similar to: Second invite after 100ms (with default t1min=100) --> canceled call problem!

On 2016-08-09 10:06, Faheem Muhammad wrote: > trip time and Call Setup time of SIP Requests. > In case of GSM Network with high delay you need to set the T1 timer a > higher value like 1000ms (500 ms default). Similarly you can reduce the > Call setup time by configuring 'T2' upto you choice as per you telephony > network. Configure t1min, timert1 and timerb according to

Hi list! I already had the problem last year, then it would be solved (surely from some technician by Deutsche Telekom on their servers), and now I have the problem again (but I didn't changed my Asterisk configuration). The problem: after 15 minutes will the call dropped, but only if the call is to another nation! If I just call another phone in Germany, I can speak longer than 15

I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one : Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> ) exten =>

Hello Joshua, thank you very much for your extremely quick answer! I really appreciate your work and your friendly and your patient support! On 06/07/2017 at 10:33 PM, Joshua Colp wrote: > On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote: >> Hello! >> >> I've got a problem to select the correct trunk if there is one provider >> and different numbers with

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hi, We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Hello! I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2. As customer of German Telekom, I have three numbers and therefore three trunks - each number is bound to one trunk. After migration, some callers complained about missing ringback tone: they didn't hear any ring tone and where surprised that they suddenly got me anyway :-). The connection afterwards was as expected. Deeper

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 Version (v): 0 Owner/Creator, Session Id (o): - 2447581897 4 IN IP4 46.17.15.23

Am 22.06.2020 um 21:30 schrieb Michael Maier: > Did you check to prevent transcoding? could you explain what do you mean and how to check it? >> On the Gateway (Banana PI), where the Asterisk server also runs, the >> load is about 0.50 during calls and it has a Gbps LAN. > > What's running on this device on parallel? What about other network > traffic - not

Hello! I'm facing ReInvites as caller from UAS despite configured session-timers=refuse (which can be seen in the SIP trace) always after 900s. (The behavior is the same if session-timers is set to accept). This just happens with one provider (German Telekom to callee at kabelbw). - The incoming ReInvite is answered immediately by asterisk (Status 100 / Status 200 - 0.02s). Media stream

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hello List, I tried to change the following parameters in sip.conf file, but looks like it cannot be changed, Defaut values: ;t1min=100 ;timert1=500 ;timerb=32000 I have changed to: ;t1min=100 timert1=100 timerb=6400 Sometime I can see too many retransmission of BYE to some of the UAs if UA is unreachable. Is there a way that I can reduce the number of retransmission of BYE message?

Hi list! So, now I have a business contract and a technician was here to check the DSL... Nothing found, except that for 50Mbps I need now vectoring. Really nice... A couple of years ago I could get 50Mbps without vectoring. Of course, Deutsche Telekom said nothing about this change... Well, I got it working, and now I have 48Mbps down and 10Mbps up. I _REALLY CAN'T_ believe, that this is

Am 22.06.20 um 16:48 schrieb Luca Bertoncello: > Hi list! > > So, now I have a business contract and a technician was here to check > the DSL... > Nothing found, except that for 50Mbps I need now vectoring. Really > nice... A couple of years ago I could get 50Mbps without vectoring. > Of course, Deutsche Telekom said nothing about this change... > > Well, I got it

Endpoint sends an INVITE Asterisk send an INVITE to the Carrier Carrier is down, does not even sends ACK PJSIP sends several INVITES End point sends <--- Received SIP request (397 bytes) from UDP XXXX::50187 ---> CANCEL sip:xxxxxxx at xxxxxxx SIP/2.0 Via: SIP/2.0/UDP xxxxxxx :50187;branch=z9hG4bK-524287-1---fbad0437cf02653d;rport Max-Forwards: 70 To: <sip:xxxxx at xxxxx> From:

Am 24.06.2020 05:05, schrieb Michael Maier: Hi > Your basic architecture looks good to me - now you have to start the Nice to hear it... > analysis of the problem with pcapsipdump and wireshark as I wrote > before to get an idea what actually happens at > all. You most probably won't come any further without doing any > analyzing. You have to learn it. It will take some, or

Am 14.06.2020 um 17:33 schrieb Luca Bertoncello: Hi So, I got a phone (Elmeg IP290) from a collegue and tested it... > What I'll do tomorrow with a test phone is: > > 1) connecting it to my Asterisk and try to make a call > 2) connecting it directly to the servers of Deutsche Telekom (using my > network) and try to make a call Absolutly *no changes* on the behaviour

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

Am 14.06.2020 um 17:05 schrieb Antony Stone: Hi Antony, > You mean that the Thomson phone is registering to Deutsche Telekom? > > I thought it was registering to your Asterisk server. Sorry, I didn't read correctly your test 2b... Normally my Thomson phone is registering to my Asterisk server. I tried to register the Thomson phone directly to Telekom's server, to check if the