RoadRunnR

Works with the App, but I also don't have the Google Login Button in there.

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?

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.

Sure, I'll send you a link to a video in private.

The "normal" Web Client / Extension login at https://<pbx>/usr_portal.htm. PBX is 65.0.5 (Debian64)

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

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

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

@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?

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

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

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.

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.

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

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

Hi, I have been through this process multiple times and it just does not work. The phone will reboot, wait about 2min with a loginuser prompt, then switch to a time zone selection screen and remain there forever. Going through the timezone selection will finally drop me to "Welcome! Press a key to log on." screen. Phone software is 8.4.35. extract from the phone's log - it is clear that the PBX is asking for credentials, I made double sure that the extension was opened for MAC based provisioning, it still insists on user credentials: [5] 24/12/2001 00:00:22: read_setting_file_list: added URL: https://192.168.13.8:443/prov/snom_3xx_fw-000000000000.xml?model=snom360 [5] 24/12/2001 00:00:22: read_setting_file_list: added URL: https://192.168.13.8:443/prov/snom_3xx_phone-000000000000.xml?model=snom360 [5] 24/12/2001 00:00:22: read_setting_file_list: added URL: https://192.168.13.8:443/prov/snom_360_buttons-000000000000.xml?model=snom360 [5] 24/12/2001 00:00:22: read_setting_file_list: added URL: https://192.168.13.8:443/prov/snom_web_lang.xml?model=snom360 [5] 24/12/2001 00:00:22: read_setting_file_list: added URL: https://192.168.13.8:443/prov/snom_gui_lang.xml?model=snom360 [5] 24/12/2001 00:00:22: Conf setup: found xml style settings [5] 24/12/2001 00:00:22: Fetching URL: https://192.168.13.8:443/prov/snom360-000413236BA0.htm [3] 24/12/2001 00:00:22: TLS: Warning: Certificate verification omitted. TLS Server authentication is switched off! [5] 24/12/2001 00:00:23: webclient::handle_challenge: http server is asking for credentials! [5] 24/12/2001 00:00:23: webclient::handle_challenge: asking user for authorization data ! [5] 24/12/2001 00:01:01: send lldp advertisment [5] 24/12/2001 00:02:01: send lldp advertisment [5] 24/12/2001 00:02:23: webclient::restart_request: got authorization reply back from user ! [5] 24/12/2001 00:02:23: webclient::restart_request: user canceled challenge request ! [1] 24/12/2001 00:02:23: Conf setup: code: 401, host: 192.168.13.8:443, file: /prov/snom360-000413236BA0.htm [5] 24/12/2001 00:02:23: Fetching URL: https://192.168.13.8:443/prov/snom_3xx_fw-000000000000.xml?model=snom360 [3] 24/12/2001 00:02:23: TLS: Warning: Certificate verification omitted. TLS Server authentication is switched off! [1] 24/12/2001 00:02:24: Conf setup: code: 404, host: 192.168.13.8:443, file: /prov/snom_3xx_fw-000000000000.xml?model=snom360 [5] 24/12/2001 00:02:24: Fetching URL: https://192.168.13.8:443/prov/snom_3xx_phone-000000000000.xml?model=snom360 [3] 24/12/2001 00:02:24: TLS: Warning: Certificate verification omitted. TLS Server authentication is switched off! [1] 24/12/2001 00:02:25: Conf setup: code: 404, host: 192.168.13.8:443, file: /prov/snom_3xx_phone-000000000000.xml?model=snom360 [5] 24/12/2001 00:02:25: Fetching URL: https://192.168.13.8:443/prov/snom_360_buttons-000000000000.xml?model=snom360 [3] 24/12/2001 00:02:25: TLS: Warning: Certificate verification omitted. TLS Server authentication is switched off! [1] 24/12/2001 00:02:26: Conf setup: code: 404, host: 192.168.13.8:443, file: /prov/snom_360_buttons-000000000000.xml?model=snom360 [5] 24/12/2001 00:02:26: Fetching URL: https://192.168.13.8:443/prov/snom_web_lang.xml?model=snom360 [3] 24/12/2001 00:02:26: TLS: Warning: Certificate verification omitted. TLS Server authentication is switched off! [5] 24/12/2001 00:02:26: read_xml_settings: found web-languages XML header [5] 24/12/2001 00:02:26: read_xml_settings: found one byte encoding: 1 [5] 24/12/2001 00:02:26: Conf setup: found xml style settings

Hi, Just upgraded from 4.5 free edition to a paid 5.1.1 and auto provisioning seems to broken. Our test snom360 is setup with a MAC in the extension, MAC is opened for provisioning without password. DHCP server option 66 points to the IP of the PBX. We then reset the phone to factory defaults and rebooted, expecting it *just* work, but nothing. The log shows that the phone contacted the pbx: [3] 20130915112856: Received SUBSCRIBE for plug and play. SIP multicast support is set to handle plug and play.[8] 20130915112858: HTTP: Received request for file snom360.htm from 192.168.13.72 [7] 20130915112858: No need to write snom_360.xml to file system [8] 20130915112858: HTTP: file snom360.htm based on template snom_360.xml is sent to 192.168.13.72 [8] 20130915112859: HTTP: Received request for file snom360-000413236BA0.htm from 192.168.13.72 [7] 20130915112859: HTTP: No Auth info in the request: Challenge 192.168.13.72 So, why is it asking for authentication when the MAC is trusted? Also, the SIP PnP request seem to broken as well, returning "noresource": [5] 20130915113353: SIP Tr udp:192.168.13.72:2048: NOTIFY sip:192.168.13.72:2048 SIP/2.0 Via: SIP/2.0/UDP 192.168.13.8:5060;branch=z9hG4bK-848c47228f3b1af977a6386fd6553f81;rport From: <sip:MAC%3a000413236BA0@pbx.local>;tag=133a5c2fd5 To: <sip:MAC%3a000413236BA0@pbx.local>;tag=83548586 Call-ID: 475330253@192.168.13.72 CSeq: 568772230 NOTIFY Max-Forwards: 70 Subscription-State: terminated;reason=noresource Event: ua-profile Content-Type: application/url Content-Length: 41 Any idea what's going on? Thanks Andreas

We have 10 Snom's, so yes, I choose to see it as a downgrade and yes I am very unhappy. Most of them are 360's and therefore not really new ones, so snomONE was not the driver for getting them, however the availability (or rather the price when adding the snomONE license into the mix) will be a driver for locking elsewhere when the next phone is needed. What about an incentive for existing snom customers? The serials and MACs of the phones should be simply enough to verify?

Glad to hear that Snom-ONE will finally Buzz-Word compliant. Are there any real benefits for the users aside from having a complicated license model and becoming more expensive?

Got it work after switching from the debian4 binary to the centos one. Seems the debian4 version was linked with an libc version that has problems with IPv4/v6 dual stack setups

Hi, It seems that runing snomONE on a system with both IPv4 and IPv6 address is broken. I'm no sure since when, I seem to remember that it worked in the first 4.5 version. With 1075 and 1090 the log file goes crazy with this: [0] 2012/09/13 11:00:49: TCP(IPv4):Could not bind socket to port 443 on IP 0.0.0.0 [0] 2012/09/13 11:00:49: FATAL: Could not open TCP port 443 for HTTP/HTTPS [0] 2012/09/13 11:00:49: (IPv6)Could not bind socket to port 443 on IP [::] [0] 2012/09/13 11:00:49: FATAL: Could not open TCP port 443 for HTTP/HTTPS [0] 2012/09/13 11:00:49: TCP(IPv4):Could not bind socket to port 389 on IP 0.0.0.0 [0] 2012/09/13 11:00:49: FATAL: Could not open TCP port 389 for HTTP/HTTPS [0] 2012/09/13 11:00:49: (IPv6)Could not bind socket to port 389 on IP [::] [0] 2012/09/13 11:00:49: FATAL: Could not open TCP port 389 for HTTP/HTTPS [0] 2012/09/13 11:00:59: (IPv6)Could not bind socket to port 80 on IP [::] It repeats over and over, connections to the webinterface are intermittend (since it shuts down port 80 and 443 and restarts them all the time). This also kills auto-provisioning. The only ports that seem to work reliably are 5060 and 5061 (SIP and RTP). Interface config: eth0 Link encap:Ethernet HWaddr 00:50:56:ae:17:48 inet addr:192.168.13.8 Bcast:192.168.13.255 Mask:255.255.255.0 inet6 addr: 2001:6f8:12d9:13::8/64 Scope:Global inet6 addr: 2001:6f8:12d9:13:250:56ff:feae:1748/64 Scope:Global inet6 addr: fe80::250:56ff:feae:1748/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:9843790 errors:0 dropped:0 overruns:0 frame:0 TX packets:8415647 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:2030992331 (2.0 GB) TX bytes:1692246693 (1.6 GB) lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:2995 errors:0 dropped:0 overruns:0 frame:0 TX packets:2995 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:369367 (369.3 KB) TX bytes:369367 (369.3 KB) System: # uname -a Linux vlx049-tpo 2.6.32-42-generic-pae #95-Ubuntu SMP Wed Jul 25 16:13:09 UTC 2012 i686 GNU/Linux Andreas

Does anybody have an answer to this?

Using ESX Server with DirectPath I/O, it would be possible to add an Sangoma card to the appliance. Is this supported and what licensee (Free, Yellow or Blue) would I need? Thanks Andreas

such mal auf eBay nach SIP und TFE, es gibt da wohl eine IP PBX von Tiptel die TFE kann. Es gibt auch diverse ISDN TFE's, aber dazu braucht man natürlich ein SIP->ISDN GW. Ansonsten sowas wie ein Relay an Seriel/Parallel/USB und einen Mini-HTTP der das ansteuern kann. Keine Ahnung ob es sowas fertig gibt, aber ein Bastler sollte so was hinbekommen. Im Home Automation Bereich könnte es das auch was geben. Edit: Schau mal hier: http://www.baudisch.de/produkte/sip-sprechstelle Andreas

So ein ähnliches Thema wurde schon mal vor einiger Zeit ohne Resultat diskutiert: http://forum.snomone.com/index.php?/topic/3312-immer-noch-das-problem-mit-der-0/ Die Lösung "ohne 0" ist in D einfach nicht akzeptabel. Es gibt hier so gut wie keine Nebenstellenanlage (ich kenne jedenfalls keine Firma/Einrichtung/Behörde bei der das nicht so ist) die ohne eine spezielle Ziffer für die Amtsholung arbeitet. Wir haben unserem ISDN->SIP Gateway (eine Asterisk) jetzt beigebracht das es vor einkommende externe Rufnummern immer eine 0 hängt und der SNOMone das sie bei raus gehenden Rufen entsprechend die 0 wieder entfernt. Soweit ist das alles ganz schön, nur besteht die SNOMone jetzt immer darauf eine doppel '0' durch ein '+' zu ersetzen. Wenn also ein Anruf von 0170-222xxx kommt, wird der als +1-702-222xxx reportet. Gibt es eine Möglichkeit dieses Verhalten zu unterdrücken? Gruß Andreas