[asterisk-users] changing port 5060 to 5061
darin iv
adariniv at gmail.com
Sun Apr 10 23:37:21 CDT 2011
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because we are already using 5060 port in router
then we cant use it again we have to configure other sip server so
please suggest me a way..........................
On 4/10/11, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
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> Today's Topics:
>
> 1. Re: asterisk-users Digest, Vol 81, Issue 27 (Steve Edwards)
> 2. Re: Asterisk FOP (Doug Lytle)
> 3. Re: Asterisk FOP (Flavio Miranda)
> 4. Re: Asterisk FOP (Doug Lytle)
> 5. Re: IAX2/0.0.29.199 (Satish Patel)
> 6. Re: Call Recording using MixMonitor - close, but would like
> some more words of wisdom. (Dan Journo)
> 7. Re: Call recording - methodology (Dan Journo)
> 8. Re: Asterisk FOP (Flavio Miranda)
> 9. Re: send voicemail to multiple emails (vip killa)
> 10. Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]
> (Tzafrir Cohen)
> 11. Re: IAX2/0.0.29.199 (Tzafrir Cohen)
> 12. AsteriskNow updated to Centos 5.6 and DAHDI doesn't work
> (Frank Tarczynski)
> 13. Re: Call recording - methodology (Silver Thorne)
> 14. Re: Call recording - methodology (Dan Journo)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sat, 9 Apr 2011 10:38:00 -0700 (PDT)
> From: Steve Edwards <asterisk.org at sedwards.com>
> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 27
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <alpine.DEB.2.00.1104091032090.20397 at localhost.localdomain>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> On Sat, 9 Apr 2011, darin iv wrote:
>
> 0) Don't re-post the entire digest back to the list it came from. Posting
> 36k of cruft to ask 'How to change SIP port number?' seems somewhat
> 'newbish.'
>
> 1) Try Google. Try 'How to change SIP port number in Asterisk?'
>
> 2) Re-post with a new, relevant Subject and you will get relevant
> responses.
>
> --
> Thanks in advance,
> -------------------------------------------------------------------------
> Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
>
>
> ------------------------------
>
> Message: 2
> Date: Sat, 09 Apr 2011 14:11:39 -0400
> From: Doug Lytle <support at drdos.info>
> Subject: Re: [asterisk-users] Asterisk FOP
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4DA0A15B.8020900 at drdos.info>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Flavio Miranda wrote:
>>
>>
>> I am truing to set up FOP but I getting the following log:
>>
> What version of FOP? 1 or 2, what OS? What version of Asterisk?
>
> Doug
>
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Sat, 9 Apr 2011 16:35:45 -0300
> From: Flavio Miranda <flaviormiranda at hotmail.com>
> Subject: Re: [asterisk-users] Asterisk FOP
> To: Asterisk Asterisk <asterisk-users at lists.digium.com>
> Message-ID: <BAY158-W16C8616B3CA40A70145B51DAA60 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> Hi,
> FOP 1
> OS Debian Lenny
> Asterisk 1.6
>
> Att,
>
>
>
> Flavio Roberto Miranda
>
> MSN:flaviormiranda at hotmail.com
> Skype: flaviormiranda
>
>
>
>> Date: Sat, 9 Apr 2011 14:11:39 -0400
>> From: support at drdos.info
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] Asterisk FOP
>>
>> Flavio Miranda wrote:
>> >
>> >
>> > I am truing to set up FOP but I getting the following log:
>> >
>> What version of FOP? 1 or 2, what OS? What version of Asterisk?
>>
>> 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 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
>
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>
> ------------------------------
>
> Message: 4
> Date: Sat, 09 Apr 2011 18:25:55 -0400
> From: Doug Lytle <support at drdos.info>
> Subject: Re: [asterisk-users] Asterisk FOP
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4DA0DCF3.9070306 at drdos.info>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Flavio Miranda wrote:
>> Hi,
>>
>> FOP 1
>>
>> OS Debian Lenny
>>
>> Asterisk 1.6
>
> All my installs are under Mandriva, running op_panel-0.30.tar.gz.
>
> Googling was inconclusive, varying from install FOP2 to having an old
> swf in your browser cache.
>
> Maybe your perl is messaged up. Give it a shot on a different machine
> (All but 1 of mine are running on different servers then the Asterisk
> 1.4 servers).
>
> Doug
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Sat, 9 Apr 2011 20:31:43 -0400
> From: Satish Patel <satish_lx at hotmail.com>
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <BLU0-SMTP181C4FDCE4683E3F819F78E90A90 at phx.gbl>
> Content-Type: text/plain; charset="utf-8"; format=flowed; delsp=yes
>
> Bump up! Please help here
>
> --
> Sent from my iPhone
>
> On Apr 8, 2011, at 2:10 PM, satish patel <satish_lx at hotmail.com> wrote:
>
>>
>> I tried to compile your version and got bunch of error on "make" and
>> it failed to compile.
>>
>> root at satish-desktop:/home/satish/issue18183# make
>> [CC] chan_iax2.c -> chan_iax2.o
>> chan_iax2.c: In function ?socket_process?:
>> chan_iax2.c:11533: error: invalid storage class for function ?iax2_p
>> rocess_thread_cleanup?
>> chan_iax2.c:11532: warning: no previous prototype for ?iax2_process_
>> thread_cleanup?
>> chan_iax2.c:11544: error: invalid storage class for function ?iax2_p
>> rocess_thread?
>> chan_iax2.c:11543: warning: no previous prototype for ?iax2_process_
>> thread?
>> chan_iax2.c:11683: error: invalid storage class for function ?iax2_d
>> o_register?
>> chan_iax2.c:11682: warning: no previous prototype for ?iax2_do_regis
>> ter?
>> chan_iax2.c:11744: error: invalid storage class for function ?iax2_p
>> rovision?
>> chan_iax2.c:11743: warning: no previous prototype for ?iax2_provisio
>> n?
>> chan_iax2.c:11796: error: invalid storage class for function ?iax2_p
>> rov_app?
>> chan_iax2.c:11795: warning: no previous prototype for ?iax2_prov_ap
>> p?
>> chan_iax2.c:11825: error: invalid storage class for function ?handle
>> _cli_iax2_provision?
>> chan_iax2.c:11824: warning: no previous prototype for ?handle_cli_ia
>> x2_provision?
>> chan_iax2.c:11864: error: invalid storage class for function ?__iax2
>> _poke_noanswer?
>> chan_iax2.c:11863: warning: no previous prototype for ?__iax2_poke_n
>> oanswer?
>> chan_iax2.c:11887: error: invalid storage class for function ?iax2_p
>> oke_noanswer?
>> ...
>> ...
>> ...
>> chan_iax2.c:14723: warning: no previous prototype for ?__reg_module?
>> chan_iax2.c:14723: error: invalid storage class for function ?__unre
>> g_module?
>> chan_iax2.c:14723: warning: no previous prototype for ?__unreg_modul
>> e?
>> chan_iax2.c:14723: error: expected declaration or statement at end
>> of input
>> chan_iax2.c:14723: warning: unused variable ?ast_module_info?
>> make[1]: *** [chan_iax2.o] Error 1
>> make: *** [channels] Error 2
>> root at satish-desktop:/home/satish/issue18183#
>>
>>
>>
>>
>>
>> > Date: Fri, 8 Apr 2011 13:16:30 -0400
>> > From: pabelanger at digium.com
>> > To: asterisk-users at lists.digium.com
>> > Subject: Re: [asterisk-users] IAX2/0.0.29.199
>> >
>> > On 11-04-08 12:56 PM, Paul Belanger wrote:
>> > > On 11-04-08 11:55 AM, satish patel wrote:
>> > >>
>> > >> @Paul - many time i am gettting following SIP error when
>> channel isn't
>> > >> available. I want to get rid on this revers thing. I tried all
>> version
>> > >> 1.8.1,1.8.2,1.8.3 but not fix :(
>> > >>
>> > > Best you can do is collect a full debug[1] log and see when the
>> issue is
>> > > introduced.
>> > >
>> > > [1]
>> > > https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
>> > >
>> > Do you mind trying the following branch[2]? Not sure if it will
>> help,
>> > but I made some changes to chan_iax2 a few months ago.
>> >
>> > [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/
>> >
>> > --
>> > Paul Belanger
>> > Digium, Inc. | Software Developer
>> > twitter: pabelanger | IRC: pabelanger (Freenode)
>> > Check us out at: http://digium.com & http://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
>> --
>> _____________________________________________________________________
>> -- 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
>
>
>
> ------------------------------
>
> Message: 6
> Date: Sat, 9 Apr 2011 20:45:58 -0400
> From: Dan Journo <dan at keshercommunications.com>
> Subject: Re: [asterisk-users] Call Recording using MixMonitor - close,
> but would like some more words of wisdom.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <31C6BA8C3525D840B022617ACBB7BC0382F3749DD4 at VMBX123.ihostexchange.net>
> Content-Type: text/plain; charset="us-ascii"
>
>> DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of
>> extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a
>> per channel basis in extensions.conf.
>
> Sorry, i forgot to mention that one.
>
>
> Dan Journo
> Kesher Communications (UK)
> Business Phone Systems<http://www.keshercommunications.com/> | Hosted
> PBX<http://www.keshercommunications.com/hostedpbx.html>
>
>
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> ------------------------------
>
> Message: 7
> Date: Sat, 9 Apr 2011 20:51:02 -0400
> From: Dan Journo <dan at keshercommunications.com>
> Subject: Re: [asterisk-users] Call recording - methodology
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <31C6BA8C3525D840B022617ACBB7BC0382F3749DD5 at VMBX123.ihostexchange.net>
> Content-Type: text/plain; charset="us-ascii"
>
>> If you don't want to record every call, you can give the operator the
>> option of press *1. We did this by adding the following to features.conf:-
>
>>
>
>> MixMonApp => *1,self/both,Macro,mixmon
>
>
>
> As brought up in another post, I forgot to add the following:-
>
>
> DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of
> extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a
> per channel basis in extensions.conf.
>
> Thanks to Warren Selby from http://www.selbytech.com for pointing that out.
>
>
> Dan Journo
> Kesher Communications (UK)
> Business Phone Systems<http://www.keshercommunications.com/> | Hosted
> PBX<http://www.keshercommunications.com/hostedpbx.html>
>
>
>
>
>
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>
> ------------------------------
>
> Message: 8
> Date: Sat, 9 Apr 2011 23:09:10 -0300
> From: Flavio Miranda <flaviormiranda at hotmail.com>
> Subject: Re: [asterisk-users] Asterisk FOP
> To: Asterisk Asterisk <asterisk-users at lists.digium.com>
> Message-ID: <BAY158-w255939A9F82A266882C97EDAA90 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
> So... something has changed now.When I run ./op_server.pl , I get the
> following verbose: all my asterisk configuration therefore, my buttons dont
> work as it should...I am wondering if my extentions.conf must have something
> different, like a hint function,or something else in order to the FOP show
> the extensions status, Thanks for any help!!
>
> Att,
>
>
>
> Flavio Roberto Miranda
>
> MSN:flaviormiranda at hotmail.com
> Skype: flaviormiranda
>
>
>
>> Date: Sat, 9 Apr 2011 14:11:39 -0400
>> From: support at drdos.info
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [asterisk-users] Asterisk FOP
>>
>> Flavio Miranda wrote:
>> >
>> >
>> > I am truing to set up FOP but I getting the following log:
>> >
>> What version of FOP? 1 or 2, what OS? What version of Asterisk?
>>
>> 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 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
>
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> ------------------------------
>
> Message: 9
> Date: Sun, 10 Apr 2011 00:56:52 -0400
> From: vip killa <vipkilla at gmail.com>
> Subject: Re: [asterisk-users] send voicemail to multiple emails
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <BANLkTikreFbYaThF_ZqWF86tPiD+j-w9vw at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I've already taken the steps you described...issue i ran into was there is
> no variables passed to "mailcmd" only STDIN... as a result i have to
> "extract" variables from STDIN...
>
> On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby <wcselby at selbytech.com> wrote:
>
>> On Fri, Apr 8, 2011 at 1:18 PM, vip killa <vipkilla at gmail.com> wrote:
>>
>>> That does not sound easy... besides these email addresses would be taken
>>> from a MySQL database.
>>>
>>>
>>>
>> It's actually what you're going to end up doing, whether you do it on the
>> MTA level or your code it into your script that you execute instead of
>> sendmail -f. Currently, there is no way to natively have asterisk send
>> one
>> voicemail to multiple email addresses.
>>
>> What's probably going to work best for you since you seem to like program
>> your own scripts (and I'm not talking an AGI here, I'm talking either pure
>> bash, php, perl, or whichever you prefer), is to change the mailcmd=
>> option
>> inside voicemail.conf and replace it with a script of your own design.
>> I'm
>> not sure off the top of my head which variables are passed to the command,
>> but you could always write a simple script that just outputs all arguments
>> to see and go from there. My guess is you're going to at the least get
>> the
>> preconfigured email address and the contents of your emailsubject and
>> emailbody options (both of which have the option of passing multiple
>> useful
>> variables).
>>
>> --
>> Thanks,
>> --Warren Selby, dCAP
>> http://www.selbytech.com
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
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>
> ------------------------------
>
> Message: 10
> Date: Sun, 10 Apr 2011 16:12:28 +0300
> From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
> Subject: [asterisk-users] Ubuntu "*-server" kernels [was: Re:
> IAX2/0.0.29.199]
> To: asterisk-users at lists.digium.com
> Message-ID: <20110410131228.GN6408 at xorcom.com>
> Content-Type: text/plain; charset=us-ascii
>
> Off-topic:
>
> On Fri, Apr 08, 2011 at 03:30:58PM +0000, satish patel wrote:
>
> [snip]
>
>> System: Linux/2.6.32-24-server built by root on
>> x86_64 2011-03-22 18:38:19 UTC
>
> Ubuntu has a separate -server kernel variant. From what I understand,
> using it is not a good idea on a Asterisk system, as it is intended to
> an application such as a file server, optimized for higher throughput.
>
> Asterisk is closer to a desktop multimedia program, which prefers low
> latency to high throughput.
>
> Is that recommendation still valid?
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.cohen at xorcom.com
> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>
>
>
> ------------------------------
>
> Message: 11
> Date: Sun, 10 Apr 2011 16:14:38 +0300
> From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
> To: asterisk-users at lists.digium.com
> Message-ID: <20110410131438.GO6408 at xorcom.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> On Fri, Apr 08, 2011 at 06:10:21PM +0000, satish patel wrote:
>>
>>
>> I tried to compile your version and got bunch of error on "make" and it
>> failed to compile.
>>
>> root at satish-desktop:/home/satish/issue18183# make
>
> How did you get that code?
>
>> [CC] chan_iax2.c -> chan_iax2.o
>> chan_iax2.c: In function ?socket_process?:
>> chan_iax2.c:11533: error: invalid storage class for function
>> ?iax2_process_thread_cleanup?
>> chan_iax2.c:11532: warning: no previous prototype for
>> ?iax2_process_thread_cleanup?
>> chan_iax2.c:11544: error: invalid storage class for function
>> ?iax2_process_thread?
>> chan_iax2.c:11543: warning: no previous prototype for
>> ?iax2_process_thread?
>> chan_iax2.c:11683: error: invalid storage class for function
>> ?iax2_do_register?
>> chan_iax2.c:11682: warning: no previous prototype for ?iax2_do_register?
>> chan_iax2.c:11744: error: invalid storage class for function
>> ?iax2_provision?
>> chan_iax2.c:11743: warning: no previous prototype for ?iax2_provision?
>> chan_iax2.c:11796: error: invalid storage class for function
>> ?iax2_prov_app?
>> chan_iax2.c:11795: warning: no previous prototype for ?iax2_prov_app?
>> chan_iax2.c:11825: error: invalid storage class for function
>> ?handle_cli_iax2_provision?
>> chan_iax2.c:11824: warning: no previous prototype for
>> ?handle_cli_iax2_provision?
>> chan_iax2.c:11864: error: invalid storage class for function
>> ?__iax2_poke_noanswer?
>> chan_iax2.c:11863: warning: no previous prototype for
>> ?__iax2_poke_noanswer?
>> chan_iax2.c:11887: error: invalid storage class for function
>> ?iax2_poke_noanswer?
>> ...
>> ...
>> ...
>> chan_iax2.c:14723: warning: no previous prototype for ?__reg_module?
>> chan_iax2.c:14723: error: invalid storage class for function
>> ?__unreg_module?
>> chan_iax2.c:14723: warning: no previous prototype for ?__unreg_module?
>> chan_iax2.c:14723: error: expected declaration or statement at end of
>> input
>> chan_iax2.c:14723: warning: unused variable ?ast_module_info?
>> make[1]: *** [chan_iax2.o] Error 1
>> make: *** [channels] Error 2
>> root at satish-desktop:/home/satish/issue18183#
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.cohen at xorcom.com
> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>
>
>
> ------------------------------
>
> Message: 12
> Date: Sun, 10 Apr 2011 09:32:03 -0400
> From: Frank Tarczynski <ftarz at mindspring.com>
> Subject: [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI
> doesn't work
> To: asterisk-users at lists.digium.com
> Message-ID: <4DA1B153.20703 at mindspring.com>
> Content-Type: text/plain; charset=windows-1252; format=flowed
>
> My AsteriskNow box was updated to Centos 5.6 (2.6.18-238.5.1.el5) and
> DAHDI doesn't want to load. I've tried building it from the sources, but
> get this error message:
> CC [M]
> /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.o
> In file included from
> /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/xpd.h:31,
> from
> /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.c:29:
> include/linux/device.h:408: error: expected identifier or ?(? before ?const?
> make[4]: ***
> [/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.o]
> Error 1
> make[3]: ***
> [/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp]
> Error 2
> make[2]: ***
> [_module_/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi]
> Error 2
> make[2]: Leaving directory `/usr/src/kernels/2.6.18-238.5.1.el5-x86_64'
> make[1]: *** [modules] Error 2
> make[1]: Leaving directory
> `/root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux'
> make: *** [all] Error 2
>
> The code in question is:
> static inline const char *dev_name(const struct device *dev)
> {
> return kobject_name(&dev->kobj);
> }
>
> Anybody else seen this problem? Any resolutions?
>
> Thanks
>
>
>
>
> ------------------------------
>
> Message: 13
> Date: Sun, 10 Apr 2011 09:35:09 -0400
> From: Silver Thorne <zoraxus at gmail.com>
> Subject: Re: [asterisk-users] Call recording - methodology
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4DA1B20D.6010106 at gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
>
> Dan et al;
>
> Okay - I have declared DYNAMIC_FEATURES=MixMonApp in the [global]
> section of my extensions.conf
>
> I dial into my trunk, the softphone rings, I answer and I press '*1' - I
> hear the tones, but I see no indication in the Asterisk CLI and I see no
> .wav file being created.
>
> I must still be missing some subtle little thing.
>
> Wow, this is taking on a life of it's own.
>
> What am I missing?
>
> Not reading the DTMF tones. Thus not executing the macro.
>
> Keep in mind, that if I execute the macro manually (put in right in my
> extension declaration in extensions.conf, it works)
>
> Let me know if you want to see anything (parameters, etc)
>
> Thanks
>
> Glen
>
> On 4/9/2011 20:51, Dan Journo wrote:
>>
>> > If you don't want to record every call, you can give the operator
>> the option of press *1. We did this by adding the following to
>> features.conf:-
>>
>> >
>>
>> > MixMonApp => *1,self/both,Macro,mixmon
>>
>> As brought up in another post, I forgot to add the following:-
>>
>> DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of
>> extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion
>> on a per channel basis in extensions.conf.
>>
>>
>> Thanks to Warren Selby from http://www.selbytech.com for pointing that
>> out.
>>
>> Dan Journo
>>
>> Kesher Communications (UK)
>>
>> Business Phone Systems <http://www.keshercommunications.com/> | Hosted
>> PBX <http://www.keshercommunications.com/hostedpbx.html>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>> http://www.asterisk.org/hello
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> ------------------------------
>
> Message: 14
> Date: Sun, 10 Apr 2011 10:28:28 -0400
> From: Dan Journo <dan at keshercommunications.com>
> Subject: Re: [asterisk-users] Call recording - methodology
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <31C6BA8C3525D840B022617ACBB7BC0382F3749DE2 at VMBX123.ihostexchange.net>
> Content-Type: text/plain; charset="us-ascii"
>
>
>> What am I missing?
>>
>> Not reading the DTMF tones. Thus not executing the macro.
>
> Start by checking you are receiving the DTMF tones.
>
> Edit logger.conf and add dtmf to the console line.
> So it looks something like this:-
>
> console => notice,warning,error,dtmf
>
> Then see if you are receiving the tones correctly.
> What method are you using to transmit the dtmf tones?
>
> Regards
>
> Dan Journo
> Kesher Communications (UK)
> Business Phone Systems<http://www.keshercommunications.com/> | Hosted
> PBX<http://www.keshercommunications.com/hostedpbx.html>
>
>
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>
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