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, [email protected] <[email protected]> wrote: > Send asterisk-users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > 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 <[email protected]> > Subject: Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 27 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: > <[email protected]> > 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 [email protected] 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 <[email protected]> > Subject: Re: [asterisk-users] Asterisk FOP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > 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 <[email protected]> > Subject: Re: [asterisk-users] Asterisk FOP > To: Asterisk Asterisk <[email protected]> > Message-ID: <[email protected]> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi, > FOP 1 > OS Debian Lenny > Asterisk 1.6 > > Att, > > > > Flavio Roberto Miranda > > MSN:[email protected] > Skype: flaviormiranda > > > >> Date: Sat, 9 Apr 2011 14:11:39 -0400 >> From: [email protected] >> To: [email protected] >> 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 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110409/fb81c30d/attachment-0001.htm> > > ------------------------------ > > Message: 4 > Date: Sat, 09 Apr 2011 18:25:55 -0400 > From: Doug Lytle <[email protected]> > Subject: Re: [asterisk-users] Asterisk FOP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > 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 <[email protected]> > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > 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 <[email protected]> wrote: > >> >> I tried to compile your version and got bunch of error on "make" and >> it failed to compile. >> >> root@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@satish-desktop:/home/satish/issue18183# >> >> >> >> >> >> > Date: Fri, 8 Apr 2011 13:16:30 -0400 >> > From: [email protected] >> > To: [email protected] >> > 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 <[email protected]> > 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 > <[email protected]> > Message-ID: > <31c6ba8c3525d840b022617acbb7bc0382f3749...@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> > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110409/11456b2e/attachment-0001.htm> > > ------------------------------ > > Message: 7 > Date: Sat, 9 Apr 2011 20:51:02 -0400 > From: Dan Journo <[email protected]> > Subject: Re: [asterisk-users] Call recording - methodology > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: > <31c6ba8c3525d840b022617acbb7bc0382f3749...@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> > > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110409/d7086a6a/attachment-0001.htm> > > ------------------------------ > > Message: 8 > Date: Sat, 9 Apr 2011 23:09:10 -0300 > From: Flavio Miranda <[email protected]> > Subject: Re: [asterisk-users] Asterisk FOP > To: Asterisk Asterisk <[email protected]> > Message-ID: <[email protected]> > 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:[email protected] > Skype: flaviormiranda > > > >> Date: Sat, 9 Apr 2011 14:11:39 -0400 >> From: [email protected] >> To: [email protected] >> 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 > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110409/bfbd159f/attachment-0001.htm> > -------------- next part -------------- > An embedded and charset-unspecified text was scrubbed... > Name: panel.txt > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110409/bfbd159f/attachment-0001.txt> > > ------------------------------ > > Message: 9 > Date: Sun, 10 Apr 2011 00:56:52 -0400 > From: vip killa <[email protected]> > Subject: Re: [asterisk-users] send voicemail to multiple emails > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > 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 <[email protected]> wrote: > >> On Fri, Apr 8, 2011 at 1:18 PM, vip killa <[email protected]> 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 >> > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110410/f7664445/attachment-0001.htm> > > ------------------------------ > > Message: 10 > Date: Sun, 10 Apr 2011 16:12:28 +0300 > From: Tzafrir Cohen <[email protected]> > Subject: [asterisk-users] Ubuntu "*-server" kernels [was: Re: > IAX2/0.0.29.199] > To: [email protected] > Message-ID: <[email protected]> > 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:[email protected] > +972-50-7952406 mailto:[email protected] > http://www.xorcom.com iax:[email protected]/tzafrir > > > > ------------------------------ > > Message: 11 > Date: Sun, 10 Apr 2011 16:14:38 +0300 > From: Tzafrir Cohen <[email protected]> > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > To: [email protected] > Message-ID: <[email protected]> > 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@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@satish-desktop:/home/satish/issue18183# > > -- > Tzafrir Cohen > icq#16849755 jabber:[email protected] > +972-50-7952406 mailto:[email protected] > http://www.xorcom.com iax:[email protected]/tzafrir > > > > ------------------------------ > > Message: 12 > Date: Sun, 10 Apr 2011 09:32:03 -0400 > From: Frank Tarczynski <[email protected]> > Subject: [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI > doesn't work > To: [email protected] > Message-ID: <[email protected]> > 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 <[email protected]> > Subject: Re: [asterisk-users] Call recording - methodology > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > 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 -- >> 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110410/37fe0484/attachment-0001.htm> > > ------------------------------ > > Message: 14 > Date: Sun, 10 Apr 2011 10:28:28 -0400 > From: Dan Journo <[email protected]> > Subject: Re: [asterisk-users] Call recording - methodology > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: > <31c6ba8c3525d840b022617acbb7bc0382f3749...@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> > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110410/e0f0d42e/attachment.htm> > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2010 - October 26-28 Washington, DC > Register Now: http://www.astricon.net/ > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 81, Issue 30 > ********************************************** >
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