Re: [asterisk-users] "dahdi show channels" no such command
Here are the commands asteriskpbx@pbxmain:/etc/dahdi$ dahdi_cfg -vvv DAHDI Tools Version - 2.7.0.2 DAHDI Version: 2.7.0.1 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08) 8 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 Setting echocan for channel 5 to mg2 Setting echocan for channel 6 to mg2 Setting echocan for channel 7 to mg2 Setting echocan for channel 8 to mg2 asteriskpbx@pbxmain:/etc/dahdi$ pbxmain*CLI> module show like chan_dahdi.so Module Description Use Count chan_dahdi.so DAHDI Telephony Driver w/PRI 0 1 modules loaded > > From: Eric Wieling >To: Joseph Towery ; Asterisk Users Mailing List - >Non-Commercial Discussion >Sent: Thursday, December 5, 2013 9:09 PM >Subject: RE: [asterisk-users] "dahdi show channels" no such command > > >Run "dahdi_cfg -vvv" to verify DAHDI kernel modules are loaded and configured >correctly. > >Verify chan_dahdi.so is built and located in your Asterisk modules directory, >usually /usr/lib/asterisk/modules/ If there is no chan_dahdi.so chances are >you built Asterisk before installing DAHDI. Rerun ./configure or run "make >distclean" and then rerun ./configure to rebuild Asterisk. > >Use the command "module show like dahdi" in the Asterisk CLI to verify >chan_dahdi.so is loaded. If it isn't loaded then chances are you have a >chan_dahdi.conf issue. > > >-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Towery >Sent: Thursday, December 05, 2013 8:41 PM >To: asterisk-users@lists.digium.com >Subject: [asterisk-users] "dahdi show channels" no such command > >I have followed the instructions in Asterisk The Definitive Guide 4th edition. > Once I load DAHDI I run the dahdi show channels command and get no such >command. I have setup all the conf files. I compiled DADHI prior to Asterisk. > >Any ideas? > > > > > > >-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "dahdi show channels" no such command
Run "dahdi_cfg -vvv" to verify DAHDI kernel modules are loaded and configured correctly. Verify chan_dahdi.so is built and located in your Asterisk modules directory, usually /usr/lib/asterisk/modules/ If there is no chan_dahdi.so chances are you built Asterisk before installing DAHDI. Rerun ./configure or run "make distclean" and then rerun ./configure to rebuild Asterisk. Use the command "module show like dahdi" in the Asterisk CLI to verify chan_dahdi.so is loaded. If it isn't loaded then chances are you have a chan_dahdi.conf issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Towery Sent: Thursday, December 05, 2013 8:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] "dahdi show channels" no such command I have followed the instructions in Asterisk The Definitive Guide 4th edition. Once I load DAHDI I run the dahdi show channels command and get no such command. I have setup all the conf files. I compiled DADHI prior to Asterisk. Any ideas? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "dahdi show channels" no such command
On Thu, Dec 5, 2013 at 8:41 PM, Joseph Towery wrote: > I have followed the instructions in Asterisk The Definitive Guide 4th > edition. Once I load DAHDI I run the dahdi show channels command and get no > such command. I have setup all the conf files. I compiled DADHI prior to > Asterisk. > > Any ideas? I am not sure. Could you provide more details? I just tried in my setup and it seems to work (though it tells me nothing exciting): [root@voip ~]# asterisk -r Asterisk 1.8.11-cert1, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Connected to Asterisk 1.8.11-cert1 currently running on voip (pid = 2575) Verbosity is at least 3 voip*CLI> da dahdi data database voip*CLI> dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service voip*CLI> quit [root@voip ~]# > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] "dahdi show channels" no such command
I have followed the instructions in Asterisk The Definitive Guide 4th edition. Once I load DAHDI I run the dahdi show channels command and get no such command. I have setup all the conf files. I compiled DADHI prior to Asterisk. Any ideas?-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging in waves.
I've been working on writing a subroutine to page groups of phones at once
and I'm having some difficulty.
My goal is to have a user call an extension, I record the page they wish to
play, I then page out that recorded file to the phones in groups.
[sub-masspage]
exten => s,1,NoOP
same => n,Answer
same => n,Set(filename=$PAGE)
same => n,Wait(1)
same => n,Record(pagequeue/${filename}%d.gsm,0,30,yk)
same =>
n,Set(DURATION=$[CEIL(${STAT(s,/var/lib/asterisk/sounds/${RECORDED_FILE}.gsm)}
/ 1650)])
same => n,Playback(one-moment-please)
same => n,Set(MUTEAUDIO(all)=on)
same => n,SIPAddHeader(Alert-Info: RingAnswer)
same => n,SipAddHeader(Call-Info: \;answer-after=0)
;same => n,Set(TIMEOUT(absolute)=${DURATION})
same =>
n,ExecIf($[${ISNULL(${ARG1})}=0]?Page(${ARG1},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG2})}=0]?Page(${ARG2},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG3})}=0]?Page(${ARG3},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG4})}=0]?Page(${ARG4},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG5})}=0]?Page(${ARG5},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG6})}=0]?Page(${ARG6},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG7})}=0]?Page(${ARG7},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG8})}=0]?Page(${ARG8},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG9})}=0]?Page(${ARG9},n(${RECORDED_FILE})is),10)
same =>
n,ExecIf($[${ISNULL(${ARG10})}=0]?Page(${ARG10},n(${RECORDED_FILE})is)10)
same => n,Set(MUTEAUDIO(all)=off)
same => n,Playback(goodbye)
same => n,TrySystem(rm -f /var/lib/asterisk/sounds/${RECORDED_FILE}.gsm)
same => n,Hangup
;end sub-masspage
The issue I'm having is the Page command is putting the phone that's doing
the paging into the meetme bridge it's creating and the dialplan stops at
the first Page command.
While I was testing it with a single phone I was using Dial instead of Page
same =>
n,ExecIf($[${ISNULL(${ARG1})}=0]?Dial(${ARG1},15,A(${RECORDED_FILE})S(${DURATION})mg)
))
same =>
n,ExecIf($[${ISNULL(${ARG2})}=0]?Dial(${ARG2},15,A(${RECORDED_FILE})S(${DURATION})mg)))
Which worked great when I sent it one phone in each argument.. Didn't think
about the fact that the first phone that answers is the only phone that
answers till I got deeper into writing it... (It's been a long day)
So I'm at a loss as what to do here, Dial almost does what I need other
than the fact that the first phone that answers is the only phone that
bridges, and Page almost does what I need other than my dialplan execution
stops after the first Page command and it's trying to include the phone I'm
calling from as one of the phones it's paging.
I'm using 1.8 so I've got Meetme and Confbridge available but I'm not sure
what to do unless I start playing with LOCAL channels. Maybe I could have
PAGE call Local Channels that have the TIMEOUT set but then I'd need a way
to pass which phones to dial in somehow.
Suggestions?
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] issue with speech in IVR
On Thu, 5 Dec 2013, Salaheddine Elharit wrote:
i have one question related to the IVR below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
how can i ask the customer to enter a password before to go to (home,s,1)
and where i must to store a password for example password 1234
if the customer enter 1234 he can Goto(home,s,1) and if the password is wrong i
playback an error message
That's 3 questions :)
You need to provide more details.
Is the password fixed or stored in a database? Is it the same as their
voicemail password?
There are examples for all these scenarios. Goggle about, read ATFOT,
visit voip-info.org or use the Asterisk 'help' commands.
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}error
Why are you fiddling with global variables? Isn't
'/var/lib/asterisk/sounds/' your 'default' sounds path?
Please don't top post.
Please trim irrelevent cruft from previous posts.
Please don't burn all your karma points asking simple questions.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] issue with speech in IVR
Look into the Authenticate application
https://wiki.asterisk.org/wiki/display/AST/Application_Authenticate
exten => 600,1,Ringing(2)
exten => 600,n,Answer
exten => 600,n,Authenticate(1234)
exten => 600,n,Goto(home,s,1)
On Thu, Dec 5, 2013 at 3:49 AM, Salaheddine Elharit <
salah.elharit...@gmail.com> wrote:
> hello list
>
> i have one question related to the IVR below
>
> exten => 600,1,Ringing()
> exten => 600,n,Wait(2)
> exten => 600,n,Goto(home,s,1)
>
> how can i ask the customer to enter a password before to go to (home,s,1)
>
> and where i must to store a password for example password 1234
>
> if the customer enter 1234 he can Goto(home,s,1) and if the password is
> wrong i playback an error message
>
> exten => 600,1,Ringing()
> exten => 600,n,Wait(2)
> the customer must enter 1234 if yes go to (home,s,1) if no go to error
> exten => 600,n,Goto(home,s,1)
>
> [error]
>
> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
> exten => s,n,Background(${sounds_path}error
>
>
> any example would be appreciated
>
>
> 2013/11/29 Mitul Limbani
>
>> Sounds cool, I suspected the echo cancel situation, these are usually
>> issue even for FAX communication on dahdi.
>>
>> Mitul
>>
>>
>> On Friday, November 29, 2013, Salaheddine Elharit wrote:
>>
>>> hello
>>>
>>> i add the following in chan_dahdi and the issue has been solved thanks
>>> a lot for your help and support now ican stop the speech and go to my
>>> context
>>>
>>> i really appreciate your help and support
>>>
>>> immediate = yes
>>> echocancel = no
>>> dtmfmode = auto
>>>
>>> -- Forwarded message --
>>> From: Salaheddine Elharit
>>> Date: 2013/11/29
>>> Subject: Re: [asterisk-users] issue with speech in IVR
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion <
>>> asterisk-users@lists.digium.com>
>>>
>>>
>>> hello
>>>
>>> i add the following in chan_dahdi and the issue has been solved thanks
>>> a lot for your help and support now ican stop the speech and go to my
>>> context
>>>
>>> i really appreciate your help and support
>>>
>>>
>>> 2013/11/29 Mitul Limbani
>>>
Try following in chan_dahdi
immediate = yes
echocancel = no
dtmfmode = auto
Mitul
On Nov 29, 2013 1:42 PM, wrote:
> Are you using a mp3 file?
> I have noticed that using control playback with a mp3 file I cannot
> use the keypad to control the playback
>
> -Original Message-
> From: Salaheddine Elharit
> Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013
> 08:05:16
> To: Asterisk Users Mailing List - Non-Commercial Discussion<
> asterisk-users@lists.digium.com>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: Re: [asterisk-users] issue with speech in IVR
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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> _
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> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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>>>
>>>
>>>
>>
>> --
>> Regards,
>> Mitul Limbani,
>> Chief Architech & Founder,
>> Enterux Solutions Pvt. Ltd.
>> 110 Reena Complex, Opp. Nathani Steel,
>> Vidyavihar (W), Mumbai - 400 086. India
>> http://www.enterux.com/
>> http://www.entvoice.com/
>> email: mi...@enterux.in
>> DID: +91-22-71967196
>> Cell: +91-9820332422
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> ___
Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)
On Thu, Dec 5, 2013 at 9:43 AM, James Bensley wrote: > If I re-route traffic away from this box (having never installed DAHDI > before) can I "simply" shutdown Asterisk, checkout the latest SVN, > compile, reload the kernel module and restart Asterisk? The dahdi project moved from svn to git about a year ago. The existing dahdi svn repo is stale and is for archival purposes only. Here is a quickstart that should get you up and running with the dahdi git repo: https://wiki.asterisk.org/wiki/display/DAHDI/Quick+Start+From+Source You might want to skip the "dahdi_genconf system" step, as it will overwrite your current /etc/dahdi/system.conf configuration. Good luck! -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with speech in IVR
What you want to use is Asterisk's dialplan "Read" command. Check it out
here. http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Regards;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine
Elharit
Sent: Thursday, December 05, 2013 5:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] issue with speech in IVR
hello list
i have one question related to the IVR below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
how can i ask the customer to enter a password before to go to (home,s,1)
and where i must to store a password for example password 1234
if the customer enter 1234 he can Goto(home,s,1) and if the password is
wrong i playback an error message
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
the customer must enter 1234 if yes go to (home,s,1) if no go to error
exten => 600,n,Goto(home,s,1)
[error]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}error
any example would be appreciated
2013/11/29 Mitul Limbani
Sounds cool, I suspected the echo cancel situation, these are usually issue
even for FAX communication on dahdi.
Mitul
On Friday, November 29, 2013, Salaheddine Elharit wrote:
hello
i add the following in chan_dahdi and the issue has been solved thanks a
lot for your help and support now ican stop the speech and go to my context
i really appreciate your help and support
immediate = yes
echocancel = no
dtmfmode = auto
-- Forwarded message --
From: Salaheddine Elharit
Date: 2013/11/29
Subject: Re: [asterisk-users] issue with speech in IVR
To: Asterisk Users Mailing List - Non-Commercial Discussion
hello
i add the following in chan_dahdi and the issue has been solved thanks a
lot for your help and support now ican stop the speech and go to my context
i really appreciate your help and support
2013/11/29 Mitul Limbani
Try following in chan_dahdi
immediate = yes
echocancel = no
dtmfmode = auto
Mitul
On Nov 29, 2013 1:42 PM, wrote:
Are you using a mp3 file?
I have noticed that using control playback with a mp3 file I cannot use the
keypad to control the playback
-Original Message-
From: Salaheddine Elharit
Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013
08:05:16
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] issue with speech in IVR
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Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
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Re: [asterisk-users] Lync and Asterisk Realtime Architecture
Remember to use sip show peer load any time you update the peer in the database. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Thursday, December 05, 2013 10:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lync and Asterisk Realtime Architecture Hi Eric, thanks for that. I hadn't been specifying a port, I'll give it a go now. On 5 December 2013 15:39, Eric Wieling wrote: If the device is not registering then you have to specify the port as well as the ip in the database entry for the peer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Thursday, December 05, 2013 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Lync and Asterisk Realtime Architecture Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer load. Has anyone got any experience of connecting to Lync using ARA? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting TDMs (Packet capture like debugging)
Hi All, Many thanks for the info, I am now finding the time to look back into this again. I have seen this page, which indicates that dahdi_pcap was pushed into the dahdi driver from version 2.4.0; https://issues.asterisk.org/jira/browse/DAHTOOL-49?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel I have also seen this page regarding installing dahdi with dahdi_pcap; http://lists.digium.com/pipermail/asterisk-ss7/2011-September/004519.html All seems simple enough. I have no test systems with E1s/PRIs connected. I have a particular production box though which is very quite and I can schedule a maintenance window in which I can fiddle with it. If I killed the box it's not really a problem as other boxes can pick up the load. So my question is this; Looking on this quiet box I see /usr/src/dahdi-linux-2.4.1.1 which doesn't include dahdi_pcap. I am thinking of checking out the latest version and compiling it to test dahdi_pcap. This box has Asterisk 11.3.0 running on Ubuntu Server 12.04.2 LTS. If I re-route traffic away from this box (having never installed DAHDI before) can I "simply" shutdown Asterisk, checkout the latest SVN, compile, reload the kernel module and restart Asterisk? What can go wrong here? Is this risky, or is it really that simple? Cheers, James, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lync and Asterisk Realtime Architecture
Hi Eric, thanks for that. I hadn't been specifying a port, I'll give it a go now. On 5 December 2013 15:39, Eric Wieling wrote: > If the device is not registering then you have to specify the port as well > as the ip in the database entry for the peer. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik > Sent: Thursday, December 05, 2013 9:21 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Lync and Asterisk Realtime Architecture > > Hi guys > > We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk > to MS Lync server. > > If I create the peer in sip.conf the trunk connects with no problem. > > However, we prefer to use ARA. > > Whenever we define the peer in our peers table, the trunk does not work, > even if we use sip show peer load. > > Has anyone got any experience of connecting to Lync using ARA? > > Thanks in advance > > Ish > > > -- > > Ishfaq Malik > Department: VOIP Support > Company: Packnet Limited > t: +44 (0)845 004 4994 > f: +44 (0)161 660 9825 > e: i...@pack-net.co.uk > w: http://www.pack-net.co.uk > > Registered Address: PACKNET LIMITED, Duplex 2, Ducie House > 37 Ducie Street > Manchester, M1 2JW > COMPANY REG NO. 04920552 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lync and Asterisk Realtime Architecture
If the device is not registering then you have to specify the port as well as the ip in the database entry for the peer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Thursday, December 05, 2013 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Lync and Asterisk Realtime Architecture Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer load. Has anyone got any experience of connecting to Lync using ARA? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lync and Asterisk Realtime Architecture
Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer load. Has anyone got any experience of connecting to Lync using ARA? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with speech in IVR
hello list
i have one question related to the IVR below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
how can i ask the customer to enter a password before to go to (home,s,1)
and where i must to store a password for example password 1234
if the customer enter 1234 he can Goto(home,s,1) and if the password is
wrong i playback an error message
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
the customer must enter 1234 if yes go to (home,s,1) if no go to error
exten => 600,n,Goto(home,s,1)
[error]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}error
any example would be appreciated
2013/11/29 Mitul Limbani
> Sounds cool, I suspected the echo cancel situation, these are usually
> issue even for FAX communication on dahdi.
>
> Mitul
>
>
> On Friday, November 29, 2013, Salaheddine Elharit wrote:
>
>> hello
>>
>> i add the following in chan_dahdi and the issue has been solved thanks a
>> lot for your help and support now ican stop the speech and go to my context
>>
>> i really appreciate your help and support
>>
>> immediate = yes
>> echocancel = no
>> dtmfmode = auto
>>
>> -- Forwarded message --
>> From: Salaheddine Elharit
>> Date: 2013/11/29
>> Subject: Re: [asterisk-users] issue with speech in IVR
>> To: Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>
>>
>>
>> hello
>>
>> i add the following in chan_dahdi and the issue has been solved thanks
>> a lot for your help and support now ican stop the speech and go to my
>> context
>>
>> i really appreciate your help and support
>>
>>
>> 2013/11/29 Mitul Limbani
>>
>>> Try following in chan_dahdi
>>>
>>> immediate = yes
>>> echocancel = no
>>> dtmfmode = auto
>>>
>>> Mitul
>>> On Nov 29, 2013 1:42 PM, wrote:
>>>
Are you using a mp3 file?
I have noticed that using control playback with a mp3 file I cannot use
the keypad to control the playback
-Original Message-
From: Salaheddine Elharit
Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013
08:05:16
To: Asterisk Users Mailing List - Non-Commercial Discussion<
asterisk-users@lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] issue with speech in IVR
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>>>
>>> --
>>> _
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>>
>>
>>
>
> --
> Regards,
> Mitul Limbani,
> Chief Architech & Founder,
> Enterux Solutions Pvt. Ltd.
> 110 Reena Complex, Opp. Nathani Steel,
> Vidyavihar (W), Mumbai - 400 086. India
> http://www.enterux.com/
> http://www.entvoice.com/
> email: mi...@enterux.in
> DID: +91-22-71967196
> Cell: +91-9820332422
>
>
>
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>http://www.asterisk.org/hello
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> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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