RE: [asterisk-users] balance anouncement
It’s telling you the sound file “Goodbye”
does not exist in the directory it looks for sounds. If you indeed have a sound
file called Goodbye then you need to either move it to the default sounds
directory or add the path line to the command. If you don’t have the
sound file you’ll need to either create one or use one you do have.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ram
Sent: Friday, September 01, 2006
11:06 AM
To:
[EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users]
balance anouncement
Hi
iam trying like this in my extension.conf
some one refered in the news group past
error in messages
Sep 1 21:31:42 WARNING[28610] file.c: File Goodbye does not exist
in any format
Sep 1 21:31:42 WARNING[28610] file.c: Unable to open Goodbye (format
ulaw): No such file or directory
Sep 1 21:31:42 WARNING[28610] app_playback.c: ast_streamfile failed on
SIP/8-3ca6 for Goodbye
exten => 888,1,Read(${CALLERIDNUM})
exten => 888,2,MYSQL(Connect connid 127.0.0.1
root password mydatabase)
exten => 888,3,MYSQL(Query resultid ${connid} select total from balance
where username=${CALLERIDNUM})
exten => 888,4,MYSQL(Fetch fetch ${resultid} AMOUNT-DUE)
exten => 888,5,MYSQL(Clear ${resultid})
exten => 888,6,Playback(Goodbye)
exten => 888,7,Hangup
iam calling from my extension to 888
Ram
On 9/1/06, John
Millican <[EMAIL PROTECTED]>
wrote:
On Friday September 01 2006 10:19 am, ram wrote:
> Hi
>
> thanks for the quick reply
>
> any documents to read to achive this
> or any examples would be great to read
>
> Ram
>
> On 9/1/06, John Millican <[EMAIL PROTECTED]
> wrote:
> > On Friday September 01 2006 9:27 am, ram wrote:
> > > Hi
> > >
> > > how can i do balance anouncement by using asterisk
> > >
> > > take example, i have table balance , user name 9, balance
200$
> > >
> > > user dial *98 or what ever, then i need anouce his balance is
200$, by
> > > reading from that row
> > >
> > > any clues how can i achive this or is this possible ?
> > >
> > > Ram
> >
> > Create an AGI script that does a db look up for the ballance and then
> > pass the
> > result back to Cepstral or Festival or your favorite text to speech
> > software.
> > John M
> >
Try google or voip-info.org and search for
Asterisk AGI should yeid some good results.
AGI can be called from the dial plan and written in your favorite language
i.e. PHP, C++, Perl, C, Java
or start here:
http://home.cogeco.ca/~camstuff/agi.html
http://asterisk.drunkcoder.com/agi.cgi
John M
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[asterisk-users] Unable to start special tone
Can anyone tell me where this is coming from? I can’t seem to find any information on it anywhere. I don’t believe I’m using “special tones” anywhere. Any ideas? Aug 23 14:30:34 WARNING[15443]: chan_zap.c:5492 ss_thread: Unable to start special tone on 15 _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] PRI and Asterisk
I have tested Redfone’s boxes. Tried two of them and was able to re-create some issues. I did not have PRI lines but a 24 channel e&m wink line so not sure if PRI is affected as well. I found that over time we had issues with hanging zap channels. Asterisk reported everything was just fine yet people got busy signals calling in and when calling out all they got was silence. The CLI never showed any incoming calls that were attempted and when dialing out it showed Dialing but nothing happened. I worked with Mark Warren at Redfone and he was very co-operative and had an idea to fix this but sadly we just didn’t have any more time to fight with it and went with Digium cards. As of this writing I am starting to get problems with inbound calls. Seems for a couple minutes no one can dial into our office and then it just clears up. No errors or anything in Asterisk to indicate a problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Varanini Sent: Tuesday, August 22, 2006 6:35 PM To: Julian Varanini Subject: RE: [asterisk-users] PRI and Asterisk Hi Everyone Any opinions on this? Thanks Julian From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Tue, 18 Jul 2006 01:29:57 + Subject: [asterisk-users] PRI and Asterisk Hi All, I am planning to order a PRI and would like to know your opinions on a devices like the Redfone redbridge. Basically any PRI to Asterisk interface that has worked well for you. Thanks, Julian ___ --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] Text to Speech
All I can find for Flite is for AAH, does it work as well with plain Asterisk? Is the setup the same? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Monday, August 21, 2006 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Text to Speech N.B.: Please use plain text when sending to this list > Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. > > > > We are willing to pay for a good one that works. Anyone tried the AT&T speech engine? The IBM ViaVoice sounds no better then Festival. You have flite that is free and, IMHO better than festival (http://nerdvittles.com/index.php?p=134). I also tried Prophecy (http://www.voxeo.com/) but I'm still waiting for the Linux version as I don't have time to babysit a Windows server :) hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Text to Speech
Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn’t cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the AT&T speech engine? The IBM ViaVoice sounds no better then Festival. Thanks for your input. _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Auto retry on Busy
You need to change:
exten => 777,4,goto(trunkretry,1,1)
to
exten => 777,4,goto(trunkretry,777,1)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Silverman
Sent: Friday, August 11, 2006 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Kevin Smith
Subject: Re: [asterisk-users] Auto retry on Busy
Kevin,
Thanks for the suggestion. I can't seem to get it to work.
This is what I put in my extensions.conf
We only have one number that we want to keep trying right now, so I
tried to set it so by calling extension 777, it would start the
system retrying. (The actual number isn't 999 :)
[trunkretry]
exten => 777,1,Dial(${TRUNK}/www1323999},10,)
exten => 777,2,gotoif[${DIALSTATUS}="BUSY"]?(LINEBUSY):(OTHER)
exten => 777,3,(LINEBUSY), Wait(15)
exten => 777,4,goto(trunkretry,1,1)
Thanks,
-N
On Aug 11, 2006, at 11:29 AM, Kevin Smith wrote:
> Why don't you just test for the dial status after the dial command
> completes? I don't really see why you want something to keep
> dialing until it gets through, but this would work.
>
> [something]
> 1,1,Dial(zap/,sip/, etc/whatever, 10)
> 1,n,gotoif[${DIALSTATUS}="BUSY"]?(LINEBUSY):(OTHER)
> 1,n(LINEBUSY), Wait(30)
> 1,n,goto(something,1,1)
> 1,n(OTHER), do something else
>
> Sure it is pretty rough, but the basics are there. Also you might
> want to read this: http://www.voip-info.org/wiki-Asterisk+variable
> +DIALSTATUS
>
> Kevin
>
>
>
> Noah Silverman wrote:
>> Hi,
>>
>> Does anybody have an easy solution for this.
>>
>> I want something that will keep trying a busy number every 30
>> seconds until it gets through.
>>
>> I've tried retrydial, but can't get it to work.
>>
>> Any suggestions?
>>
>> Thanks,
>>
>> -N
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RE: [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk
My Asterisk is 1.2.9.1 but I've recreated this on 1.2.7 and 1.2.8. Not tried
1.2.10 yet.
This only happens on forwarded calls for me as well. I've not let it run too
long to see if the server dies eventually. I don't believe it will because
once the caller hangs up the errors stop and my server carries on. I've not
had it die on me yet but I've no told our users NEVER to forward outside the
building.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of M D
Sent: Thursday, August 10, 2006 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fwd: Dropping incompatible frame killing
Asterisk
Hi
Sorry, I should have mentioned that we're only running SIP. Our calls
to the PSTN are routed through a VoIP carrier and all of our clients
are SIP.
Which version of Asterisk are you using? Is this killing your box? If
it is, have you established why? CPU being killed, memory starvation,
something else?
It is only happening on forwarded calls, though. I'll have to try your
workaround.
Thanks,
Mark
On 10/08/06, Kevin Savoy <[EMAIL PROTECTED]> wrote:
> This is an issue I'm having as well. Here is what I've discovered.
>
> Call comes in on a T1 line. Call is sent to a SIP phone (say 4000) based
on
> the extensions.conf setup. User of phone 4000 has set a forward in the
phone
> to an external number, 1-555-555-. There is nothing telling Asterisk
to
> Dial(Zap/g1) so the call does not get converted back to slin to send along
> the T1 lines out of the building. Since SIP can't be sent the frame is
> incompatible and is dropped. I know this probably isn't as technical as it
> should be but in essence it is what is happening. I've had to do a
> workaround and set up an extension that dials the number that the phone
was
> to be forwarded too. I set up extension 500. The user forwards the phone
to
> 500. extensions.conf says Dial(Zap/g1/155).
>
> Band-aid solution. I've seen on the bug reports it is a known issue but
not
> resolved yet. Last update was July 5th.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of M D
> Sent: Thursday, August 10, 2006 8:50 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Fwd: Dropping incompatible frame killing
Asterisk
>
> Hi there
>
> We're running Asterisk 1.2.1 (I know, it's old; we have an upgrade
> planned but can't do it just yet) on Debian testing. Every now and
> Asterisk and the box are dying -- no SSH login, no calls, nothing. The
> last lines logged are:
>
> Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Executing
> Dial("SIP/5060-0843a7f0", "SIP/123456|30")
> Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Called 123456
> Jul 31 14:23:31 VERBOSE[14085] logger.c: -- Got SIP response 302
> "Moved Temporarily" back from 85.189.x.x
> Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Now forwarding
> SIP/5060-0843a7f0 to 'Local/[EMAIL PROTECTED]' (thanks to
> SIP/123456-2241)
> Jul 31 14:23:31 VERBOSE[32701] logger.c: -- Executing
> Dial("Local/[EMAIL PROTECTED],2",
> "SIP/[EMAIL PROTECTED]:5070") in new stack
> Jul 31 14:23:31 VERBOSE[32701] logger.c : -- Called
> [EMAIL PROTECTED]:5070
> Jul 31 14:23:31 VERBOSE[32701] logger.c: --
> SIP/outbound.gateway:5070-550a is ringing
> Jul 31 14:23:31 VERBOSE[32696] logger.c: --
> Local/[EMAIL PROTECTED],1 is ringing
> Jul 31 14:23:31 VERBOSE[32701] logger.c: --
> SIP/outbound.gateway:5070-550a is making progress passing it to
> Local/[EMAIL PROTECTED],2
> Jul 31 14:23:31 VERBOSE[32696] logger.c: --
> Local/[EMAIL PROTECTED] _110-7282,1 is making progress passing it to
> SIP/5060-0843a7f0
> Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
> frame on Local/[EMAIL PROTECTED],2 of format slin since our
> native format has changed to alaw
> Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
> frame on Local/[EMAIL PROTECTED],2 of format slin since our
> native format has changed to alaw
>
> The last lines are repeated until the server dies.
>
> The phone appears to be a SNOM and should be using only g.711 alaw or
ulaw.
>
> I inherited this box with Asterisk running as root so I've changed it
> to a non-privileged user but assuming the server is dynig through
> resource starvation I doubt it'll help.
>
> So, any ideas what this traffic is? What can we do to stop it? Clearly
> I need to upgrade Asterisk but a cursory glance at the changelog
> doesn't suggest a bug was reported with these symptoms which would
> have been fixed in a lat
RE: [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk
This is an issue I'm having as well. Here is what I've discovered.
Call comes in on a T1 line. Call is sent to a SIP phone (say 4000) based on
the extensions.conf setup. User of phone 4000 has set a forward in the phone
to an external number, 1-555-555-. There is nothing telling Asterisk to
Dial(Zap/g1) so the call does not get converted back to slin to send along
the T1 lines out of the building. Since SIP can't be sent the frame is
incompatible and is dropped. I know this probably isn't as technical as it
should be but in essence it is what is happening. I've had to do a
workaround and set up an extension that dials the number that the phone was
to be forwarded too. I set up extension 500. The user forwards the phone to
500. extensions.conf says Dial(Zap/g1/155).
Band-aid solution. I've seen on the bug reports it is a known issue but not
resolved yet. Last update was July 5th.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of M D
Sent: Thursday, August 10, 2006 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk
Hi there
We're running Asterisk 1.2.1 (I know, it's old; we have an upgrade
planned but can't do it just yet) on Debian testing. Every now and
Asterisk and the box are dying -- no SSH login, no calls, nothing. The
last lines logged are:
Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Executing
Dial("SIP/5060-0843a7f0", "SIP/123456|30")
Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Called 123456
Jul 31 14:23:31 VERBOSE[14085] logger.c: -- Got SIP response 302
"Moved Temporarily" back from 85.189.x.x
Jul 31 14:23:31 VERBOSE[32696] logger.c: -- Now forwarding
SIP/5060-0843a7f0 to 'Local/[EMAIL PROTECTED]' (thanks to
SIP/123456-2241)
Jul 31 14:23:31 VERBOSE[32701] logger.c: -- Executing
Dial("Local/[EMAIL PROTECTED],2",
"SIP/[EMAIL PROTECTED]:5070") in new stack
Jul 31 14:23:31 VERBOSE[32701] logger.c : -- Called
[EMAIL PROTECTED]:5070
Jul 31 14:23:31 VERBOSE[32701] logger.c: --
SIP/outbound.gateway:5070-550a is ringing
Jul 31 14:23:31 VERBOSE[32696] logger.c: --
Local/[EMAIL PROTECTED],1 is ringing
Jul 31 14:23:31 VERBOSE[32701] logger.c: --
SIP/outbound.gateway:5070-550a is making progress passing it to
Local/[EMAIL PROTECTED],2
Jul 31 14:23:31 VERBOSE[32696] logger.c: --
Local/[EMAIL PROTECTED] _110-7282,1 is making progress passing it to
SIP/5060-0843a7f0
Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
frame on Local/[EMAIL PROTECTED],2 of format slin since our
native format has changed to alaw
Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
frame on Local/[EMAIL PROTECTED],2 of format slin since our
native format has changed to alaw
The last lines are repeated until the server dies.
The phone appears to be a SNOM and should be using only g.711 alaw or ulaw.
I inherited this box with Asterisk running as root so I've changed it
to a non-privileged user but assuming the server is dynig through
resource starvation I doubt it'll help.
So, any ideas what this traffic is? What can we do to stop it? Clearly
I need to upgrade Asterisk but a cursory glance at the changelog
doesn't suggest a bug was reported with these symptoms which would
have been fixed in a later release.
Cheers,
Mark
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RE: [asterisk-users] Operator in Voicemail
It is an o as in operator. That's what the manuals say. O extension is operator. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, July 24, 2006 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Operator in Voicemail Are you sure this is saying "exten = 0" with a ZERO and not an Oh? Looks like a lowercase Oh to me below. Kevin Savoy wrote: > This doesn't solve the problem. Still the same. Any other ideas? > > > > > > This is what I am using: > > > > exten = o,1,Answer() > > exten = o,2,GoTo(default,3000,1) > > exten = o,3,Hangup() > > > > Hope this helps, > > > > Henk > > > > > > *From:* [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony > Davis > *Sent:* maandag 24 juli 2006 18:20 > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* RE: [asterisk-users] Operator in Voicemail > > > > I'm having the exact same problem here. I originally thought it was a > context problem. > > However, to troubleshoot I tried placing the following in every context > (default, from-inside, from-outside, etc) in extensions.conf with no luck: > > / exten => o,1,DIAL(SIP/100,100)/ > > > > Like Kevin, it works fine for our internal users, just doesn't work for > callers coming from the PSTN. > > > > Thanks, > > -AntD > > > > *From:* [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] *On Behalf Of *Kevin Savoy > *Sent:* Monday, July 24, 2006 7:37 AM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* [asterisk-users] Operator in Voicemail > > > > I've got an odd problem. I have set in Voicemail.conf operator=yes as a > default. This is so that when a caller is in the voicemail system they > can press 0 and be sent to the operator. This works fine when the caller > is internal to the system but NOT when the caller is calling in from the > PSTN. Instead the caller gets the message Press 1 to accept the > recording. Pressing 0 again deletes the message. How do I get this to > work for outside callers calling in?? ___ --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] Operator in Voicemail
This doesn’t solve the problem. Still the same. Any other ideas? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henk Sent: Monday, July 24, 2006 12:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Operator in Voicemail This is what I am using: exten = o,1,Answer() exten = o,2,GoTo(default,3000,1) exten = o,3,Hangup() Hope this helps, Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Davis Sent: maandag 24 juli 2006 18:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Operator in Voicemail I’m having the exact same problem here. I originally thought it was a context problem. However, to troubleshoot I tried placing the following in every context (default, from-inside, from-outside, etc) in extensions.conf with no luck: exten => o,1,DIAL(SIP/100,100) Like Kevin, it works fine for our internal users, just doesn’t work for callers coming from the PSTN. Thanks, -AntD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy Sent: Monday, July 24, 2006 7:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Operator in Voicemail I’ve got an odd problem. I have set in Voicemail.conf operator=yes as a default. This is so that when a caller is in the voicemail system they can press 0 and be sent to the operator. This works fine when the caller is internal to the system but NOT when the caller is calling in from the PSTN. Instead the caller gets the message Press 1 to accept the recording. Pressing 0 again deletes the message. How do I get this to work for outside callers calling in?? Thanks _________ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Operator in Voicemail
I’ve got an odd problem. I have set in Voicemail.conf operator=yes as a default. This is so that when a caller is in the voicemail system they can press 0 and be sent to the operator. This works fine when the caller is internal to the system but NOT when the caller is calling in from the PSTN. Instead the caller gets the message Press 1 to accept the recording. Pressing 0 again deletes the message. How do I get this to work for outside callers calling in?? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Email notification of voicemail
Sheesh, tried that and I'm STILL getting email attempts. This doesn't make sense here. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Friday, July 14, 2006 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Aha - get rid of the leading comma for each entry.. => ,Front Desk => .. A. On Jul 13, 2006, at 1:00 PM, Kevin Savoy wrote: > I've X'd out the extensions and passwords but this is all I have in > there. > Thanks > > [default] > =>,,Front Desk,, ___ --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: [BULK] Re: [Asterisk-Users] how to decrease answer time !
The world is full of smart alecks. Thank the lords because what a boring world this would be without us :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Thursday, July 13, 2006 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [BULK] Re: [Asterisk-Users] how to decrease answer time ! Importance: Low When you hear the phone ring, run faster so you reach it more quickly. On 13/07/06, Pablo Mora <[EMAIL PROTECTED]> wrote: > > > > > > > > Pablo Mora, Ing. > > GERENTE DE OPERACIONES > > ESPOLTEL S.A. > > Malecón 100 y Loja > > Telf.:2514477 > > > ___ > --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 > > > -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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] Email notification of voicemail
I've X'd out the extensions and passwords but this is all I have in there. Thanks [general] attach=no format=wav49 skipms=3000 nextaftercmd=yes maxsilence=10 silencethreshold=128 maxlogins=3 tz=central operator=yes [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/CST6CDT|'vm-received' Q 'digits/at' IMp mountain=America/Phoenix|'vm-received' q 'digits/at' IMp pacific=America/Los_Angeles|'vm-received' q 'digits/at' IMp military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' [default] =>,,Front Desk,, =>,,Eric Kelley =>,,Justin Hall =>,,Jason Smestad XXXX=>,,Cumi Everson =>,,Glenda Cusker =>,,Laura Sanford =>,,Gary Sundet =>,,Kevin Penner =>,,Kevin Savoy =>,,Jeff Garaas =>,,Natalie Thompson =>,,Jolene Ross =>,,Ralf Patterson =>,,Mike Satterlee =>,,Michelle Siverson =>,,Kathy Evenson =>,,Kathy Michels -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Thursday, July 13, 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Can you send me (or pastebin) your voicemail.conf? A. On Jul 13, 2006, at 12:35 PM, Kevin Savoy wrote: > Thanks for replying. Have tried that. If I don't specify an email > address it > then takes the first name and last name and then the domain of the > pbx. For > example > > 1234 => 1234,Bob Smith > > I then get: > > [EMAIL PROTECTED] > > Which of course fails because that address doesn't exist. > > Any other ideas? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Anthony > Rodgers > Sent: Thursday, July 13, 2006 2:24 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Email notification of voicemail > > Try having nothing after the name in your voicemail.conf: > > 1234 => 1234,The Marquis de Sade ___ --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] Email notification of voicemail
Thanks for replying. Have tried that. If I don't specify an email address it then takes the first name and last name and then the domain of the pbx. For example 1234 => 1234,Bob Smith I then get: [EMAIL PROTECTED] Which of course fails because that address doesn't exist. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Thursday, July 13, 2006 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Try having nothing after the name in your voicemail.conf: 1234 => 1234,The Marquis de Sade Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jul 12, 2006, at 11:17 AM, Kevin Savoy wrote: > I have attach=no in my voicemail.conf so that can't be doing it. Not > sure > where that sendmail command is. Don't see it in voicemail.conf or any > other > config in the asterisk directory. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of VoIP > Street > Sent: Wednesday, July 12, 2006 12:01 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Email notification of voicemail > > Kevin Savoy wrote: > > Asterisk is trying to send an email to users when they receive a > > voicemail. Can this be shut off? I have not entered any email > addresses > > in voicemail.conf so it tries to send to > [EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>. This of course gets > rejected > > since the user does not exist and the root users mailbox on linux > gets > > full of these rejection notices. I can't seem to find anywhere to > tell > > Asterisk to stop notifying people they have voicemails. > > > > > > > > I'm using 1.2.9.1 of Asterisk. Thanks > > > > > > > > _ > > > > > > > > **Kevin Savoy** > > > > **Business Unit Telecom Analyst** > > > > 2218 4th Ave W > > > > Williston, ND 58801 > > > > Ph: 701-774-4023 > > > > Fax: 701-774-2901 > > > > http://www.novo1.com > > > > Novo 1 is a service mark of Novo 1, Inc > > > > > > > > > > > --- > - > > > > ___ > > --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 > > You could try commenting out: > > attach=yes > > Also, if you don't want any emails sent ever for any voice mail users > you could probably uncomment the following line and give it a bogus > path > to the mailer. > > ;mailcmd=/usr/sbin/sendmail -t > > There is probably a better way to do this but we have never needed to > turn it off so I am not sure. > > Hope this helps. > > -- > VoIP Street > Origination/Termination with SUPERIOR customer service! > http://www.VoIPstreet.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 > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] quad T1 pri
Should be span 1 for the for T1 and span 2 for the second T1 in your config. They are both span 1. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Thursday, July 13, 2006 12:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] quad T1 pri I am connecting my Quad T1 card to the Siemens HIpath 4000 with pri. Port 1 is working just fine. But port 2 is not working. I think I have my configuration correct (see below). Is there something special about configuring 2 PRI? I have done it with dual T1 no problem. I am using asterisk 1.2.9.1 libpri-1.2.3 and zaptel 1.2.6. THanks, Jerry -- zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=1,1,0,esf,b8zs bchan=25-47 dchan=48 zapata.conf: signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel => 1-23 signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel => 25-47 ___ --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] Voicemail & CallerID
Would this work?
exten => 3299,1,VoicemailMain(${EXTEN})
This way it would check the voicemail of the extension doing the dialing?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peder @
NetworkOblivion
Sent: Thursday, July 13, 2006 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail & CallerID
I've got a question about voicemail and callerid and I can't quite
figure it out. I've got extensions 100, 101 and 102. For outbound
callerID (calls from the phones to the PSTN), I want the callerid to say
100 on all phones, so under sip.conf, I added:
callerid="Bill" <100>
The problem is that when they go to check voicemail, it looks at their
callerID and it drops them into mailbox 100 (calls to them still go into
their own specific mailbox, it is just when they hit their messages
button). Any idea how to get around that? Or do I just have to send
them to voicemail without having it automatically enter their extension?
This is what my voicemail does:
exten => 3299,1,VoicemailMain(${CALLERIDNUM})
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RE: [asterisk-users] Email notification of voicemail
I have attach=no in my voicemail.conf so that can't be doing it. Not sure where that sendmail command is. Don't see it in voicemail.conf or any other config in the asterisk directory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Street Sent: Wednesday, July 12, 2006 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Kevin Savoy wrote: > Asterisk is trying to send an email to users when they receive a > voicemail. Can this be shut off? I have not entered any email addresses > in voicemail.conf so it tries to send to [EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>. This of course gets rejected > since the user does not exist and the root users mailbox on linux gets > full of these rejection notices. I can't seem to find anywhere to tell > Asterisk to stop notifying people they have voicemails. > > > > I'm using 1.2.9.1 of Asterisk. Thanks > > > > _ > > > > **Kevin Savoy** > > **Business Unit Telecom Analyst** > > 2218 4th Ave W > > Williston, ND 58801 > > Ph: 701-774-4023 > > Fax: 701-774-2901 > > http://www.novo1.com > > Novo 1 is a service mark of Novo 1, Inc > > > > > > > ___ > --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 You could try commenting out: attach=yes Also, if you don't want any emails sent ever for any voice mail users you could probably uncomment the following line and give it a bogus path to the mailer. ;mailcmd=/usr/sbin/sendmail -t There is probably a better way to do this but we have never needed to turn it off so I am not sure. Hope this helps. -- VoIP Street Origination/Termination with SUPERIOR customer service! http://www.VoIPstreet.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 ___ --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] Email notification of voicemail
Asterisk is trying to send an email to users when they receive a voicemail. Can this be shut off? I have not entered any email addresses in voicemail.conf so it tries to send to [EMAIL PROTECTED]. This of course gets rejected since the user does not exist and the root users mailbox on linux gets full of these rejection notices. I can’t seem to find anywhere to tell Asterisk to stop notifying people they have voicemails. I’m using 1.2.9.1 of Asterisk. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Unable to configure my DID number
Are you sure they are sending you all 10
digits and not just the last four? Our provider just sends the last four digits
on DID. If this is the case you would have this:
exten => 4567,1,Answer()
exten => 4567,1,DIAL(SIP/user,20)
Hope this helps.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Monday, July 10, 2006 7:45
AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users] Unable
to configure my DID number
Hi friends,
At present, I am making outgoing calls using Teliax service with Asterisk. But,
I am unable to receive calls. My DID number is: 3031234567. I am using SIP
Server (Asterisk) setup, which is provided on Teliax website support. I have
replaced my DID number i.e., 3031234567 in YOURNUMBER. But, I am unable to
receive calls.
My configuration file in extensions.conf File:
exten => _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten => 3031234567,1,Answer()
exten => 3031234567,1,DIAL(SIP/user,20)
---
I hope the above configuration is proper, If not please suggest the
modifications.
In addition, I have some doubts.
1) How should I configure my DID number in extensions.conf file to recevice
incoming calls?
2) Are they any modifications required in "Features" option in my
account on Teliax website?
3) To receive incoming calls, do I need to make any kind of modifications to
other configuration files in "Asterisk" and setup DID number?
4) Do I need to set Public IP in my Asterisk server or our local IP is enough?
4) After configuring DID number, where can I receive the phone call (ring)?
5) How can I setup IVR (Interactive Voice Response) system to my DID number.
(i.e., If someone calls to my DID number, then our IVR (Welcome message) should
respond and ask for extension number.)
Please respond to this message ASAP. Looking forward to your response.
Thank you.
Regards,
Chandra.
How low will we go? Check out Yahoo! Messenger’s low PC-to-Phone
call rates.
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RE: [asterisk-users] Zap Channel not hanging up on Telco side
No I have not using bristuff. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stoffell Sent: Thursday, July 06, 2006 3:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Channel not hanging up on Telco side On 7/6/06, Kevin Savoy <[EMAIL PROTECTED]> wrote: > I'm having an issue where Asterisk hangs up a call (either party hangs up) but the telco side > of the T1, both the local company and AT&T, does not receive the hangup signal from > Asterisk. Therefore Asterisk thinks the channel is available but it's still off-hook on the telco I have not experienced this on a standard asterisk yet, but I did on a bristuff version (the latest 0.3.0pre-1 series, being n, o, p and q). However, there's a patch to libpri for this in the mailing list. Are you using the bristuff'ed version? cheers ___ --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] Zap Channel not hanging up on Telco side
Good point. I am using Asterisk 1.2.9.1, Zaptel 1.2.6. libpri 1.2.3. I had the same issue with Asterisk 1.2.7.1, Zaptel 1.2.5 and libpri 1.2.2. My Zapata.conf looks like this: [channels] context=default musiconhold=default resetinterval=60 ;AT&T T1's group=1 switchtype=national signalling=em_w channel => 1-96 ;Local T1's group=2 switchtype=national signalling=em_w channel => 97-168 My Zaptel.conf looks like this. We are using RedFone's Fonebridges hence the dynamic channels but I was able to re-create this with the wct4xxp card we have as well. # #AT&T # dynamic=eth,eth0/00:0C:42:03:63:0F/0,24,1 e&m=1-24 dynamic=eth,eth0/00:0C:42:03:63:0F/1,24,2 e&m=25-48 dynamic=eth,eth0/00:0C:42:03:63:0F/2,24,3 e&m=49-72 dynamic=eth,eth0/00:0C:42:03:63:0F/3,24,4 e&m=73-96 # #Local # dynamic=eth,eth0/00:0C:42:03:63:17/0,24,1 e&m=97-120 dynamic=eth,eth0/00:0C:42:03:63:17/1,24,2 e&m=121-144 dynamic=eth,eth0/00:0C:42:03:63:17/2,24,3 e&m=145-168 #dynamic=eth,eth0/00:0C:42:03:63:17/3,24,4 #e&m=169-192 loadzone=us defaultzone=us -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Thursday, July 06, 2006 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Channel not hanging up on Telco side Kevin Savoy wrote: > > I'm having an issue where Asterisk hangs up a call (either party hangs > up) but the telco side of the T1, both the local company and AT&T, > does not receive the hangup signal from Asterisk. Therefore Asterisk > thinks the channel is available but it's still off-hook on the telco > > Does us no good, not knowing what version of Asterisk, or seeing how you have it configured. How about showing your zaptel, zapata and relevant configs? Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Channel not hanging up on Telco side
I’m having an issue where Asterisk hangs up a call (either party hangs up) but the telco side of the T1, both the local company and AT&T, does not receive the hangup signal from Asterisk. Therefore Asterisk thinks the channel is available but it’s still off-hook on the telco side. I have confirmed this with AT&T that the channels are open on their side but not in Asterisk. Any ideas what is causing this? The CLI does show the call hanging up. This is not on every call and I’ve not been able to determine yet when it locks and when it doesn’t. My guess at this point is when a call is placed but the caller does not wait for the answering party to answer and hangs up. Then the called party line keeps ringing, I’ve confirmed this calling my cell phone, and then the called party answers. At this point I see the CLI state that it is starting a simple switch, as if the call is reconnecting to Asterisk and then hangs up since I have the s,1,Hangup() and Asterisk at this point doesn’t know where the call should go. I believe at this point the call is hungup in Asterisk but NOT on the telco side. Any ideas how to get around this??? Thanks _____ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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: Asterisk in Seattle
I'm in Williston, North Dakota and we have an office in Billings, MT. He's right. We are 500 miles form civilization! :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, July 05, 2006 10:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.com Subject: RE: [asterisk-users] RE: Asterisk in Seattle It can't be that bad there in Anchorage. I'm in Billings, MT, which is about half the size of Anchorage, and sometimes (no, wait... most times) it seems like I'm 500 miles from civilsation. Wait, I AM 500 miles from civilisation! -Original Message- From: Josh Reineke [mailto:[EMAIL PROTECTED] Sent: Wed 7/5/2006 8:03 PM To: asterisk-users@lists.digium.com Cc: Subject: [asterisk-users] RE: Asterisk in Seattle I work for a medium size business in Anchorage, AK running two installations with about 30 handsets a piece. They've both been in service for a couple of years. I'm in Seattle fairly frequently, being it's the metropolis closest to Anchorage. I'd be jazzed if there was a user group there and would be willing to help in it's formation. Josh Message: 15 Date: Wed, 5 Jul 2006 14:00:35 -0600 From: "Douglas Garstang" <[EMAIL PROTECTED]> Subject: [asterisk-users] Asterisk in Seattle To: "Asterisk Users Mailing List - Non-Commercial Discussion" Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" All, Anyone know of any companies (small, large) that are using, experimenting with, deploying, and so on, Asterisk in Washington state, most likely in and around Seattle? I'm curious from an employment perspective. :) Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Possible Bug?
I believe I’ve found a possible bug in the Zaptel channel drivers. I’ve been able to recreate this on a couple of servers. One with Asterisk 1.2.7.1 and 1.2.9.1,and also Zaptel 1.2.5 and Zaptel 1.2.6. I was testing a server configuration with some T1’s we have with AT&T. When I disconnected the AT&T T1’s we were no longer able to check voicemail. I noticed then that no messages from the server could be heard. Playback, voicemail or any other message the server would play to the caller on the phone. It would display on the CLI that it was playing the message but nothing could be heard and it would hang the call until it timed out. It took a lot of trial and error before I figured out that if I unloaded all related files to the Zaptel drivers the messages could then be heard. Reactivated the Zaptel with no T1’s attached and again it killed the messages. One server has a Wct4xxp card and the other using dynamic with Redfone’s from Fonbridge. Both reacted the same way. This would not be good if we had a fiber cut somewhere and then our users could no longer use their voicemail or any other application that played sounds. Can anyone else recreate this? Do any of you think this is a bug? Should I submit a bug report? I’ve heard to submit a bug to send it to mantis. What is this and where or how do I do this? Thanks _____ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Dropping incompatible voice frame
This didn’t work for me either. I tried using the patch at the link below and it didn’t work either. If I were to guess what was happening here, it would be when the call is forwarded by the phone Asterisk doesn’t know which device to send the call to. How does it know to open a Zap channel and dial the command? What tells Asterisk to open Zap channel and dial the number the phone had it it’s forward? Am I off track here? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Wednesday, June 28, 2006 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dropping incompatible voice frame This is known issue, we fixed it by putting an answer() in the dial plan before it gets forwarded, the fix transcode_via_sln=no (detailed in the bug tracker) didn't work for me. YMMV. http://bugs.digium.com/view.php?id=4101 On 6/28/06, Kevin Savoy <[EMAIL PROTECTED]> wrote: Sorry if this has been posted before but I'm having an issue where I get the following on my CLI. ast_read: Dropping incompatible voice frame on Local/XX of format ulaw since our native form has changed to slin A call comes in on our main to toll free number on an AT&T T1 line and is sent to phone 4000. This is our secretary's desk. If she leaves the desk she forwards the phone to one of our sister companies so that they would answer the call. This call is sent back out the AT&T T1. If she answers the call and then forwards outside the building it works fine but if she forwards her phone outside the building to auto forward the call when she is away from her desk we get the above error. I have recreated this on my own phone (both hers and mine are Polycom 501's) and with a Cisco 7960. I also tried a different toll free number with the same results. I searched the internet and found four people having the same issue but none have gotten responses on how to fix it. Each time it was something similar where the call was redirected. I know the T1's are configured correctly because all other incoming and outgoing calls work fine until this error occurs. Then nothing works. I am using Asterisk 1.2.7.1 with Zaptel 1.2.5 and Libpri 1.2.2 . I have tried using both a digium Wctxxp 4 port and RedFone's Fonebridges and have gotten the same result both ways so the problem is within Asterisk itself. I also tried allow=all in sip.conf as well as specifically listing allow= slin and all other formats to no avail. Also when this happens the channel is no longer usable even though Asterisk thinks it is available. When the next call is placed it times out because that channel has been locked by the above error. The only way out is a complete reboot and reset of all systems. Not good. Any help would be greatly appreciated. If I had hair left I'd be pulling it out about now. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] bristuff hangup issue
I can also add that this happens on em_w lines as well. I've had issues where callers start getting dead air when dialing out. Talking with the phone company the lines were in an off-hook state even though Asterisk hung up the call. I done exactly as below where I hang up before the other party answers the call. I've also had where after I hang up the CLI shows the call hanging up and then another call starts, starting simple switch, as if the call was re-established but Asterisk doesn't know what to do with the call and executes the s,1,hangup() on the call. This does NOT however always hang up the call on the AT&T side. The T1 still shows the call as off-hook even though it's not in use. It seems random (at least I haven't figured out the pattern yet) as to when the channel gets hung up properly on AT&T's side and when it's not. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stoffell Sent: Thursday, June 29, 2006 3:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] bristuff hangup issue hi, Just wanted to inform everyone, if you're using the latest bristuff's you might (depends on the country!) have hangup issues. The issue appears every time you dial an external number, and hangup after letting it ring for a few times. Then the remote party keeps ringing. In some situations (we only encountered this while dialing to other * servers) it keeps the line open on the telco-side. Meaning.. you pay for it! The cdr on the calling asterisk (with the bug) doesn't indicate a long connection time. However, the cdr on the called asterisk does.. (I've seen several durations of over 20 hours) A show channels doesn't indicate any active calls. A quick fix has been posted a while ago by Marcel van der Boom (in libpri/q931.c), this works. According to the release notes this should have been applied to the latest bristuff, but be careful, the problem still exists on bristuff-0.3.0-PRE-1q. I have emailed junghanns.net to let them know. Best regards, stoffell ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
Sorry if this has been posted before but I’m having an issue where I get the following on my CLI. ast_read: Dropping incompatible voice frame on Local/XX of format ulaw since our native form has changed to slin A call comes in on our main to toll free number on an AT&T T1 line and is sent to phone 4000. This is our secretary’s desk. If she leaves the desk she forwards the phone to one of our sister companies so that they would answer the call. This call is sent back out the AT&T T1. If she answers the call and then forwards outside the building it works fine but if she forwards her phone outside the building to auto forward the call when she is away from her desk we get the above error. I have recreated this on my own phone (both hers and mine are Polycom 501’s) and with a Cisco 7960. I also tried a different toll free number with the same results. I searched the internet and found four people having the same issue but none have gotten responses on how to fix it. Each time it was something similar where the call was redirected. I know the T1’s are configured correctly because all other incoming and outgoing calls work fine until this error occurs. Then nothing works. I am using Asterisk 1.2.7.1 with Zaptel 1.2.5 and Libpri 1.2.2. I have tried using both a digium Wctxxp 4 port and RedFone’s Fonebridges and have gotten the same result both ways so the problem is within Asterisk itself. I also tried allow=all in sip.conf as well as specifically listing allow=slin and all other formats to no avail. Also when this happens the channel is no longer usable even though Asterisk thinks it is available. When the next call is placed it times out because that channel has been locked by the above error. The only way out is a complete reboot and reset of all systems. Not good. Any help would be greatly appreciated. If I had hair left I’d be pulling it out about now. Thanks _____ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Turning AAAH into a call-center
Speaking as one of those call centers we are looking at doing a turn over to Asterisk from our Nortel systems and are doing it ourselves. We've looked at a lot of packages from Fonality, Signate, Aheeva and others and none fit our needs. Each has good aspects but none have all of what we need. Below he states extensive reporting in QueueMetrics. Yes and no. Extensive but limited to a few areas. Agent level reporting doesn't go deep enough. There is no DNIS reporting. In our business that is critical. We have 40 to 50 queues and some queues can have up to 1,000 toll free numbers pointed at it. Our clients want to know how many calls on each individual toll free and all the statistics that go with it. Talk time, abandons, thresholds, etc. The commercial products are fine products but not in depth enough for a larger call center. We have decided to look into doing it ourselves and found that it wasn't really that difficult to find all the numbers we needed. It left me feeling that these companies didn't really spend a lot of time on reporting as we found all we needed pretty quickly. The previously stated companies do fine with small call centers or call centers with only a few clients. We're having more issues with finding a phone to fit our business then we are software for Asterisk. I'm disappointed that all these phone manufacturers have never considered the call center when designing phones. Nortel and Avaya have phones specifically designed for call centers. They don't have handsets. Only headsets. Sadly none of the VoIP phones are designed that way. Anyways that's my 2 cents on this. (if anyone cares) :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: Tuesday, May 16, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Turning AAAH into a call-center I believe there are quite different levels for the Asterisk market, so most people who run call centers wont feel confident in downloading a couple of ISOs from the internet and setting things up themselves. l. In data Mon, 15 May 2006 21:47:09 +0200, Steve Totaro <[EMAIL PROTECTED]> ha scritto: > I bet Signate will love this. > > Lenz wrote: >> Hello list, >> we have prepared a short tutorial that will teach you to turn your >> [EMAIL PROTECTED] box into a full-fledged call center within minutes, with >> both always-on and callback agents available and the very extensive >> reporting facilities that QueueMetrics provides. >> >> You can download it from the donwloads page at >> http://queuemetrics.loway.it/download.jsp >> >> QueueMetrics is a full-fledged call-center monitoring system and it is >> available for free to home users, SOHOs and individual enthusiasts. >> >> Any comment on the document is welcome. >> l. >> >> >> --Loway Research - Home of QueueMetrics >> http://queuemetrics.loway.it >> >> ___ >> --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 -- Assum est, versa et manduca. ___ --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] Call Center Phone with Auto Answer
This is an idea I've had for long term. Might have to push it up a bit. Not with astGUI but a similar CTI idea. Using the inbound DNIS to trigger screen pops. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Tuesday, May 09, 2006 6:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Hello, It may be more than you want, but astGUIclient/VICIDIAL has screen pops for inbound calls through a web browser. It is also Open-source. http://astguiclient.sf.net MATT--- On 5/9/06, Steve Totaro <[EMAIL PROTECTED]> wrote: > I do but I didn't write it and it was on company time so I cannot > share. Sorry. Just throwing out ideas and possibilities. > > > > Junaid Uppal wrote: > > Hi Steve , > > > > I was actually looking forward for the same thing , do y ou have > > something like this , as an example? > > > > regards > > > > Junaid Uppal > > > > > > On 5/9/06, * Steve Totaro* <[EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>> wrote: > > > > Use an activex screenpop. > > > > Thanks, > > Steve Totaro > > > > > -Original Message- > > > From: Kevin Savoy [mailto:[EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>] > > > Sent: Monday, May 08, 2006 3:32 PM > > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > > Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer > > > > > > This may be the way to go but not the best. Our agents frankly > > aren't > > the > > > brightest people and I can see them forgetting it as soon as it is > > said to > > > them, or they are not paying attention and missing the announcement > > but it > > > is something to look into. Thanks > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]> > > > [mailto: [EMAIL PROTECTED] > > <mailto:[EMAIL PROTECTED]>] On Behalf Of Time > > Bandit > > > Sent: Monday, May 08, 2006 2:23 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer > > > > > > > Ok I can get this to work now the next problem is since the agent > > stays > > > > "off-hook" when a call is presented to them there is no indication > > of > > > what > > > > call this is. Being an inbound call center we have 100's of > > clients. > > > 1,000's > > > > of toll frees and DNIS. We use the Asterisk callerID function to > > assign > > > a > > > > name to each call so that when the call is presented to the > > agent it > > > > displays which company the call is for. With AgentLogin all the > > agent > > > gets > > > > is the number they dialed to log in. No idea which client this > > call > > is > > > for. > > > > Any ideas there? > > > When you send the caller to the queue, you can pass the name of the > > > audio file to be played as the announcement to the agent when he > > gets > > > the call. Maybe you could use that and pre-record the name of the > > > customer, passing that audio file > > > > > > something like "exten => > > > 8000,n,Queue(8000|t||FilenameOfTheCustomerName|300)" > > > > > > maybe you could also use festival > > > > > > hth > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com > > <http://Easynews.com> -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > <http://lists.digium.com/mailman/listinfo/asterisk-users> > > > > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com > > <http://Easynews.com> -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > >
RE: [Asterisk-Users] Call Center Phone with Auto Answer
I see two problems with doing it this way. One the agent is not paying attention and misses the announcement. Two we have some queues that can have as many 800+ toll free numbers pointed at the same queue. The agents need to be able to know which of the toll free numbers was dialed. We do that in our current system by putting the last four digits of the toll free on the phone display. With the below scenario it will not only have to have a variable as to which account but also say out the digits of the last four but only on certain accounts. That is possible I realize but messy. I need a combination of the two scenarios. AgentCallBackLogin so that I can have a phone display but AgentLogin so that there is no Call Back and just presents the call. Unless I can come up with something I’m guessing we are going to have to write some kind of program that can do this instead. Anyone have anything else to add? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Junaid Uppal Sent: Tuesday, May 09, 2006 2:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Hi Steve , I was actually looking forward for the same thing , do y ou have something like this , as an example? regards Junaid Uppal On 5/9/06, Steve Totaro <[EMAIL PROTECTED]> wrote: Use an activex screenpop. Thanks, Steve Totaro > -Original Message- > From: Kevin Savoy [mailto:[EMAIL PROTECTED]] > Sent: Monday, May 08, 2006 3:32 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer > > This may be the way to go but not the best. Our agents frankly aren't the > brightest people and I can see them forgetting it as soon as it is said to > them, or they are not paying attention and missing the announcement but it > is something to look into. Thanks > > -Original Message- > From: [EMAIL PROTECTED] > [mailto: [EMAIL PROTECTED]] On Behalf Of Time Bandit > Sent: Monday, May 08, 2006 2:23 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer > > > Ok I can get this to work now the next problem is since the agent stays > > "off-hook" when a call is presented to them there is no indication of > what > > call this is. Being an inbound call center we have 100's of clients. > 1,000's > > of toll frees and DNIS. We use the Asterisk callerID function to assign > a > > name to each call so that when the call is presented to the agent it > > displays which company the call is for. With AgentLogin all the agent > gets > > is the number they dialed to log in. No idea which client this call is > for. > > Any ideas there? > When you send the caller to the queue, you can pass the name of the > audio file to be played as the announcement to the agent when he gets > the call. Maybe you could use that and pre-record the name of the > customer, passing that audio file > > something like "exten => > 8000,n,Queue(8000|t||FilenameOfTheCustomerName|300)" > > maybe you could also use festival > > hth > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Center Phone with Auto Answer
This may be the way to go but not the best. Our agents frankly aren't the brightest people and I can see them forgetting it as soon as it is said to them, or they are not paying attention and missing the announcement but it is something to look into. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Monday, May 08, 2006 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer > Ok I can get this to work now the next problem is since the agent stays > "off-hook" when a call is presented to them there is no indication of what > call this is. Being an inbound call center we have 100's of clients. 1,000's > of toll frees and DNIS. We use the Asterisk callerID function to assign a > name to each call so that when the call is presented to the agent it > displays which company the call is for. With AgentLogin all the agent gets > is the number they dialed to log in. No idea which client this call is for. > Any ideas there? When you send the caller to the queue, you can pass the name of the audio file to be played as the announcement to the agent when he gets the call. Maybe you could use that and pre-record the name of the customer, passing that audio file something like "exten => 8000,n,Queue(8000|t||FilenameOfTheCustomerName|300)" maybe you could also use festival hth ___ --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] Call Center Phone with Auto Answer
Ok I can get this to work now the next problem is since the agent stays “off-hook” when a call is presented to them there is no indication of what call this is. Being an inbound call center we have 100’s of clients. 1,000’s of toll frees and DNIS. We use the Asterisk callerID function to assign a name to each call so that when the call is presented to the agent it displays which company the call is for. With AgentLogin all the agent gets is the number they dialed to log in. No idea which client this call is for. Any ideas there? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 08, 2006 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer That is correct. Just use IAX trunking and speex. You will be fine. Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 12:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Because we will have many of these phones in remote locations and we don’t want to be chewing up bandwidth with agents not on calls. Am I making the right assumption here that phones that are idle will not be taking up bandwidth where ones with MOH playing would be? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 08, 2006 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Why not just use AgentLogin and let them listen to music until a call comes in? Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 11:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Correct. We have to hit the “answer” button. In a call center environment such as ours we don’t want to give the agents the option of not answering the call when they are logged in. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 08, 2006 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Does the phone ring, just not auto-answer? Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Firstly the auto-answer on both the 301 and 501 phone is set to on, but it doesn’t seem to have an effect. I’ll have to look into this _ALERT_INFO variable. Not much experience with it here. Could you give me a dial plan example that would work? Here is what we have now. exten=>3472,1,Answer() exten=>3472,2,Wait(1) exten=>3472,3,Playback(this-call-may-be-monitored-or-recorded) exten=>3472,4,SetCallerID(ICS) exten=>3472,5,Queue(ICS) What can I add to this to make the phone auto-answer? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer Sent: Friday, May 05, 2006 6:29 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of examples, or design your dialplan such that the appropriate _ALERT_INFO variable is set where the queue is concerned. p From: "Kevin Savoy" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Date: Fri, 5 May 2006 15:31:41 -0500 Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer The problem with what is in wiki is that these calls are being sent to a queue. There is no way to have the queue dial the preceding digit that I can think of that would trigger this. In the example shown he has an 8 dialed before the extension. How would I get Asterisk to dial an 8 before sending the call to the logged in agent in the queue? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Friday, May 05, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Kevin Savoy wrote: > Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agent
RE: [Asterisk-Users] Call Center Phone with Auto Answer
Because we will have many of these phones in remote locations and we don’t want to be chewing up bandwidth with agents not on calls. Am I making the right assumption here that phones that are idle will not be taking up bandwidth where ones with MOH playing would be? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 08, 2006 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Why not just use AgentLogin and let them listen to music until a call comes in? Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 11:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Correct. We have to hit the “answer” button. In a call center environment such as ours we don’t want to give the agents the option of not answering the call when they are logged in. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 08, 2006 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Does the phone ring, just not auto-answer? Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Firstly the auto-answer on both the 301 and 501 phone is set to on, but it doesn’t seem to have an effect. I’ll have to look into this _ALERT_INFO variable. Not much experience with it here. Could you give me a dial plan example that would work? Here is what we have now. exten=>3472,1,Answer() exten=>3472,2,Wait(1) exten=>3472,3,Playback(this-call-may-be-monitored-or-recorded) exten=>3472,4,SetCallerID(ICS) exten=>3472,5,Queue(ICS) What can I add to this to make the phone auto-answer? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer Sent: Friday, May 05, 2006 6:29 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of examples, or design your dialplan such that the appropriate _ALERT_INFO variable is set where the queue is concerned. p From: "Kevin Savoy" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Date: Fri, 5 May 2006 15:31:41 -0500 Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer The problem with what is in wiki is that these calls are being sent to a queue. There is no way to have the queue dial the preceding digit that I can think of that would trigger this. In the example shown he has an 8 dialed before the extension. How would I get Asterisk to dial an 8 before sending the call to the logged in agent in the queue? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Friday, May 05, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Kevin Savoy wrote: > Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it. > My Polycom phones support auto-answer. This link should get you started. http://www.voip-info.org/wiki-Polycom+auto-answer+config Mike ___ --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 Get amazing travel prices for air and hotel in one click on Yahoo! FareChase ___ --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] Call Center Phone with Auto Answer
Correct. We have to hit the “answer” button. In a call center environment such as ours we don’t want to give the agents the option of not answering the call when they are logged in. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 08, 2006 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Does the phone ring, just not auto-answer? Thanks, Steve Totaro From: Kevin Savoy [mailto:[EMAIL PROTECTED] Sent: Monday, May 08, 2006 10:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer Firstly the auto-answer on both the 301 and 501 phone is set to on, but it doesn’t seem to have an effect. I’ll have to look into this _ALERT_INFO variable. Not much experience with it here. Could you give me a dial plan example that would work? Here is what we have now. exten=>3472,1,Answer() exten=>3472,2,Wait(1) exten=>3472,3,Playback(this-call-may-be-monitored-or-recorded) exten=>3472,4,SetCallerID(ICS) exten=>3472,5,Queue(ICS) What can I add to this to make the phone auto-answer? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer Sent: Friday, May 05, 2006 6:29 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of examples, or design your dialplan such that the appropriate _ALERT_INFO variable is set where the queue is concerned. p From: "Kevin Savoy" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Date: Fri, 5 May 2006 15:31:41 -0500 Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer The problem with what is in wiki is that these calls are being sent to a queue. There is no way to have the queue dial the preceding digit that I can think of that would trigger this. In the example shown he has an 8 dialed before the extension. How would I get Asterisk to dial an 8 before sending the call to the logged in agent in the queue? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Friday, May 05, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Kevin Savoy wrote: > Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it. > My Polycom phones support auto-answer. This link should get you started. http://www.voip-info.org/wiki-Polycom+auto-answer+config Mike ___ --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 Get amazing travel prices for air and hotel in one click on Yahoo! FareChase ___ --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] Call Center Phone with Auto Answer
Firstly the auto-answer on both the 301 and 501 phone is set to on, but it doesn’t seem to have an effect. I’ll have to look into this _ALERT_INFO variable. Not much experience with it here. Could you give me a dial plan example that would work? Here is what we have now. exten=>3472,1,Answer() exten=>3472,2,Wait(1) exten=>3472,3,Playback(this-call-may-be-monitored-or-recorded) exten=>3472,4,SetCallerID(ICS) exten=>3472,5,Queue(ICS) What can I add to this to make the phone auto-answer? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philippe Lindheimer Sent: Friday, May 05, 2006 6:29 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of examples, or design your dialplan such that the appropriate _ALERT_INFO variable is set where the queue is concerned. p From: "Kevin Savoy" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Date: Fri, 5 May 2006 15:31:41 -0500 Subject: RE: [Asterisk-Users] Call Center Phone with Auto Answer The problem with what is in wiki is that these calls are being sent to a queue. There is no way to have the queue dial the preceding digit that I can think of that would trigger this. In the example shown he has an 8 dialed before the extension. How would I get Asterisk to dial an 8 before sending the call to the logged in agent in the queue? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Friday, May 05, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Kevin Savoy wrote: > Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it. > My Polycom phones support auto-answer. This link should get you started. http://www.voip-info.org/wiki-Polycom+auto-answer+config Mike ___ --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 Get amazing travel prices for air and hotel in one click on Yahoo! FareChase ___ --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] Call Center Phone with Auto Answer
We are using agent login but we don't want MOH on the line at all times as some of these phones could and probably will be connected in remote locations. We don't want to stream MOH across frames chewing up bandwidth when there are no calls to present to that phone. We do have the ackcall=no in the agents.conf and it seems to have no affect. Am I missing something here? Appreciate any help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cook Sent: Friday, May 05, 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Kevin Savoy wrote: > > Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it. > > Thanks > May be I am not understanding... Why not use agentlogin and have the agents always logged in with MOH... they get a beep and they are connected.. Change ackcall=no in agents.conf Then you don't need auto-answer ___ --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] Call Center Phone with Auto Answer
The problem with what is in wiki is that these calls are being sent to a queue. There is no way to have the queue dial the preceding digit that I can think of that would trigger this. In the example shown he has an 8 dialed before the extension. How would I get Asterisk to dial an 8 before sending the call to the logged in agent in the queue? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Friday, May 05, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Kevin Savoy wrote: > Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it. > My Polycom phones support auto-answer. This link should get you started. http://www.voip-info.org/wiki-Polycom+auto-answer+config Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Center Phone with Auto Answer
Can anyone recommend a phone to use in an inbound call center environment that has an auto answer feature? We don’t want the agents having to acknowledge the call. The call should just activate on the headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. None of these support it. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Dumping queue_log to MySQL
Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don’t get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. We will be using this for a call center and need more reliability. Anyone got one working? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Auto Logout from queue
I have tried using the autologoff in the agents.conf and it sort of works. I set it to 5 seconds to test it and it has taken anywhere from 35 to 60 seconds to actually do something at which point it does indeed log out the agent. I don't want to be pestering agents with test calls to see if they are indeed there so the below scripting isn't really practical in our environment. Can anyone tell me why the agents.conf file setting doesn't work as described? If it is set to 5 it should log them off after 5 seconds or so not 30 - 60 seconds. I don't really want the call sitting at a logged out agents phone for anymore then 5 seconds when there are other agents out there waiting to take that call. Any ideas? Thanks _____ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Tuesday, April 25, 2006 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Auto Logout from queue Via dialplan maybe? exten => xxx,1,Dial(SIP/101_Queue,20,tr) exten =>xxx,2,RemoveQueueMember(Comercial_Queue,SIP/101_Queue,1) Kerry Garrison escribió: > Yes, that is the functionality I am looking for, just not sure how exactly > to pull that off. > > > _ > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Alexander > Lopez > Sent: Tuesday, April 25, 2006 12:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Auto Logout from queue > > > Use the local channel to call the agent first, and if there is no answer, > log them out. > > > > _ > > From: [EMAIL PROTECTED] on behalf of Kerry Garrison > Sent: Tue 4/25/2006 2:38 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] Auto Logout from queue > > > i have a client that wants a function that will automatically logout an > agent from a queue if they do not answer a call. This would prevent future > calls from being sent to that phone if the agent forgot to logout. Any > ideas? > > Kerry Garrison > Director of Technical Services > Tech Data Pros - Orange County's Mobile IT Service Provider > (949) 502-7819 x200 - <mailto:[EMAIL PROTECTED]> > [EMAIL PROTECTED] > <http://www.techdatapros.com/> http://www.techdatapros.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 > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Status of Queue
Is there variable or a way I can check to ensure that an agent is logged into a particular queue? I don’t want to queue a call up to a queue that no one is logged into. I would like to have the call redirected to another extension if there are no queue members. Thanks for any insight _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] PRI blocking on incoming calls
We tried the pri intese debug last night but this only showed us what was going on with the Nortel trunk not the MCI T1 side. Sadly it was of no use. Any other ideas? Anyone?? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles Sent: Tuesday, April 18, 2006 2:48 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PRI blocking on incoming calls Ok. First of all , be sure Redfone ethernet link and the Asterisk ethernet link are both on the same switch segment. Then try an “pri intense debug” on asterisk console. I believe (not sure), this link is not at IP level but ethernet level 2 It can help to determine if packets get stucked into the redphone or it is an issue related to Asterisk. Hope it helps Ing. Oscar Andrés Carriles Presidente InFoDaX Consultants Nicolás Jorge 994 (B1706AVA) Haedo Buenos Aires, Argentina Tel: 54 11 4650 1775 Fax: 54 11 4650 4295 www.infodax.com.ar -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy Enviado el: Martes, 18 de Abril de 2006 04:03 p.m. Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] PRI blocking on incoming calls We have a crossover from telco to the CSU and a crossover from the CSU to the RedFone and then a regular Ethernet cable from the RedFone to the Asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles Sent: Tuesday, April 18, 2006 2:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PRI blocking on incoming calls I believe it is important to determine if the issue arrives at TDMoE level (RedPhone uses it to avoid a direct T1 link with the box ) Or at PRI level. I am not clear what kink of link bridges your redPhone to Asterisk. Is it an ethernet link or a T1 crossover? Ing. Oscar Andrés Carriles Presidente InFoDaX Consultants Nicolás Jorge 994 (B1706AVA) Haedo Buenos Aires, Argentina Tel: 54 11 4650 1775 Fax: 54 11 4650 4295 www.infodax.com.ar -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy Enviado el: Martes, 18 de Abril de 2006 03:30 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] PRI blocking on incoming calls Ok here is our setup. We are using Asterisk 1.2.6 and Zaptel 1.2.5. We are using RedFone’s FoneBridge’s. We also have a Nortel Option 11C that we have hooked up to the Asterisk. We have 3 T1’s from MCI into one FoneBridge on ports 1 to 3 using d4 and ami signaling. Then we have a local Qwest T1 on port 4 using esf and b8zs. All four are configured with em_w. We are, at this point, only using Asterisk as an IVR with plans to move off the Nortel in the future if we can make this work. We have a second FoneBridge with four PRI’s connected to our Nortel 11C using esf and b8zs and pri_net. The telco T1’s do not have D-Channels but the Nortel do. Calls come into the first FoneBridge and into Asterisk. They are played a message about call recording and then the call is transferred to the Nortel system to be processed by an agent. When we first fire this up all seems to work just fine, calls come in, get the message and then transfer to the Nortel and on to an agent. Everybody is happy. The problem is after 5-20 minutes calls on the MCI lines start getting busy signals. The Qwest line NEVER stops working. We would place a few test calls on the MCI and get busy signals and then they start going through again. A few minutes later they get busy signals again. When we get the busy signals there is no response on the Asterisk CLI with verbose at 10. It’s as if the Asterisk is not ever seeing the call. What is annoying is that it works fine for a bit and then starts hiccupping. Can anyone shed any light on where to look? Any help would be desperately appreciated. Please help. _____ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 ___ --Bandwidth and Coloc
RE: [Asterisk-Users] PRI blocking on incoming calls
We have a crossover from telco to the CSU and a crossover from the CSU to the RedFone and then a regular Ethernet cable from the RedFone to the Asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Carriles Sent: Tuesday, April 18, 2006 2:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] PRI blocking on incoming calls I believe it is important to determine if the issue arrives at TDMoE level (RedPhone uses it to avoid a direct T1 link with the box ) Or at PRI level. I am not clear what kink of link bridges your redPhone to Asterisk. Is it an ethernet link or a T1 crossover? Ing. Oscar Andrés Carriles Presidente InFoDaX Consultants Nicolás Jorge 994 (B1706AVA) Haedo Buenos Aires, Argentina Tel: 54 11 4650 1775 Fax: 54 11 4650 4295 www.infodax.com.ar -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kevin Savoy Enviado el: Martes, 18 de Abril de 2006 03:30 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] PRI blocking on incoming calls Ok here is our setup. We are using Asterisk 1.2.6 and Zaptel 1.2.5. We are using RedFone’s FoneBridge’s. We also have a Nortel Option 11C that we have hooked up to the Asterisk. We have 3 T1’s from MCI into one FoneBridge on ports 1 to 3 using d4 and ami signaling. Then we have a local Qwest T1 on port 4 using esf and b8zs. All four are configured with em_w. We are, at this point, only using Asterisk as an IVR with plans to move off the Nortel in the future if we can make this work. We have a second FoneBridge with four PRI’s connected to our Nortel 11C using esf and b8zs and pri_net. The telco T1’s do not have D-Channels but the Nortel do. Calls come into the first FoneBridge and into Asterisk. They are played a message about call recording and then the call is transferred to the Nortel system to be processed by an agent. When we first fire this up all seems to work just fine, calls come in, get the message and then transfer to the Nortel and on to an agent. Everybody is happy. The problem is after 5-20 minutes calls on the MCI lines start getting busy signals. The Qwest line NEVER stops working. We would place a few test calls on the MCI and get busy signals and then they start going through again. A few minutes later they get busy signals again. When we get the busy signals there is no response on the Asterisk CLI with verbose at 10. It’s as if the Asterisk is not ever seeing the call. What is annoying is that it works fine for a bit and then starts hiccupping. Can anyone shed any light on where to look? Any help would be desperately appreciated. Please help. _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.2/314 - Release Date: 16/04/2006 ___ --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] PRI blocking on incoming calls
Ok here is our setup. We are using Asterisk 1.2.6 and Zaptel 1.2.5. We are using RedFone’s FoneBridge’s. We also have a Nortel Option 11C that we have hooked up to the Asterisk. We have 3 T1’s from MCI into one FoneBridge on ports 1 to 3 using d4 and ami signaling. Then we have a local Qwest T1 on port 4 using esf and b8zs. All four are configured with em_w. We are, at this point, only using Asterisk as an IVR with plans to move off the Nortel in the future if we can make this work. We have a second FoneBridge with four PRI’s connected to our Nortel 11C using esf and b8zs and pri_net. The telco T1’s do not have D-Channels but the Nortel do. Calls come into the first FoneBridge and into Asterisk. They are played a message about call recording and then the call is transferred to the Nortel system to be processed by an agent. When we first fire this up all seems to work just fine, calls come in, get the message and then transfer to the Nortel and on to an agent. Everybody is happy. The problem is after 5-20 minutes calls on the MCI lines start getting busy signals. The Qwest line NEVER stops working. We would place a few test calls on the MCI and get busy signals and then they start going through again. A few minutes later they get busy signals again. When we get the busy signals there is no response on the Asterisk CLI with verbose at 10. It’s as if the Asterisk is not ever seeing the call. What is annoying is that it works fine for a bit and then starts hiccupping. Can anyone shed any light on where to look? Any help would be desperately appreciated. Please help. _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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
