Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

Size: - one location - 15 IP Phones ( 1 dect) - Create new voip trunk (current are ISDN) (30 number block) - LTS is important - an SLA is optional at the moment there is no one On 11.12.2017 22:31, Ron Wheeler wrote: > You might want to add some details > - size of the project >  -- number of

Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

You might want to add some details - size of the project  -- number of locations  -- number of extensions - are you converting your trunks? - what are your thoughts on hardware - brands, type of stations - what type of long-term support do you want the consultant to provide? - what size company

Re: [asterisk-users] Asterisk / FreePBX Support / Reseller

Hello, we plan to move a PBX to asterisk and searching for Support and a Phonehardware Reseller in Germany. The should be no license costs per User / Server. - Install Configure Asterisk for our specification - Install FreePBX or similar (optional) - Resell Hardware Thanks for any suggest.

[asterisk-users] Asterisk BackgroundDetect and the talk extension

I am using AMI to issue a BackgroundDetect on a channel. Everything works great, I receive the result and the variables on the channel. I am running into one issue though. After calling that function on AMI, when I send the next command on AMI for that channel. For example, a Playback. This

[asterisk-users] Asterisk 13.18.3, 14.7.3, 15.1.3 and Certified Asterisk 13.13-cert8 Now Available

The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available security releases are released as versions 13.13-cert8, 13.18.3, 14.7.3 and 15.1.3. These releases are available for immediate download at

Re: [asterisk-users] asterisk-users Digest, Vol 160, Issue 5

Get Outlook for Android From: asterisk-users-boun...@lists.digium.com on behalf of asterisk-users-requ...@lists.digium.com Sent: Monday, November 6, 2017 6:00:01 PM To:

[asterisk-users] Asterisk 15.1.2 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.1.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.1.2 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.18.2 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.18.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.18.2 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 14.7.2 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 14.7.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.7.2 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.8 compile error

I just tried to compile the latest Asterisk 13.8.0 and it stopped with several errors on pjsip. So FYI if you run the install_prereq script and then use ./configure --with-pjproject-bundled you will have the same problem because the prereq script installs an older version of pjproject.

Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

On Wednesday 01 November 2017 at 12:15:08, Michael Maier wrote: > On 11/01/2017 at 10:14 AM Antony Stone wrote: > > On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote: > >> > >> I'm facing the following scenario: > >> > >> - Initial call opened to asterisk: SDP g722,alaw,ulaw > >>

Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

On 11/01/2017 at 10:14 AM Antony Stone wrote: > On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote: > >> Hello! >> >> I'm facing the following scenario: >> >> - Initial call opened to asterisk: SDP g722,alaw,ulaw >> >> - Outgoing call to provider started with Invite / SDP alaw, g726

Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

On 11/01/2017 10:14, Antony Stone wrote: > > If you don't have a G.729 licence, don't offer G.729 to the peer. AFAIK the g729 patents have expired or are granted royalty free for the holder's declaration: http://www.sipro.com/G729.html -- Guido Falsi --

Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote: > Hello! > > I'm facing the following scenario: > > - Initial call opened to asterisk: SDP g722,alaw,ulaw > > - Outgoing call to provider started with Invite / SDP alaw, g726 and > g729. So, you're claiming to the provider that

[asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

Hello! I'm facing the following scenario: - Initial call opened to asterisk: SDP g722,alaw,ulaw - Outgoing call to provider started with Invite / SDP alaw, g726 and g729. - Provider sends 183 Session progress SDP: g729, alaw - Provider sends g729 rtp packages But: there is no license to

[asterisk-users] Asterisk 15.1.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.1.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 14.7.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 14.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.7.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.18.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.18.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk EOL Announcement

Dearly Beloved, We have gathered here today to mourn the passing of a deeply regarded branch of Asterisk - Asterisk 11. As of today, it has officially reached its end of life. It was a good branch, having served 5 years faithfully in the service of its users. As far as history goes, 11.0.0 was

[asterisk-users] Asterisk 14 Security Fix Only Mode

Hey all, For those who may not be aware Asterisk 14 transitioned from bug fix mode to security-fix-only mode a few weeks ago (Sept 26th). For those of you that are still on this release, it's a good time to consider building an upgrade plan for moving to 15.x.x. I sincerely apologize for the

Re: [asterisk-users] Asterisk chan_sip registration attempts

Thank you, Matt. Is it possible that Asterisk (by mistake or misconfiguration or something else) stops sending REGISTER requests or is it possible for the ISP server to send some kind of SIP-message to stop Asterisk's registering attempts (I'm not very familiar with all SIP message types)? On

Re: [asterisk-users] Asterisk chan_sip registration attempts

Maybe the provider has added an extra gateway and it is not processing accounts correctly. If they had one before and now two then 40-60% registration fails would show that. Kind regards, Matt > On Oct 10, 2017, at 06:27, Dmitriy Ermakov wrote: > > Hello! > > Could

[asterisk-users] Asterisk chan_sip registration attempts

Hello! Could you help me with Asterisk 11.21.2 and AsteriskNow platform. The problem is: My Asterisk PBX has SIP (chan_sip) trunk to provider. Asterisk periodically loses trunk registratrion: *sip show registry:* /Host    dnsmgr Username   Refresh State

Re: [asterisk-users] Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP

The problem the 183 is received with mode sendonly and playbacks an audio, so when the destination playbacks audio, the origin was put on hold. | | | INVITE | |-->| | 183 Session Progress/SDP

Re: [asterisk-users] Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP

I think it is normal, the call is placed on hold as soon as the remote media address is null. It makes sense because when a 183 is sent, some media is supposed to be sent as with a 200, so placing the call on hold when no media is available sounds logic. Le 06/10/2017 à 03:56, Rafael dos

[asterisk-users] Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP

Hi Is it a normal behavior of Asterisk put a call on hold when receive a Session Progress with media address 0.0.0.0 in SDP? I believe the call on hold should be initiate with a re-invite. Thanks -- Att, Rafael Saraiva -- _

Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations

On Tue, Sep 26, 2017, at 05:53 PM, marek cervenka wrote: > Dne 26/09/2017 v 22:33 Joshua Colp napsal(a): > > On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: > >> hi, > >> > >> i want use asterisk+pjsip as voip client with multiple registrations > >> (perf testing) > >> > >> i'm using this

Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations

Dne 26/09/2017 v 22:33 Joshua Colp napsal(a): On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308

Re: [asterisk-users] Asterisk pjsip registration issues - Solved

quot; <dpl...@radagast.org> Sent: Tuesday, September 26, 2017 3:28 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk pjsip registration issues > Hey all > > I am trying to register a PJSIP server on our office to an Asterisk 11 > chan_sip server

Re: [asterisk-users] asterisk pjsip as voip client with multiple registrations

On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote: > hi, > > i want use asterisk+pjsip as voip client with multiple registrations > (perf testing) > > i'm using this example configuration for one account > > [308] > type=registration > outbound_auth=308 >

[asterisk-users] asterisk pjsip as voip client with multiple registrations

hi, i want use asterisk+pjsip as voip client with multiple registrations (perf testing) i'm using this example configuration for one account [308] type=registration outbound_auth=308 server_uri=sip:3...@example.com:5060 client_uri=sip:3...@example.com:5060 [308](auth-userpass) username=308

Re: [asterisk-users] Asterisk pjsip registration issues

> Hey all > > I am trying to register a PJSIP server on our office to an Asterisk 11 > chan_sip server in a datacenter. > > I keep getting > WARNING[18084]: res_pjsip_outbound_authenticator_digest.c:178 > digest_create_request_with_auth_from_old: Host: 'XXX.XXX.XXX.XXX:5060': > Unable to

Re: [asterisk-users] Asterisk pjsip registration issues

On Tue, Sep 26, 2017, at 10:33 AM, Bryant Zimmerman wrote: > Hey all > > I am hoping someone can assist I have now spent over a week trying to > figure out what is going on with PJSIP registrations. > > I am able to register handsets against an asterisk 13 server running > pjsip, but I

[asterisk-users] Asterisk pjsip registration issues

Hey all I am hoping someone can assist I have now spent over a week trying to figure out what is going on with PJSIP registrations. I am able to register handsets against an asterisk 13 server running pjsip, but I am not able to get pjsip to register out to an older chan_sip asterisk

Re: [asterisk-users] Asterisk 15, Jack, streams, speech recognition… so many questions!

> On 22/09/2017, at 8:08 AM, Jonathan H wrote: > > Removing the "record in Asterisk/store as file/convert file/upload > file <> receive stream/save file/convert file/playback in Asterisk" > part of the sequence would save vital seconds of silence and caller > annoyance.

[asterisk-users] Asterisk 15, Jack, streams, speech recognition… so many questions!

I know Asterisk has a speech recognition interface built in, but I need to go beyond that, with APIs like Lex, Wit or Luis etc. There's also the very cheap/free high quality speech synthesis services like Amazon Polly, which can also return an audio stream object (or save a file). These APIs can

[asterisk-users] Asterisk 11.25.3, 13.17.2, 14.6.2, Asterisk 11.6-cert18, Asterisk 13.13-cert6 Now Available (Security Release)

The Asterisk Development Team has announced security releases for Asterisk 11, 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6. These releases are available for immediate download at

[asterisk-users] Asterisk and Virtual routing and forwarding (VRF)

Hello, Reading [1], Asterisk's PJSIP requires a unique IP+port combination when configuring multiple transport instances. Linux now supports Virtual routing and forwarding (VRF) (see [2]). Is there a way to set a given PJSIP transport to use a given interface and doing so, brings VRF-awareness

Re: [asterisk-users] Asterisk 13.X with multiple IP addresses: Can I force a given chan_sip peer to a given IP address ?

2017-09-06 15:53 GMT+02:00 Dovid Bender : > Oliver, > > Not per peer. > That confirms what I thought (feared ?). Maybe, I'm gonna try with PJSIP channel instead as you can specify a peer-specific transport with PJSIP. I'll check if if using a specific IP is PJSIP transport

Re: [asterisk-users] Asterisk 13.X with multiple IP addresses: Can I force a given chan_sip peer to a given IP address ?

Oliver, Not per peer. What you can do is if you have a few spare IP's is set up OpenSiPS and bounce the calls that way. The call will be Carrier -> OpenSipS -> Asterisk. Asterisk will see the IP of OpenSipS and you can decide there what to do with the call. On Wed, Sep 6, 2017 at 5:39 AM,

Re: [asterisk-users] Asterisk 11.25.2

Marcelo, I had to drop back to 11.25.1 yesterday. Had other things all day yesterday. I have not had a chance to try anything yet with the suggestion from Josh. Jerry -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Asterisk 13.X with multiple IP addresses: Can I force a given chan_sip peer to a given IP address ?

Hello, I'm quite sure this question has already be asked previously but before diving into it with a lab setup, I would like to re-ask here the thereafter question. I've got a bunch of very old Asterisk boxes (lastest Asterisk version is 1.6.1.X), all belonging to the same network, I would like

Re: [asterisk-users] Asterisk 11.25.2

Hello Jerry. Does the Joshua's tips helped you to solve your issues or are you still facing audios problems? I am asking you because I need to update some servers but I can't have this kind of problems. Thanks. Regards, On 5 Sep 2017 2:02 pm, "Joshua Colp" wrote: > On Tue,

Re: [asterisk-users] Asterisk 11.25.2

On Tue, Sep 5, 2017, at 09:56 AM, Jerry Geis wrote: > My setup using 11.25.1 was working. When I installed 11.25.2 I now get > "sort of" working. > > I am using NAT in the setup. When I have an internal phone and call out I > get audio both ways. > But when I call IN my phone rings but I have no

[asterisk-users] Asterisk 11.25.2

My setup using 11.25.1 was working. When I installed 11.25.2 I now get "sort of" working. I am using NAT in the setup. When I have an internal phone and call out I get audio both ways. But when I call IN my phone rings but I have no audio. Is there a new setting I need to tweek ? Thanks, Jerry

Re: [asterisk-users] Asterisk bugs make a right mess of RTP

That is correct. have a look at rtp.conf. On Fri, Sep 1, 2017 at 10:54 AM, Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Friday 01 September 2017 at 16:48:17, Dovid Bender wrote: > > > On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp wrote: > > > On Fri, Sep 1,

Re: [asterisk-users] Asterisk bugs make a right mess of RTP

On Friday 01 September 2017 at 16:48:17, Dovid Bender wrote: > On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp wrote: > > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote: > > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ > As Josh mentioned this is an issue

Re: [asterisk-users] Asterisk bugs make a right mess of RTP

On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp wrote: > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote: > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ > > This specific issue exists in a lot of different implementations and > devices. Unfortunately there's

Re: [asterisk-users] Asterisk bugs make a right mess of RTP

On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote: > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ This specific issue exists in a lot of different implementations and devices. Unfortunately there's nothing within SDP that guarantees or provides what the source of media should

[asterisk-users] Asterisk bugs make a right mess of RTP

http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ -- Dave Topping e: i...@dntopping.uk t: 03445 888 888 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

Re: [asterisk-users] Asterisk Voicemail changes

7 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk Voicemail changes > > On Friday 01 Sep 2017, Tim Turpin wrote: >> Is there a way that I can modify the source code for the voicemail >> application? I need to chan

Re: [asterisk-users] Asterisk Voicemail changes

-users] Asterisk Voicemail changes On Friday 01 Sep 2017, Tim Turpin wrote: > Is there a way that I can modify the source code for the voicemail > application? I need to change some of the options in the user’s > interface to make it work like an existing system that I’m replacing. $ vi

Re: [asterisk-users] Asterisk Voicemail changes

On Friday 01 Sep 2017, Tim Turpin wrote: > Is there a way that I can modify the source code for the voicemail > application? I need to change some of the options in the user’s interface > to make it work like an existing system that I’m replacing. $ vi /usr/src/asterisk-*/apps/app_voicemail.c

Re: [asterisk-users] Asterisk Voicemail changes

-boun...@lists.digium.com] On Behalf Of Jonathan H Sent: Thursday, August 31, 2017 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Voicemail changes What about just using the built-in options? In http://doxygen.asterisk.org/trunk

Re: [asterisk-users] Asterisk Voicemail changes

rs-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *On Behalf Of *Jonathan H > *Sent:* Thursday, August 31, 2017 6:13 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk Voicemail changes > &

Re: [asterisk-users] Asterisk Voicemail changes

, August 31, 2017 6:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Voicemail changes Well, yes, anyone can recompile anything! But what exactly is it that the current voicemail can't do or be modified to do through normal dialplan and config

Re: [asterisk-users] Asterisk Voicemail changes

an H > *Sent:* Thursday, August 31, 2017 4:13 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk Voicemail changes > > > > What about MiniVM? http://doxygen.asterisk.org/trunk/App_minivm.html > > E

Re: [asterisk-users] Asterisk Voicemail changes

] On Behalf Of Jonathan H Sent: Thursday, August 31, 2017 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Voicemail changes What about MiniVM? http://doxygen.asterisk.org/trunk/App_minivm.html Example: http://doxygen.asterisk.org/trunk

Re: [asterisk-users] Asterisk Voicemail changes

What about MiniVM? http://doxygen.asterisk.org/trunk/App_minivm.html Example: http://doxygen.asterisk.org/trunk/Config_minivm_examples.html That said, I don't know if it's actually actively developed or stable (docs last updated 2015 - Asterisk team?) Also make sure your Asterisk is up to date

[asterisk-users] Asterisk 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, 13.13-cert5 Now Available (Security Release)

The Asterisk Development Team has announced security releases for Asterisk 11, 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5. These releases are available for immediate download at

[asterisk-users] Asterisk Voicemail changes

Is there a way that I can modify the source code for the voicemail application? I need to change some of the options in the user's interface to make it work like an existing system that I'm replacing. Thanks. Tim -- _ --

[asterisk-users] Asterisk 15 Beta Released

It is with great pleasure I wish to inform you of the first beta release of the new Asterisk 15 branch. It's a very exciting time to be a user of Asterisk! Asterisk 15 is arguably the biggest release of Asterisk that has happened in the last 10 or so years. There has been a lot of work done in the

Re: [asterisk-users] Asterisk 15.0.0-beta1 Now Available

On Wed, Aug 2, 2017, at 02:28 PM, Ira wrote: > Re: [asterisk-users] Asterisk 15.0.0-beta1 Now AvailableHello Asterisk, > > Wednesday, August 2, 2017, 9:20:19 AM, you wrote: > > > > The Asterisk Development Team would like to announce the first beta > > of

Re: [asterisk-users] Asterisk 15.0.0-beta1 Now Available

Title: Re: [asterisk-users] Asterisk 15.0.0-beta1 Now Available Hello Asterisk, Wednesday, August 2, 2017, 9:20:19 AM, you wrote: > The Asterisk Development Team would like to announce the first beta of Asterisk 15.0.0. > This beta is available for immediate download at

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

On 2017-08-02 07:08, Nathan Anderson wrote: Richard Kenner wrote: But the question here was *Asterisk*, not kernels. User-level code has *way* fewer dependencies. *Precisely*. Unless we're talking DAHDI here (which we're not), Linux & ESXi are red herrings. Carlos Chavez wrote: I

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

Richard Kenner wrote: > But the question here > was *Asterisk*, not kernels. User-level code has *way* fewer > dependencies. *Precisely*. Unless we're talking DAHDI here (which we're not), Linux & ESXi are red herrings. Carlos Chavez wrote: > I am having a very tough time trying to

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

On Tue, Aug 1, 2017 at 12:53 PM, Carlos Chavez wrote: > I am having a very tough time trying to replace an Elastix 2.X > install running as a virtual machine on ESXI 4. I tried using the Freepbx > 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I keep

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

> There are certain versions of the Linux kernel that have no support > under the older version of ESXI. We started having issues under our > ESXI v4 setup with RH Enterprise and vmware's response was, "It's > not supported" "not supported" and "does not work" are not the same thing. ESXI

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

>>> On Aug 2, 2017, at 6:45 AM, Richard Kenner ken...@gnat.com wrote: >>> I wouldn't believe it either. I'd be quite surprised if something won't >>> work with any ESXI version. *Perhaps* there's a configuration issue, but >>> I'd be surprised about that too. There are certain versions of the

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

> The version is licensed and the customer does not want to invest on new > hardware/software at the moment. If the ESXI version is too old I need > to give them definitive proof that the segfaults are caused by that but > since the old elastix has been running there for years they do not

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

On 2017-08-01 15:48, Doug Lytle wrote: I am having a very tough time trying to replace an Elastix 2.X install running as a virtual machine on ESXI 4 Licensed or free ESXI? I want to say your version is too old. I'm currently running ESXI 6.0 update 3 at home and Asterisk in a VM under debian

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

>>> I am having a very tough time trying to replace an Elastix 2.X >>> install running as a virtual machine on ESXI 4 Licensed or free ESXI? I want to say your version is too old. I'm currently running ESXI 6.0 update 3 at home and Asterisk in a VM under debian without issue. Doug --

[asterisk-users] Asterisk 13 on old VMware ESXI 4

I am having a very tough time trying to replace an Elastix 2.X install running as a virtual machine on ESXI 4. I tried using the Freepbx 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I keep getting random segfaults: [175711.476685] asterisk[2942]: segfault at 188 ip

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Floimair Florian Gesendet: Donnerstag, 13. Juli 2017 11:52 An: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Betreff: Re: [asterisk-users] Asterisk re

Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

- Non-Commercial Discussion <asterisk-users@lists.digium.com> *Subject:* Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264) Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. Why not try removing all codecs

Re: [asterisk-users] Asterisk install_prereq

On Fri, Jul 21, 2017, at 10:32 PM, Carlos Chavez wrote: > Is there a reason for Asterisk 13.17.0 to download and install > pjproject-devel-2.3-6.el7.x86_64.rpm when you run the install_prereq > script? Since most people will compile asterisk using the bundled > version of pjproject this

[asterisk-users] Asterisk install_prereq

Is there a reason for Asterisk 13.17.0 to download and install pjproject-devel-2.3-6.el7.x86_64.rpm when you run the install_prereq script? Since most people will compile asterisk using the bundled version of pjproject this may cause confusion. And it is also an older version than the

Re: [asterisk-users] Asterisk 13.16.0 segfault

On Thursday 20 July 2017 at 20:46:30, Marcelo Terres wrote: > I don't have much knowledge about freepbx, but if some day I had to use it, > I would prefer to use the Asterisk compiled from source, unless it comes > with an Asterisk package (rpm, supposing it is running CentOS). FreePBX (as a

Re: [asterisk-users] Asterisk 13.16.0 segfault

I don't have much knowledge about freepbx, but if some day I had to use it, I would prefer to use the Asterisk compiled from source, unless it comes with an Asterisk package (rpm, supposing it is running CentOS). On 20 Jul 2017 5:08 pm, "Carlos Chavez" wrote: > On 7/20/17

Re: [asterisk-users] Asterisk 13.16.0 segfault

On 7/20/17 8:47 AM, Marcelo Terres wrote: Which version of Asterisk are you using? Are you compiling it with the bundle pjproject ? --with-pjproject-bundled Regards, Marcelo H. Terres > IM: mhter...@jabber.mundoopensource.com.br

Re: [asterisk-users] Asterisk 13.16.0 segfault

Which version of Asterisk are you using? Are you compiling it with the bundle pjproject ? --with-pjproject-bundled Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres

Re: [asterisk-users] Asterisk 13.16.0 segfault

On 7/19/17 2:37 AM, Marcelo Terres wrote: This is the pjsip library. Is it possible to you to update pjsip for the latest version to test if it solves the problem? On 18 Jul 2017 3:52 pm, "Carlos Chavez" > wrote: I am getting frequent

Re: [asterisk-users] Asterisk 13.16.0 segfault

This is the pjsip library. Is it possible to you to update pjsip for the latest version to test if it solves the problem? On 18 Jul 2017 3:52 pm, "Carlos Chavez" wrote: > I am getting frequent segfaults on a new Asterisk installation. So far > the only message I see is: >

[asterisk-users] Asterisk 13.16.0 segfault

I am getting frequent segfaults on a new Asterisk installation. So far the only message I see is: Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 7fb2d535723f sp 7fb25a11b5c0 error 4 in libasteriskpj.so.2[7fb2d52e5000+18] Jul 18 09:17:00 pbxbogota kernel:

Re: [asterisk-users] Asterisk realtime in combination with ARI - error while trying to prepare SQL statement for writing into database

On Thu, Jul 13, 2017, at 10:01 AM, Floimair Florian wrote: > Hey guys! > > I successfully got Asterisk realtime (14.6.0) with MariaDB (MySQL fork) > running on Debian 9. > > I will document the steps to do so shortly (the main difference is > default encoding and the odbc connector & its

[asterisk-users] Asterisk realtime in combination with ARI - error while trying to prepare SQL statement for writing into database

Hey guys! I successfully got Asterisk realtime (14.6.0) with MariaDB (MySQL fork) running on Debian 9. I will document the steps to do so shortly (the main difference is default encoding and the odbc connector & its configuration). What I’m trying to do now is to use ARI to create PJSIP

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

Terres Gesendet: Mittwoch, 12. Juli 2017 22:55 An: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Betreff: Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script Please open a Ticket (https://issues.asterisk.org), to let the

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

Gesendet: Mittwoch, 12. Juli 2017 13:50 > An: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > Betreff: [asterisk-users] Asterisk realtime - Error with index length in > alembic script > > Hi! > > I just tried setting u

[asterisk-users] Asterisk 14.6.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 14.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.6.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.17.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.17.0 resolves several issues reported by the community and would have not been

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

: Mittwoch, 12. Juli 2017 13:50 An: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Betreff: [asterisk-users] Asterisk realtime - Error with index length in alembic script Hi! I just tried setting up Asterisk realtime database following the wiki a

[asterisk-users] Asterisk realtime - Error with index length in alembic script

Hi! I just tried setting up Asterisk realtime database following the wiki article https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a Debian 9 machine (which switched from MyQSL to MariaDB). One has to install mariadb-plugin-connect, python-mysqldb and alembic packages

Re: [asterisk-users] Asterisk Wiki down?

On Sun, Jul 9, 2017, at 05:21 PM, Joshua Colp wrote: > On Sun, Jul 9, 2017, at 04:58 PM, Jonathan H wrote: > > Definitely not just you - not working for me either, and tested from a > > few ping sites too > > I've created a ticket internally to get this looked into. This has now been resolved

Re: [asterisk-users] Asterisk Wiki down?

On Sun, Jul 9, 2017, at 04:58 PM, Jonathan H wrote: > Definitely not just you - not working for me either, and tested from a > few ping sites too I've created a ticket internally to get this looked into. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW -

Re: [asterisk-users] Asterisk Wiki down?

Definitely not just you - not working for me either, and tested from a few ping sites too On 9 July 2017 at 20:39, Dovid Bender wrote: > I am tryint to get to > https://wiki.asterisk.org/wiki/display/AST/Function_REGEX both via V6 and v4 > and it seems to be timing out. > >

[asterisk-users] Asterisk Wiki down?

I am tryint to get to https://wiki.asterisk.org/wiki/display/AST/Function_REGEX both via V6 and v4 and it seems to be timing out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

Re: [asterisk-users] asterisk ari dialer

i solved problem of missing incoming channel using local channel curl -X POST "http://my_pbx:8088/ari/channels?endpoint=Local%2F300%40originate=555666777=originate=1=7=30_key=apikey; (%2F is /, %40 is @) extensions.conf [originate] exten => 300,1,noop(originate) same => n,answer

Re: [asterisk-users] Asterisk sip_autodestruct messages - extensions locked

Based on the line number of that error in chan_sip.c, it looks like you're running Asterisk 1.8 or earlier. AFAIK, The issue you are seeing was fixed years ago, but not THAT many years ago! If I'm right, you should upgrade to fix that issue. Cheers, Steve On Fri, 30 Jun 2017 at 13:39 Stefan

[asterisk-users] Asterisk sip_autodestruct messages - extensions locked

Hi guys Does anybody have any opinion on what causes tens of thousands of these messages per hour to pop up in the CLI: [Jun 30 14:24:59] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct: Autodestruct on dialog '7e9597ae6ce95fef23374f4b380a9b70@192.168.0.1:5060' with owner

Re: [asterisk-users] asterisk ari dialer

my use case is for performace testing scenario asterisk14 - sip - tested asterisk - sip - clients (asterisk 14) i have working ari push configuration now i want create a call where call leg A will be some media file. call leg B will be channel to tested asterisk i dont have an incoming

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