Walter
-
Posts
47 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Walter
-
-
No.
just that you'll need a https fax provider too. (doesn't need to be the sip trunk provider)
OH OK, getting the idea now.
Do you know who provides HTTPS fax? and the adapters?
-
What I like about ubuntu is the large hcl they have. I use desktop so I can remote into the pbx and use it's web browser to work on the phones.
Same with debian with he Graphic interface installed. Remote into it. and see the desktop.
Since it will only be the PBX and the load of it is less than UBUNTU and windows open faster than in UBUNTU I made the selection.
Try it , I installed both in the same pc so I know the difference on speed is not the PC.
Debian will give you the option to install the graphic interface at the first screen, and you can also install the email server and use it to send Voice mail to email if PC is strong enough for it.
-
REFER is usually used in a call transfer situation. So typically it is not an issue for fax, door phones, etc.
Ok but still a little confused over this REFFER thing, Sorry.
The ATA will be getting an extension number from the PBX and the PBX will be redirecting the DID to that extension.
That still is not considered Transfer? Is not the same way a Vice call would be routed though the PBX?
Remember my PBX is in the cloud because there are a number of stores I have to give DFax and VOice extensions to.
-
So then I guess it will requiere that my SIP Trunk provider can provide HTTP for fax codecs and I will have to terminate on the AC ATA?
-
I had to remove DHCP. DHCP caused an issue with allowing a static to work. After I removed it, it worked fine.
Glad to hear I wasn't the only one, well actually google the problem and found a lot of ways tog et around it.
Debian looks almost the same, menus are in the same place, and I didn't have that problem, it disconnects the Network and reconnects it under the new IP.
and the nominal Memory and Processor usage s less when idle or working in Debian that what it was in Ubuntu.
-
If you run the PBX in the "cloud" jitter is much higher than on CPE, so the virtualization overhead is probably neglectable. But I would be careful that the server is not too virtual, the service provider must not move the VM around from one phsical host to another all the time as this will create major service interruptions for your users.
150 extensions should be okay. The more important number is how many calls you expect. I think the important point here is that you have the option to add more CPU horsepower if you need it (relocate the server on a different physical device if you run short on resources).
And you are right, FAX is something important. Apart from using the PBX and T.38 for FAX, you can also consider using HTTPS for the FAX and bypass the whole FAX problem. PAP2 is an option, but I think you can also take a look at AudioCodes and Mediatrix, especially if you are interested inthe HTTP solution.
Does HTTP fax works better? I can still go thru the snomONE?
-
it is important to realize that even any ATA ports suppplied by Sangoma/snom Plus are considered 3rd party.
But for fax the non-snom limitation shouldn't be much of a problem as the main limitation on non-snom is no SIP REFER and fax extension shouldn't generally need that.
How can I make sure that my ATA is not using SIP REFER? I didn't know about that.
-
Hello, I'm not very familiar with linux, so I wanted to ask if snomeOne will install under Ubuntu Desktop v10 ?
Also if will it be much difference using server ubuntu than Desktop ubuntu? I know server doesn't have a graphic interface.
Thank you.
Sorry I dind't thank you earlier.
I did try it in Ubuntu, but now I installed Debian witht he graphic interface and is working faster than in Ubuntu, plus I had a weird problem in Ubuntu when I switched to static IP it never did switch, it kep using DHCP, I think it was something to do with Gnome or whatever, but was odd, that you can't make a simple change like that in the graphic enviroment. IN Debian works like a charm.
Thank you.
-
Hello I'm trying to find info on How many None snom devices can be used on each snomONE version, and also to know if ATA are in the same bag as none snom, since snom doesn't make any ATA device.
I need to setup several ATA for fax and it can break my deal of using snomONE as the PBX. Which I wont like.
Thank you
-
Hello guys I need some help .
I have a new project that will put the snomOne in a cloud environment (Virtual Server)
I'll go for the 150 version. All remote phones.
Client has many stores, usually 2 m9 Handsets and 1 fax line/ credit card machine ;-)
So what is the best Fax ATA device for the snom One? I was thinking Linksys PAP2. ?
How big of a virtual server do I need for 150 extensions most will have the 2 handsets registered on the same extension.
Fax is what I'm not comfortable about. What are your thoughts?
-
Hello, I'm not very familiar with linux, so I wanted to ask if snomeOne will install under Ubuntu Desktop v10 ?
Also if will it be much difference using server ubuntu than Desktop ubuntu? I know server doesn't have a graphic interface.
Thank you.
-
You can use !([0-9]{10}$)!\1!t!100 or !([0-9]{10}$)!\1!u!100, where 't' refers to the number from the "To" header, 'u' is the number from the "Request-URI". 100 is the default account(say an AA or ACD or Hunt group) if nothing is matched.
Please refer http://kiwi.pbxnsip.com/index.php/Inbound_Calls_on_Trunk for details on how to route the inbound calls.
That one worked out great with the t.
Thanks
-
So in the paid extensions I can only use 5 or 10 at the same time? from 20 to 150 extensions available?
I was hoping I could use all of them and save some bandwidth setting up the PBX in the cloud, since most of the extensions are branch offices.
-
If you set the domain country code to 1, then most of the 10/11 digit dialing issues will be taken care of.
it is set, it was set to 1
My idea was to create a Page with the full DID as name and route each DID to Hunt groups or IVR or extensions depending the needs.
SO what expressiong will you use? for a 10 digit number?
This is what I got:
[5] 2011/03/29 01:28:00: Domain trunk Broadvox@pbx.xxxxx.net could not identify user for +12142223333
-
Hi there, I've tried many ways and I can't make it work with regular expressions.
My number comes in as +1214-222-3333 +1214-223-3333 +1469-333-1234 and so on, Numbers don't match at all.
I guess my problem is the + sign but my provider uses it. And I need to use almost the full number except the 1.
I tried something like this ([0-9]{10}$)\1 and many others but the call fails.
I have many trunks and many DID
s that I'll have to route so it is crucial to have an expression that works for me.
Thank you for the help.
-
I want to use G729 in all my extensions, but it doesn't work.
I'm trying to find info about this.
1- I heard that I need to buy the 729 Licenses, which I'm ok with but I need to know where?
2- Also if I buy the same amount of licenses that the snom one I can handle as extensions would they work?
3- Do I need a 729 License for the Trunk also?
Thank you.
-
Thanks
-
2 Questions in one, (one of them might be really easy for the experienced guys)
Easy one first, Extensions vs Bandwidth.
It is ok to think that a registered extension does not utilize bandwidth, only when the extension comes alive on a call internal or external?
Second question.
My Test PBX is set and I can make outbound calls but I can receive,
My provider tells me the PBX is trying to receive inbound on port TCP and I can see the logs
I went to ADMIN and ports and took out the TCP 5060 just left it blank and reboot the pbx and the system altogether but it still does the same.
What am I missing?
5] 2011/03/22 01:11:31: SIP Tx tcp:xxx.xxx.xxx.xxx:5060:
REGISTER sip:dn01-03.fs.broadvox.net SIP/2.0
Via: SIP/2.0/TCP 192.168.2.202:49158;branch=z9hG4bK-f6e1c3ba33143afb3d6115bbe69c5524;rport
From: <sip:1112223333@dn01-03.fs.broadvox.net>;tag=32567
To: <sip:1112223333@dn01-03.fs.broadvox.net>
Call-ID: r63k9oso@pbx
CSeq: 18585 REGISTER
Max-Forwards: 70
Contact: <sip:1112223333@192.168.2.202:49158;transport=tcp;line=c81e728d>;+sip.instance="<urn:uuid:76e78aa8-1c75-44fb-aa67-df375e927127>"
User-Agent: snom-PBX/2011-4.2.0.3981
Supported: outbound
Authorization: Digest realm="dn01-03.fs.broadvox.net",nonce="2c27b68e-3e38-47d7-ba6c-9ae3c2baf14a",response="05ea6f8c40acc4410eb9212717f407c5",username="1112223333",uri="sip:dn01-03.fs.broadvox.net",qop="auth",nc=00000013,cnonce="e7e63c4a",algorithm=MD5
Expires: 3600
Content-Length: 0
-
Right, for inbound calls you can use this link: http://wiki.snomone.com/index.php?title=Inbounds_Calls
For outbound calls it is essentially on how you service provider or PSTN gateway wants it represented: http://wiki.snomone.com/index.php?title=Outbound_Calls
Thank you for that link, explained it for me.
Thank you for your time.
-
Inbound and outbound are different topics. For inbound, the "alias" is the way to go. For outbound the "ANI" is the important setting. In both cases, you also need to adjust the trunk settings to deal with the operator's (or PSTN gateway) style of dealing with these items.
Ok so I create an Alias for each DID I have but do I need to setup something in the Trunk ? how would the call will go from Trunk to DID to Hunt List to extensions?
I was working with Thirdlane before and I got used to see the Reoutes and the DID's there, so I'm a bit confused now.
-
Hello I'm testing a snomOne pbx right now, and I'm having a hard time figuring out the DID usage.
I get that I can create an Extension with Alias as DID but how can I route Incoming calls dialing to that DID when they are all coming into the same Trunk?
snomOne & Ubuntu
in Linux-related Topics
Posted
Here the link, let me know what U think.