Re: [asterisk-users] Combine audio and video from two different sources

On Tue, Jul 6, 2021 at 2:14 AM Jean Aunis wrote: > Le 30/06/2021 à 16:10, Ryan Press a écrit : > > [...] > [from-internal-custom] ; Doorbell video bridge > exten => doorbell_rtsp,1,Answer() same => n,RTSP-SIP(rtsp:// > admin:12345@192.168.24.53:554/live/sub,0,asterisk,506

[asterisk-users] Combine audio and video from two different sources

I can easily configure without writing a bunch of new code. Thanks, Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New

Re: [asterisk-users] I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones

I've found the SPA525G2's much easier to deal with than the 7960's. Probably more money, but worth it in my opinion. From: asterisk-users on behalf of Turritopsis Dohrnii Teo En Ming Sent: Friday, December 18, 2020 10:36 AM To: jnov...@stromberg-carlson.org

Re: [asterisk-users] Modems

I can't speak to the other items, but it's always better to have a dedicated FXS to answer the modem calls on an analog line. I've had to do this for ATT network router for their own management and it's always been fine. From: asterisk-users On Behalf Of John T. Bittner Sent: Tuesday,

Re: [asterisk-users] Doing weird bouncing of IAX trunk calls on purpose

;core show channels" and can't be hung up. Any ideas? This just started a month or so ago when we started forwarding all calls as stated below. /shrug From: asterisk-users On Behalf Of Ryan, Travis Sent: Friday, July 26, 2019 11:01 AM To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Doing weird bouncing of IAX trunk calls on purpose

Ok, so this might seem weird, but hang with me on this. I have two sites, Indy and Lafayette that each have their own Asterisk server. They each have their own outside PRI line. They are also trunked internally via and IAX tunnel over a private fiber line. I've recently been asked to have the

Re: [asterisk-users] Paging systems?

You need more than an ATA. You need something with an FSO and FXO. I've used Linksys/SPA3102-3.3.6 and been happy with it. From: asterisk-users On Behalf Of Sebastian Nielsen Sent: Thursday, March 21, 2019 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re:

Re: [asterisk-users] PJSIP DNS ISSUE

Can't you just reference everything in IPs? If not, then hardcode the IPs in your /etc/hosts file. I think that's a bad idea, but that's one way to ensure you always have the Ip of a domain name. From: asterisk-users On Behalf Of John T. Bittner Sent: Wednesday, February 20, 2019 11:30 AM To:

Re: [asterisk-users] DAHDI fax detection

Yes it's very easy. Mine is using a simulated PRI over an ATT Flex line. I just followed the many tutorials out there. I answer the call, then it takes 6-7 seconds (you can add a wait if you want) and then it detects it and drops it to the fax extension in the same context. Also, until

Re: [asterisk-users] Realtime pjsip issues

Any weirdness with realtime has almost always gone back to schema issue for me. Just my experience… On 9/15/17, 10:48 AM, "asterisk-users-boun...@lists.digium.com on behalf of Joshua Colp" wrote: On Fri, Sep 15,

Re: [asterisk-users] corosync and Asterisk 13

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis Sent: Wednesday, July 19, 2017 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] corosync and Asterisk

Re: [asterisk-users] corosync and Asterisk 13

13.13. I REALLY need some help figuring this out.  Thanks! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis Sent: Wednesday, July 19, 2017 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <aster

Re: [asterisk-users] corosync and Asterisk 13

Anyone else using corosync with Asterisk 13 and Ubuntu 16.04 or higher? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis Sent: Wednesday, July 19, 2017 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] corosync and Asterisk 13

s://twitter.com/mhterres https://linkedin.com/in/marceloterres On 19 July 2017 at 14:46, Ryan, Travis <ry...@oscarwinski.com<mailto:ry...@oscarwinski.com>> wrote: I want to use corosync and installed it via ubuntu repository. I guess there is a version 1 and 2 of corosync. For some

[asterisk-users] corosync and Asterisk 13

I want to use corosync and installed it via ubuntu repository. I guess there is a version 1 and 2 of corosync. For some reason ./configure for Asterisk (13) isn't recognizing I have corosync installed. I can't enable the res_corosync module in menuselect. Any ideas? Thanks! Travis --

Re: [asterisk-users] BLF sharing between Asterisk 11 and 13

: Re: [asterisk-users] BLF sharing between Asterisk 11 and 13 On Sun, Jul 16, 2017, at 02:38 PM, Ryan, Travis wrote: > So any phone that wants just state information needs to have an > account on all the servers it needs that information from? Guess I can > do that, but seems to

Re: [asterisk-users] BLF sharing between Asterisk 11 and 13

haring between Asterisk 11 and 13 BLF with pjsip is a little bit different. Did you read the https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+for+Presence+Subscriptions? On 16 Jul 2017 3:38 am, "Ryan, Travis" <ry...@oscarwinski.com<mailto:ry...@oscarwinski.com&g

Re: [asterisk-users] BLF sharing between Asterisk 11 and 13

...@lists.digium.com] On Behalf Of Joshua Colp Sent: Sunday, July 16, 2017 5:34 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] BLF sharing between Asterisk 11 and 13 On Sat, Jul 15, 2017, at 11:37 PM, Ryan, Travis wrote: > I have servers setup in versions 11 and 13. Betw

[asterisk-users] BLF sharing between Asterisk 11 and 13

/pjsip_distributor.c:347 log_unidentified_request: Request from '"Travis Ryan" <sip:6...@yyy.xxx.com>' failed for '10.1.2.XXX:5060' (callid: 8c79c540-c0710...@10.1.2.xxx) - No matching endpoint found How do I make a server allow another extension on another server see

[asterisk-users] Core dump still happening

Ok, so a few years ago, when 13 first came out, I was having a core dump (crash) issue with asterisk 13. I worked with Josh some and even used my Digium subscription for support. Never was able to get it fixed at that time so let it go. Well now I am trying on the same server, after a

Re: [asterisk-users] TDM400P takes too long to ring

format that you have not set up for, this is likely the cause of the delay (looking for caller ID). All the best, David On 27 April 2017 at 12:48, Ryan, Travis <ry...@oscarwinski.com<mailto:ry...@oscarwinski.com>> wrote: Hey all, I have a setup with two analog lines coming into a

[asterisk-users] TDM400P takes too long to ring

Hey all, I have a setup with two analog lines coming into and Asterisk 13 box with a TDM400P and it takes a lot of rings before asterisk takes over. I've traced this same box on two different analog providers so it probably isn't a problem with them. I DO have callerid enabled and not sure I

Re: [asterisk-users] Asterisk inside network. What phone works well?

s with phones outside the network. My go to phones are Polycom VVX series or X-Lite / Bria softphones. The key is to make sure you have configured Asterisk sip.conf with the externip= and nat=yes settings. Additionally on the NAT routers that the outside phones are behind SIP ALG should be disabled. Rya

[asterisk-users] cloud solution?

So if someone has their own hardware and infrastructure but wants a software (not FreePBX but perhaps similar) what options do we have? Would like to virtualize it and not stuck with any one virtualization technology. Discuss... :) Travis --

[asterisk-users] Recommendations for free virtual server tech and Asterisk?

What is the best virtual server tech (and most stable, etc) to use for a asterisk virtual hosting environment? I have a client that wants to do virtual hosting of Asterisk (only SIP or IAX, no PRI, etc) and I'm wondering if Xen or something else would be best? We'd like to stay away from the

[asterisk-users] Create conference bridge via outlook

Is there any way to have a meeting request in Outlook allow someone to attach/setup a conference bridge, time, etc for Asterisk? Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street Lafayette, IN 47905 ry...@oscarwinski.com<mailto:ry...@oscarwinski.

Re: [asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street Lafayette, IN 47905 ry...@oscarwinski.com (765) 742-1102 We're not the IT departmentWe're the I-TEAM department! > -Original Message- > From: asterisk-users-boun...@lists.digi

[asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

script) here is what happens. http://pastebin.com/3GFe6fG9 Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street Lafayette, IN 47905 ry...@oscarwinski.com<mailto:ry...@oscarwinski.com> (765) 742-1102 We're not the IT departmentWe're the I-TEAM depa

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

ess you have an Exchange Enterprise setup in which case I would suggest exploring unified messaging Thanks, Neeraj On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis <ry...@oscarwinski.com<mailto:ry...@oscarwinski.com>> wrote: I am wondering what the best solution is for initiating a call f

[asterisk-users] Dial your phone and contact phone from within outlook?

this? Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street Lafayette, IN 47905 ry...@oscarwinski.com<mailto:ry...@oscarwinski.com> (765) 742-1102 We're not the IT departmentWe're the I-TEAM depa

[asterisk-users] Error making dahdi linux compete 2.11.0

Getting the some errors making dahdi 2.11.0. Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1=96455 In that link they say to use 2.10.2 but that's from December. Is there a fix yet for this? Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North

Re: [asterisk-users] WhatsApp VoIP in Asterisk integration?

That would be cool. Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street Lafayette, IN 47905 ry...@oscarwinski.com (765) 742-1102 We're not the IT departmentWe're the I-TEAM department! > -Original Message- > From: asterisk-user

Re: [asterisk-users] sql schema without alembic

. https://alembic.readthedocs.org/en/latest/offline.html [Ryan, Travis] I’m also very interested. I have tables that are already named the same as alembic uses, so it causes me issues on upgrades. -- _ -- Bandwidth and Coloc

Re: [asterisk-users] Detected alarm on channel 3: Red Alarm

> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Vitor Mazuco > Sent: Tuesday, January 05, 2016 9:21 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Detected

Re: [asterisk-users] Detected alarm on channel 3: Red Alarm

e: [asterisk-users] Detected alarm on channel 3: Red Alarm > > Humm, if I put a filter in this lines, maybe back? > > > > 2016-01-05 12:36 GMT-02:00, Ryan, Travis <ry...@oscarwinski.com>: > > > >> -Original Message- > >> From: asterisk-u

Re: [asterisk-users] weather.agi

http://www.wunderground.com/weather/api/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of d...@donkelly.biz Sent: Wednesday, December 16, 2015 9:20 AM To: 'Asterisk Users Mailing List - Non-Commercial

[asterisk-users] PJSIP and RTT in realtime

So I am using PJSIP realtime with Asterisk 13. I set the qualify_frequency column AORS and it now shows the RTT in milliseconds in the console. I want to be able to display that in a webpage, and was hoping the RTT would be updated in one of the realtime tables, but I don't see it. The old

Re: [asterisk-users] Issues with

Sorry, figured out i had to add ulaw to my tables for my realtime PJSIP setup on the device trying to use it. Thanks, Travis From: <asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>> on behalf of Travis Ryan <ry...@oscarwinsk

[asterisk-users] Issues with

I am trying to get my Linksys/Cisco SPA3102 to connect to asterisk 13 PJSIP. It is registered just fine but when I dial one of my known extensions on the server. As far as I can tell it should be able to translate as also pasted below. Can anyone help me? res_pjsip_sdp_rtp.c:324

Re: [asterisk-users] does res_pjsip support ZRTP?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dmitriy Serov Sent: Tuesday, October 06, 2015 10:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] does res_pjsip support ZRTP?

Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Sunday, October 04, 2015 12:44 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling

Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC

not pulling from ODBC On 15-10-05 09:16 AM, Ryan, Travis wrote: [snip] > > > So should anyone using realtime PJSIP be using the registrations line? Even > if it's not used for any trunking? A registrations line in sorcery.conf for res_pjsip would do absolutely nothing. If you

[asterisk-users] Answering analog call

I have a TDM400P analog card in my asterisk server. I haven't used analog for a while. The caller hears at least two rings before my 312 extension gets rang internally. Does it usually take that long? Below is my relevant dialplan. Also callerID isn't working but that might just be the test

[asterisk-users] MWI and PJSIP

Does something change with MWI when moving from SIP to PJSIP? Seems my phone isn't be alerted of its new VM. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Losing my mind on MWI

Can't get MWI working with PJSIP and my Cisco phones and realtime. I have "mailboxes" populated in the endpoints and aors tables, with 312@default which is the voicemail context. I'm not sure what else to try. Please help! :) Travis --

[asterisk-users] caller id spoofing/setting on analog

I've not used analog for quite some time. It seems it's not possible in asterisk to spoof a phone number/name on an analog call? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Losing my mind on MWI

Of Ryan, Travis Sent: Friday, September 25, 2015 5:28 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Losing my mind on MWI Can't get MWI working with PJSIP and my Cisco phones and realtime. I have "mailboxes" populated in the endpoints and aors tables, with 312@def

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

:23 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04 On 15-09-23 10:25 PM, Ryan, Travis wrote: > Ok I did all that and it's still crashing. I did find some other areas > I think that shouldn't have had any of those files, so I t

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

14.04 On 15-09-23 10:25 PM, Ryan, Travis wrote: > Ok I did all that and it's still crashing. I did find some other areas > I think that shouldn't have had any of those files, so I thought it > would work, because I got rid of ALL of them per your instructions and > completely

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

um.com> wrote: >Ryan, Travis wrote: >>> >> I think i¹m down to the right set of pj and only have one of the files >>for >> pkg-config but now asterisk doesn¹t see that it¹s installed. Also >>ldconfig >> is showing right info. >> >> > >Wh

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

On 9/24/15, 1:30 PM, "asterisk-users-boun...@lists.digium.com on behalf of Joshua Colp" <asterisk-users-boun...@lists.digium.com on behalf of jc...@digium.com> wrote: >Ryan, Travis wrote: >>> >> That folder doesn¹t have any libpj files in it. How do I make it f

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

On 9/24/15, 8:10 AM, "asterisk-users-boun...@lists.digium.com on behalf of Joshua Colp" <asterisk-users-boun...@lists.digium.com on behalf of jc...@digium.com> wrote: >On 15-09-24 08:54 AM, Ryan, Travis wrote: >> >> travis@pcimphone1:~/downloads/ast

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

Did something change DB-wise with PJSIP and realtime between 13.3.2 and 13.5.0? I'm getting an unknown column error and unsure where I need that column and the type it needs to be. Thanks! [Sep 24 15:32:41] -- Attempted to remove non-existent contact 'sip:312@10.1.1.201:5060' from AOR

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

Yes, the schema can change between versions. Following the instructions on https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime#SettingupPJSIPRealtime-InstallingandUsingAlembic will cause alembic to upgrade the tables. -- Joshua Colp Digium, Inc. | Senior Software Developer 445

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

, 2015 10:12 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU > 14.04 > > Ryan, Travis wrote: > > Ok so now I'm getting this when doing a make in asterisk... > > > > travis

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Ryan, Travis > Sent: Wednesday, September 23, 2015 9:55 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: R

[asterisk-users] problems with PJSIP install on UBUNTU 14.04

I've built PJSIP a few months ago on a server that was 12.04 and can't remember how I got past this same issue. I've looked at the links I'll put below and the comments section where others had the issue, but those tips aren't helping either. Basically everything seems to compile and install

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

Spoke too soon. Same thing. Josh, any other ideas? > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Ryan, Travis > Sent: Wednesday, September 23, 2015 10:50 AM > To: Asterisk Users Mai

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

e: [asterisk-users] problems with PJSIP install on UBUNTU > 14.04 > > Ryan, Travis wrote: > > I've built PJSIP a few months ago on a server that was 12.04 and > can't > > remember how I got past this same issue. I've looked at the links > I'll > > put below

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

essage- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Ryan, Travis > Sent: Wednesday, September 23, 2015 10:01 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] proble

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

roblems with PJSIP install on UBUNTU > 14.04 > > On 15-09-23 12:14 PM, Ryan, Travis wrote: > > Spoke too soon. Same thing. > > > > Josh, any other ideas? > > Not really, that's the exact configure line I use. You may have to do a > "make distclean" on

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

s-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Ryan, Travis > Sent: Wednesday, September 23, 2015 11:46 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU > 14.04 > > &

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

m > Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU > 14.04 > > On 15-09-23 07:17 PM, Ryan, Travis wrote: > > Getting constant segfaults now... > > > > [ 157.894809] asterisk[1424]: segfault at c ip 7f8b2fbcfd04 sp > > 7f8b9172

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

with PJSIP install on UBUNTU > 14.04 > > On 15-09-23 07:53 PM, Ryan, Travis wrote: > > > > I'm not sure what that means. I just built it how the wiki says too, > > and earlier messages in this thread. J > > It means not all instances of PJSIP were removed

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

with PJSIP install on UBUNTU > 14.04 > > On 15-09-23 07:36 PM, Ryan, Travis wrote: > > I've got the backtrace, but how much of the info do you want? > > Ideally everything. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan D

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

roblems with PJSIP install on UBUNTU > 14.04 > > On 15-09-23 07:17 PM, Ryan, Travis wrote: > > Getting constant segfaults now... > > > > [ 157.894809] asterisk[1424]: segfault at c ip 7f8b2fbcfd04 sp > > 7f8b91722010 error 4 in res_hep_pjsip.so[7f8b2fba2000+4500

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

14.04 On Wed, Sep 23, 2015 at 5:43 PM, Ryan, Travis <ry...@oscarwinski.com<mailto:ry...@oscarwinski.com>> wrote: > -Original Message- > From: > asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> > [mailto:asteris

Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04

install on UBUNTU 14.04 On 15-09-23 08:00 PM, Ryan, Travis wrote: > I ran all the uninstall commands and the rm commands. Made sure that ldconfig > -p had no pj stuff in it. > > That's not enough? What did I miss? "make uninstall" uses the configure parameters. You can do:

Re: [asterisk-users] Asterisk uses Anonymous, but why?

the AMI Originate action or a call file. You can specify a caller id there. You cannot specify one from the command line. Richard [Ryan, Travis] Are you sure? I have no issue with a PRI line and using the set command like so…Unless it’s a toll free number. Set(CALLERID(num)=765637

Re: [asterisk-users] showing sip number insted of pri number

I¹m not quite sure I understand everything you typed, but I¹ll say the following. You can spoof any phone number on any call outbound through your PRI, except any toll free numbers, etc. Basically if someone else is paying for the phone call, like a toll free number is, then the payer (called

Re: [asterisk-users] Asterisk 13 / realtime voicemail creation

Ø From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oceanet - Cédric BASSAGET Sent: Thursday, July 09, 2015 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 13 / realtime

Re: [asterisk-users] tls on asterisk 13

Asterisk13 can do native tls with each phone? Nice. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ricky gutierrez Sent: Wednesday, July 08, 2015 3:06 PM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] CDR in an MySQL-Database

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis Sent: Monday, July 06, 2015 4:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] CDR in an MySQL

Re: [asterisk-users] DTMF issue

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees Sent: Monday, July 06, 2015 5:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF issue Hello folks, We have an issue with several Cisco SPA512G phones

Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

I've seen this before. It can be done by calling an AGI script when placing the outgoing call. You'd then prompt and make sure the code matches and do your billing logic, etc there. Then place the call if it's valid. -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] asterisk email to fax

I hope his mother in law doesn't live with him. That's a support issue for sure. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen Sent: Thursday, June 25, 2015 2:50 PM To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] error trying to get PJSIP working

I'm doing an upgrade from Asterisk 11 to 13. I'm following the guide at https://wiki.asterisk.org/wiki/display/ast/setting+up+PJSIP+REaltime to setup realtime, as I use realtime on Asterisk 11 too. I'm getting the following error when trying to connect the peer to the server. Help? :) Thanks,

Re: [asterisk-users] small pbx for the office [it was: small homebrew pbx]

or a Cisco router with FXO card and DSP modules. I have deployed both and haven't had any complaints. They just work once configured. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Strange Polycom Issue

Polycom will fix the issue in 5.3 in a few months.. Thanks David Could this be a 5.2.x issue only? I have a hundred of the VVX 400 phones running 4.1.7 and haven't heard of this issue yet from our users. Thanks, Ryan

Re: [asterisk-users] chan_sip and 2 devices under same extension - transferring call endpoint(s)

Asterisk then dials the outbound number. No need for any transferring. You could also look at Asterisk call files to originate the call. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Asterisk LTS segment faults

rock solid performance. Ryan -- _ -- 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

[asterisk-users] Certified Asterisk 11.6 Menuselect

Has there been a change in the way certified Asterisk is being packaged? Starting with certified Asterisk 11.6 has all the extended options are checked by default in menuslect? Certified Asterisk 11.2 does not have them checked and neither does certified Asterisk 1.8.15? Thanks, Ryan

Re: [asterisk-users] DUNDi with SIP Mapping

On Wed, Apr 16, 2014 at 10:20 AM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: You are a bit outside of what I have done, but this looks like it might be what you want to do with SIP: http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP I had looked at that guide

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

| grep filename | wc -l Thanks, Ryan -- _ -- 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] DUNDi with SIP Mapping

to dundi) in new stack Is there a way to configure DUNDi to use SIP or does it only work with IAX? Thanks, Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] DUNDi with SIP Mapping

is going to become cumbersome. I wanted to move to DUNDi to simplify the setup. It looks like I need to switch to IAX trunks to be able to do this. Thanks, Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450

and version 4.1.5 on a Polycom VVX 400. Buddies work on all three phones. The firmware is for both SIP and Lync. You change the base profile option accordingly. Look in the Polycom UC Software Admin Guide for more information. Ryan

Re: [asterisk-users] Maximum number of users

for the majority of the day, and a total of 8-10k calls processed per day. A few times a week I will see the last minute load at 20 and the 5 min load at 7. This seem to happen when there are a high volume of new calls as the FreePBX dialplan is complex. Ryan

[asterisk-users] Asterisk not sending bye message to original UA

UA leaving the first phone stuck in a holding state. Am I missing something here? Here is the refer sip message (see attached) Thank you! -- Ryan Tilton Seattle Event Disc Jockey www.DJRT.com Toll Free: 1-877-411-DJRT Cell: 206-409-3906 Fax: 206-922-6199 -- Reviews at DJRT.COM

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

be ulaw this for channel. However I'm not sure how to make this change as I don't know my way around the interaction with the Asterisk core and the channels. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

a higher bandwidth codec. I looked at this a year ago on Asterisk 1.8 and ended up using ulaw for everything but remote phones. The remote phones end up transcoding g729 to ulaw for most calls. Ryan -- _ -- Bandwidth

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

with pjsip will have a better solution. Ryan -- _ -- 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] Why doesn't Asterisk try to prevent transcoding

transcoding? Ryan -- _ -- 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

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

On Sat, Dec 14, 2013 at 10:31 PM, Ryan Wagoner rswago...@gmail.com wrote: Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw

Re: [asterisk-users] Asterisk on Windows

ESXi server with Windows VMs for AD and Exchange and Linux VMs for Asterisk and Web / FTP. Asterisk with Exchange UM for voicemail is a winning combination and works seamlessly. It is essentially a private cloud of the customer. Why not use the OS that works for the task at hand? Ryan

Re: [asterisk-users] Trouble with upgrading - RBS T1

. Although this might be tricky depending on the OS version. Ryan On Mon, Dec 9, 2013 at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote: Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5

Re: [asterisk-users] SIP Presence across two servers

I haven't tried it, but the res_corosync module states it will sync device state across servers. https://wiki.asterisk.org/wiki/display/AST/Corosync On Thu, Nov 14, 2013 at 3:54 AM, Leandro Dardini ldard...@gmail.com wrote: Aligning presence over multiple servers is not simple and require

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

form of raid for redundancy. I usually go with two 15K SAS drives in raid 1 or four 7.2k SATA drives in raid 10. Performance between the two should be similar. With drives being as cheap as they are skip raid 5. Ryan -- _ -- Bandwidth

Re: [asterisk-users] T.38 gateway patch against Asterisk 1.8.11.0

On Fri, Mar 30, 2012 at 1:16 PM, Bryant Zimmerman brya...@zktech.comwrote: What does this patch fix? Why is it not in Jarr? Thanks Bryant It looks like the patch is a backport of the t.38 gateway functionality in Asterisk 1.10. Ryan

Re: [asterisk-users] Stuck DAHDI Lines

, DAHDI 2.5, and libpri 1.4.12. I'm currently up to 16 weeks, 2 days of uptime and 914,745 calls processed without a stuck channel. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

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