Topic: SAIL Caller ID (CallerID, CLID)

[sonoracomm]

Hi folks,

Has anyone had any trouble setting outbound CLID on a VOIP trunk?  Is there a trick to it?

I have set the Outbound Callerid: field and I checked with Telasip to make sure they had 'unlocked' the caller ID so we can specify it.

Is there any documentation or are there any notes anywhere on this?

Thanks,

G

[SARK devs]

Hi

The later releases of SAIL allow you to set CLID on otbound trunks.  Earlier versions set the CLID based upon the Trunk DiD.

Either way, whether the carrier accepts and honours it or not varies from carrier to carrier and from country to country depending upon your national laws.  

SAIL will put any CLI you like onto the line.  What arrives at the other end may or may not be the same thing. You will need to check with your carrier and the telecoms regulator in your country.



Regards

Selintra

[sonoracomm]

Solved.

I needed to do two things to make CLID work for me.

1) For Telasip, I had to add this to the Telasip peer stanza:

Code: [Select]

sendrpid=yes
I also edited the

2) I had to specify a phone number, as opposed to a (alpha) name, in the 'Outbound Callerid' field.

My Telasip Carrier definition now looks like this:

Host URL: sip.sfo.telasip.com
Registration Template (Optional): username:password@sip.sfo.telasip.com

[peer]
type=peer
host=
qualify=3000
canreinvite=no
username=
fromuser=
secret=
insecure=very
sendrpid=yes

[SARK devs]

Good work and thanks for the info

I'm glad you got it working OK.

Kind Regards

J

[del]

Hi sonoracomm,

My Telasip is like this:

Host Url:gw4.telasip.com

Yours is:

Host URL: sip.sfo.telasip.com

Do I need to change it to try the CLID or should I just add:

sendrpid=yes

to the Peer.
Thanks.

Del

[sonoracomm]

Hi Del,

Telasip is moving from statically assigned gateways to a more dynamic method as they expand their coverage.  I forgot exactly what they explained to me, but no, it is not necessary to use the San Francisco gateway.  You might ask them about this issue...

Just use the sendrpid line.

Also, you have to request that they unlock the callerID or you will have no control.

Speaking of control, I'd like to be able to callerID name as well as the callerID number.  Is that possible?

G

[SARK devs]

I'd like to be able to callerID name as well as the callerID number. Is that possible?

Short answer is - don't know.  CLID is a bit of a mess from a standards point of view.  If you want to learn more about it you can have a look here...

http://www.ainslie.org.uk/callerid/cli_faq.htm

CallerId NAME seems maybe to be more prevalent in Bellcore networks than anywhere else.  In the UK, we are fortunate in that BT openly publishes exactly how its network operates in a series of Supplier Information Notes or, rather appropriately, SINs.    According to BT SIN 227 there is provision to send a name in the BT CLIP (BT speak - stands for Calling Line Identity Presentation) but I've never seen or heard of it being done and I'm not sure if any UK CPE (Customer Premises Equipment) would pick it up or whether zaptel has support for it.  You can reference the BT SIN library here....

http://www.sinet.bt.com/

if you are as sad as we are at Selintra, you may even enjoy reading what you find...

Kind Regards

Selintra

[sonoracomm]

Hey Selintra,

Thanks for getting back to me so quickly.

Does the selintra AGI implement this function?

Code: [Select]

CALLERID(name)
Here is a URL to some info...

http://www.asteriskguru.com/tutorials/calleridname_function.html

Here in the US, it seems that the CallerID NAME function seems to mostly work.  I think it gets less support between RBOCs (regional bell operating companies - also known as fiefdoms) and wireless carriers.

In the end, I don't even know if it would work properly even IF SAIL supported it...I just wanted to try.

G

[SARK devs]

Does the selintra AGI implement this function?
Code:
CALLERID(name)

Don't know without looking at the code but I don't think so (I haven't got access to svn where I am).  Wouldn't be more than a few minutes work to put in tho'.  Why not try a Dial statement in a custom app first to see if it works?

try something like this....  

Use a number you own (to see if it works) - I've used the Selintra office number just to demonstrate, - and make sure you've set the callerid field in the trunk you are testing.

exten=>01924918076,1,Set(CALLERID(name)=name)
exten=>01924918076,2,agi(selintra,OutTrunk,trunkname-you-want-to-test)
exten=>01924918076,3,Hangup



Best

Selintra

[sonoracomm]

OK,

I've gotten back to this and done some more testing.

I've gotten CLID (numeric) working on my system.

I added a custom app as above, substituting my cell number.

From Asterisk extension 4101, I dial the cell number while watching the Asterisk console in a very verbose manner.

Asterisk does _not_ dial (all-busy (?) tones) and the console shows this output:

Code: [Select]

   -- Executing AGI("SIP/4101-0a147548", "selintra|OutCluster|5206611342") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing Set("SIP/4101-0a147548", "CALLERID(name)=Sonora Comm") in new stack
    -- Executing AGI("SIP/4101-0a147548", "selintra|OutTrunk|Telasip") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing Hangup("SIP/4101-0a147548", "") in new stack
  == Spawn extension (default, 5206611342, 3) exited non-zero on 'SIP/4101-0a147548'
    -- Executing Hangup("SIP/4101-0a147548", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/4101-0a147548'
Without documentation of the AGI script, I don't know why it didn't dial.  It says "completed, returning 0"...

Any clues?  

Thanks much,

G
p.s. I also tested dialing out on a land line to the mobile and the caller ID passes both the number and the name to the mobile phone.

[SARK devs]

You can watch what the agi is doing by turning on agi debugging....

agi debug

turn it off again by doing

agi no debug

Kind Regards

Selintra

[del]

Hi All,

Sorry about reviving this thread but I thought it would save starting another and repeating some of the stuff  Any how here is my question, I use voipdiscount.com to make outgoing calls and it sends a different number each time I use it! I was wondering if I could make it send my name instead using a custom app? Any input is welcome.

Regards,
Del

[hervep]

Hi All,

Sorry about reviving this thread but I thought it would save starting another and repeating some of the stuff  Any how here is my question, I use voipdiscount.com to make outgoing calls and it sends a different number each time I use it! I was wondering if I could make it send my name instead using a custom app? Any input is welcome.

Regards,
Del

Hi Del,

I don't know voipdiscount, but I assume they are using standard gateways to relay your sip calls to the PSTN. As a result they send the Calling Line Identity of the PSTN line you 'randomly' use to make your call. PSTN does not support 'calling name' as a standard feature, only few PSTN protocols are supporting this kind of extension.
Most of the time, it is technically possible to relay a number as CLI, again it depends of the provider and local legal rules : if it is open, you can make calls using the 'identity' of someone else ... that's the point !.

Best,

Hervé