[asterisk-users] asterisk ari dialer

hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

Hello, Yes. When I today understood to set rtcp_mux=yes, at least Chrome (60.0 beta) worked (quickly tested) as expected. I'm sure that some day dtls_rekey can be set to the other value than 0 as well with Chrome. Best regards, Teijo 10.4.2017, 16.57, Matt Fredrickson kirjoitti: On

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

ellens *Sent:* Friday, April 21, 2017 10:18 AM *To:* Asterisk Users Mailing List - 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

[asterisk-users] Asterisk crashes when storing voicemail via odbc

Hi all, I'm working on migrating all of my servers to store voicemail in a mysql database via odbc. I've got a development server that I can reconfigure and test at will. When it's configured to store vm on the file system, it seems to be rock solid. However, when I ONLY change it to store

Re: [asterisk-users] Asterisk 13 attended transfer alcatel

On Tue, Jun 20, 2017, at 09:50 AM, Jason TOMLINSON wrote: > Hi, I've put the sip output here : https://pastebin.com/W7M4zxHA > Thanks It certainly shows it happening but nothing stands out as to why. Looks like a bug, go ahead and file one on the issue tracker[1]. [1]

Re: [asterisk-users] Asterisk 13 attended transfer alcatel

Objet : Re: [asterisk-users] Asterisk 13 attended transfer alcatel On Fri, Jun 9, 2017, at 04:59 AM, Jason TOMLINSON wrote: > Hello, > > Since upgrading from asterisk 11 to asterisk 13 (I have tested up to > the latest 13.16.0 release), we have a problem with attended transfers > to

Re: [asterisk-users] asterisk 13.16. - sigseg during negotiation

On 06/18/2017 at 12:11 PM, Joshua Colp wrote: > On Sun, Jun 18, 2017, at 06:00 AM, Michael Maier wrote: >> Hello! >> >> unchanged asterisk crashes during udptl / t.38 negotiation with telekom >> - they do not support t.38 / udptl. > > All Asterisk issues need to go through the issue tracker[1].

Re: [asterisk-users] asterisk 13.16. - sigseg during negotiation

On Sun, Jun 18, 2017, at 06:00 AM, Michael Maier wrote: > Hello! > > unchanged asterisk crashes during udptl / t.38 negotiation with telekom > - they do not support t.38 / udptl. All Asterisk issues need to go through the issue tracker[1]. In this case we'd also need to see the full SIP debug so

[asterisk-users] asterisk 13.16. - sigseg during negotiation

Hello! unchanged asterisk crashes during udptl / t.38 negotiation with telekom - they do not support t.38 / udptl. In detail: fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server Fax server sends t.38 reinvite via asterisk to easybell. Session Description Protocol

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/17/2017 at 02:18 PM, Michael Maier wrote: > On 06/16/2017 at 04:00 PM, Joshua Colp wrote: >> On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: >> >> >> >>> >>> t38modem and asterisk are using >>> >>> m=image 35622 udptl t38 >>>^ >>> >>> Provider uses >>> >>>

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Sat, Jun 17, 2017, at 09:18 AM, Michael Maier wrote: > > I can proof, that UDPTL vs. udptl is the problem. After "fixing" > asterisk and opal both using UDPTL, the negotiation works flawlessly. > See attached patches. > > Sending t38 faxes internally works fine. Externally via provider

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/16/2017 at 04:00 PM, Joshua Colp wrote: > On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: > > > >> >> t38modem and asterisk are using >> >> m=image 35622 udptl t38 >>^ >> >> Provider uses >> >> m=image 35622 UDPTL t38 >>^ >> >> Could this be

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: > > t38modem and asterisk are using > > m=image 35622 udptl t38 >^ > > Provider uses > > m=image 35622 UDPTL t38 >^ > > Could this be a problem? If I'm sending internal only, it's always >

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

Am 16.06.2017 um 11:12 schrieb Joshua Colp: On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: Has anybody any idea why asterisk drops the media stream in the 200 OK? The channel has been T38_ENABLED before! Or is it necessary to add more debug code? Who does the negotiating? Only

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: > Has anybody any idea why asterisk drops the media stream in the 200 OK? > The channel has been T38_ENABLED before! Or is it necessary to add more > debug code? Who does the negotiating? > Only asterisk or is pjsip doing some parts, too?

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/15/2017 at 08:15 AM Michael Maier wrote: > On 06/14/2017 at 10:17 PM, Joshua Colp wrote: >> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: >> >> >> >>> >>> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >>> Just one exception - and that's the package in

Re: [asterisk-users] Asterisk 1.6.2 how to debug T.38 udptl problems

On Thu, Jun 15, 2017 at 12:11:36PM +0200, Benoit Panizzon wrote: > Or does anyone have an idea over what the asterisk is stumbling? What if you set the maxdata in asterisk to a value lower than the other side? e.g. sip.conf: t38pt_udptl = yes,fec,maxdatagram=400 --

[asterisk-users] Asterisk 1.6.2 how to debug T.38 udptl problems

Hi all I know, a fairly old asterisk installation. Is there any way to debug T.38 handshaking issues? We have a C3 Voice Switch with link to the Asterisk server. I see this Dialogue: C3 => Asterisk => Invite g711 <= 200OK C3 detects Fax and send re-invite => Invite T.38 Version:0

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > I can now say, that asterisk / pjsip seams to work *mostly* as expected. > Just one exception - and that's the package in question, which can't be > seen in tcpdump. > > I extended the above patch by adding the info at the last

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.0 >>

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > I added this patch to see, if really all packages are are freed after > they have been processed: > > --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.0 > +0200 > +++ a/res/res_pjsip/pjsip_distributor.c 2017-06-13

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 06:51 PM Joshua Colp wrote: > On Sun, Jun 11, 2017, at 01:47 PM, Joshua Colp wrote: >> The distributor is in res/res_pjsip/pjsip_distributor.c, the distributor >> function being the entry point. That function returning PJ_TRUE >> indicates to PJSIP that it has been handled and no

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 11:35 PM Daniel Tryba wrote: > On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote: >> Let's go into details: >> I'm starting at the point where asterisk / fax client receives the T.38 >> reininvite from the server from the provider 195.185.37.60:5060 for the >> fax

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 04:34 PM Michael Maier wrote: > On 06/11/2017 at 01:29 PM Joshua Colp wrote: >> On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote: >> >> >> >>> I did some further investigations and fixed a local problem. Now, >>> asterisk is able to successfully activate T.38 -

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote: > Let's go into details: > I'm starting at the point where asterisk / fax client receives the T.38 > reininvite from the server from the provider 195.185.37.60:5060 for the > fax client (extension 91): I'm running Asterisk 11 on my

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Sun, Jun 11, 2017, at 01:47 PM, Joshua Colp wrote: > On Sun, Jun 11, 2017, at 01:31 PM, Michael Maier wrote: > > On 06/11/2017 at 04:39 PM Joshua Colp wrote: > > > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: > > > > > > > > > > > >>> > > >>> PJSIP uses a dispatch model. The

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Sun, Jun 11, 2017, at 01:31 PM, Michael Maier wrote: > On 06/11/2017 at 04:39 PM Joshua Colp wrote: > > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: > > > > > > > >>> > >>> PJSIP uses a dispatch model. The request is queued up, acted on, and > >>> then that's it. The act of acting

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 04:39 PM Joshua Colp wrote: > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: > > > >>> >>> PJSIP uses a dispatch model. The request is queued up, acted on, and >>> then that's it. The act of acting on it removes it from the queue. >> >> That's the *expected* behavior

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: > > > > PJSIP uses a dispatch model. The request is queued up, acted on, and > > then that's it. The act of acting on it removes it from the queue. > > That's the *expected* behavior ... . I rechecked again and again. All > existing

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/11/2017 at 01:29 PM Joshua Colp wrote: > On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote: > > > >> I did some further investigations and fixed a local problem. Now, >> asterisk is able to successfully activate T.38 - unfortunately just for >> very shot time because of a phantom

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote: > I did some further investigations and fixed a local problem. Now, > asterisk is able to successfully activate T.38 - unfortunately just for > very shot time because of a phantom package it receives! What was the local problem? >

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 09:32 PM Joshua Colp wrote: > On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: >> On 06/05/2017 at 06:29 PM, Joshua Colp wrote: >>> On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: Do you have any idea where to start to look at? Adding additional output

Re: [asterisk-users] Asterisk 13 attended transfer alcatel

How are both machines connected to each other ? Through a SIP trunk ? A TDM one ? 2017-06-09 9:59 GMT+02:00 Jason TOMLINSON : > Hello, > > > > Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the > latest 13.16.0 release), we have a problem with

Re: [asterisk-users] Asterisk 13 attended transfer alcatel

On Fri, Jun 9, 2017, at 04:59 AM, Jason TOMLINSON wrote: > Hello, > > Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the > latest 13.16.0 release), we have a problem with attended transfers to an > alcatel pbx in which the call being transferred still has music on hold >

[asterisk-users] Asterisk 13 attended transfer alcatel

Hello, Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the latest 13.16.0 release), we have a problem with attended transfers to an alcatel pbx in which the call being transferred still has music on hold even after the transfer has completed. Is this a known issue? Any new

Re: [asterisk-users] asterisk server - no sound

And it is worst (and that could be the reason of the error). 127.0.0.1 is configured in 2 interfaces (lo and venet0), just with different network masks. Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br

Re: [asterisk-users] asterisk server - no sound

Well, based on the config that you sent, your server just have the localhost IP (127.0.0.1) Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at

Re: [asterisk-users] asterisk server - no sound

I am using version: 14.5.0 No, Im not using Dundi. Can you a bit more informative when you say I "need to configure the IPs in your server"? thanks! a On 06/06/2017 07:47 PM, Marcelo Terres wrote: > I think you need to configure the IPs in your server. You just have > localhost... > Marcelo H.

Re: [asterisk-users] asterisk server - no sound

I think you need to configure the IPs in your server. You just have localhost... Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 16:27, andre

Re: [asterisk-users] asterisk server - no sound

Looks like it comes com pbx_dundi.c. Why are you using dundi? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 18:43, Marcelo Terres

Re: [asterisk-users] asterisk server - no sound

Which Asterisk version are you using? Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June 2017 at 18:32, andre castro wrote: >

Re: [asterisk-users] asterisk server - no sound

Any ideas. After configuring port forwarding on the server (machine making nat) to forward connections originated from external clients to the machine running asterisk, as explained in https://www.voip-info.org/wiki/view/port+forwarding My peers were unable to register. And When running

Re: [asterisk-users] asterisk server - no sound

Try to use the echo app. If you can listen your echo, so it is something in the network. Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 6 June

Re: [asterisk-users] asterisk server - no sound

Thanks Anthony. I did it on the server, according to https://www.voip-info.org/wiki/view/port+forwarding However after doing it, when running Asterisk I get the following message sudo asterisk -vvr No ethernet interface found for seeding global EID. You will have to set it manually. Unable

Re: [asterisk-users] asterisk server - no sound

On Tuesday 06 June 2017 16:57:07 andre castro wrote: > On 06/06/2017 04:36 PM, Antony Stone wrote: > > > > Tell us about your networking arrangement - are both phones and the > > Asterisk machine on the same network? > > Nop. They are in 2 different networks. The phones in one and the >

Re: [asterisk-users] asterisk server - no sound

Thank you Daniel for pointing out the errors and debug option. Both fixed and on. It made no difference. There are no errors printed and still no sound on ppers Now to Antony questions: On 06/06/2017 04:36 PM, Antony Stone wrote: > On Tuesday 06 June 2017 15:18:32 andre castro wrote: > >> I

Re: [asterisk-users] asterisk server - no sound

Le 06/06/2017 à 16:25, Daniel Tryba a écrit : On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote: extensions.conf: [home] exten = 102,1,Answer() same = n,Wait(1) If this is copy and paste, then your dialplan is broken (= should be =>) Well, no. = or => are the same. -- Daniel

Re: [asterisk-users] asterisk server - no sound

On Tuesday 06 June 2017 15:18:32 andre castro wrote: > I just installed asterisk in a debian server. > All seems to be running fine, but the audio sent by the server. > But I hear nothing at the peer's end. > > When one peer calls another, sound comes through just fine. Tell us about your

Re: [asterisk-users] asterisk server - no sound

On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote: > extensions.conf: > [home] > exten = 102,1,Answer() > same = n,Wait(1) If this is copy and paste, then your dialplan is broken (= should be =>) But to debug, enable logging (core set verbose 5), when needed debugging (core set debug

[asterisk-users] asterisk server - no sound

hello folks, this might be a simple question... I just installed asterisk in a debian server. All seems to be running fine, but the audio sent by the server. If I have one of my registered peers call and extension (102) that plays back audio (extension.conf and sip.conf coffee-pasted below),

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: >> >> Do you have any idea where to start to look at? Adding additional output >> in the source code? Which functions could be interesting? I may add own >> debug code to see why things are

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > > Do you have any idea where to start to look at? Adding additional output > in the source code? Which functions could be interesting? I may add own > debug code to see why things are happening as they happen here. The logic for T.38

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 06:10 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote: >> On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: >>> On 06/05/2017 at 11:30 AM, Joshua Colp wrote: On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: > On 06/04/2017 at 01:41

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: > > On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > > > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: > > >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > > >>> Just

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 05:00 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: >> On 06/05/2017 at 11:30 AM, Joshua Colp wrote: >>> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > Just a guess

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: > On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: > >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > >>> Just a guess (without knowing about your network), but are the two

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another?

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: > On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > > Just a guess (without knowing about your network), but are the two ends > > points on public networks and visible to one another? If not the reinvite > > may be passing an

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > Just a guess (without knowing about your network), but are the two ends > points on public networks and visible to one another? If not the reinvite > may be passing an internal (nat'ed) address to the other and the connection > will

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier Sent: Sunday, June 4, 2017 3:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjs

[asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

Hello! I'm still trying to get a working t.38 configuration w/ pjsip. I'm now able to send t.38 faxes to my own extension: hylafax -> t38modem -> extension -> extension -> t38modem -> hylafax. The fax is sent by t38modem. The receiving part of t38modem accepts the call, sends ReInvite for

[asterisk-users] Asterisk 14.5.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 14.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.5.0 resolves several issues reported by the community and would have not been

[asterisk-users] Asterisk 13.16.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 13.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.16.0 resolves several issues reported by the community and would have not been

Re: [asterisk-users] Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?

Will do, thanks for the confirmation! Chris. On 2017-05-30 18:08, Joshua Colp wrote: > On Tue, May 30, 2017, at 01:55 PM, Christopher van de Sande wrote: > >> Cool, I've attached 2 sip trace examples, 14.3.1 and 14.4.1. Both are >> using identical configurations. >> >> The biggest

Re: [asterisk-users] Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?

On Tue, May 30, 2017, at 01:55 PM, Christopher van de Sande wrote: > Cool, I've attached 2 sip trace examples, 14.3.1 and 14.4.1. Both are > using identical configurations. > > The biggest difference I can see happens right after the UPDATE message > from MicroSIP. > > The SDP headers on

Re: [asterisk-users] Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?

On Tue, May 30, 2017, at 01:07 PM, Christopher van de Sande wrote: > Hi first post, so hope I'm not violating protocol. > > Been using Asterisk as home phone and hobby use for nearly 10 years. I > love this project. > > Anyway, would someone mind verifying my pjsip.conf ? This seems to work >

[asterisk-users] Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?

Hi first post, so hope I'm not violating protocol. Been using Asterisk as home phone and hobby use for nearly 10 years. I love this project. Anyway, would someone mind verifying my pjsip.conf ? This seems to work well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade to

Re: [asterisk-users] Asterisk 14 audio quality with remote files

On Tue, May 16, 2017 at 3:00 PM, Tiago Ferreira wrote: > Anyone? > I tried converting the file to g722 with ffmpeg and got the same result. > > regards > Tiago > On 12-05-2017 12:10, Tiago Ferreira wrote: > > Hello everyone, > > I am using the Asterisk REST API in

[asterisk-users] Asterisk 13.13-cert4, 13.15.1, 14.4.1 Now Available (Security Release)

The Asterisk Development Team has announced security releases for Certified Asterisk 13.13 and Asterisk 13 and 14. The available security releases are released as versions 13.13-cert4, 13.15.1, and 14.4.1. These releases are available for immediate download at

Re: [asterisk-users] Asterisk 13 queue and DND phones

...@lists.digium.com] On Behalf Of Héctor Royo Sent: May 17, 2017 08:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13 queue and DND phones Hi. I will try to give you an idea: You can remap de 'Do not disturb' key to do some actions. (Best examples

Re: [asterisk-users] Asterisk 13 queue and DND phones

Hi. I will try to give you an idea: You can remap de 'Do not disturb' key to do some actions. (Best examples I've found so far: http://community.polycom.com/t5/VoIP/FAQ-Using-Enhanced-Feature-Keys-EFK-macros-to-change-key/td-p/5705 ) I once tried to remap de DND on a Polycom IP 550 to something

[asterisk-users] Asterisk 13 queue and DND phones

Hi, I've noticed that when I set a phone on DND (phone-side DND, meaning it rejects calls with a busy status, SIP 486 response code I believe) the queue keeps on trying the phone over and over again. This creates issues in terms of CDR entries - in a scenario where there is only one phone

Re: [asterisk-users] Asterisk 14 audio quality with remote files

Anyone? I tried converting the file to g722 with ffmpeg and got the same result. regards Tiago On 12-05-2017 12:10, Tiago Ferreira wrote: Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am

[asterisk-users] Asterisk 14 audio quality with remote files

Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above

Re: [asterisk-users] asterisk 13.15.0 stopping/crashing

On Tue, May 9, 2017, at 11:06 AM, marek cervenka wrote: > i can upgrade asterisk to DONT_OPTIMIZE version at night > > before that, do you see something strange? > > is it known issue? The only issue that looks like it could be related is ASTERISK-26969[1]. Once you have an unoptimized

Re: [asterisk-users] asterisk 13.15.0 stopping/crashing

i can upgrade asterisk to DONT_OPTIMIZE version at night before that, do you see something strange? is it known issue? [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/sbin/asterisk -f -C

Re: [asterisk-users] asterisk 13.15.0 stopping/crashing

when run from console without systemd i found its segfaulting turned core dump on because it was off Dne 09/05/2017 v 13:52 marek cervenka napsal(a): hi, i have strange problem with asterisk 13.15.0+pjsip bundled/centos 7/systemd start script we are using chan_pjsip only for webrtc

[asterisk-users] asterisk 13.15.0 stopping/crashing

hi, i have strange problem with asterisk 13.15.0+pjsip bundled/centos 7/systemd start script we are using chan_pjsip only for webrtc endpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days ago today i have problems with stopping/crashing asterisk

Re: [asterisk-users] Asterisk 13 on CentOS 6

http://linoxide.com/tools/install-setup-asterisk-13-pbx-centos-7/ 2017-04-27 2:15 GMT+08:00 Jerry Geis : > > yum install jansson* > > This works for CentOS 7 but not CentOS 6. > > Thanks, > > Jerry > > > > -- >

Re: [asterisk-users] Asterisk 13 on CentOS 6

> yum install jansson* This works for CentOS 7 but not CentOS 6. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] Asterisk 13 on CentOS 6

yum install jansson* Jerry Geis 于2017年4月26日 周三下午8:32写道： > >It can't be disabled. jansson is a required dependency for Asterisk 13 > >as JSON is used internally for things. > > > Ok thanks - that is a little confusing since there are entries in the > configure script that

Re: [asterisk-users] Asterisk 13 on CentOS 6

>It can't be disabled. jansson is a required dependency for Asterisk 13 >as JSON is used internally for things. Ok thanks - that is a little confusing since there are entries in the configure script that lead one to think it can be a configure time switch. I'll go the other route and install

Re: [asterisk-users] Asterisk 13 on CentOS 6

On Wed, Apr 26, 2017, at 09:24 AM, Jerry Geis wrote: > Trying to install asterisk 13 on CentOS 6. > > The ./configure tells me: > configure: error: *** JSON support not found (this typically means the > libjansson development package is missing) > > I don't really need JSON so I thought I would

[asterisk-users] Asterisk 13 on CentOS 6

Trying to install asterisk 13 on CentOS 6. The ./configure tells me: configure: error: *** JSON support not found (this typically means the libjansson development package is missing) I don't really need JSON so I thought I would just disable it. ./configure --with-jansson=no does not work

[asterisk-users] Asterisk 11 EOL 6 Month Notice

Greetings, This is your friendly 6 month warning that Asterisk 11 will be reaching an official end of life state on October 25, 2017. As many of you know, for the past 6 months Asterisk 11 has been in security fix only mode. This means it currently does not receive bug fixes, but it does

Re: [asterisk-users] asterisk name in mysql

I will try to reinstall everything according to your instructions and i will come back. I might need a couple of weeks due to a business trip though Στις 24 Απρ 2017 6:54 μ.μ., ο χρήστης "John Kiniston" < johnkinis...@gmail.com> έγραψε: > Well, My suggestion was to use FUNC_ODBC but instead you

Re: [asterisk-users] Asterisk download stats

-users Subject: Re: [asterisk-users] Asterisk download stats I have tried to find these in the past, best I came up with was using Shodan.io search Looking for Asterisk I get: TOTAL RESULTS 42,036 TOP COUNTRIES United States 12,914 Russian Federation 3,173

Re: [asterisk-users] Asterisk download stats

gt; > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Subject: [asterisk-users] Asterisk download stats > Message-ID: > <CAM3TTh1d7Zn71O1by+25tGj7r9saqU_K1XOViTL+k7F1d5qv-g@mail. > gmail.com&

Re: [asterisk-users] asterisk name in mysql

Well, My suggestion was to use FUNC_ODBC but instead you went with APP_MYSQL which has been depricated. Did you compile APP_MYSQL? It's not enabled by default. On Sat, Apr 22, 2017 at 1:25 PM, Atux Atux wrote: > Thanks a lot for the reply. > I did follow that already, but i

Re: [asterisk-users] asterisk name in mysql

In article , Atux Atux wrote: > > Thanks a lot for the reply. > I did follow that already, but i do have a problem. Here is my > extensions.conf part for that particular number > exten =>

Re: [asterisk-users] asterisk-users Digest, Vol 153, Issue 28

risk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Subject: Re: [asterisk-users] asterisk name in mysql > Message-ID: > <CACoLBwUid-NOos3Qp9x3CP+fFmrKzk+QJuYXTQnnOw4BS2thaw@ > mail.gmail.com> > Content-Type: text/pla

[asterisk-users] Asterisk download stats

Hi, Are there any stats on where Asterisk is downloaded from based on the country? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] asterisk name in mysql

On Saturday 22 April 2017 at 22:25:52, Atux Atux wrote: > Thanks a lot for the reply. > I did follow that already, but i do have a problem. Here is my > extensions.conf part for that particular number > exten => 6912345678,1,Answer() > exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root

Re: [asterisk-users] asterisk name in mysql

Thanks a lot for the reply. I did follow that already, but i do have a problem. Here is my extensions.conf part for that particular number exten => 6912345678,1,Answer() exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypasswd asterisk) exten => 6912345678,n,MYSQL(Query resultid

Re: [asterisk-users] asterisk name in mysql

You can use func_odbc to do this. https://wiki.asterisk.org/wiki/display/AST/Getting+Asterisk+Connected+to+MySQL+via+ODBC2 There is a good chapter in the Asterisk book about using ODBC for hotdesking that may help you understand ODBC as well.

[asterisk-users] asterisk name in mysql

hi. currently i am running the phonebook in astdb with *database put cidname 0123456789 "name_surname"* and i retrive it with *exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})* Now, my system has mysql and i got all my contacts in there in a database is called *asterisk

Re: [asterisk-users] asterisk as non root

the output of ls -l is root@pbx: ~ $ ls -l /var/run/asterisk/asterisk.ctl srwxr-xr-x 1 asterisk asterisk 0 Apr 20 19:47 /var/run/asterisk/asterisk.ctl root@pbx: ~ $ On Thu, Apr 20, 2017 at 7:46 PM, Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Thursday 20 April 2017 at

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

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Friday, April 21, 2017 10:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] As

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