similar to: Fail2ban for asterisk 16 PJSIP

Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for '192.168.32.116:5060' (callid: 0_1409534529 at 192.168.32.116) - No matching endpoint found or [Sep 9 12:56:14] NOTICE[10163]

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> 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 19:44:16.000000000 >> +0200 >> +++

09.09.2016 13:45, Joshua Colp ?????: > Dmitry Melekhov wrote: >> Hello! >> >> >> Upgraded 13.10 to 13.11.1 today and now I see messages in log: >> >> >> [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request >> 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for >>

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> 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

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 subsequent modules >> should be called by that running thread. The

I have servers setup in versions 11 and 13. Between two 11 servers, I had no issues sharing BLF, and assigning the hints on my Cisco 525G2 phones. I've upgraded to 13 on one of these servers, and now can't share BLF. I get something like... [2017-07-15 22:35:49] NOTICE[3483]: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from '"Travis Ryan" <sip:612

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: > > > > <snip> > > > >>> > >>> 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

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 interesting? I may add own > >> debug code to see why things

Right now it reply 401 Unauthorized with message in log "No matching endpoint ..." on Content 0 should reply 200 OK I guess <--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 ---> OPTIONS sip:10.30.100.27:5080 SIP/2.0 Via: SIP/2.0/UDP 10.30.100.41;branch=z9hG4bKf5eb.1ac76487000000000000000000000000.0 To: <sip:10.30.100.27:5080> From: <sip:vprx00 at

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried

Hello, I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway. I am able to make calls outbound through the gateway, but I am not able to make calls into the PBX from external PSTN. Specifically, an incoming call is _received_ by Asterisk, but it is not able to route the call internally owing to the following error: [Feb 18 21:08:47] NOTICE[4606]:

The Asterisk Development Team has announced the first release candidate of Asterisk 14.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues

Hi all, In my case I using realtime, here is how it looks in plant [10001] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600 at 192.168.1.1:5060 client_uri=sip:600 at 192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all

Hi, Info !!! not a question !!! the pjsip logger is different: [Sep 15 07:33:27] NOTICE[65267] res_pjsip/pjsip_distributor.c: Request from '"1001" <sip:1001 at 81.20.137.222>' failed for '85.25.197.23:5071' (callid: 1bfa1fcfee1e20dbe9bbbcac5d7bdffc) - No matching endpoint found and here the RegEx for fail2ban to catch this log: |NOTICE.* .*: Request from

I have two problems. The first is that when I dial my number from a phone on the Internet or any phone outside my LAN, Asterisk does not respond in any way, which means somehow my system is not picking up the fact that there's an incoming call to it. The second problem is that I thought I'd try an internal phone to see if I could get the hello-world stuff working at the least. I

Hello Everyone, How to configure PJSIP to reply 200 OK from upstream sip proxy on keepalive packet ? proxy ~> Keepalive OPTIONS ~> asterisk <~ 200 OK <~ volga629 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171203/8b9bc701/attachment.html>

Thanks George, for your mighty quick response. I made the changes (re: server_uri_pattern etc.) and still, no luck--it fails for the same error. BTW, there is nothing for transport (but this is the same config from my SIP/UDP + Twilio days, which worked): *CLI> pjsip show transport twilio-siptrunk Unable to find object twilio-siptrunk. *CLI> pjsip show identifies No objects found. I did

I have a trunk set up for the DID from my provider: [my_provider] type=registration outbound_auth=my_provider server_uri=sip:sip.example.com client_uri=sip:my_username at sip.example.com retry_interval=60 [my_provider] type=auth auth_type=userpass password=123456 username=my_username [my_provider] type=aor contact=sip:sip.example.com:5060 [my_provider] type=endpoint context=from-my_provider

Hi! On x86_64, changeset 18081, running/building on rhel5, trying to use the HVM direct boot causes the domain to reboot immediately and then the log says the domain is restarting too rapidly. No obvious hints as to where the problem is. If I build the in-tree ioemu code, things work. Any clues as to where to look or how to get some useful debugging output? Thanks, John Byrne