Hi,
after couple of months of SER experience I'm now facing a problem with CLIR type of supplementary service which I'm not able solve by myself, hopefully I'll receive some hints from seruser list readers.
The case is that in pstn connection I have Cisco as5300 gw in use and when calling from pstn side I can omit the calling line id e.g. by typing the prefix #31# just before the b-sub uri in number format --> calleed party (being now sip uac registered in to SER) will see anonymous caller via pstn gw, OK (the same can be monitored from the sip messages coming from gw to ser). But how do the same when calling from sip uac (registered to SER) to pstn ? I have understood that one way is to use Append_rpid_hf (in september posts there were nice descriptions for the usage of it, but I'm not sure if I understood it thoroughly). Before playing with prefix and suffix I just tested how the append_rpid_hf would work and added the line in my ser.cfg...only that when I make call from sip client and monitor the sip messages there is no Remote-Party id line appended to the initial sip invite header ? btw, I'm using radius auth and ser 0.8.11
reg. timlaa
............................................................ Maksuton sähköposti aina käytössä http://luukku.com Kuukausimaksuton MTV3 Internet-liittymä www.mtv3.fi/liittyma
I've been toying with a couple ideas on terminating an in process call. I'm thinking that a BYE instruction send to the FIFO device will do just that. For example:
UA -> SER -> Gateway (call setup worked fine, call in progress).
Then, at a later time, inject a BYE to the UA and to the Gateway for the call ID. If I save a packet, say the initial INVITE, can I simply increment the SEQ, change the METHOD to BYE, and squirt it into the SER FIFO towards the endpoints? Has anyone done this?
Thanks, ---greg
Greg,
Theoretically it is possible, but you must examine every message within that dialog because they increment CSeq. What you need to remember is From, To, Contact, CSeq, and Route set for both sides.
Jan.
On 19-11 10:22, Greg Fausak wrote:
I've been toying with a couple ideas on terminating an in process call. I'm thinking that a BYE instruction send to the FIFO device will do just that. For example:
UA -> SER -> Gateway (call setup worked fine, call in progress).
Then, at a later time, inject a BYE to the UA and to the Gateway for the call ID. If I save a packet, say the initial INVITE, can I simply increment the SEQ, change the METHOD to BYE, and squirt it into the SER FIFO towards the endpoints? Has anyone done this?
Thanks, ---greg
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Indeed. A phone from an unnamed vendor likes sending in-dialog OPTIONs, others love to send INFO,and so on and so forth. If you don't watch for follow-up requests, you may end up running out of sequence or racing with parallel requests. Eventually, the recepient will report out-of-order requests and deny BYEs.
So to get it right, you will end up with a call stateful server. We have such, I will be happy to take it off-line if you are interested.
-jiri
At 05:36 PM 11/19/2003, Jan Janak wrote:
Greg,
Theoretically it is possible, but you must examine every message within that dialog because they increment CSeq. What you need to remember is From, To, Contact, CSeq, and Route set for both sides.
Jan.
On 19-11 10:22, Greg Fausak wrote:
I've been toying with a couple ideas on terminating an in process call. I'm thinking that a BYE instruction send to the FIFO device will do just that. For example:
UA -> SER -> Gateway (call setup worked fine, call in progress).
Then, at a later time, inject a BYE to the UA and to the Gateway for the call ID. If I save a packet, say the initial INVITE, can I simply increment the SEQ, change the METHOD to BYE, and squirt it into the SER FIFO towards the endpoints? Has anyone done this?
Thanks, ---greg
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Jiri Kuthan http://iptel.org/~jiri/
Do you have Sip-Rpid attribute in the radius database as described in the radius howto:
http://iptel.org/ser/doc/ser_radius/ser_radius.html#AEN165
Did you authenticate the message before you call append_rpid_hf ?
If you do not authenticate the message, or there is no Sip-Rpid attribute in the radius then the header field will be not appended.
Jan.
On 19-11 14:48, timlaa@luukku.com wrote:
Hi,
after couple of months of SER experience I'm now facing a problem with CLIR type of supplementary service which I'm not able solve by myself, hopefully I'll receive some hints from seruser list readers.
The case is that in pstn connection I have Cisco as5300 gw in use and when calling from pstn side I can omit the calling line id e.g. by typing the prefix #31# just before the b-sub uri in number format --> calleed party (being now sip uac registered in to SER) will see anonymous caller via pstn gw, OK (the same can be monitored from the sip messages coming from gw to ser). But how do the same when calling from sip uac (registered to SER) to pstn ? I have understood that one way is to use Append_rpid_hf (in september posts there were nice descriptions for the usage of it, but I'm not sure if I understood it thoroughly). Before playing with prefix and suffix I just tested how the append_rpid_hf would work and added the line in my ser.cfg...only that when I make call from sip client and monitor the sip messages there is no Remote-Party id line appended to the initial sip invite header ? btw, I'm using radius auth and ser 0.8.11
reg. timlaa
............................................................ Maksuton sähköposti aina käytössä http://luukku.com Kuukausimaksuton MTV3 Internet-liittymä www.mtv3.fi/liittyma
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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Greg Fausak -
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Jiri Kuthan -
timlaa＠luukku.com