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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: domainwizard on February 03, 2008, 04:48:59 PM
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The following in the login info for the us carrier vitelity, could i get some assistance in how to configure a trunk to properly utilizing their [inbound] and [outbound] needs, i put the unique user login, password and number in yellow
thx
Please add the following configuration to your /etc/asterisk/sip.conf
Add the following at the bottom of your sip.conf
[vitel-inbound]
type=friend
host=inbound5.vitelity.net
context=inbound
username=xxxxxxxx
secret=xxxxxxxx
allow=all
insecure=very
[vitel-outbound]
type=friend
host=outbound1.vitelity.net
context=outbound
username=xxxxxxxx
fromuser=xxxxxxxx
trustrpid=yes
sendrpid=yes
secret=xxxxxxxxx
allow=all
Please add the following configuration to your /etc/asterisk/extensions.conf
Add the following to the bottom of your extensions.conf
[outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound)
exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound)
; e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911
exten => _911,1,Dial(SIP/911@vitel-outbound)
[inbound]
exten => NXXNXXXXXX,1,Answer
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Hi,
You can create a new carrier in SAIL and plug the stanza information into the user and peer entries. (you don't need the user stuff that they've specified but it won't hurt to plug it in anyway).
Instructions how to create a carrier here
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter259
I notice they don't require registration (or you don't mention it) so they must have some sort of web-panel or manual registration requirement to find you. Their inbound extensions.conf entry is also rather stupid. Ignore what they've suggested and specify an absolute DiD otherwise it will play hell with any other carriers you might have set up. Unless you really know what you are doing, you should never specify wild card dialplans on inbound.
Kind Regards
S
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i am thinkin i misunderstand or somehow do not get it when the carrier requires a separate inbound and outbound trunk..
on a totally different not, is it possible to block CID on outbound calls with *67?
b
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i am thinkin i misunderstand or somehow do not get it when the carrier requires a separate inbound and outbound trunk..
SIP trunks are logical constructs. They don't necessarily behave like a regular pots line. In general, you need a SIP trunk to make an outbound call. With asterisk you can receive an inbound SIP call with no "trunk" at all. However, the carrier/caller has to know where you are in order to send you a call in the first place, either through registration or some kind of set-up routine (usually browser based).
*67 is for US domestic PSTN lines AFAIK. You would have to speak to your VoIP carrier to see if they honour it or not.
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I am trying to setup vitelity at the moment - I have about 10 DID's with them, all under the same user.
Vitelity does require registration - in the past with another product I would register using:
username:secret@inbound5.vitelity.net:5060
Outbound goes via outbound1.vitelity.net as the original poster has outlined.
When I set them up as a carrier, do I provide the registration info there -- or leave it to the trunkline? I presume with the trunkline, setup one for each DID with the DID specified in the DID Number SIP/IAX Name field to allow for internal routing of each DID?
Would be interested in finding out how to best handle vitelity. Apologies if I've overlooked anything.
The vitelity DID's and outbound calling is the last carrier I have to transfer to SAIL. The other carriers have worked out smoothly.
Thanks!
Steve
Hi,
You can create a new carrier in SAIL and plug the stanza information into the user and peer entries. (you don't need the user stuff that they've specified but it won't hurt to plug it in anyway).
Instructions how to create a carrier here
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter259
I notice they don't require registration (or you don't mention it) so they must have some sort of web-panel or manual registration requirement to find you. Their inbound extensions.conf entry is also rather stupid. Ignore what they've suggested and specify an absolute DiD otherwise it will play hell with any other carriers you might have set up. Unless you really know what you are doing, you should never specify wild card dialplans on inbound.
Kind Regards
S
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Hi Steve-o
As we discussed above, Asterisk will pretty much accept a sip call from anywhere provided you have an "extension" in extensions.conf to catch it.
Registration is pretty much exactly as you decribe for Tbox, or whatever you were using before. For this carrier, you can just use the "generalSIP" predefined carrier (already in SAIL). It is already set up to do most of what you need.
You only need register a carrier once, so you only need to create one carrier trunk, no matter how many DiDs you have. To create the DiDs just use the PTT_DiD_Group trunk. This won't actually create a trunk, it will just create entries in the correct context within extensions.conf. However, you already told the carrier where you were when you registered the one carrier trunk, so there isn't any need to register any more. Moreover, if your DiDs are consecutive, you can create them all at once with the PTT_DiD_Group trunk.
When you create your trunk (specifying generalSIP), SAIL will ask you for 5 items... Here's how we would fill it out
Trunk Name: Vitelity1
DiD Number: Your Vitelity UID
Carrier's URL/IP Address: inbound5.vitelity.net
Username : your Vitelity UID
Password : your Vitelity pwd
Hit Commit.
Now go edit your new trunk..
SAIL will already have filled out your regsitration correctly for you. Don't change what it has done.
Next; in the left hand window... the only two things you might want to add to what SAIL has already generated for you are...
trustrpid=yes
sendrpid=yes
I'm not sure that the sendrpid and trustrpid values are necessary but you can try it with and without. DO NOT be tempted to add a "context= " statement. SAIL has already got this covered and you will screw SAIL up if you include your own.
You shouldn't need anything at all in the right-hand window so you can just leave it empty (depite what the carrier says in her literature).
Finally, create your DiD trunks (using PTT_DiD_Group) and route them to wherever you want each call to end up.
That should about do it. You don't need to add anything to extensions.conf, SAIL will take care of that for you.
Kind Regards
S