Topic: [Announce] Selintra-sail-2.1.13-256

[del]

Hi All,

Well I have now successfully installed asterisk, I can make internal ext to ext calls and can even dial out on sipdiscount, so far so good  
I would still like to go further if possible, so can anyone help/point me in the right direction to the following: can I get asterisk extensions to automatically dial 001 or 0044 so that the numbers can be dialed as normal (without the international codes), can someone tell me of a good (preferably lowcost) place to get a DiD number to point at my sipdiscount account (or any sip account) and finally (I think) can the Digium X100P be used for attaching an analogue device , such as a fax or is it only for use with an analogue PSTN line? Thanks again for all the help.

Regards,
Del

[SARK devs]

(can) I get asterisk extensions to automatically dial 001 or 0044 so that the numbers can be dialed as normal (without the international codes),

Yes you use masks in the trunk definition.  It depends what your carrier requires in the way of numbers but most will accept E164 format (which is country-code+area-code+subscriber number).  To set up a trunk to send E164 - put the following mask in  the trunk
 

00: 0:44

Then just dial all your numbers as you would from a normal domestic phone.

 

can someone tell me of a good (preferably lowcost) place to get a DiD number to point at my sipdiscount account (or any sip account)

Lots to choose from.  Just have a google around looking for companies that will do inbound numbers.  In the UK you can try Telappliant (www.voiptalk.org) or Gradwell (www.gradwell.org) or even (soon) www.selintra.com - we've just signed a deal with one of the Tier 2 carriers to enable us to offer a range of geographic and non-geographic numbering services ourselves (sorry for the ad - but you did ask).

 

can the Digium X100P be used for attaching an analogue device , such as a fax or is it only for use with an analogue PSTN line?

PSTN Line only.  The X100P can operate only as an FXO device (at least as far as asterisk and the zaptel drivers are concerned).  For FAX you need an FXS termination, either using an ATA, like the sipura/linksys devices or using a TDM400 board with one or more FXS modules fitted.


Kind Regards
[/quote]

[del]

Hi Selintra,
Thanks for the reply, as I am located in the US am I right in thinking that

00: 0:44 should be 00: 0:1

Also could you point me in to the place to do this.
Thanks again.

Regards,
Del

[del]

Hi Selintra,

Another look in the server-manager panel reveals a box titled: Transformation Mask I assume that this where you enter the 00: 0:1
I will give it a try anyhow.

Regards,
Del

[SARK devs]

Hi Del,

No - it's different in the states.   Tell me what you're trying to do.  Is it to do with the requirements of the carrier?

Best

Jeff

[del]

Hi Jeff,

I am trying to use sipdiscount for my UK and US calls. At the moment I have to dial 00 plus country code for all calls, even local calls need the full international and area code. I would like to be able to dial local numbers without it and all other US numbers if possible.  
Example: Say my area code is 407,  I currently have to dial 001407xxxxxxx for local calls, with a normal telco I would not need the 001 or the 407, if I dial another area code say 286 I would need to dial 001286xxxxxxx, with a normal telco it would be 1286xxxxxxx, is it possible for me to dial like a normal telco? I understand that I have to dial the full code for the UK.
Thanks in advance.

Regards,
Del

[JonB]

Del,

It is a requirement of sipdiscount that you use dial 00+countrycode+areacode etc.

For local calls you would need to use a transformation mask

407:001407

i.e when you dial 407XXXXX the dial string sent to sipdiscount is actually 001407xxxxxxx

This would however get really messy as you would have to create transformation masks for every area code so you could create another transformation mask

1:001

so if you dialed 1286xxxxx the string sent to sipdiscount will be 001286xxxxx


Jon

[del]

Hi JonB,

Thanks for the info, I now realise that it is not practical to enter a mask for every US area code, but the 407:001 would make dialing locally a bit easier. Can I enter more than 1 mask in the box? If so how would I seperate them in the box. What I am thinking is I could just put in 9:00 that would be in effect the same as dialing 9 for an outside line if I am thinking correctly, then dialing other areas would be the same ie. (9 for a line) and 1-286xxxxxxx Anyhow I  am gratefull for your input. I will try out a few things and see what works.

Regards,
Del

[JonB]

Del,

1 - the transform mask needs to be 407:001407 i.e 407 is replaced in the dial string by 001407

2 - yes you can have more than one transform mask seperated by a white space. The documentation explains this

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter096#Transformation_mask

You could also use a transform mask of :001  i.e 001 will be prepended or added to any number you dial.

Jon

[del]

Hi JonB,

I have put in this mask 407:001407 but I still can't dial out using just 407xxxxxxx I have even put in _407 in the routes but still no good. I have read the docs as per your link and I can't seem to see what I am missing. It is getting late here so I will try again tomorrow. thanks again for your help.

Regards,
Del

[del]

Hi All,

I have given up with the mask thing for the time being, it is no hardship to dial the full number. I am however struggling to setup to receive calls, I have a DiD number from stanaphone, I have setup stanaphone as a carrier and trunkline. If someone dials my DiD it doesn't ing on the asterisk. A search on google seems to point to me having to do something in the sip.conf file. Can someone confirm that this is the place to be looking? Thanks.

Regards,
del

[SARK devs]

Hello Del

Here is a mask for you...

:001 0019:001407

You will have to dial "9" for locals but it's better than dialing it all I think.

A normal US national number of say 513+662+2300 (don't dial the leading 1 - just dial it as shown) will be transformed by the mask to 001+513+662+2300 (my old number in Cincinnati - ).

For locals you will have to dial 9.  So 9+618+2345 will be transformed twice by the mask - first to 0019+618+2345 then to 001407+618+2345.

This works fine here - see how you get on with it.

Now as to stanaphone - I need to see what is happening when you dial inbound.  First tho' I will take a look at the Stanaphone site to see how they are set-up.

Be back with you shortly.

Kind Regards

Selintra

[fpausp]

Hi Selintra,

A few days ago i installed sail pbx (Version: 2.1.13-261) on my sme7, i use it with a software client Firefly with iax. Everything is working i can dial in and out from/to my lan ...

Now i like to use a isdn card to dial to and from the isdn-network but i have problems to understand the technical english doc, sorry.

Today i got my isdn card (Acer ISDN 128 Surf PCI), the card was identified well. Could you be so kind and tell me a few steps.

regards
fpausp

[SARK devs]

Hello fpausp,

I am pretty sure your card uses HFC PCI so it should be OK but you should upgrade to SAIL-2.1.13-272 and the new anabri rpms.  

Here are the ones you need...

anabri-asterisk-1.2.9.1-3
anabri-asterisk-zappri-1.2.6-3

available from here

http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/AsteriskForSail-2.1.13/ISDNHFC/

selintra-sail-2.1.13-272

available from here

 ftp://81.149.154.14/CurrentStableRelease/

Remove your existing rpms with rpm -e and install these instead  (you should not lose your phone or trunk definitions).

Then follow the steps here to recognise your card...

http://81.149.154.14/docs/cgi-bin/view/Main/SysVersion2Release1Issue13

Create a route for your ISDN line and now you should be able to make and receive calls.

Kind Regards

Selintra

[del]

Hi Selintra,

Thanks for the mask, I will try it and see. As for stanaphone, please don't spend too much time as I am only using them for testing purposes   , I really want to get a local number (their's are New York codes) but don't want to pay until I am sure I can get it working   I have not found a local DiD yet. Perhaps you could point me to the general area of setting a sip line for receiving incoming calls?   Thanks for all your help.

Regards,
Del