[asterisk-users] How to change SIP header TO: ?

Hello friends. I have a softswitch in which I cannot create a list of blocked source numbers; So, I have thought to use Asterisk and return a 302 message when the number can make the call, my dialplan is as follows: [from-external]   exten => _AX.,1,Verbose(===> ${CALLERID(num)} to ${EXTEN})

Re: [asterisk-users] asterisk mysql contacts

On Wed, 17 Jan 2018 09:26:28 -0700 John Kiniston wrote: > use func_odbc, create a new function that does a lookup. > > [CALLERID] > prefix=LOOKUP > dsn=MyDB > readsql=SELECT CALLERID from MyNames where CallerIdNum = > '${SQL_ESC(${ARG1})}' > > exten =>

Re: [asterisk-users] queue peridiodic-announce-frequency

On Wed, 17 Jan 2018 12:08:40 +0100 Antony Stone <antony.st...@asterisk.open.source.it> wrote: > On Wednesday 17 January 2018 at 11:59:21, Paul Neuwirth wrote: > > > Hello group, > > > > I tried a lot to enlarge the frequency (i.e. more announces, low > >

[asterisk-users] queue peridiodic-announce-frequency

Hello group, I tried a lot to enlarge the frequency (i.e. more announces, low wait between). according to config, every 30 seconds the announcement should take place. In fact, the first periodic announce is done after 2 minutes? What is my fault? Thank you Regards Paul # zypper if asterisk

Re: [asterisk-users] remote Asterisk console

On Tue, 16 Jan 2018 18:18:18 +0200 Tzafrir Cohen <tzafrir.co...@xorcom.com> wrote: > On Tue, Jan 16, 2018 at 11:05:01AM +0100, Paul Neuwirth wrote: > > Hello group, > > > > what is the preferred method to connect to asterisk cli over > > network? I need to r

[asterisk-users] remote Asterisk console

Hello group, what is the preferred method to connect to asterisk cli over network? I need to run asterisk cli commands remotely. Sharing the unix socket through NFS, if that's working? Or any other approaches, despite using SSH or rlogin, rsh. Thank you Paul

Re: [asterisk-users] load-balancing AMI and load-balancing FastAGI?

Anyone? -- _ -- 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

[asterisk-users] load-balancing AMI and load-balancing FastAGI?

events/response across multiple processes (multiple AMI connections on the same asterisk machine), should the ami events/response should be pushed into RabbitMQ so the proess can read from RabbitMQ ? Thanks Paul -- _ -- Bandwidth

Re: [asterisk-users] small homebrew pbx

On 15/06/15 07:46, lu...@sulweb.org wrote: Hello all, Given the requirements above, what's a cheap but working PCIe card / USB adapter I could buy for this kind of PBX? Do I need things like echo cancellation? Do I need FXS ports? Thanks in advance, Lucio. I would get hold of some

Re: [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.

, chan_ooh323 from my side to another asterisk SIP chan_sip on both sides. Just because everything work OK, I , definitely, can comment out this error message, but... Could you give me any idea why this error can appear? If you haven't create an issue on Jira, this is a bug. -- Paul Belanger

Re: [asterisk-users] WebRTC phone

here. We also do this and it works quiet well. Kudos. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] Is Asterisk a Linux only system?

It should be noted, we did have a FreeBSD and Ubuntu systems running the testsuite back in 2010. FreeBSD was donated to the project. I personally had a PowerPC system running asterisk / testsuite, on debian. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC

Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers

storage device. Alternatively, you could use something like Puppet to deploy the files to all the servers. This is basically what we do, we use puppet to help distribute files to remote servers while still using app_queue. Shared network drive also works. -- Paul Belanger | PolyBeacon, Inc. Jabber

Re: [asterisk-users] Asterisk Java API - Up to date

phase. You should be able to google Asterisk dialers to see some example that people have done. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] queue show queue-name vs queue log for calculating average hold time

On Wed, Jan 28, 2015 at 1:37 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show

Re: [asterisk-users] queue show queue-name vs queue log for calculating average hold time

as to what it means? Thanks in advance Welcome to business logic embedded into app_queue. The issue with the queue show command rendering stats, is what timeframe are the stats aggregated over? IIRC, the calculations are using a moving average[1]. -- Paul Belanger | PolyBeacon, Inc. Jabber

Re: [asterisk-users] Cannot get my first WebRTC experiment to work.

chrome. I hope someone can intersperse the output with comments? Pastebin the fill debug, you've delete an important piece of information. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https

Re: [asterisk-users] ${HASH(SIP_CAUSE,channel-name)}

cause = ) in new stack [Oct 30 14:48:03] -- Executing [h@pbx-routing:6] NoOp(SIP/SipAT01-0015, sip cause = ) in new stack Can anyone tell me how this should be used ? sip.conf: storesipcause=yes -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger

Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

On Oct 29, 2014, at 2:45 PM, Ben Klang bkl...@mojolingo.com wrote: On 10/28/2014 06:03 PM, Ben Langfeld wrote: On 28 October 2014 19:47, Derek Andrew derek.and...@usask.ca wrote: What is the alternative to the dial plan? Is everyone talking about getting rid of the statements like: exten

Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 29, 2014, at 4:26 PM, Matthew Jordan mjor...@digium.com wrote: On Wed, Oct 29, 2014 at 2:31 PM, Paul Albrecht palbre...@glccom.com wrote: On Oct 28, 2014, at 5:03 PM, Ben Langfeld b...@langfeld.me wrote: On 28 October 2014 19:47, Derek Andrew derek.and...@usask.ca wrote: What

Re: [asterisk-users] Asterisk 12 - zombie processes

an external process? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation

Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

AMI/AGI.” or this Paul: take away apps, and whatever is in the core is what we should care about.” -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE

Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

to share their vision with the rest of the Asterisk community. On Oct 27, 2014, at 2:32 PM, Jeffrey Ollie j...@ocjtech.us wrote: On Mon, Oct 27, 2014 at 2:04 PM, Paul Albrecht palbre...@glccom.com wrote: The reason the dial plan can never be deprecated is because Asterisk wouldn’t be Asterisk

[asterisk-users] AppKonference 2.6

I have released an updated AppKonference that compiles with Asterisk 13. You can download the latest code from source forge: sourceforge.net/projects/appkonference That said Asterisk 13 doesn’t get that much attention because I use Asterisk 1.4 + some hacks. Here’s a link to my Asterisk 1.4

Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)

. On Oct 24, 2014, at 11:48 AM, Jeffrey Ollie j...@ocjtech.us wrote: On Fri, Oct 24, 2014 at 10:09 AM, Paul Albrecht palbre...@glccom.com wrote: When Matt says deprecating the dial plan would be difficult and would take a long time it seems to me he’s being evasive and misleading. He doesn’t

Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 23, 2014, at 1:58 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: From: Paul Albrecht palbre...@glccom.com Seems like now is as good a time as any to raise these issues, in fact, sooner is better than later because once developers start down a path it’s very difficult

Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 3:27 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: From: Paul Albrecht palbre...@glccom.com Here’s a link to the minutes: https://wiki.asterisk.org/wiki/ display/AST/AstriDevCon+2014 It has you saying: Leif: we're in a transition, moving from dialplan

Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 3:39 PM, Matthew Jordan mjor...@digium.com wrote: On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht palbre...@glccom.com wrote: On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote: On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht palbre...@glccom.com

Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 23, 2014, at 1:55 AM, Olle E Johansson o...@edvina.net wrote: It is critical that a group of developers ask themself questions along these lines - what if??? - What if we removed AGi and AMI? - What if we made a pluggable PBX? - What if we restarted working on a SIP channel? -

[asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

Really? Shouldn’t something this major affecting the entire Asterisk community get discussed on the lists? Any idea what Leif is talking about when he says the community is in transition, moving from dial plan model to external control? Here’s a link to the notes posted on the Asterisk wiki:

Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote: Paul Albrecht wrote: Really? Shouldn’t something this major affecting the entire Asterisk community get discussed on the lists? Any idea what Leif is talking about when he says the community is in transition, moving from dial

Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda itemDeprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote: On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht palbre...@glccom.com wrote: On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote: Paul Albrecht wrote: Really? Shouldn’t something this major affecting

Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item DeprecateAMI/AGI(Ben Klang)

On Oct 22, 2014, at 11:47 AM, BJ Weschke bwesc...@btwtech.com wrote: On 10/22/14, 12:14 PM, Paul Albrecht wrote: On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote: Paul Albrecht wrote: Really? Shouldn’t something this major affecting the entire Asterisk community get

Re: [asterisk-users] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)

On Oct 22, 2014, at 2:26 PM, Leif Madsen lmad...@thinkingphones.com wrote: On 22 October 2014 14:55, Paul Albrecht palbre...@glccom.com wrote: On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote: This is an open source project. Communication is done in an open

Re: [asterisk-users] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)

justification for such a profound change to a mature product interface than some vague desire by unknown persons who know best for the entire Asterisk community. So, to answer your question, yes, and no. On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht palbre...@glccom.com wrote: On Oct 22

Re: [asterisk-users] how to make voip client cannot use same username?

with the other user? Since what you describe is a valid for SIP, you'll have to drop the packets at the network level (firewall). Or use the ACL system in asterisk to restrict it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https

Re: [asterisk-users] log caller hangup events

Hi, I am mostly concerned with inbound calls. Would it work the same? Regards, Paul From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnan N gopalakrishnan...@gmail.com Sent: Monday, August 18

Re: [asterisk-users] Error opening file for reading: Permission denied

Mitch, Is it the below error? if ((fd = open(filename, O_RDONLY)) 0) { ast_log(LOG_WARNING, Cannot open file '%s' for reading: %s\n, filename, strerror(errno)); return NULL; } Regards, Paul From: asterisk

[asterisk-users] log caller hangup events

advice would be most welcome! Regards, Paul -- _ -- 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

Re: [asterisk-users] Asterisk peer definition registration

to check if there's any chance I could ask Asterisk not to register when I reset. Or is there any other possible solution for this? No, only reload after your ITSP brute force timer has expired. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode

Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

. I also believe there are some open issue with dtls + srtp too. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

On Fri, Aug 15, 2014 at 12:17 PM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Thanks Paul, I appreciate your thoughts. I understand your way, it's logical in your environment. I prefer to use LTS versions of Asterisk so I'm guessing what I want to do is not quite possible

Re: [asterisk-users] Asterisk on CentOS7

Discussion Subject: Re: [asterisk-users] Asterisk on CentOS7 On Thursday, August 14, 2014 03:15:16 AM Paul Greenberg wrote: Hi Anthony, That script does not work. My guess is that it is related to the way asterisk interacts with CentOS environment. Best Regards, Paul Greenberg, Esq

Re: [asterisk-users] Asterisk on CentOS7

manually. Best Regards, Paul Greenberg, Esq. Law Office of Paul Greenberg 530 Main Street, Suite 102 Fort Lee, NJ 07024 E-mail: p...@greenberg.pro Tel: 201-402-6777 Fax: 201-301-8876 Web: http://www.greenberg.pro From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hi, Wow, thanks Paul, realizing the problem makes a lot of sense. So I setup Kamailio as a peer, but if I disable chan_sip module completely, I can't do it in sip.conf like I'd otherwise assume to do. I

Re: [asterisk-users] Asterisk on CentOS7

Hi Anthony, That script does not work. My guess is that it is related to the way asterisk interacts with CentOS environment. Best Regards, Paul Greenberg, Esq. Law Office of Paul Greenberg 530 Main Street, Suite 102 Fort Lee, NJ 07024 Tel: 201-402-6777 Fax: 201-301-8876 Web: http

Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Thank You Paul for your reply, The registrations in my setup are not duplicated, the 'secret' field in the realtime table is empty, which causes Asterisk to not authenticate requests from my

Re: [asterisk-users] Asterisk 12 on Debian Wheezy

or something else ? libpjsua.so (libc6) = /usr/lib/libpjsua.so You will likely need to pass the pjproject directory to configure. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https

Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

a kamailio peer and away you go. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth

[asterisk-users] Asterisk IP7960 and MWI Issue

is ON for another 20-30 minutes. It seems that there is a poll interval of some kind. I am not sure whether it is a setting on the phone or the asterisk. Any ideas? Best Regards, Paul Greenberg, Esq. Law Office of Paul Greenberg 530 Main Street, Suite 102 Fort Lee, NJ 07024 E-mail: p...@greenberg.pro Tel

Re: [asterisk-users] Notification when queue member's phone rings

seconds. It's only happened once in 2 years that I know of, so may not be worth worrying about. AMI will raise the AgentCalled[1] event. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_AgentCalled -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com

Re: [asterisk-users] Live Recording on the Storage Server?

On Thu, Apr 17, 2014 at 7:25 AM, binary dreamer dreamer.bin...@gmail.com wrote: hi. I would not do that due to network issues. My approach is to record everything locally and every hour or so to move everything to a storage. +1 save yourself the headache and do this. -- Paul Belanger

Re: [asterisk-users] Asterisk 11 under VMware?

could leverage snapshots in VM ware for the purpose or migrating or back ups. I don't think it is a waste per say, just different requirements. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter

Re: [asterisk-users] Asterisk CLI Banner

. Or can you express your creativity by fiddling with ASTERISK_PROMPT? If you really want to do it: 1) create a wrapper to asterisk -r 2) pipe the welcome message to /dev/null 3) ??? 4) profit you didn't modify Asterisk. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC

Re: [asterisk-users] asteriskdocs.org says 3rd ed. is latest

the site now. Currently only the 3rd edition is published online. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] AMI Proxy

want to use. We used starpy for a while, but ended up rewriting our own version. Currently we're connecting AMI to a message bus and passing events across the bus. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com

Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

, now when a user changes their password, secret.conf gets updated not voicemail.conf. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not using realtime. anyway to prevent AppVoicemail ro auto generate files? passwordlocation = spooldir Read voicemail.conf about how to use it. -- Paul Belanger

Re: [asterisk-users] Need more meetme users -- hitting some limit

to change it with modprode. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth

Re: [asterisk-users] Which is more efficient for 1 to many broadcasting?

are you looking for? We did some load testing recently and found less people in a bridge is better then more. Audio source location didn't really matter much. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger

Re: [asterisk-users] High Availability with Asterisk

goes live. Correct, in this case para-virt is not the way to go. You'll want to use a virtualization platform that does support multi-hardware with live migration support. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https

Re: [asterisk-users] High Availability with Asterisk

, auto-scaling VoIP setup. :) +1 to this post. A lot of good information here. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] Updating to 11.7.0

. While we're at it, what's the recommended alternative method to replace using asterisk -rx in bash scripts now? cheers, Paul. Jerry -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com http://www.digium.com

Re: [asterisk-users] Updating to 11.7.0

On 05/03/14 12:56, Paul Hayes wrote: I appreciate that and I do understand why but that setting doesn't work as described, it seems to do nothing. While we're at it, what's the recommended alternative method to replace using asterisk -rx in bash scripts now? cheers, Paul. Apologies

Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

On Thu, Feb 27, 2014 at 10:55 PM, Darryl Moore dar...@moores.ca wrote: On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com wrote: No such thing as 'free open source g729 license', if you actually read the site: There is regarding the copyright on the code. The fact

Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

material on the subject however, I am still in need of some definitive answers. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] SIP 603 Declined error message

,hangup,1) -- Executing [hangup@app-blackhole:1] NoOp(SIP/trunk503in-010b, Blackhole Dest: Hangup) in new stack -- Executing [hangup@app-blackhole:2] Hangup(SIP/trunk503in-010b, ) in new stack -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger

Re: [asterisk-users] G729 - what happens if licences used up?

. If you offer a both g729 and ulaw, then ulaw will be used. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] Changing gateway address

of a Linux support issue then specific to Asterisk. Depending on your OS, will dictate how to change your gateway. check /etc/network/inferfaces if you are ubuntu / debian. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https

Re: [asterisk-users] How does extensions.lua compares to extensions.conf ?

On Thu, Feb 13, 2014 at 1:04 AM, George Joseph george.jos...@fairview5.com wrote: On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote: Hello, How does extensions.lua compares

Re: [asterisk-users] How does extensions.lua compares to extensions.conf ?

or memcached. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation

[asterisk-users] answering machine screening with MixMonitor

briefly at apps/app_mixmonitor.c and main/file.c but I don't fully understand the code. Is mixmonitor forking an external conversion process to generate the audio data? thanks for any insights! -- G. Paul Ziemba FreeBSD unix: 9:06AM up 10 days, 11:05, 4 users, load averages: 1.39, 1.50, 1.54

Re: [asterisk-users] Asterisk as a media gateway

away from your core Asterisk box. I suggest picking up the book[1] and reading the chapter on connecting multiple Asterisk boxes together. [1] http://www.asteriskdocs.org/ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https

Re: [asterisk-users] Asterisk not receiving call from VPN address

On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried

Re: [asterisk-users] Asterisk not receiving call from VPN address

On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server

Re: [asterisk-users] Asterisk not receiving call from VPN address

, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 At this point in time, you'll need to show us a .pcap on the Asterisk box, when you make a call to it via Kamailio. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan

Re: [asterisk-users] Asterisk Fax detection *11.7

to be reliable. If you need fax, you should be using T.38. Your codec is likely the issue. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] Asterisk not receiving call from VPN address

. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Asterisk not receiving call from VPN address

On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine? Well, you need to use tcpdump on each

Re: [asterisk-users] How to install TEST_FRAMEWORK(E) ?

and on). TEST_FRAMEWORK is an option selectable under the Compiler Flags - Development menu in menuselect. ./configure --enable-dev-mode -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com

Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

and insist using SIP. If your ITSP cannot accommodate your request, thank them and look for another provider. H323 is Asterisk is basically dead, sure there is a module, sure it might compile, but you'll be going down the path of zero help. -- Paul Belanger | PolyBeacon, Inc. Jabber

Re: [asterisk-users] Does cdr adaptive odbc re-connect automatically after a long idle time?

logs and see what Asterisk is doing when the odbc connection is down. EG: it should be attempting to reconnect. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

, if you want anybody to call you, you need to leave it open to the public. Meaning, you can't really secure it. Obviously, don't have any outbound trunks configured on the box so that the only location some could dial would be your extension. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan

Re: [asterisk-users] How to read IRQs and timing slips values

, clocking, etc. In the end I had to upgrade dahdi to 2.7+ and the issue went away. Never did figure out the real problem, but to this day I think the issue was a delay on the frames from the PCI bus into the software. All that to say, try upgrading DAHDI and see what happens. -- Paul Belanger

Re: [asterisk-users] Dropped call on new CISCO router for no reason!

max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly appreciated, Nick. Show us the problem, give us a SIP trace[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul

Re: [asterisk-users] AppKonference 2.5

On Dec 17, 2013, at 1:29 AM, virendra bhati virbh...@gmail.com wrote: Good Paul, I used Konference a lot very nice apps, but will this work with asterisk latest version or not ? It should work on the latest asterisk version. I used asterisk 1.4,1.8 but didn't work on 11

[asterisk-users] AppKonference 2.5

is mixed and whispered to the spyee. -- Paul Albrecht -- _ -- 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

Re: [asterisk-users] A Question about Management/Control Protocol Licensing

management/control protocols to this list: ARI, and the ExternalIVR interface. If not, it might be instructive to learn why! Would also like to see this update to include ARI. We talked a little about it at astridevcon, and I think it is likely an oversight. -- Paul Belanger | PolyBeacon, Inc

Re: [asterisk-users] Call Queue advise

? Options 1 - log the agent out, they don't get the next call. Option 2 - Set up weights for your agents, as answer a new call, increment then up so they don't get the next. Either way, I see issues with the setup. Best ways is to rethink your queue strategy and stop using ring all. -- Paul

Re: [asterisk-users] Asterisk on Windows

footing on performance. I don´t mean another slow cygwin port, I man a native Asterisk for windows. In fact, I would invest on the project if somebody wants to do it. Do you just sit around and think shit up to blame Digium all day? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan

Re: [asterisk-users] issue with speech in IVR

entered during prompt) exten = _X,2,Goto(project,s,1) Then you have a DTMF issue, Background will allow DTMF to interrupt the prompts. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https

Re: [asterisk-users] Asterisk uses 105% CPU

tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 55 threads listed. First thing, prune your Asterisk configuration and don't load any modules you don't need to use. Are you really using chan_mgcp, chan_skinny, res_calender, etc. -- Paul Belanger | PolyBeacon

Re: [asterisk-users] issue with speech in IVR

,s,1) my problem when the customor call the number 600 and press 1 in order to go to the project menu he must wait all the speech music1 music2 and music 3 if there is any way to go to project menu during the speech thanks and regards -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan

Re: [asterisk-users] Asterisk uses 105% CPU

there is some trancoding when using voicemail... How can I find out if there is trancoding ?? Maybe explain what your dialplan is doing. Are you making system calls to a database or AGI? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github

Re: [asterisk-users] CEL for attented transfer

. Then make an educated guess about what is happening. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

Re: [asterisk-users] calendar.conf include

On 13-11-13 10:20 AM, Jonas Kellens wrote: Hello, can I use include-statements in the calendar.conf configuration file ? You _should_ be able to use it will every .conf file, otherwise it is a bug. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger

Re: [asterisk-users] Unix connections not always disconnecting

. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Capture dead phone?

the issue at the source. Spend the money for a UPS at each desktop, convert your phones to PoE and install a UPS in your server room. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

for me (no affiliation). Or maybe your Internet link sucks and you need to change your ISP. ^ this Like others said, you really need to drill down and find out where your audio issues are. Local is easy to do, since you control the network, remote is harder. -- Paul Belanger | PolyBeacon, Inc

Re: [asterisk-users] Is this big of new modification in Asterisk Events Objects values ?

is was *[CallerIDnum] * *So 'n' is now 'N' * Asterisk AMI got basically a rewrite[1] of how it works, so there are some breaking changes moving forward. Read ChangeLog and UPGRADE.txt in the source tree for more information. [1] https://wiki.asterisk.org/wiki/display/AST/AMI+1.4+Specification -- Paul

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