Dose Asterisk 1.4 support sip-isup

recently i met a big truble.we have a asterisk (1.4.21 version ) which supply CallingCard service,out bound calls were routed to a Partiner’s GW.Call could be terminated successfully.but the ANi could not be transmitted,i mean,the called person could get the Original caller phone number.
the partiner said,just because my SIp server (Asterisk) did not send the Nature of caller number(such as :international ,national or customer).that should be ISUP parameter.i dont know if Asterisk can send those kind of Nation to outboud GW through SIP messenge.
thanks for any suggestions you can give.i am wating ,thank you …

It looks to me as though the simplest method is to use + format numbers (like full international format numbers on mobile phones).

At least for 1.6, type of number is available from CALLERID function.

Also 1.6 has a usereqphone sip.conf flag which will cause numbers not prefixed with + to be marked as national/network format (although the RFC doesn’t seem to allow the two sub-cases to be distinguished).

Thanks david55,i will have a try.then i will tell you the result.

hi,devid55.just now ,i use Asterisk 1.6 to do test of function Callerid,but failed or bad testing method.following as log,pls help to give some suggestion. thank you.

86200001@Server-A -> Called ->8611111111@Server-B
Asterisk CLI verbose
– Executing [8611111111@from_test:1] NoOp(“SIP/86200001-09f65d48”, “”) in new stack
– Executing [8611111111@from_test:2] Set(“SIP/86200001-09f65d48”, “CALLERID(TON)=1”) in new stack
– Executing [8611111111@from_test:3] Dial(“SIP/86200001-09f65d48”, “SIP/8611111111@Server-B,30”) in new stack
Sip invite message from Server-A to Server-B.
i think if A want to tell B about N(umber)O(f)T(ype),it should be in the Invite package.
any suggestion ??

U 2010/08/19 10:24:28.407450 Server-A:5060 -> Server-B:5060
INVITE sip:8611111111@Server-B SIP/2.0
Via: SIP/2.0/UDP Server-A:5060;branch=z9hG4bK685a673e;rport
Max-Forwards: 70
From: “86200001” sip:86200001@Server-A;tag=as2a81c75a
To: sip:8611111111@Server-B
Contact: sip:86200001@Server-A
Call-ID: 2be4f67c781b20eb75f0a5dd034c21ff@Server-A
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 19 Aug 2010 10:24:28 GMT
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 312

o=root 391336112 391336112 IN IP4 Server-A
s=Asterisk PBX
c=IN IP4 Server-A
t=0 0
m=audio 13242 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

When I said + format, I actually meant include the +. RFC 3398 says that you need to end up with either sip:+86…@domain, or tel:+86…

I don’t know if Asterisk can generate tel: URI’s.

I don’t believe ton is ever used for outbound SIP calls. I have a feeling that even for outbound ISDN you have to hard code the type of number against the channel. I think it is only meaningful for inbound ISDN calls.

Given your name, I’m assuming the 86 is the country code for 中国, not an area code。