for more lines yet. But there again you still made a judgment call about the two lines. If I tell this is the setup then there is nothing to question. Sometimes all tech guys have a problem with assuming sometimes there are more to a situation than what was presented. I am guilty as well. LOL!! Yes, my answer was a little vague. So was the information you provided. Now had you bothered to read the 2nd and 3rd paragraph, you might have noticed that I provided a few methods that you could consider. My intention for doing this was simple. Maybe one of the ways mentioned would spark a response from you that would help to clarify the right way. Now suppose for a moment that you had actually read the reply. Let's also pretend that in reading it you realized that, yes, you have two pots lines, but what you had meant to say was that you had two unused pots lines along with some other form of incoming trunks. Then maybe you would have responded with an email to clarify that, to which I could have suggested that maybe you could look into a two port cell phone gateway to keep the incoming lines free and still keep connected to your sales guy. Can you see how we could have used that information to consider the right option? Here again had you read my first statement you would have understood that I appreciated your reply. John thanks for the input. forget about my right way ok! I made a mistake on putting this in...this is what I was really looking for: explained later down.. Considering that this list is for non commercial discussion, our only form of payment here is in the repayment of our debt to others that have gone before us and helped us out. Next time please appreciate the fact that someone else took time out of their busy day to consider and to reply your request for information. Now if you would like to provide a little more detail with your request, I'm fairly sure that someone here will likely respond to it. There again making assumptions are not right because I did try your first option. But John do not get me wrong as I have been thinking about the second so before you can say please appreciate the response lets try to get the facts straight.
Re: [asterisk-users] Transfer Call to Cell Phone
John thanks for the input. forget about my right way ok! by the way selling does not depend on the amount of lines you have and we are very productive trust me I have seen a million dollar corp work off four lines so your statement is quite vague... Otis John Faubion wrote: I have two pots lines into my current Asterisk Box. I have an outsides sales guy who wants to work off his cell phone or transfer his calls from his extension and the main sales extensions. How can I do this right? Do it right? You really haven't provided enough information to make the right decision. Do you have more than two lines? Surely you have more than two lines. You mention his extension and the main sales extensions. I can't imagine a sales department with only two lines. Well I can, but they don't sell much! 8) If you have other lines available, such as through an ITSP, T1/E1, or etc, then you only need to map his extension to an outside line. This could be done either through a follow-me, call forwarding, fixed routing, or etc. As an example, we have several agents (we're a real estate brokerage office), that only come into the office occasionally. Since most of them use their cell phones for nearly all of their business, I have fixed routing to send calls to them. I will soon have an IVR for them to be able to change that fixed routing on their own. We also have some agents that have a regular desk here in the office. For them, the use call forward unanswered at the phone to route the calls to their cell phones when they are out of the office. The owner uses follow-me to route her calls to the office phone, her home phone and her cell phone. Another way to do it would be to install a SIP/IAX/TDM to TDMA/GSM gateway. Make sure the provider is the same as the salesman's cell phone provider and your mobile to mobile minutes can be free. If you have more than a couple salesmen, this route will likely entail a multi-port gateway but the idea is still the same. As far as the right way, that depends on way to many factors tat you haven't addressed. John Faubion ___ --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 1.2.16 - No Caller ID
Giorgio, That does not work it just shows up as useincomingcalleridonzaptransfer I set the following: callerid=useincomingcalleridonzaptransfer. Are you referring to something else like useincomingcalleridonzaptransfer=yes Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Giorgio Incantalupo wrote: Hi, have you tried different values of callerid? Maybe setting *useincomingcalleridonzaptransfer* to yes can help you. Giorgio Incantalupo OCOSA ListAcc wrote: Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 channel=2 ___ --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] Asterisk 1.2.16 - No Caller ID
so to fix the no caller id thing will need to adjust the rx gain and tx gain? Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Eric ManxPower Wieling wrote: OCOSA ListAcc wrote: Eric, I have watched the CLI before and it said nothing although I did change the position of the callerid = asreceived to right below and nothing it still shows up on the phones asterisk and in voice mail sent via e-mail unknown caller: Here is an output from a while back but it stopped so I do not know what is up. -- Starting simple switch on 'Zap/1-1' Apr 6 16:54:39 ERROR[4726]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-92) Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6255 ss_thread: CallerID feed failed: Success Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/1-1' These errors usually indicate that your rxgain for the FXO ports is either too high or too low. Change the rxgain in /etc/asterisk/zapata.conf in increments of 2 either up or down until, but you generally don't want it to be less than -10 or greater than 10. reload chan_zap.so should apply the gain changes. ___ --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] Asterisk 1.2.16 - No Caller ID
Eric, Thanks when I took the rx and tx to 0.0 on both the caller id showed up I guess I will play with. My main reasoning for adjusting the rx and tx was to get rid of the echo...What other tips do you suggest or anyone out there? Thank you! Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Eric ManxPower Wieling wrote: OCOSA ListAcc wrote: Eric, I have watched the CLI before and it said nothing although I did change the position of the callerid = asreceived to right below and nothing it still shows up on the phones asterisk and in voice mail sent via e-mail unknown caller: Here is an output from a while back but it stopped so I do not know what is up. -- Starting simple switch on 'Zap/1-1' Apr 6 16:54:39 ERROR[4726]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-92) Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6255 ss_thread: CallerID feed failed: Success Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/1-1' These errors usually indicate that your rxgain for the FXO ports is either too high or too low. Change the rxgain in /etc/asterisk/zapata.conf in increments of 2 either up or down until, but you generally don't want it to be less than -10 or greater than 10. reload chan_zap.so should apply the gain changes. ___ --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
