Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API
On Thu, Jan 28, 2016 at 5:34 PM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > Hi, > > I am using Asterisk 13.6.0 and was wondering if I can programmatically add > users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk > server using API of some sort. > > You can use the Asterisk Manager Interface to modify the config files and reload. https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4817239 -- _ -- 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] PJSIP Stun/ICE
On Thu, Jan 28, 2016 at 6:58 PM, James Cloos wrote: > > "AS" == A J Stiles writes: > > AS> If you are paying for a business-grade Internet connection, you > AS> should get a static IP address -- or a block of them -- as > AS> standard. Maybe you need to change your ISP? > > In some places (including here) static ip is not affordable. > Please create a JIRA issue and let me know what the number is. I've just posted a patch for review that allows reloading transports from the command line. I'd like to know what else you actually need. -- _ -- 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] PJSIP Stun/ICE
> "AS" == A J Stiles writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API
Hi, I am using Asterisk 13.6.0 and was wondering if I can programmatically add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk server using API of some sort. Please do let me know. Thanks, Sonny. -- _ -- 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] 11.21.0 : echo woes : can't install canceller
On 01/28/2016 03:39 PM, sean darcy wrote: i've got calls coming into an 11.21.0 box. The internal phones are analogue off a TDM400 board, and SIP extensions. Using an analogue internal phone, the remote party always hears an echo on it's side. We do not hear an echo. Doesn't matter who is the calling party. But if we use a SIP extension, no echo. I've built /lib/modules/4.3.3-303.fc23.x86_64/dahdi/dahdi_echocan_oslec.ko And requested oslec echo cancel: grep oslec /etc/dahdi/system.conf echocanceller=oslec,1,2,4 grep echo /etc/asterisk/chan_dahdi.conf echocancel=yes echocancelwhenbridged=no echotraining=yes but it's never loaded: # lsmod | grep echo [root@asterisk ~]# dahdi_cfg -vvv DAHDI Tools Version - 2.10.0 DAHDI Version: 2.11.0 Echo Canceller(s): Configuration == Channel map: 0 channels to configure. I can manually insert the oslec module using modprobe. Thats seems to work. CLI> dahdi show version DAHDI Version: 2.11.0 Echo Canceller: OSLEC But it's not persistent across reboots. sean And even if I do manually load the oslec kernel module, I don't think it's actually being used cat /proc/dahdi/1 Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) 1 WCTDM/4/0 FXOKS (In use) 2 WCTDM/4/1 FXOKS (In use) 3 WCTDM/4/2 Reserved 4 WCTDM/4/3 FXSKS (In use) RED AFAIK, the echo canceller should show up here. sean -- _ -- 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] 11.21.0 : echo woes : can't install canceller
i've got calls coming into an 11.21.0 box. The internal phones are analogue off a TDM400 board, and SIP extensions. Using an analogue internal phone, the remote party always hears an echo on it's side. We do not hear an echo. Doesn't matter who is the calling party. But if we use a SIP extension, no echo. I've built /lib/modules/4.3.3-303.fc23.x86_64/dahdi/dahdi_echocan_oslec.ko And requested oslec echo cancel: grep oslec /etc/dahdi/system.conf echocanceller=oslec,1,2,4 grep echo /etc/asterisk/chan_dahdi.conf echocancel=yes echocancelwhenbridged=no echotraining=yes but it's never loaded: # lsmod | grep echo [root@asterisk ~]# dahdi_cfg -vvv DAHDI Tools Version - 2.10.0 DAHDI Version: 2.11.0 Echo Canceller(s): Configuration == Channel map: 0 channels to configure. I can manually insert the oslec module using modprobe. Thats seems to work. CLI> dahdi show version DAHDI Version: 2.11.0 Echo Canceller: OSLEC But it's not persistent across reboots. sean -- _ -- 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] Resource List Subscriptions/BLF List and Aastra phones
Hello, I'm giving a try to Resource List Subscriptions feature also called BLF List in several phone vendors documentation (see [1]). I could successfully configured this feature woth Yealink phones but I've got some issues with Aastra phones (6757i with 3.3.1 firmware). Before diving deeper into this, can you share your own experience here ? More specifically, here are my current results: - dedicated keys (softkey type) are automatically populated - when pressing one dedicated key when matching extension is idle, nothing is dialed (speed dialing not working) - when pressing one dedicated key when matching extension is ringing, nothing is dialed (directed pickup not working) Comments ? Suggestions ? Regards [1] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158 -- _ -- 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] Caller ID Sent in PAI header.
Hello, Thanks for your reply. Is this mentioned in any RFC ? I checked RFC3325 for PAI and RFC3261, but nothing mentioned there. Best regards On Thu, Jan 28, 2016 at 2:50 PM, Laurent Schweizer < laurent.schwei...@peoplefone.com> wrote: > Hello, > > > > Usually in the P-Asserted you have the network number and in the From the > preferred number. > > > > In this case the Preferred (from) number is displayed. > > > > > > BR > > > > Laurent > > > > *De :* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *De la part de* Aziz TestAccount > *Envoyé :* jeudi 28 janvier 2016 15:46 > *À :* asterisk-users@lists.digium.com > *Objet :* [asterisk-users] Caller ID Sent in PAI header. > > > > Hi All, > > When receiving an invite containing two different caller ID, one in FROM > header and the other in "P-Asserted Identity" Header, Which one will be > used by the callee ? I couldn't find any RFC specifying this detail. > > > Thank you. > > -- > _ > -- 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
Re: [asterisk-users] Caller ID Sent in PAI header.
Hello, Usually in the P-Asserted you have the network number and in the From the preferred number. In this case the Preferred (from) number is displayed. BR Laurent De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Aziz TestAccount Envoyé : jeudi 28 janvier 2016 15:46 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Caller ID Sent in PAI header. Hi All, When receiving an invite containing two different caller ID, one in FROM header and the other in "P-Asserted Identity" Header, Which one will be used by the callee ? I couldn't find any RFC specifying this detail. Thank you. -- _ -- 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] Caller ID Sent in PAI header.
Hi All, When receiving an invite containing two different caller ID, one in FROM header and the other in "P-Asserted Identity" Header, Which one will be used by the callee ? I couldn't find any RFC specifying this detail. Thank you. -- _ -- 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 13 mixmonitor - random missing syllables
when you say load - how many concurrent calls? Is there transcoding happening? sip / PRIs ? what load? On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka wrote: > Dne 27.1.2016 v 17:50 A J Stiles napsal(a): > >> On Wednesday 27 Jan 2016, Marek Červenka wrote: >> >>> Dne 27.1.2016 v 13:14 A J Stiles napsal(a): >>> On Wednesday 27 Jan 2016, Marek Červenka wrote: > hi, > > i have strange problem with asterisk 13 mixmonitor, recording to wav > (centos6) > when the system is under load, there are sometimes missing syllable > > there arent BIG spikes on cpus > recordings are to ramdisk (/dev/shm) > > any hints? > First, try recording to a real disk (preferrably a separate drive, so nothing else will be seeking the heads about; and connected by SATA, not USB, for full speed). Does that work any better? >>> i tried before. IO is not the problem >>> >> Are you saying that it records fine when you use a real disk, but not >> with a >> ramdisk? >> >> And why are you using a ramdisk for your mixmonitor recordings? >> >> > i have problem in both scenarios > im using ramdisk because is faster and IO cannot be problem > > -- > --- > Marek Cervenka > === > > > -- > _ > -- 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
Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables
Dne 27.1.2016 v 17:50 A J Stiles napsal(a): On Wednesday 27 Jan 2016, Marek Červenka wrote: Dne 27.1.2016 v 13:14 A J Stiles napsal(a): On Wednesday 27 Jan 2016, Marek Červenka wrote: hi, i have strange problem with asterisk 13 mixmonitor, recording to wav (centos6) when the system is under load, there are sometimes missing syllable there arent BIG spikes on cpus recordings are to ramdisk (/dev/shm) any hints? First, try recording to a real disk (preferrably a separate drive, so nothing else will be seeking the heads about; and connected by SATA, not USB, for full speed). Does that work any better? i tried before. IO is not the problem Are you saying that it records fine when you use a real disk, but not with a ramdisk? And why are you using a ramdisk for your mixmonitor recordings? i have problem in both scenarios im using ramdisk because is faster and IO cannot be problem -- --- Marek Cervenka === -- _ -- 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