Vodia PBX
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So where dose the timezone.xml file go in the directory structure.
You need to put that file into the html directory (make it if it does not exist). After changing the file or loading it there the PBX service must be restarted...
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A semicolon might actually be a reasonable choice - it will make the match pratically impossible. I would be careful with numbers, maybe someone really dials that number and then you have "funny" effects.
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There is some interesting information on http://wiki.snom.com/Features/LED_Remote_Control that explains some of the background.
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Check out http://wiki.pbxnsip.com/index.php/Localization#Time_Zones - if you have a time zone that we should include in the standard distribution we are happy to put it in!
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I am looking at sip server software that will work with our existing SNOM 370 phones and have seen at least one thread that mentions that the dialog-info settings on the phone work. If my understanding of this is correct, if I set a right-side button to monitor an extension, when that extension receives a call the actual call info will display on my screen show who the call is to and from. Then I can press that blinking button and 'steal' the call to my phone. Is that correct?
Thanks,
Brian
Yes, stealing calls is possible. If you want to avoid this, just use the speed dial mode in the buttons select box on the PBX (see http://wiki.pbxnsip.com/index.php/Assigning_Buttons). Then you will just monitor the extension status - if you press it you will dial it (that's why it is called speed dial). This is the original BLF idea.
And there is always the dialog permission. With this setting you can also control the access rights to information about calls on the respective extension.
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No Problem is http://wiki.pbxnsip.com/index.php/SNMP the full and complete MIB?
So far yes. 2.2 will also add a memory sensor (number 12), but is not released yet.
HQ experienced a FAILED Switch, thus killing the PBX for a bit, and we saw a call in the call log in the remote office trying to make a trunk call.Ouch. Network equipment like switches and routers are also not bug free. We have to keep this in mind when tracking down problems.
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Did you use a higher priority for the next entry in the dial plan?
http://wiki.pbxnsip.com/index.php/Trunk_Se...tbound_Settings was not clear on this, has just been updated.
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In the replacement, you can use a pattern like sip:\*43@\r to make the PBX dial *43. In the pattern you need to use something else than the star character, because that is the code that tells the PBX to look for it's own codes.
Maybe you can use a pattern like 012* as prefix and sip:\*\1@\r as replacement, then if someone dials 01243 the PBX would dial *43 on the trunk.
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Whow how did you get these HUGE screenshots?
First, I would always use the outbound proxy. Many SIP devices don't like it when the PBX says send it to "localhist"...
Don't use the STUN server (STUN is nonsense anyway).
Question: Why do you use a trunk on the PBX? I think you would register the Grandstream as an extension. Who is registering where? PBX trunks send REGISTER out, but do not receive REGISTER messages.
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If you can, use an SNMP tool to poll the PBX for data like how many calls are on. If the PBX does not respond, there is trouble going on.
It is *not* a good idea to stress the system with lots ot http/https requests to see if it is (still) responding. SNMP is very lightweight and has practically no impact on the operation.
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Ehh... FXO is not FXO... Maybe we really need to think about a couple of settings, like "check polarity change", "loop disconnect timeout time".
We are using this new Comcast service and I love it. No such problems, just great audio over a very short FXO cable.
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But the fact is that the cell is also included in the call even after the "office hours". It seems to me that the service flag setup on the extension is totally ignored.
Did you check if the time is correct on the system? Are you using the right time zone for the system, domain and extension?
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If I don't have a buddy list on the Polycom then I have a (slight) problem now. Below the line button there is a image popping up from time to time, which is a little bit annoying. Still on 2.2.2. When I provision a buddy there things look okay. Running 2.1.6.2448, of course.
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Yes, I provided them the same info and will let you know what they say. From your standpoint this is something that should work and the issues appears to be on their side?
Well, sending to port 0 is not an option in IP, so that must be changed anyway. If that was the last obstable? You never know!
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*BroadVox transitions to T38
[7] 2008/02/24 16:57:58: SIP Rx udp:64.152.60.75:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport=5060
From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02
To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4
Call-ID: 1459988973_58241008@64.152.60.75
.
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m=image 0 udptl t38
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPFEC
a=sendrecv
Yes that is definitevely a problem. Broadvox is usually quite responsive, would be great to have them provide this kind of service! Did you also ask them if they can do anything about it?
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Cool.
Make sure you are using 7.1.30 or higher on the. Previous versions have their problems!
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The problem is a remote phone at neither sighting trying to register at the city1.domain.com location may be doing some DNS lookup stuff, and the master site domain.com has dns-srv records pointing to domain.com and I think this is causing that phone a problem.
If you want that a SIP device only uses DNS A or DNS AAAA then specify the port number behind the domain name (e.g. domain.com:5060). This is a little RFC3262 trick.
[7]22/2/2008 11:18:02: Trusted IP Addresses: udp:11.222.195.38[2]22/2/2008 11:18:03: Registrar 45@city1.domain.com refused with code 482
482 is really strange. IMHO the PBX should never do that on a REGISTER request. Is 11.222.195.38 the address that you expect?
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I just enabled ENUM on the phone, ...
I would not enable ENUM on the phone just to be able to dial regular numbers. ENUM is a long long topic that you probably don't want to touch just to solve that problem. I would solve the problem with that dial plan on the PBX above.
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No password for the service flag, please! The caller has to authenticate himself e.g. in a calling card and then he can call the service flag also from home or on the road.
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However, the call from the AA to the extension rings forever instead of being transferred to voicemail. Hitting reject on the Polycom phone should force the caller to voicemail (as it does on an extension to extension call, and as it does on our other test server coming from the AA), but instead gives the caller the audio_en/ex_wrong_id.wav 'this number could not be found'.
I think the problem is that the Polycom sends a 6xx code, which is a very high priority code?
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So you want to have two locations that can call each other? You should set up two trunks between them and then use the dial plan to route the calls between them. Just pick a nice prefix that they can use to call each other, or just pick the PSTN number of the PBX as prefix.
Not sure what that has to do with phones...
GPX 2000 transfer issue
in Grandstream Phones
Posted
Hmm. First question of course is what version you are running on the phone (and the PBX).
Second it would be good to have a trace so that we can see what is going wrong. If there is a probability involved, it sounds to me like a race condition. Maybe the phone behaves differently if it received a BYE message during the transfer - or not.