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Topic: Two simple questions
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Hi all,
I have come across two questions: 1.If I use the Test Agent,how do I create a re-Invite request? 2.If I use a soft-phone client,how could I make a phone via dialing telephone number?And I find that I can't add a new Tel Key in DNS, if I need a third test user, what can I do? Thanks for solving my problem. /Nick | |||||||
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Hi Nick,
1. re-INVITE: You have to create it manually in TA 2. You have to set the tel URI in the HSS for the user you want to call /Philippe | |||||||
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Hi Philippe,
Thanks for your reply. But what is the "Tel Keys" in DNS?I can't add new user there.It does not work,if I just add a tel URI in HSS.The request To header through X-Lite sent to Server is "100001 <sip: 100001@ericsson.com>" rather than "Alice <sip: alice@ericsson.com>". Or do you means that I should change the user's name to Number? /Nick | |||||||
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Hi Nick,
You also need to create the DNS entry to associate the tel uri to the sip uri /Philippe | |||||||
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Hi Philippe,
I also think that I should add something in DNS,but I can't. When I choose the "Tel Keys" tab,click "Add" button,but nothing happens. And in the fact,in the "Tel Keys" tab,I have set the URI. e.g. ---URI -------------------------------------DNS Record ---URI:tel:+8602810001-------------------Record Type:SIP URI ---Country Code:86------------------------SIP URI: sip:alice@ericsson.com ---Area Code:028 ---Phone: 10001 And I also set the Alice's "User Profile" in HSS,"Tel URI" as "tel:+8602810001". The similar setting as Bob. But if I dial the number 8602810001 on the X-Lite as Bob,I got the Response from server "Request Timeout".And the CSCF capture the Invite request as below [SDS] [INFO ] <-- Sent message on UDP [Local: 0.0.0.0:5082 | Remote: 127.0.0.1:5060] [[[ INVITE sip:8602810001@ericsson.com SIP/2.0 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO CSeq: 1 INVITE Content-Length: 515 Record-Route: <sip:127.0.0.1:5082;from-tag=bd35ec5a;lr> Contact: <sip:bob@127.0.0.1:1112> User-Agent: X-Lite release 1011a stamp 39982 P-Asserted-Identity: <sip:bob@ericsson.com>,<tel:+8602810002> Route: <sip:test010.ericsson.com;lr;call=orig>,<sip:127.0.0.1:5082;lr;original-dialog-id=z9hG4bKbc319e1bac85f1cee28b859aaa83c9fa-INVITE> To: "8602810001"<sip:8602810001@ericsson.com> From: "bob"<sip:bob@ericsson.com>;tag=bd35ec5a Call-ID: ZWMyNDgyYzdmY2IxYmUzZWViMDU0NmJhYTZiZmNkMjM. Content-Type: application/sdp<br>Via: SIP/2.0/udp 127.0.0.1:5082;branch=z9hG4bKbc319e1bac85f1cee28b859aaa83c9fa,SIP/2.0/UDP 127.0.0.1:1112;rport=1112;branch=z9hG4bK-d87543-2b73c31b71766a51-1--d87543-;received_port_ext=5081;received=127.0.0.1 Here ignoring Bob's SDP So,I have met two problems. 1.I can not set/add "Tel Keys" tab in DNS for a new User(e.g.Coco) 2.I have set the "Tel Keys" in DNS for "Bob" and "Alice",but I still can not get through "Bob" to "Alice" with dialling Alice's phone number by X-Lite regitered with "Bob". Thanks. /Nick | |||||||
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Hi Nick,
This is a bug. For now, I cannot give you a resolution date /Philippe | |||||||
如果不能实现使用数字拨号,那还搞什么呢... 不过如果用户名就是数字ID...还是可以继续... Ah,faint! |