Author Topic: calling a SIP URI  (Read 11682 times)

valentin_nils

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Re: calling a SIP URI
« Reply #15 on: January 02, 2011, 05:14:32 AM »
dicodread,

I like your ideas and input , but I think this is a complete different topic  from OP.
You want to discuss about some dial plan logic.
The arguments you give are all sound.

Can you open please your own topic. with it ?
Otherwise new readers will not find the topic.
In addition I encourage you to bring any good ideas forward.
(its much appreciated)

1) either by directly contacting Grandstream and giving feedback or
2) posting in my feature wishlist topic. or the "Vote for the GXV3140" Poll.

Thanks in advance.
« Last Edit: January 02, 2011, 05:22:49 AM by valentin_nils »
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dicodread

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Re: calling a SIP URI
« Reply #16 on: January 02, 2011, 06:16:18 AM »
Thanks but I feel you are missing the main point I am making. I have been with callcentric since they started in 2006 so I know all about their peering capabilities via sip broker and their phone book sip entries called by *75XX.
My main point is if I try to call a sip uri that isnt part of the list in sipbroker or one that I dont want to add as a phone book entry in callcentric, I cant.

And yes even the cheap 40.00 linksys pap2 ata can do what I am asking.

valentin_nils

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Re: calling a SIP URI
« Reply #17 on: January 02, 2011, 09:59:51 AM »
dicodread,

I got your point, that is exactly the reason why I suggest to open  a new topic and post it to Grandstream.
If you want your voice to be heard, than you should raise it visibly.
If you just post  as one reply of many in a related but different topic (thats what you do right now) than changes that Grandstream evt. find it are small.

The reason why I made the suggestion is because our involvement within this forum started when we discovered that it took 18 months for Grandstream to get where we are now. Now you can wait another 18 months and hope that they "might" integrate something that is posted as a reply (out of many)  somewhere in the forum OR you actually bring it to their attention and allow them to act on it. This is especially true if you submit a support ticket. This way you get a reply from them and know if there is the possibility to implement what you are asking.



« Last Edit: January 02, 2011, 08:53:01 PM by valentin_nils »
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TheFug

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Re: calling a SIP URI
« Reply #18 on: January 03, 2011, 10:08:05 AM »
Thanks but I feel you are missing the main point I am making. I have been with callcentric since they started in 2006 so I know all about their peering capabilities via sip broker and their phone book sip entries called by *75XX.
My main point is if I try to call a sip uri that isnt part of the list in sipbroker or one that I dont want to add as a phone book entry in callcentric, I cant.

And yes even the cheap 40.00 linksys pap2 ata can do what I am asking.
My guess is, this is a dialplan syntax problem, also that it's  too limited now, to define this in the dialplan (options)

dicodread

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Re: calling a SIP URI
« Reply #19 on: January 03, 2011, 03:18:08 PM »
That could be a start.
When the dial plan allows for the "@" symbol to be processed we will be good.

valentin_nils

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Re: calling a SIP URI
« Reply #20 on: January 04, 2011, 04:43:26 AM »
dicodread, why dont you create your own dialplan ?
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TheFug

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Re: calling a SIP URI
« Reply #21 on: January 04, 2011, 09:37:43 AM »
dicodread, why dont you create your own dialplan ?
Is this possible with the GXV3140 ? I see only limited options in the FAQ or manual(s) There should also be a kind of priority setting be possible.
Linksys allows gw1, gw2, options in dialplan, (for selecting a voip provider account) line1, line2, etc.... could be a solution for the GXV3140.

valentin_nils

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Re: calling a SIP URI
« Reply #22 on: January 04, 2011, 04:31:36 PM »
my understanding is that dicodraed can not use the "@" mark.
He should get " The number you dialed doesnt match dial plan" error message.

So he just needs to modify the regular expression in the web interface "Account / Call Settings / DialPlan" accordingly
However I dont know how the peering/ call charging itself will be done by provider.
« Last Edit: January 04, 2011, 04:33:45 PM by valentin_nils »
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TheFug

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Re: calling a SIP URI
« Reply #23 on: January 05, 2011, 12:06:58 PM »
So this could be the same syntax as used for a SPA3102 ATA ?
what I understand out of the guides, FAQ, or manual, this is not the case.
Also, to complete a phone-book entry one has to select an account to dial de person/entry.... so the sip-uri will be dialed with an account ?
Also, phone-book entries can only be added at the phone, and not via the http inlog of the phone, or i just couldn't find it......

valentin_nils

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Re: calling a SIP URI
« Reply #24 on: January 05, 2011, 03:25:02 PM »
TheFug My understanding is that you always have to choose an account for making a call.

If you choose the DirectIP call application than I assume your partner will only see "Anonymous".
As far as I can see its only good for testing on internal networks .
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TheFug

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Re: calling a SIP URI
« Reply #25 on: January 06, 2011, 10:51:36 AM »
Thanks. well the whole sip-uri calling will be for most phones a problem, in the past i was also trying this, outgoing is a less problem, but incomming is the hard part,
You need a static IP address from your ISP,(or need a [external] DNS service) and this isn't normal in most cases.
Most of the time a firewall will block it, and the router at the receiving end should respond correctly, and then also for the xxxxx@  part, So i gave up on it at a certain moment, I guess not many people will use it, also because it is hard to set it up, for a normal user. (having a voip server would help, but is too much, for a feature like this)

JennyMay

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Re: calling a SIP URI
« Reply #26 on: May 04, 2011, 10:23:00 PM »
Hello everyone, I am new in this forum and I want to know more about  Toll Free Numbers and also how to install SIP URI.I hope I can find someone who can I ask questions too.


Have a nice day ahead.

TheFug

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Re: calling a SIP URI
« Reply #27 on: May 05, 2011, 09:22:55 AM »
You can forget about sip-uri calling for the 3140, it will not work, in general it is hard to setup anyway, this is also the reason why not many people use it, better use something like Sipbroker.
If you register for a free Voxalot account, you can dial the free 0018xxxx numbers by replacing the zero┬┤s for a asterisk symbol.
Some Voipproviders also allow you to make free 800 calls, but you need to do this by ways of making a P2P call, how or if this is allowed, you should check with their support.

xiaochun3612

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Re: calling a SIP URI
« Reply #28 on: June 07, 2011, 02:01:50 AM »
The settings look o.k to me. Lets give it a try when you are ready.

Just call me via skype or Ekiga.