Vodia PBX

There is no point in trying out unless you know what the problem is. E.g. you can check the registry on the server if there was a limit, and unless you change it, there will be point in restarting the server. Another problem that is causing such problems are usually firewalls. For example if those phones are behind NAT with a router that accepts only 64 connections, it would perfectly explain the problem. In that case, you would have to get a new router or change the setting on that router, if there is one. From the PBX point of view, there are installations with more than 64 connections and usually there is no problem.

Maybe there is a limitation set by the server. I searched the Internet and found this one, may be good as a quick checklist: https://social.technet.microsoft.com/Forums/windowsserver/en-US/fb7a6ef7-5a70-43a4-b0fe-c0252877467b/concurrent-tcp-connection

There is a setting "Maximum number of SIP connections" in admin/SIP/Settings. Make sure that this set to something higher than 60.

Well Agent Groups have a lot more features than hunt groups (e.g. the MoH). Hunt groups are essentially a way to have more than one phone ring in an incoming call, but not much more.

Agent groups are described on http://vodia.com/documentation/agentgroups please check it out.

In a nutshell, use a ACD (agent group). This will take care about lining up callers, even if the only "agent" for the ACD is a single extension.

We did that in the old versions. However we often ended up in a chaos, essentially because of two problems (1) Excel import and export was buggy, for example when quotes and newlines are in the text and chaos when changes need to be added; and (2) many translations were made out of context. That is why we "invested" in the translation right on the page, with the PBX uploading the translations to a central database. So today you don't even need a XML file; I think the only thing we need to do is to add the Arabic language to the drop down, even if it currently just an empty shell. And we need to take a look at the HTML where we can do the right to left direction.

Caller-ID has two directions, inbound and outbound. In the PBX the term "DID" is used for inbound and "ANI" for outbound. I assume we are talking about inbound here. The easiest is in a call center environment to use the ACD. In the ACD there is a special tab for representing the caller-ID ("Numbers"). There you can define what numbers the ACD serves, and even upload WAV files that are played back to the agent when the call connects, so that they don't even need a display to know how they should answer the phone.

From what I know a FAX device that wants to send a page always sends the FAX tone, no matter what the other side says. This is how you can multiplex a regular landline with a FAX machine. For example if the call hits the auto attendant, it can detect the fax tone and then redirect the call into a mailbox. Maybe you can record the call and attach the WAV here so that we get a better understanding what is going on.

Not sure if 5.1.3 had that already, but in the mailbox settings you can set the mailbox to be a FAX mailbox. Then the PBX is sending 10 seconds of digital silence (seems like that is what you are experiencing). The PBX waits for the FAX tone during that time, and that might be the problem: Chicken/egg. If your sender is also waiting for a FAX tone, both sides are sitting there and waiting for the other side to start the conversation. I am not even sure how this is supposed to work, who should start the tone or if it matters at all.

Maybe file system full or quota exceeded? Can you manually create lets say a text file in the recording directory with 4 KB?

Hmm. Are you able to record for example the name for the user or the personal announcement (*98). Maybe your file system is not writable for whatever reason.

Uploading in the forum is a little bit complicated. You need to switch into the "more reply options" mode where you can manage your attachments then you can put the link into the text. You can open a ticket at https://vodia.zendesk.com/home

Should be possible to attach something here (maybe you have to put it into a ZIP). Otherwise just open a ticket and attach it there with a reference to the forum topic.

I would try to get a PCAP for the call, I think 5.2.3 already supports generating PCAP for the extension and/or the trunk the call is involved with. Maybe it is related to silence suppression. If there is a long period of silence without any RTP packet, the PBX might think that the call is "dead" and kills it. In that case you would get an admin email with the SIP trace attached.

The PCAP looks okay. What happens if someone misses a call? Does the PBX send out an email for the missed call? Maybe this is an email problem an not a voicemail problem. Can you listen to the voicemail message if you retrieve it with *97?

Call Recording kann auf verschiedenen Ebenen gesteuert werden: Domäne und Nebenstelle. Auf Domänen-Ebene geht am einfachsten (http://vodia.com/documentation/domain_settings "Recording Default for this Domain"); dort kann man einstellen welche Typen aufgezeichnet werden sollen. Das Ganze setzt voraus dass die PBX für Call-Recording lizenziert ist.

Hmm. I think according to the RFC the PBX does the right thing. But I just don't see the practical relevance. The registration process should make sure that the PBX picks a random address (if there are more than one) and then stick to it. I think this is something that we should change; I don't even think that we need a setting for that.

I guess you hear "There was no answer to your request"? In that case your are right, the PBX sends only if the call was ringing. We will add that with the next build.

Did you already use the drop-down setup from the PBX trunk setup web page? Usually those problems should be solved with the outbound proxy.

It does not matter what carrier you are using, unless there is a carrier that sends 18x without SDP (which would not be considered "good"). I guess we could change the software that resets the ringback logic when the failover happens, so that on the next trunk you would also have the ringback experience.

It is a tricky topic. Generally, the PBX tries to use the remote ring back tone that comes from a trunk, e.g. when you call another country you would hear that country ringback tone or think about vanity ringback tones. If the trunk sends a 18x code without SDP that strategy obviously failed and it uses its own ringback tone from the file ringback.wav. Now when it fails over and it did not have remote SDP, it should continue playing that ringback file; only if the failover happens on a trunk that provides early media the PBX will be in trouble as it has already decided to use the remote early media). BTW the same problem occurs if the remote trunk provides the SDP, but does not provide the actual media, e.g. because of NAT problems. Not sure what I should recommend here. Failover is really only for failure cases and it should be an exception, where such glitches like early media problems are happening only in a few cases.

After the phone has it's IP address it will be visible. No need to wait longer than a minute.

Actually, today I would not recommend to use the SIP multicast or rely on it. The PBX now supports a much easier way to find devices in the LAN just by scanning the network. After the phone powers up, you can assign it to an extension and domain just through the web interface. The problem with the multicast PnP is that things get terrible when there are more than one PBX in the LAN and that you have to assign the MAC to an extension before you can start the phone. Plus the mechanism works only with a few phones models. See http://www.vodia.com/documentation/pnp_lan_prov for more information.

LoL you mean jail breaking snom phones? I think that can be done using TFTP update, http://wiki.snom.com/FAQ/How_can_I_set_the_phone_back_to_admin_mode_or_factory_defaults_respectively