Re: [asterisk-users] Asterisk now available with bundled pjproject!

On 2016-03-13 02:30, Recursive wrote: On 07.03.2016 20:28, George Joseph wrote: The current Asterisk 13 and master git branches have a new feature that will be included in 13.8.0: The ability to compile and run Asterisk with a bundled version of pjproject. [...] PLEASE TRY THIS!! I'd love

Re: [asterisk-users] Asterisk now available with bundled pjproject!

On 07.03.2016 20:28, George Joseph wrote: > The current Asterisk 13 and master git branches have a new feature that will > be included in 13.8.0: The ability to compile and run Asterisk with a > bundled version of pjproject. > [...] > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is

Re: [asterisk-users] Asterisk now available with bundled pjproject!

Hi George, Le 07/03/2016 12:53, George Joseph a écrit : > Le 07/03/2016 09:28, George Joseph a écrit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. I don't think this is related to the bundled version, but I got PJSIP_ERXOVERFLOW when initiating a WebRTC video

Re: [asterisk-users] asterisk-users Digest, Vol 140, Issue 15

It does not work. That was the first think I tried. Maybe we need a patch? I don't want to file a bug if there is a workaround. On Sat, Mar 12, 2016 at 1:00 PM, wrote: > Send asterisk-users mailing list submissions to >

Re: [asterisk-users] Asterisk now available with bundled pjproject!

On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard wrote: > Hi, > > Le 07/03/2016 09:28, George Joseph a écrit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. > > I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got: > > [pjproject]

Re: [asterisk-users] Asterisk now available with bundled pjproject!

Hi, Le 07/03/2016 09:28, George Joseph a écrit : > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got: [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2 [pjproject] Applying patches and custom files

[asterisk-users] Asterisk now available with bundled pjproject!

The current Asterisk 13 and master git branches have a new feature that will be included in 13.8.0: The ability to compile and run Asterisk with a bundled version of pjproject. ​​ Why would you want to do this? Several reasons: - Predictability: When built with the ​bundled

Re: [asterisk-users] Asterisk 13 Realtime MusicOnHold

March 4 2016 3:48 PM, "Carlos Chavez" wrote: > I am having a problem trying to use the realtime database for > musiconhold for Asterisk 13. Everything is setup and I can see the mapping: > > ===> musiconhold (db=general, table=musiconhold) > > Only what is in the

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

um.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Patrick Laimbock > Sent: Friday, March 04, 2016 10:58 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Asterisk 13.5 and higher (asterisk > 13.7.2) quitting > > Hi Travis, > > On 0

[asterisk-users] Asterisk 13 Realtime MusicOnHold

I am having a problem trying to use the realtime database for musiconhold for Asterisk 13. Everything is setup and I can see the mapping: ===> musiconhold (db=general, table=musiconhold) Only what is in the musiconhold.conf file appears in Asterisk and the database is completely

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

Hi Travis, On 04-03-16 15:23, Ryan, Travis wrote: I start asterisk 13.7.2 and it dies before I can rasterisk into it. I’ve tried getting a coredump, but it doesn’t coredump. I know there are a lot of errors in the log below, but most of those just say it’ll not load a module, and no big deal.

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

I start asterisk 13.7.2 and it dies before I can rasterisk into it. I've tried getting a coredump, but it doesn't coredump. I know there are a lot of errors in the log below, but most of those just say it'll not load a module, and no big deal. When launching from commandline (not service

Re: [asterisk-users] Asterisk Call Forwarding

On Fri, 4 Mar 2016, Madushan Geethanga wrote: What is redacted means? same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void) Censored. Ususally for political reasons. In this case, the OP didn't want to put a real phone number in a public list. -- Thanks in advance,

Re: [asterisk-users] Asterisk Call Forwarding

Hi What is redacted means? same => n,GotoIf($["${CALLERID(num)}"="**"]?divert:void) Thanks Best Regards, Madushan On Thu, Mar 3, 2016 at 10:58 PM, Madushan Geethanga wrote: > > Hi, > > Thanks Phil, I will implement this and get back to you. > > Best Regards, >

Re: [asterisk-users] Asterisk Call Forwarding

Hi, Thanks Phil, I will implement this and get back to you. Best Regards, Madushan On Thu, Mar 3, 2016 at 4:12 PM, Phil Reynolds < phil-aster...@tinsleyviaduct.com> wrote: > On Thu, 3 Mar 2016 08:21:14 +0530 > Madushan Geethanga wrote: > > > Hi > > I have to setup call

Re: [asterisk-users] Asterisk Call Forwarding

On Thu, 3 Mar 2016 08:21:14 +0530 Madushan Geethanga wrote: > Hi > I have to setup call forwarding. How do we setup Call forwarding in > asterisk?. Eg. user dials a number and insert some mobile number for > forwarding and dial another number to cancel the forwarding.

[asterisk-users] Asterisk Call Forwarding

Hi I have to setup call forwarding. How do we setup Call forwarding in asterisk?. Eg. user dials a number and insert some mobile number for forwarding and dial another number to cancel the forwarding. thanks a lot. Best Regards, Madushan --

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

I'm discovering WebRTC and I think it's a technology that is quite difficult to integrate with so many changing interfaces. I think this is typically the kind of subject where the community could positively contribute to keep wiki pages updated. As I'm quite interested in this topic, I'm

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

Olivier wrote: 2016-02-19 12:01 GMT+01:00 Marek Červenka >: on my own server Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-19 12:01 GMT+01:00 Marek Červenka : > on my own server > Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and

[asterisk-users] Asterisk 13.6.0: ChannelDtmfReceived message generated twice towards the ARI application

I have an ARI application that is registered for Stasis in the dialplan. One of the events I reap in my application is a ChannelDtmfReceived. The thing is, Asterisk 13.6.0 sends me two DTMF for each DTMF pressed (have tried both SIP phones and landlines). That is, I receive two ChannelDtmfReceived

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

on my own server i want try jssip https://github.com/versatica/JsSIP it looks like a lot "livelier" than sipml5 any experience with jssip? Dne 18.2.2016 v 16:01 Olivier napsal(a): 2016-02-18 15:42 GMT+01:00 Marek Červenka >: my

Re: [asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement

That makes sense, so its not possible to have option 'tT' in DIAL() and have directmedia at the same time. Thanks Richard, Regards, Sammy On Thu, Feb 18, 2016 at 4:42 PM, Richard Mudgett wrote: > > > On Thu, Feb 18, 2016 at 3:05 PM, SamyGo wrote: > >>

Re: [asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement

On Thu, Feb 18, 2016 at 3:05 PM, SamyGo wrote: > Hi All, > I've been wondering if I can instruct asterisk in the dialplan to engage > the Media handling for a particular call or not. > > I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf > setting

[asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement

Hi All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: George, May I propose we improve the documentation on the Asterisk Wiki? I thought I would have spent far less time here (though you folks have been mightily helpful, and thanks again!) should the documentation for the TCP transport be improved in both the Wiki and

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

George, May I propose we improve the documentation on the Asterisk Wiki? I thought I would have spent far less time here (though you folks have been mightily helpful, and thanks again!) should the documentation for the TCP transport be improved in both the Wiki and specifically, in

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 15:42 GMT+01:00 Marek Červenka : > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

my experience with pjsip for webrtc http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html Dne 18.2.2016 v 15:36 Olivier napsal(a): 2016-02-18 14:57 GMT+01:00 Simon Hohberg >: Is it

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 14:57 GMT+01:00 Simon Hohberg : > > Is it implied here that both HTTPS and WSS must also come from the same >> server (Same Origin Policy) ? >> > No, the same origin policy does not apply to web sockets. > > Then, can I also install my own WebRTC demo

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ? No, the same origin policy does not apply to web sockets. Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ? If

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

Thank you much for yor reply. 2016-02-18 13:30 GMT+01:00 Simon Hohberg : > Hi Oliver, > > On 02/18/2016 12:10 PM, Olivier wrote: > > Hello, > > I'm trying to have my first calls with WebRTC. > My server has asterisk 13.7.0. > > I'm following the instructions from

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

Hi Oliver, On 02/18/2016 12:10 PM, Olivier wrote: Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type

[asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > Wow. Incredible. That worked. The backslash is important there; I kept > trying with no backslash and followed the instructions in > pjsip_wizard.conf.sample (in configs/samples) and it says we have to say

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Wow. Incredible. That worked. The backslash is important there; I kept trying with no backslash and followed the instructions in pjsip_wizard.conf.sample (in configs/samples) and it says we have to say transport=tcp ; the only example however talks about ipv4. Is this documented somewhere and I

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > I made some progress. The first thing I have realized is that it is my > Twilio configuration in pjsip_wizard.conf that was killing me. I have since > removed that entire file from /etc/asterisk and I am

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

I made some progress. The first thing I have realized is that it is my Twilio configuration in pjsip_wizard.conf that was killing me. I have since removed that entire file from /etc/asterisk and I am able to make "from-internal" context calls (i.e., calls that do not leave the VoIP island).

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Yes, it is enabled on port 5060. I do receive a TCP ACK back from the server, so I know the TCP segment is received at the server hosting the Asterisk build. On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles wrote: > On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: >

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: > OK. Let me ask this. Is anything else necessary, except choosing TCP as the > preferred protocol on the client, to make TCP w Asterisk work? At the > moment, I have only changed one line in pjsip.conf from my working UDP > setup: > >

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: OK. Let me ask this. Is anything else necessary, except choosing TCP as the preferred protocol on the client, to make TCP w Asterisk work? At the moment, I have only changed one line in pjsip.conf from my working UDP setup: [transport-tcp] type=transport protocol=tcp ;

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

OK. Let me ask this. Is anything else necessary, except choosing TCP as the preferred protocol on the client, to make TCP w Asterisk work? At the moment, I have only changed one line in pjsip.conf from my working UDP setup: [transport-tcp] type=transport protocol=tcp ; <--- only this

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

OK. I will report with my findings. It appears increasingly likely that I have done something very silly on my side. It is a little perplexing that the EXACT setup (on the same machine) worked for UDP ... On Wed, Feb 17, 2016 at 8:23 AM, Joshua Colp wrote: > Sonny Rajagopalan

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: Sorry, I was not being very clear, Joshua, and thanks for your patience with this issue. I had set pjsip set logger on and core set debug 99. See absolutely zilch on asterisk CLI. Or in /var/log/asterisk/messages. If the messages are not reaching Asterisk, what could be

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sorry, I was not being very clear, Joshua, and thanks for your patience with this issue. I had set pjsip set logger on and core set debug 99. See absolutely zilch on asterisk CLI. Or in /var/log/asterisk/messages. If the messages are not reaching Asterisk, what could be the issue? I am a little

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: Is there a specific place where I can set logger to log incoming TCP segments from L4? $ netstat -tulpn | grep asterisk | grep LISTEN: tcp0 0 0.0.0.0:8088 0.0.0.0:* LISTEN 10313/asterisk tcp0 0

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Is there a specific place where I can set logger to log incoming TCP segments from L4? $ netstat -tulpn | grep asterisk | grep LISTEN: tcp0 0 0.0.0.0:80880.0.0.0:* LISTEN 10313/asterisk tcp0 0 0.0.0.0:50600.0.0.0:*

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: I receive a TCP ack back from that port (5060; owned by Asterisk) --confirmed by wireshark on the Asterisk server. That's from Wireshark, but what is Asterisk seeing? If Asterisk doesn't show the connection or the traffic then something else is up (firewall, etc).

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

I receive a TCP ack back from that port (5060; owned by Asterisk) --confirmed by wireshark on the Asterisk server. What else should I be looking for? This is on a machine on AWS that was running a UDP based Asterisk fine (I did not make ANY other change other than changing protocol=tcp). I also

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: I can confirm that the server is receiving the SIP request, but simply doesn't do anything with it (log from the server below). Does this have anything to do with how PJSIP was compiled or configured?: TCP support is enabled in PJSIP by default. If you do "pjsip set

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

I can confirm that the server is receiving the SIP request, but simply doesn't do anything with it (log from the server below). Does this have anything to do with how PJSIP was compiled or configured?: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:11.12.13.14 SIP/2.0

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Nope, there are no contacts to show that pertain to these endpoints (only my SIP trunks show up). On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp wrote: > Sonny Rajagopalan wrote: > >> Does this help: >> > > Yes, the transport parameter is in the Contact header so it's

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Does this help: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:1.2.3.4;transport=TCP SIP/2.0 Method: REGISTER Request-URI: sip:1.2.3.4;transport=TCP Request-URI Host Part: 1.2.3.4 [Resent Packet: False] Message Header Via:

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: Does this help: Yes, the transport parameter is in the Contact header so it's interesting it didn't work. If you use pjsip show contacts what is the contact for the AOR? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:;transport=TCP SIP/2.0 That's the request URI, not the Contact header. The Contact contains the URI that the server should dial to reach the client. The full message

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp wrote: > Sonny Rajagopalan wrote: > > > > > *CLI> pjsip set logger on >>

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan wrote: *CLI> pjsip set logger on PJSIP Logging enabled [Feb 15 18:28:12] WARNING[30075]: pjsip:0 : tsx0x7f14b0003 .Temporary failure in sending Request msg OPTIONS/cseq=52002 (tdta0x7f14b0001470), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) [Feb 15

[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

This question was asked by Chirag on March 4 2015 earlier, but I am following exactly the same procedure here and I cannot even get my clients to register on Asterisk. Here's my PJSIP.conf: [transport-tcp] type=transport protocol=tcp bind=0.0.0.0:5061 ... [endpoint_internal](!) type=endpoint

[asterisk-users] Asterisk 11.21.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 11.21.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.21.2 resolves an issue reported by the community and would have not been possible

[asterisk-users] Asterisk 13 realtime static not working

I am trying to port our Asterisk front end to Asterisk 13 but I cannot get realtime static to work. Realtime for PJSIP, Voicemail and Queues is working fine so I know res_odbc is configures properly. In past versions of Asterisk I was using Mysql (res_config_mysql) to load realtime

[asterisk-users] Asterisk & Docker

Given that SIP packets have embedded IP addresses in them and when behind NAT they need to have IP settings to deal with it. How does one set up Asterisk when it is in a docker container? E.g. Docker Container IP <==> Docker Host <==> Gateway <==> Internet 172.17.0.1/16 <==> 192.168.50.1 <==>

[asterisk-users] Asterisk 13.7.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.7.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.7.2 resolves an issue reported by the community and would have not been possible without

[asterisk-users] Asterisk 11.6-cert12, 11.21.1, 13.1-cert3, 13.7.1 Now Available (Security Release)

The Asterisk Development Team has announced security releases for Certified Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1. These releases are available for immediate download at

Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

t;> -- >> *From*: "Sonny Rajagopalan" <sonny.rajagopa...@gmail.com> >> *Sent*: Thursday, January 28, 2016 7:35 PM >> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < >> asterisk-users@lists.digium.com

Re: [asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup

On Tue, Feb 2, 2016 at 11:11 AM, Richard Mudgett wrote: > > Since you didn't specify the channel driver, I took a quick look at the > chan_dahdi, chan_sip, and chan_pjsip channel drivers to see if they > set any default groups. I didn't see any of those channel drivers set

Re: [asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup

On Tue, Feb 2, 2016 at 11:32 AM, John Kiniston wrote: > > Should setting a namedcallgroup & namedpickupgroup supersede numeric > callgroups and pickupgroup ? > No. They operate in parallel. > > I've got 5 peers on my 13.7.0 box, > > Three of them have a namedcallgroup

[asterisk-users] Asterisk not matching peer of incoming call

I'm having an issue with my Asterisk 1.8.21 server and hairpinning a call. Any help would be appreciated. My Asterisk server sends a call out to my proxy. The proxy then routes the call back to Asterisk because it recognizes that the destination is on that same Asterisk server. When the call

[asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup

Should setting a namedcallgroup & namedpickupgroup supersede numeric callgroups and pickupgroup ? I've got 5 peers on my 13.7.0 box, Three of them have a namedcallgroup & namedpickupgroup of 'kiniston' and Two of them have a namedcallgroup & namedpickupgroup of 'sanday'. I'm not specifying a

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables (SOLVED)

solved - https://issues.asterisk.org/jira/browse/ASTERISK-25734?filter=13140 Dne 29.1.2016 v 11:46 Brian :: napsal(a): 12 calls isn't under any type of load. Someone with better understanding of Asterisk internals may chime in here. Could it be vmware timing? Is timing critical when using

Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

28, 2016 7:35 PM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > *Subject*: [asterisk-users] Asterisk 13.6.0: Is there a way to create > PJSIP users and dialplans programmatically using API > > Hi,

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

12 calls isn't under any type of load. Someone with better understanding of Asterisk internals may chime in here. Could it be vmware timing? Is timing critical when using mixmonitor? I've seen > 100 concurrent calls being recorded wtihout issue. On Fri, Jan 29, 2016 at 10:39 AM, Marek

Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

From: "Sonny Rajagopalan" <sonny.rajagopa...@gmail.com> Sent: Thursday, January 28, 2016 7:35 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Subject: [asterisk-users]

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

Dne 28.1.2016 v 13:37 Brian :: napsal(a): when you say load - how many concurrent calls? Is there transcoding happening? sip / PRIs ? what load? 12 concurrent calls no transcoding SIP under 1.5 with 4x 1Ghz vcpus (its vmware VPS) On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka

[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. -- _

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

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

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

[asterisk-users] Asterisk 13.7.0 AutoMixMonitor

On my older Asterisk installs I'm still using Automon because I can set MONITOR_EXEC to run my post process command and use MONITOR_EXEC_ARGS to send it some options I need by adding those to my sip.conf entries with SetVar lines. On my Asterisk 13.7.0 box I want to use the

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

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

[asterisk-users] asterisk 13 mixmonitor - random missing syllables

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? -- --- Marek Cervenka

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

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

[asterisk-users] Asterisk 13.7.0 losing database connection

Hi everyone, I upgraded from Asterisk 13.5.0 to 13.7.0 and I am having database connection problems. I am doing Asterisk realtime with PJSIP 2.4.5 and it works perfectly in 13.5.0. But now I am losing my database connection (running on a virtual box) and I am stuck! I spent all day

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

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

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 19

On Monday 25 Jan 2016, waqas.mehmood90 wrote: > I am working on asterisk ivr .i am facing problrm in crontab.when i run > example it give bash 5:command not found then i check and found that no > crontab for root user kindly guide me please Hello, is that the vet? One of my animals is poorly.

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 19

nd IAX2. But I find no way to configure t38 on a IAX2 > channel. This would be incorrect. IAXMODEM does not support T.38. I'm currently using HylaFAX+ in several of our facilities that are connected by IAX2, what I do is to receive the incoming fax, convert to a print stream or pdf and print the f

[asterisk-users] Asterisk 13.7.0 failed to start - PJSIP 2.4.5

Hello, We installed the subject detailed versions on a uptodate debian wheezy. When starting Asterisk we get Loading chan_pjsip.so. == Registered RTP glue 'PJSIP' == Registered channel type 'PJSIP' (PJSIP Channel Driver) 18:26:10.812 sip_endpoint.c !Module "mod-refer" registered

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 19

On Mon, 25 Jan 2016, waqas.mehmood90 wrote: I am working on asterisk ivr .i am facing problrm in crontab.when i run example it give bash 5:command not found then i check and found that no crontab for root user kindly guide me please If you start your thread with a relevant subject you may

Re: [asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

Gh! 15 minutes after reading your answer, I had it working perfectly! Thank you! Before I type it up, here's what works for me - can you see any obvious flaws or hidden dangers here? - pjsip.conf

[asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate

Re: [asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

Jonathan H wrote: Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat.

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 10

Hello sir have a nice day and sory to distrb you again . I am unable to get solution of my problem Unable to get user extension no from cid Sent from my Samsung Galaxy smartphone. Original message From: asterisk-users-requ...@lists.digium.com Date:15/01/2016 11:00 PM

[asterisk-users] Asterisk 13.7.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.7.0 resolves several issues reported by the community and would have not been possible

[asterisk-users] Asterisk 11.21.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 11.21.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.21.0 resolves several issues reported by the community and would have not been possible

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 8

How to get user extention no in agi php scrip from which he's calling on ivr i am using cid and able to get his name but not his extention no please help me Sent from my Samsung Galaxy smartphone. Original message From: asterisk-users-requ...@lists.digium.com

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 8

On Wednesday 13 Jan 2016, waqas.mehmood90 wrote: > How to get user extention no in agi php scrip from which he's calling on > ivr i am using cid and able to get his name but not his extention no > please help me Within the dialplan, what you are looking for would be ${CALLERID(num)} . So you

Re: [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11

On Mon, 2016-01-11 at 14:52 +0100, Juergen Sauer wrote: > It seems to be, that this fw can not deal with not-numeric-sip accounts. > I entered the extension number as name, account and it works. Glad to hear that. Very interesting. Good to know! > Solved by my self, using Try-and-error Metodic.

Re: [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11

What version of the ST2030 firmware are you using? On Thu, Jan 7, 2016 at 8:59 AM, Juergen Sauer wrote: > Am 07.01.2016 um 10:55 schrieb Frank: > > On Wed, 2016-01-06 at 17:03 +0100, Juergen Sauer wrote: > Thx, 4answer. :) > > >> with in my sip.conf, I have got for

Re: [asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11

Am 11.01.2016 um 14:21 schrieb Scott Griepentrog: > What version of the ST2030 firmware are you using? Hardware Information HW version V3 Software Information Boot Code version V1.01 DSP version V1.00 4 way. App version V1.66 It seems to be, that this fw can not

[asterisk-users] Asterisk 11 and old Thomson 2030S Hardphone => SIP Register/Auth Problem against V11

Hi! I wish you all e Happy New Year first! Allthough, I'm relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. :) Platform is Debian 8/Asterisk Packages (11) from Debian Repo. But I am running into problems setting up 2

Re: [asterisk-users] Asterisk Behind Firewall

I have a /29 to use for the network. My immediate go-to set-up will be to put the asterisk server on a public IP off the /29 and harden the IPtables along with other monitoring scripts and lock down methods. Then add the router on a different /29 IP and have all the phones register through the

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