Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

On Tue, Dec 22, 2015 at 09:30:52AM +, Luca Bertoncello wrote: > Zitat von Sebastian Kemper <sebastian...@gmx.net>: > > Hi Sebastian > > > I tried with > > sip set debug 42 > sip set verbose 42 > > The result was in my first E-Mail... Hi

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

On Tue, Dec 22, 2015 at 09:42:04AM +, Luca Bertoncello wrote: > Is it not this: > > http://lists.digium.com/pipermail/asterisk-users/2015-December/288078.html > > ? > > sip set debug 42 should be a little trick to enable more debugging... > So I got in this list some months ago...

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

On Tue, Dec 22, 2015 at 07:19:47AM +0100, Luca Bertoncello wrote: > "Brian ::" schrieb: > > > sip trace? > > Could you please explain? I'm not a VoIP-expert... > > Thanks > Luca Bertoncello > (lucab...@lucabert.de) Hi Luca, Brian suggests to check the SIP traces. You can

[asterisk-users] Re-Invite to Native Bridge

Hello all, My Asterisk is between my ITSP and a SIP phone. I cannot do direct media between the provider and the SIP phone, but I would like Asterisk to locally RTP bridge the two channels using native_rtp. Example: > Bridge cfb56606-6b40-4da7-a6fa-6499e183cdbb: switching from simple_bridge >

Re: [asterisk-users] Update peer IP address

Am 16. September 2015 18:48:16 MESZ, schrieb Daniel Heckl : >Sebastian, > >If I have understood you correctly, the SIP communication is now via >NAT instead forwarded ports. For safety, it is much better. > >I think it is not because of a UDP timeout, but rather because of

Re: [asterisk-users] Update peer IP address

On Tue, Apr 14, 2015 at 08:26:07AM +0200, Sebastian Kemper wrote: > On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote: > > I do not want set allowguest=yes. The problem is, there is no official > > list with ip addresses of Telekom Germany. But I think all ip > >

Re: [asterisk-users] Asterisk 11 and pulseaudio setup as local user

Am 3. Juli 2015 13:17:34 MESZ, schrieb Jerry Geis ge...@pagestation.com: alsa_card_init^[[0m: snd_pcm_open failed: Connection refused soundcard_init^[[0m: Problem opening alsa capture device These are the errors I get. I changed the following: chown -R myuser:myuser /var/log/asterisk chown -R

Re: [asterisk-users] Logging in local time

Am 5. Juni 2015 16:29:21 MESZ, schrieb Luca Bertoncello lucab...@lucabert.de: Hi again! I just noticed, that my Asterisk (running on an OpenWRT-Switch) writes the logs using GMT... On the Switch the time is right configured and a date says me the current LOCAL time. I didn't found in

Re: [asterisk-users] Signaling incoming call

Am 31. Mai 2015 10:58:56 MESZ, schrieb Luca Bertoncello lucab...@lucabert.de: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Guenther Boelter gboel...@gmail.com schrieb: -BEGIN PGP SIGNED MESSAGE- Hash: SHA256 On 05/31/2015 02:31 PM, Luca Bertoncello wrote: Hi list! Now all

Re: [asterisk-users] Debugging dialplan

On Fri, May 29, 2015 at 07:24:45AM +0200, Luca Bertoncello wrote: Since I think, I have a problem in my dialplan, how can I debug it? It would be very useful a command in Asterisk CLI to ask Asterisk what it would do if the number X call the number Y. Something like exim -bt, if someone here

Re: [asterisk-users] Debugging dialplan

Hi Luca, It's not the A number you have to look at if you want to know how a call comes into the dialplan and then goes out again. You want do know in which context a call arrives. That depends on things like the IP address (peer), username/password (friend) or other things. I suggest to read

Re: [asterisk-users] Update peer IP address

On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote: I do not want set allowguest=yes. The problem is, there is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. Hello Daniel, Judging by the lists I found I think

Re: [asterisk-users] Update peer IP address

should be prepared for changes. You must enable the dnsmgr. If DNS resolves a new ip, the peer is updated. Hello Daniel, Thanks for the tip. I've enabled the DNS manager. Let's see how it goes. Kind regards, Sebastian Am 14.04.2015 um 08:26 schrieb Sebastian Kemper sebastian...@gmx.net

Re: [asterisk-users] Update peer IP address

On Tue, Mar 31, 2015 at 12:36:34PM +0200, Daniel Heckl wrote: Hello Sebastian, I had already seen this list of the hosts, but it is not active. All servers with which my Asterisk has been communicated are not listed. A port scan, to eventually update the list, found hundreds of servers

Re: [asterisk-users] Update peer IP address

On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote: On 4/1/15 10:48 AM, Daniel Heckl wrote: John, thank you four your answer. I think you have misunderstood the problem. It’s about a ip address change of the sip trunk, not of my asterisk server. You would probably benefit by

Re: [asterisk-users] Update peer IP address

On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: Hello I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain it understandable. Hello Daniel, I'll find myself in the same

[asterisk-users] Asterisk on OpenWrt (first time user)

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently