Hi all.
I'm trying to get my call working working and I have a question.
Suppose a SIP subscriber has voicemail and call forwarding is enabled.
When an incoming call is destined for this SIP user, SER will reply with a 302 redirect message.
Next the caller's phone should construct a brand new INVITE to the redirected destination.
My question is this:
If the forwarded destination is busy, then the call should be sent to the voicemail of the SIP subscriber that was originally called. This is the way the RBOCs call forwarding works.
If I'm using sl_send_reply("302", "Temporarily Moved") back to the caller, how can I make the voicemail piece work?
Regards, Paul
Why wouldn't you just rewrite the hostname and port to be that of the voicemail server in your "302 failure route handling logic" and relay the call?
Java Rockx wrote:
Hi all.
I'm trying to get my call working working and I have a question.
Suppose a SIP subscriber has voicemail and call forwarding is enabled.
When an incoming call is destined for this SIP user, SER will reply with a 302 redirect message.
Next the caller's phone should construct a brand new INVITE to the redirected destination.
My question is this:
If the forwarded destination is busy, then the call should be sent to the voicemail of the SIP subscriber that was originally called. This is the way the RBOCs call forwarding works.
If I'm using sl_send_reply("302", "Temporarily Moved") back to the caller, how can I make the voicemail piece work?
Regards, Paul
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Thanks Steve.
The only reason I was using 3xx responses was because I believed that was the correct way to handle call forwarding.
The downside of just relaying the message after rewriting the hostname/port is that validation must be done on the forwarded destination.
For example, if a SIP subscriber __does not__ have the free-pstn ACL and they forward to a PSTN number, they could circumvent the ACL and make PSTN calls by dialing their own number.
sl_send_reply("302", "Redirect") causes a whole new INVITE to be processed in the normal fashion.
I suppose if there is no way to failover to voicemail then I'll have to validate forwarded numbers.
Another problem with sending out an INVITE is realized when dealing with PSTN gateways. A nasty situation creeps up when a SIP phone is forwared to a PSTN destination. Then when an incoming call from some other PSTN phone to the SIP phone occurs, SER would try to send an INVITE back out the PSTN to the forwarded destination and many PSTN gateways will reply with "482 Loop Detected".
Regards, Paul
On Mon, 28 Feb 2005 11:29:37 -0500, Steve Blair blairs@isc.upenn.edu wrote:
Why wouldn't you just rewrite the hostname and port to be that of the voicemail server in your "302 failure route handling logic" and relay the call?
Java Rockx wrote:
Hi all.
I'm trying to get my call working working and I have a question.
Suppose a SIP subscriber has voicemail and call forwarding is enabled.
When an incoming call is destined for this SIP user, SER will reply with a 302 redirect message.
Next the caller's phone should construct a brand new INVITE to the redirected destination.
My question is this:
If the forwarded destination is busy, then the call should be sent to the voicemail of the SIP subscriber that was originally called. This is the way the RBOCs call forwarding works.
If I'm using sl_send_reply("302", "Temporarily Moved") back to the caller, how can I make the voicemail piece work?
Regards, Paul
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
--
ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
Java Rockx wrote:
Thanks Steve.
The only reason I was using 3xx responses was because I believed that was the correct way to handle call forwarding.
The downside of just relaying the message after rewriting the hostname/port is that validation must be done on the forwarded destination.
For example, if a SIP subscriber __does not__ have the free-pstn ACL and they forward to a PSTN number, they could circumvent the ACL and make PSTN calls by dialing their own number.
Ok. In my case the failure_route doing the 302 handling is only hit for voicemail.
sl_send_reply("302", "Redirect") causes a whole new INVITE to be processed in the normal fashion.
I suppose if there is no way to failover to voicemail then I'll have to validate forwarded numbers.
Another problem with sending out an INVITE is realized when dealing with PSTN gateways. A nasty situation creeps up when a SIP phone is forwared to a PSTN destination. Then when an incoming call from some other PSTN phone to the SIP phone occurs, SER would try to send an INVITE back out the PSTN to the forwarded destination and many PSTN gateways will reply with "482 Loop Detected".
Regards, Paul
On Mon, 28 Feb 2005 11:29:37 -0500, Steve Blair blairs@isc.upenn.edu wrote:
Why wouldn't you just rewrite the hostname and port to be that of the voicemail server in your "302 failure route handling logic" and relay the call?
Java Rockx wrote:
Hi all.
I'm trying to get my call working working and I have a question.
Suppose a SIP subscriber has voicemail and call forwarding is enabled.
When an incoming call is destined for this SIP user, SER will reply with a 302 redirect message.
Next the caller's phone should construct a brand new INVITE to the redirected destination.
My question is this:
If the forwarded destination is busy, then the call should be sent to the voicemail of the SIP subscriber that was originally called. This is the way the RBOCs call forwarding works.
If I'm using sl_send_reply("302", "Temporarily Moved") back to the caller, how can I make the voicemail piece work?
Regards, Paul
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
--
ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu