Vodia PBX
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Seems that v3 f/w has a bug when running more than 4+ domains in a hosted setup.
All outbound calls remain ok but inbound calls were getting dropped and sometimes even stuck in a loop.
Backdating to v2 f/w removed the issue.
Anyone else experienced this?
This is probably because of the tel:-alias changes. Check if your trunks are set to "global".
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Has the SIP IP Replacement list and IP Routing List been improved in v3 f/w?
I can see in your release notes it says that "The SIP routing subsystem was improved. Especially in cases when the system was using advanced routing mechanisms like IP tables, the old way could lead to wrong IP addresses", has this issue been resolved in the new f/w?
No, thoso two lists have not been touched. What has been touched is the way the PBX looks at the OS to find out which interface should be used for sending a request. Especially in Linux and BSD (VPN and ipchains!) that could cause problems.
Those two lists are evil anyway. I regret that we added them. It is just a support nightmare - if a customer does not have a public IP address, the ITSP must have either offer a SBC or they must have a public IP address. Period. Everything else is crazy with support and troubleshooting. Frustrating for every one.
One nice day hopefully we laugh about it and just use IPv6.
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Okay, please give the following image a shot:
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I am running 3.0.0.2998 on a CS410. The customer has 2 phones in an agent group. One of the 2 phones is almost always logged out. They claim that there have been 3 times now where the phone has logged itself in. When I look at the extension in the web interface, the phone status shows it is logged in, however the agent group sometimes will not show the time range that the phone was available for, and when it does, it will often show a start time when no one was in the building. Is there a better way to track this?
The only thing I can think of is to send an email when the status changes. Then at least the agent gets aware of the status change. Also, the 3.0 version supports AJAX in the web interface (user mode) where the admin of the queue can see who is logged in an who is logged out.
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Could it have anything to do with <http_cert_file/> <http_tls_web/> in the pbx.xml file? These are not set, obviously.
Voice mail messages are being sent through the server to customer email servers without issue. I am receiving system messages such as CPU Limit, for example.
I think most SMTP servers do not require a client certificate, so I don't think that is the problem. We just need a test account then we can give it a try.
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When do you think this will be implented?
Does this perhaps also include saying that you're caller 4 in the line of waiting callers?
Yepp!
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We are looking into this. There is a problem somewhere.
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If I put in "901; 903" as participants will this start the conference by ringing 901 and 903?
If not this would be a really usefull feature.
Well, this automatic invitation into the conference has a lot of problems. If you think on how probable it is that the other side really picks up. The biggest problems are redirects to cell phones and mailboxes, where the PBX has no idea if it is a real person or a machine. Then the admin has the problem that the calls need to be torn down again and so on. Muddy area...
At least we can say 3.0 offers attended transfer into a conference room. That means the organizer can call people up, call the conference and then perform an attended transfer into the conference.
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We are testing Agent groups and its working perfectly.
Is there a way of announcing how many customers are waiting inline ahead of you?
And can the PBX play a (custom/own) message stating that the waitingtime would be longer then 10 minutes and asking the person to hang and try later.
Currently we don't estimate the waiting time. We have the prompts ready for queue annoucements for that, but the logic is missing right now.
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hi support,
any update on above request? thx
regards,
kelvin
Treat calls to other domains just like calls to other companies. As a trunk you can use the outbound proxy "127.0.0.1" - which loops the request back to itself. Make sure that this trunk is a "global" trunk, then you need only one.
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We are having a problem with inbound call routing. I setup individual proxies on our SBC for each of the domains on the PBX, all on different IP addresses, but that calls are still not being associated with the correct domain. All calls are giving me a message "Received incoming call without trunk information and user has not been found".
The only thing that helped was putting all the trunks that have inbound calls coming to them into "Global" mode. We are currently running pbxctrl-centos5-3.0.0.2991. Do you know if there is a new build that fixes this trunk association problem?
The "outbound proxy" is also the "inbound proxy" for the trunk. You should set an outbound proxy (even if you just want to receive calls) so that the PBX can tell by the IP address and the port where the call comes from. The PBX also supports DNS resolution, that means if you ahve a DNS SRV record, the PBX will access requests from any of the potential destination addresses.
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Does pbxnsip support HD voice?
Yes, the PBX supports G.722 (which runs at 64 kbit/s). However, as the prompts are all in 8 kHz you should not expect miracles when taking to the mailbox. But for calls that pass through the PBX the call quality should be great.
We don't support G.722.1 (see http://en.wikipedia.org/wiki/G.722.1). That is a different codec. Seems the licensing terms are reasonable and even the CPU overhead is relatively low. So that might mean one nice day we can add this codec as well.
Please note that even for G.722, transcoding means loosing quality. If you are transcoding between G.711 and G.722 you will have worse quality than with a pure G.711 call. This is a little bit surprising in the beginning, but it becomes understandable that representing an audio signal with 64 kbit/s in two different ways can result only in a common denominator quality. One more reason to avoid transcoding where ever possible.
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After checking several of our servers:
Windows 2003 servers (ver 2.1.6.2450)
Upgraded to 3.x - pbx.xml does not have the email_<variables> added.
Linux servers (pbxctrl-rhes4-2.1.14.2498)
Upgraded to 3.x - pbx.xml has the email_<variables> added
Can I add the variables manually to the xml file?
I would not manually add them. Just change a global setting from the web interface, this will rewrite the pbx.xml file. Then either edit it there (need restart) or use the web interface to change it (described on the wiki page I mentioned above).
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Did anyone figure out how to create a button on Snom phones to dial *90?
So customer can press the button and then hit the extension number to intercom an extension?
Seems like every system I sell the customer asks about this.
At the moment I would say the answer is no. You can program specific destinations, but you cannot have the user press a key and then enter the remaining digits.
The only thing would be entering something like "9" in the beginning (well, that's also a key!) and then deal with the local dial plan on the phone to generate a *90 out of it.
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Okay, I found the email_ settings in the xml file but which one specifically will turn off these alerts: "Source address for sip:" and still allow me to receive the CPU stats that are sent out nightly?
That would be "email_address_change".
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I am trying to setup multicast paging using Snom phones as the destination. I have Multicast enabled on the phones with the multicast IP/port matching what I have entered in the paging account. When I call the extension the phone connects but the page doesn't go through. Doing a packet capture shows the audio stream going from the phone to the PBX, but I do not see any multicast traffic leaving the pbx. Are there any other places I need to enable multicast support to enable the paging?
I am using 3.0.0.2998 on the cs410 and 7.1.35 on the phones.
If you have several IP addresses, you might have to specify on what IP address the multicast traffic should be set. There is a setting called "Multicast IP-Addresses" in the admin/ports section where you can enter the private IP address. You need to restart the PBX after changing that setting.
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Has a new options been slipstreamed into release 2993 release?
We have added STARTTLS, which is probably causing the problem. It works with google, but that seems to be only half the battle. We have another case where it does not work, but maybe you can also send me (PM) a test account for that server and we can test it from the lab.
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The phone registers and shows its inside IP behind NAT, but the SIP packets are addressing it properly via the real-world address .. when sending RTP to the phone it will send it to the inside IP that is behind NAT which is obviously unroutable .. any ideas .. would a packet trace help?
For media-related issues, PCAP is the preferred way to see what is really going on. Maybe in this case the routing table of the PBX would also be useful.
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Whenever I user * for the pattern and 100 as the replacement. It trys to dial 100 instead of the dialed number. Is there any workaround for this.
Well, what header are you looking at? The Request-URI must be 100, because that's what has been registered.
The To-Header should contain the destination number that the gateway should dial.
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Hi PBXnSIP, where is the NAT in this topology?
No NAT here. Just wanted to make the general point with TCP that the contact header in SIP is useless for a user-agent ("Contact: <sip:lala@ip;transport=tcp>"), especially behind NAT. Instead of that, they should have taken something like a connection ID.
But without NAT is is almost the same problem, because also phones with a routable address typically do not accept incoming TCP connections, because of DoS and general programming pain in embedded environments.
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Not sure what you mean PBXNSIP.
Well, my Wireshark shows that the UDP packet checksums are incorrect... The reason is defintively not the ITSP, as the UDP checksums are generated by the router (as UDP is layer 3). Maybe something stupid like bits flipping around, bytes being chopped off, or just a bad firmware on the router. Checksums are there for a reason and they should be correct!
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The reason we are registering the fxo port is because it is behind a non configurable router that changes ip addresses.
Yea, that's the reason why we introduced that "call extension" feature...
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I have my Audiocodes 114 fxo registered to pbxnsip with extention 100. I was wondering how I dialout from that extention. I see options for "call extention" in my dial plan. But I don't know what patern and replacement to put in there. I want everything to go out my fxo port. I was thinking you put * for the pattern and 100 for the replacement. But it trys to dial 100 instead of the number. Any Ideas.
Is there any reason why you want to register the FXO? Usually you just use the outbound proxy on both the PBX and the gateway to point to each other and use a trunk.
If you want to use the "call extension" feature, the replacement is the extension number (e.g. "100", no "sip" or domain name before or after that).
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Where are the flags located?
Ehh.... they are hidden, but available. Check the pbx.xml file for names starting with "email_". You can change them in the pbx.xml file (which would require a restart) or use the trick in http://wiki.pbxnsip.com/index.php/Global_Configuration_File.
Hunt Group--Auto Attendant--Hunt Group Fails to get to Final Stage
in Hunt Group Setup
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That sounds like a DTMF problem to me. Do you see anything in the log (media, level 6) like "Received DTMF 2"?