[asterisk-users] Voicemail: saycid without prefix
Hi list! Yesterday I set up a voicemail on my Asterisk 1.8. It works as expected, but I'd like to have the CID without unnecessary prefix... Right now, if I call from my mobile phone I hear the complete prefix for my mobile number, indeed without "00". So I hear "message from 49177...". How can I set Asterisk to just read the prefix if it's necessary (so that calls from german numbers will not have "0049")? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Action Originate in Asterisk 13 creates 2 calls in core show channels
On Fri, Jul 3, 2015 at 1:46 PM, Alonso Genis wrote: > Hello, > > I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with > success. > > I have an application that sends an action Originate to AMI for > calling, it's working well, but when i see to Asterisk's CLI, i see 2 > calls for just one originate: > > pftestes40copiabh*CLI> core show channels verbose > Channel Context ExtensionPrio State > Application Data CallerIDDuration > Accountcode PeerAccount BridgeID > SIP/1903-00091903_aux 1 Up > AppDial (Outgoing Line) 190300:00:12 1902 > 19027866921b-4675-4823-8 > Local/1902@1902_in-0 macro-atende s1008 > Ringing Dial SIP/1902,30,t 1903 > 00:00:15 19029428460a-f4e7-46d1-b > Local/1902@1902_in-0 macro-atende s1008 Up > Dial SIP/1903,30,t 190200:00:15 1902 > 19027866921b-4675-4823-8 > SIP/1902-00081902_aux 1 Up > AppDial (Outgoing Line) 190200:00:15 1902 > 9428460a-f4e7-46d1-b > 4 active channels > 2 active calls > > In fact, just one call is up. > > Somebody knows if this is ok, or it's a bug? May be someday asterisk > will create just one call for one originate? > > Thanks in advanced for your answers! It isn't a bug. The output of 'core show channels' reports a 'call' (which is not a concept that is represented well anywhere in Asterisk) as a channel with a PBX thread running. In this case, that's the two channels in your output that are not outbound channels, i.e., the Local channels that dialled your SIP channels. That fact that you have two different SIP channels means that something either performed two Originates, or you have done a parallel Dial. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] Action Originate in Asterisk 13 creates 2 calls in core show channels
> On 3Jul, 2015, at 13:46, Alonso Genis wrote: > > Hello, > > I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with > success. > > I have an application that sends an action Originate to AMI for > calling, it's working well, but when i see to Asterisk's CLI, i see 2 > calls for just one originate: It looks like you’re originating with local channels. Try with /n at the end (not sure if that helps anymore) and also send us your originate line. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] Asterisk 11 and pulseaudio setup as local user
On 3Jul, 2015, at 06:17, Jerry Geis wrote: > > alsa_card_init^[[0m: snd_pcm_open failed: Connection refused > soundcard_init^[[0m: Problem opening alsa capture device > If the problem is a permission thing I’d confirm that. Run Asterisk as root and see if the problem goes away. If so then you’ll have to work your way through devices and drivers and check who owns them and is able access them. I’d also try temporarily disabling SELinux to see if that helps. As with the first option this would be temporary - go back to running Asterisk as myuser or whatever and reenable SELinux regardless of the outcome. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] Action Originate in Asterisk 13 creates 2 calls in core show channels
Hello, I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success. I have an application that sends an action Originate to AMI for calling, it's working well, but when i see to Asterisk's CLI, i see 2 calls for just one originate: pftestes40copiabh*CLI> core show channels verbose Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode PeerAccount BridgeID SIP/1903-00091903_aux 1 Up AppDial (Outgoing Line) 190300:00:12 1902 19027866921b-4675-4823-8 Local/1902@1902_in-0 macro-atende s1008 Ringing Dial SIP/1902,30,t 1903 00:00:15 19029428460a-f4e7-46d1-b Local/1902@1902_in-0 macro-atende s1008 Up Dial SIP/1903,30,t 190200:00:15 1902 19027866921b-4675-4823-8 SIP/1902-00081902_aux 1 Up AppDial (Outgoing Line) 190200:00:15 1902 9428460a-f4e7-46d1-b 4 active channels 2 active calls In fact, just one call is up. Somebody knows if this is ok, or it's a bug? May be someday asterisk will create just one call for one originate? Thanks in advanced for your answers! Alonso. -- _ -- 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] Asterisk 11 and pulseaudio setup as local user
On Fri, 3 Jul 2015, Jerry Geis wrote: Ok digging deaper... I was always trying to run the session as su myuser -c "asterisk -fn" This does not seem to work. If I login as myuser and run "asterisk fn" it worked... I got a lot of crackly noise that I normally dont have but it worked. Any thoughts on why I cannot run the command as 'su myuser -c "asterisk -fn"' ? Maybe there is a difference in the environment -- like a search [XXX]PATH or resource limit? Does comparing the output from: su myuser -c 'set | sort --unique >set-su' sudo --user=myuser sh -c 'set | sort --unique >set-sudo' set | sort --unique >set-shell # logged in as myuser su myuser -c 'ulimit -a | sort >ulimit-su' sudo --user=myuser sh -c 'ulimit -a | sort >ulimit-sudo' ulimit -a | sort >ulimit-shell # logged in as myuser yield any clues? Maybe the 'crackly noise' is because myuser cannot access resources the same way root can -- like an elevated priority or 'real time' or ??? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST-- _ -- 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] Asterisk 11 and pulseaudio setup as local user
On Friday 03 Jul 2015, Jerry Geis wrote: > Ok digging deaper... I was always trying to run the session as > su myuser -c "asterisk -fn" > > This does not seem to work. > > If I login as myuser and run "asterisk fn" it worked... I got a lot of > crackly noise that I normally dont have > but it worked. > > Any thoughts on why I cannot run the command as 'su myuser -c "asterisk > -fn"' ? This sounds as though you are running afoul of SELinux permissions. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Asterisk 11 and pulseaudio setup as local user
Ok digging deaper... I was always trying to run the session as su myuser -c "asterisk -fn" This does not seem to work. If I login as myuser and run "asterisk fn" it worked... I got a lot of crackly noise that I normally dont have but it worked. Any thoughts on why I cannot run the command as 'su myuser -c "asterisk -fn"' ? Thanks jerry -- _ -- 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] Asterisk 11 and pulseaudio setup as local user
Am 3. Juli 2015 13:17:34 MESZ, schrieb Jerry Geis : >alsa_card_init^[[0m: snd_pcm_open failed: Connection refused >soundcard_init^[[0m: Problem opening alsa capture device > >These are the errors I get. > >I changed the following: >chown -R myuser:myuser /var/log/asterisk >chown -R myuser:myuser /var/lib/asterisk >chown -R myuser:myuser /var/spool/asterisk >chown -R myuser:myuser /var/run/asterisk >chown -R myuser:myuser /usr/lib/asterisk >edited asterisk.conf and uncommented runuser and rungroup and changed >to >myuuser > >Then I get the above error when running > > >Jerry > > > Sounds more like wrong perms on the sound devices in /dev. Is your user in the audio group, or whatever it's called on your box? -- _ -- 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] Asterisk 11 and pulseaudio setup as local user
alsa_card_init^[[0m: snd_pcm_open failed: Connection refused soundcard_init^[[0m: Problem opening alsa capture device These are the errors I get. I changed the following: chown -R myuser:myuser /var/log/asterisk chown -R myuser:myuser /var/lib/asterisk chown -R myuser:myuser /var/spool/asterisk chown -R myuser:myuser /var/run/asterisk chown -R myuser:myuser /usr/lib/asterisk edited asterisk.conf and uncommented runuser and rungroup and changed to myuuser Then I get the above error when running Jerry -- _ -- 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
