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RoadRunnR

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  1. Works with the App, but I also don't have the Google Login Button in there.
  2. I didn't change any of the front end templates. 65.0.7 still shows the same problem. The "Signin with Google" button is shown very briefly and then turns to "Signed In..." and redirects to the user page. Logout throws me back into that loop. I did the video in incognito because that is the only way not to be logged in automatically. Could that be related to the fact that chrome itself is also signed in to the Google account?
  3. http://portal.vodia.com/downloads/pbx/version-65.0.xml currently (2020-05-20T08:50+00:00) points to the unannounced 65.0.6. The binary for Debian64 referenced in there is broken and prevents the PBX from starting.
  4. Sure, I'll send you a link to a video in private.
  5. The "normal" Web Client / Extension login at https://<pbx>/usr_portal.htm. PBX is 65.0.5 (Debian64)
  6. Hi, I've successfully configure the Google login. Google seems to have changed the configuration setting, making the instructions in the doc incomplete. After creating the OAuth project as described in the docs, you have to go into the console and add the PBX URI to the "Authorized JavaScript-Sources setting. But the real problem is that the logout no longer works. This means that you can not switch accounts (e.g. go from an extension to the admin account). The problem is most likely that the logout button is not performing a Google OAuth logout. Is there a way to modify the user portal page to make logout perform a Google logout as well? Thanks, Andreas
  7. Das Einrichtungstemplate in der 65.0.5 setzt leider immer noch einige Werte falsch. DT benutzt Rufnummern im +49... format. Alle settings die sich darauf beziehen sollten entsprechend eingestellt sein (mindestens Rewrite global numbers im SIP Trunk) Encrypt media sollte auf Default stehen. Der Assistent stellt das auf "off". Wenn SIP per TLS benutzt wird (Default), *muss* auch RTP+DTLS benutzt werden. Ein mix von secure/insecure wird nicht unterstützt (steht so auch in der technischen Doku von DT, [1], Abschnitt 11.31) PRACK sollte an sein, geht aber auch ohne (lt. [1]) Ein paar Nacharbeiten sind je nach Anschluss noch wichtig Destination for incoming calls sollte umgestellt werden wenn man mehr als eine Rufnummer / Extension hat Die Codec preference sollte angepasst werden, G.722, G.711(A/U) überprüfen [1]: https://geschaeftskunden.telekom.de/internet-dsl/tarife/festnetz-internet-dsl/deutschlandlan-sip-trunk/sip-trunk-technische-unterlage
  8. I did manage to get our phones (SNOM 720/725/760) to use SNI by forcing the use of DNS names instead of IPs with Admin Level / Phones / Settings /Use domain name instead of IP address = ON
  9. @Vodia PBX still monitoring this thread? I've run into a similar problem and it turns out that it appears to be impossible to replace the certificate that is presented when a HTTPS request does not include a SNI (server name indication). When a SNI is present the correct ACME (or manual) certificate is used, but when no SNI extension is present it always falls back to the build in Vodia certificate. Is there any way to make the PBX always use the provisioned ACME certificate?
  10. Some weeks ago we notice that ongoing calls have random audio drop outs. I've managed to track this down to the PBX not sending audio for some 100 milliseconds, sometime up to one second. Upon closer inspection it turned out that this always happens when a new SIP TLS connection is established. Depending on the cipher, the delay between the TLS client hello and the server hello can be something from 100ms up to 800ms. This delay corresponds to audio dropout. I can trigger this manually by using something like: openssl s_client -connect pbx:5061 I have verified that this is not an issue with insufficient entropy, the kernel has sufficient. System details: Intel(R) Core(TM)2 CPU 6300 @ 1.86GHz) PBX release 57.0 Debian 7.9 (wheezy) I know the system is somewhat slow and the release is old. But it should still not need that long to build the TLS server hello and the SIP TLS processing should not block the normal audio path. Has this been addressed in newer releases or will they have the same issue? Or in other words, would upgrading solve this issue? Thanks, Andreas
  11. I have a SIP FXS (Telefon/FAX machine to SIP) gateway that refuses to play with a SNOM ONE. Incoming calls work, but outgoing calls get rejected by the SNOM ONE with 403 Forbidden. I already tried to configure it a SIP Trunk/Gateway, but then calls from the FXS box, through the SNOM OUT to our SIP trunk get rejected with 404 Not found. This seems to be a case of Trunk-To-Trunk routing not working, but I can't find any information on how to route between SIP trunks. Any hints? Here is a trace with the extension only config: [5] 17:35:14.575 PACK: SIP Rx udp:192.168.13.181:5060: INVITE sip:6255555@192.168.13.8 SIP/2.0 From: <sip:sip@192.168.13.8>;tag=c0a80db5-13c4-3a4fd3be-2bed72-5ead To: <sip:6255555@192.168.13.8> Call-ID: 78f738-c0a80db5-13c4-3a4fd3be-2bed6e-ad7@192.168.13.8 CSeq: 1 INVITE Via: SIP/2.0/UDP 192.168.13.181:5060;branch=z9hG4bK-3a4fd3be-2bed76-35e2 Max-Forwards: 70 Contact: <sip:6255555@192.168.13.181:5060> Content-Type: application/SDP Content-Length:171 v=0 o=SIP 25021527 1492432404 IN IP4 192.168.13.181 s=- c=IN IP4 192.168.13.181 t=0 0 a=ptime:30 m=audio 2076 RTP/AVP 4 18 0 101 a=rtpmap:101 telephone-event/8000 [5] 17:35:14.576 PACK: SIP Tx udp:192.168.13.181:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.13.181:5060;branch=z9hG4bK-3a4fd3be-2bed76-35e2 From: <sip:sip@192.168.13.8>;tag=c0a80db5-13c4-3a4fd3be-2bed72-5ead To: <sip:6255555@192.168.13.8>;tag=034a7cf432 Call-ID: 78f738-c0a80db5-13c4-3a4fd3be-2bed6e-ad7@192.168.13.8 CSeq: 1 INVITE Content-Length: 0 [5] 17:35:14.577 PACK: SIP Tx udp:192.168.13.181:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.13.181:5060;branch=z9hG4bK-3a4fd3be-2bed76-35e2 From: <sip:sip@192.168.13.8>;tag=c0a80db5-13c4-3a4fd3be-2bed72-5ead To: <sip:6255555@192.168.13.8>;tag=034a7cf432 Call-ID: 78f738-c0a80db5-13c4-3a4fd3be-2bed6e-ad7@192.168.13.8 CSeq: 1 INVITE User-Agent: Vodia-PBX/5.2.6 Content-Length: 0 [5] 17:35:15.077 PACK: SIP Tr udp:192.168.13.181:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.13.181:5060;branch=z9hG4bK-3a4fd3be-2bed76-35e2 From: <sip:sip@192.168.13.8>;tag=c0a80db5-13c4-3a4fd3be-2bed72-5ead To: <sip:6255555@192.168.13.8>;tag=034a7cf432 Call-ID: 78f738-c0a80db5-13c4-3a4fd3be-2bed6e-ad7@192.168.13.8 CSeq: 1 INVITE User-Agent: Vodia-PBX/5.2.6 Content-Length: 0
  12. RoadRunnR

    5.1.3

    Tried the factory reset, same result. The problem also happens when the time on the phone is correct (normal reboot). The 1970 time setting happens only after a power-on reset.
  13. RoadRunnR

    5.1.3

    snom360 having problems registering after upgrade from 5.1.1 to 5.1.3. Log on the phone shows: 1/1/1970 01:01:53 [ERROR ] PHN: Warning: Certificate verification omitted. TLS Server authentication is switched off! 1/1/1970 01:01:55 [NOTICE] LID: TCP: SIP-0 1 connect timeout 1/1/1970 01:01:55 [ERROR ] PHN: SIP: transport error: 1000000 -> tls:192.168.13.8:5061 1/1/1970 01:01:55 [NOTICE] PHN: SIP: Add dirty host: tls:192.168.13.8:5061 (0 sec) 1/1/1970 01:01:55 [NOTICE] PHN: SIP: final transport error: 1000000 -> tls:192.168.13.8:5061 1/1/1970 01:01:55 [ERROR ] PHN: SIP: transport error 1000000: generating fake 597 After going to Settings / Identity / Reregister User it succeeds.
  14. Hi, Thanks for the excellent help! For everyone with the same problem: It turned out that the old version had some manual customization for snom 710 phones in the html and webpages sub-directory. That customization where not visible in web frontend (at least I couldn't find it) and had to be removed manually. After that everything works like a charm. Andreas
  15. TFTP and PnP event level 8 log: [8] 2013/09/17 09:20:57: HTTP: Received request for file snom360.htm from 192.168.13.72 [8] 2013/09/17 09:20:57: HTTP: file snom360.htm based on template snom_360.xml is sent to 192.168.13.72 [8] 2013/09/17 09:20:58: HTTP: Received request for file snom360-000413236BA0.htm from 192.168.13.72 [7] 2013/09/17 09:20:58: HTTP: No Auth info in the request: Challenge 192.168.13.72 [8] 2013/09/17 09:22:59: HTTP: Received request for file snom_3xx_fw-000000000000.xml from 192.168.13.72 [7] 2013/09/17 09:22:59: HTTP: Error finding snom_3xx_fw-000000000000.xml, Send back 404 Not Found to 192.168.13.72 [8] 2013/09/17 09:23:00: HTTP: Received request for file snom_3xx_phone-000000000000.xml from 192.168.13.72 [7] 2013/09/17 09:23:00: HTTP: Error finding snom_3xx_phone-000000000000.xml, Send back 404 Not Found to 192.168.13.72 [8] 2013/09/17 09:23:01: HTTP: Received request for file snom_360_buttons-000000000000.xml from 192.168.13.72 [7] 2013/09/17 09:23:01: HTTP: Error finding snom_360_buttons-000000000000.xml, Send back 404 Not Found to 192.168.13.72 the contents of the generated/pbx.../224 folder for this extension: -rw-r--r-- 1 root root 71 Oct 26 2012 snom360-000413236BA0.htm -rw-r--r-- 1 root root 465 Oct 26 2012 snom360.htm -rw-r--r-- 1 root root 768 Oct 26 2012 snom_3xx_fkeys.xml -rw-r--r-- 1 root root 183 Oct 26 2012 snom_3xx_fs.xml -rw-r--r-- 1 root root 179 Oct 26 2012 snom_3xx_fw.xml -rw-r--r-- 1 root root 5222 Oct 26 2012 snom_3xx_phone.xml -rw-r--r-- 1 root root 1022 Oct 26 2012 snom_gui_lang.xml -rw-r--r-- 1 root root 1575 Oct 26 2012 snom_web_lang.xml I have now removed that extension completely from the pbx, delete the generate folder for it and recreated it. The extension folder in the generated directory has not been recreated. I tripple checked the MAC, even cut&pasted it from the log, no change, the remaining time is fine also. The html folder contains only a snom_710.xml file. @snom ONE: I can give you access if you send me a ssh public key, I'll drop you a PM when I have external forwarding for the webif setup Thanks Andreas
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