[asterisk-users] Upgraded server crashes on voicemail storage

Hi all, I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've discovered that my server crashes as soon as I leave a voicemail message. I'm using odbc voicemail storage as well as mysql dynamic configuration. I'm using unixODBC 2.3.2-r2 with myodbc 5.2.7-r1 I suspect that

[asterisk-users] RTP / NAT question with IPv6/IPv4 problem

Hello, All my asterisk systems use only IPv4 currently. I have one phone which is on T-Mobile network, and this network is only IPv6 now. The phone can register fine, because T-Mobile does NAT64 and it connects fine to my IPv4 asterisk server. But in the SDP for a call setup, this

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

Re: [asterisk-users] OT: DMARC enabled domains on this list

On Tue, Jun 06, 2017 at 08:23:33AM -0400, James B. Byrne wrote: > > The reports are there to tell you something isn't right (like on this > > mailing list). Disabling them is only hiding the problem, people might > > be replying with the correct answer to a problem, but the OP might > > never gets

[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] IAX port 4569

Run this command: tcpdump -pni any -s0 -vvv port 4569 -w /tmp/iax.pcap Try to authenticate and do calls, after that stop the command and send to us the file /tmp/iax.pcap that i can see and help you. If you want add-me at whatsapp Att, Hélvio Junior dCAA - Digium Certified Asterisk

Re: [asterisk-users] OT: DMARC enabled domains on this list

On Mon, June 5, 2017 15:30, Daniel Tryba wrote: > > The reports are there to tell you something isn't right (like on this > mailing list). Disabling them is only hiding the problem, people might > be replying with the correct answer to a problem, but the OP might > never gets that message. >

Re: [asterisk-users] Extensions of sip trunk

On Tue, Jun 06, 2017 at 12:40:21AM +0200, Hans-Peter Jansen wrote: > > Yes, something like if they can't fix the R-URI: > > exten => X_.,n,Set(TO=${CUT(SIP_HEADER(To),@,1)}) > > exten => X_.,n,Set(TO=${CUT(TO,:,2)}) > > exten => X_.,n,Goto(somewhereelsetopreventloops${TO},1) > > Sorry for the

Re: [asterisk-users] IAX port 4569

In article <87af2f00-9973-f338-1cbc-9ce0a5bf7...@sys-concept.com>, wrote: > Doesn't matter how much I increase the verbose output > asterisk -vvr > asterisk will not even print a single line. Check the settings in /etc/asterisk/logger.conf, for example: ...