Vodia PBX

Is that because you want to run multiple domains on multiple IP addresses? Are you using UDP or TCP? We recently solved a problem where the contact header for TCP connections was using the OS routing table instead of the local name of the socket. Just checking if this problem is related.

It would be interesting to see where the problem is. I know that for a phone it can be tricky to dynamically switch between 8 kHz and 16 Khz; especially when the phone advertizes 16 kHz in the beginning then receives 8 kHz. Wideband is still a little bit bleeding edge.

I believe that was in the beginning the business of gateway companies; at least for Vegastream. Businesses had a PBX and they saw no reason to change it; the only thing was that they wanted home users to connect via Internet.

What replacement are you using? Maybe also use 00* as replacement. Otherwise the PBX will cut off the first two zeros and then I can understand the carrier rejects the call.

I also don't know. There are many things I don't know with the .cnf.xml files.....

Maybe you should try the pattern xxxxxxxxxx; make sure that you use different priorities. When you set the log level to 9, you should see what expressions the PBX tries to match.

Factory reset or restart system?

You can log into a extension mailbox and then dial from there. Going into the mailbox: While hearing the prompt, enter your PIN. Dialling the star code: Just dial the code and then terminate it with the pound sign (e.g. *98*123#).

I guess you already rebooted the device? Run wireshark on your PC to see if the CS410 sends out a bootp (DHCP) request.

No, one hunt group for every different destination.... Okay, maybe not very practical. Probably it is easier to leave that part to the phone. What phones are you using?

Did you change the domain name? Make sure it contains the name "192.168.1.3" or "localhost". The domain name is important because the PBX can handle a lot of domains, and needs to figure out which domain to choose. "localhost" is a wildcard matching anything.

Different ways: You can delegate that job to the phone; or you can employ a hunt group in front of the phone to set the ring melody. I guess we should think about adding a settings for an extension just like for hunt groups and agent groups that specifies how to ring.

Turn SIP logging on and check if you see the REGISTER packets in the log file. If that is not the case, you are still wrestling with the IP subsystem setup.

I would kill everything listing on the SIP port to make sure there is no old stuff running. SIP runs actually on port 5060 UDP, 5060 TCP and 5061 TCP/TLS. The same ports are allocated for IPv4 and IPv6, so you might see six ports being allocated. That's fine. This is the way the IETF wants it!

You don't need to change the script. Just hit the save button, and the new port will be stored in the config file. Next time when it starts it will stick to the port that you specify in the web interface. Windows seems to be more "tolerant" to these kind of problems. But the behavior is very erratic. Only one application should listen on one port IMHO.

Try this: ./pbxctrl --dir /usr/local/pbxnsip/ --config config.xml --log log.file --no-daemon

Try to start it manually, then set the log to write to a file, and check for error messages. Maybe there is a port that cannot be allocated. Or just "chmod a+x" missing on a file.

A routing proxy can help, but you don't need it. You can also use a trunk; failover can be used to continue processing the dial plan when the number is not local. The original idea of drilling a shortcut was flawed; think about only about the CDR. In which domain will you see the CDR? Stuff like that. We made the mistake to listen to a customer who did not care much about any other problems than solving just his specifc problem. I guess we have to come up with an example TAR configuration that demonstrates how to set up multiple domains that can call each other without external proxy.

The point here is that there are customers who set up extensions with numbers like 911 and then nobody can call for help any more. Therefore, when someone calls 911 and there is a extension with the name 911, the PBX will skip that extension and attempt and outbound call. Of course, it is not 100 % speed dial. The point is that in many VoIP installations you cannot just pass 911 to the PSTN gateway or ITSP, because the user is located in another location than the PSTN gateway. Then you have to dial the ten-digit emergency number which resolves to the local "police station" (or whoever is providing that service). If the service provider does that already for you, you can just pass the 911 and things are very easy.

I assume you were talking about calling into another domain using a name with "tel:" in the beginning. The "tel:" has been replaced with a "+" now and it really represents a global phone number. What is your use-case? What are you doing when you are calling into another domain? Maybe there is a simple workaround in version 3.

Yes, thanks! There was a space missing. Will be fixed in the next version! Until then, you can use the attached version. Put it into the html directory of the PBX (make it if it does not exist yet) and frop the file there. Don't forget to remove it when doing an upgrade later. polycom_phone.xml

The background information is that domains are "mobile" in 3.0. Think about a server farm where you have 1000 domains, running on 100 servers, where you want to be able to move a domain from one server to another. In this setup, you would use trunks from one domain to another, probably involving DNS to locate the other server.

Well, now you have to set up trunks to call from one domain to another. Just think that the other domain is running on a different server.

Do you now see a log message?

I think such files must be listed in the master file (see below). <?xml version="1.0" standalone="yes"?> <!-- Default Master SIP Configuration File--> <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="{http-url}/polycom_phone_{mac}.cfg, {http-url}/polycom_sip_{mac}.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY="" LICENSE_DIRECTORY=""> <APPLICATION_SPIP300 APP_FILE_PATH_SPIP300="sip_212.ld" CONFIG_FILES_SPIP300="phone1_212.cfg, sip_212.cfg"/> <APPLICATION_SPIP500 APP_FILE_PATH_SPIP500="sip_212.ld" CONFIG_FILES_SPIP500="phone1_212.cfg, sip_212.cfg"/> </APPLICATION> Currently we don't list them there. Hmm, if we list them there what happens if they are not there? I guess we need to take a look the version 3.1 (of the Polycom phones) templates to see where we can fit them in.