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Zitat von Sebastian Kemper <sebastian_ml at gmx.net>: Hi Sebastian > Brian suggests to check the SIP traces. You can either enable SIP > debugging in Asterisk like so: > > sip set debug on > > Or you could run tcpdump and capture the SIP traffic. > > The first option is probably the easiest. I tried with...

...port and allow only your local phones and the sip trunk ip range. I think srvlookup must be set to yes to place outbound calls if there is an ip address change. I think with the restriction of the firewall that should be a secure solution. > Am 01.04.2015 um 19:23 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: > > 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 th...

...your local phones and the sip trunk ip range. I think srvlookup must be set to yes to place outbound calls if there is an ip address change. > > I think with the restriction of the firewall that should be a secure solution. > > > Am 01.04.2015 um 19:23 schrieb Sebastian Kemper <sebastian_ml at gmx.net <mailto:sebastian_ml at gmx.net>>: > > > > 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 misunders...

Zitat von Sebastian Kemper <sebastian_ml at gmx.net>: > I don't remember seeing anything looking like a SIP trace in your first > mail. Try > > sip set debug on > > instead of > > sip set debug 42 > > I don't think there's a sip debugging level like 42 in Asterisk. You can > either switch i...

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 > > addresses comes from the ip range 217.0.0.0/13. > > Hello Daniel, > > Judging by the lists

...er solution. If I change insecure to insecure=port,invite - could that be a solution? Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? Has there anyone experience with dynamic ip addresses of Asterisk? Daniel > Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: > > 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 u...

...es and the sip trunk ip range. I think srvlookup must be set to yes to place outbound calls if there is an ip address change. >> >> I think with the restriction of the firewall that should be a secure solution. >> >> > Am 01.04.2015 um 19:23 schrieb Sebastian Kemper <sebastian_ml at gmx.net <mailto:sebastian_ml at gmx.net>>: >> > >> > 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...

...runk ip range. I think srvlookup must be set to yes to place outbound calls if there is an ip address change. >>> >>> I think with the restriction of the firewall that should be a secure solution. >>> >>> > Am 01.04.2015 um 19:23 schrieb Sebastian Kemper <sebastian_ml at gmx.net <mailto:sebastian_ml at gmx.net>>: >>> > >>> > On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote: >>> >> On 4/1/15 10:48 AM, Daniel Heckl wrote: >>> >>> John, >>> >>> >>> >>> thank...

...nsecure to insecure=port,invite - could that be a solution? > > Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no > problem)? Has there anyone experience with dynamic ip addresses of Asterisk? > > Daniel > > Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: > > 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. &g...

Hi list! 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 know the SMTP-daemon Exim... Is there such an option in Asterisk? Thanks Luca Bertoncello (lucabert at lucabert.de)

"Brian ::" <bc at iptel.co> schrieb: > sip trace? Could you please explain? I'm not a VoIP-expert... Thanks Luca Bertoncello (lucabert at lucabert.de)

...ared 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_ml at gmx.net>: > > > >> 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...

...cal phones and the sip trunk ip range. I think srvlookup must > be set to yes to place outbound calls if there is an ip address change. > > I think with the restriction of the firewall that should be a secure > solution. > > > Am 01.04.2015 um 19:23 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: > > > > 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?...

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 enabling the DNS Manager to allow for dynamic IP changes: # cat dnsmgr.conf [general] enable=yes ; enable creation of managed DNS

...ecure to insecure=port,invite - could that be a solution? > > Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? Has there anyone experience with dynamic ip addresses of Asterisk? > > Daniel > >> Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: >> >> 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. >>&g...

...runk ip range. I think srvlookup must >> be set to yes to place outbound calls if there is an ip address change. >> >> I think with the restriction of the firewall that should be a secure >> solution. >> >> > Am 01.04.2015 um 19:23 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: >> > >> > 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 >&g...

...solution? >>> >>> Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? Has there anyone experience with dynamic ip addresses of Asterisk? >>> >>> Daniel >>> >>>> Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: >>>> >>>> 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 probl...

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. For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de <http://tel.t-online.de/>), the message is answered with OK and the peer is registered. Usually INVITES comes now from this ip address. All

...rvlookup must >>> be set to yes to place outbound calls if there is an ip address change. >>> >>> I think with the restriction of the firewall that should be a secure >>> solution. >>> >>> > Am 01.04.2015 um 19:23 schrieb Sebastian Kemper <sebastian_ml at gmx.net >>> >: >>> > >>> > 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 t...

...tion. If I change insecure to insecure=port,invite - could that be a solution? Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? Has there anyone experience with dynamic ip addresses of Asterisk? Daniel Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: 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 mys...