Re: [OpenSIPS-Users] PUA module functions and XCAP functionality
Hi, On May 19, 2011, at 8:13 AM, Jayesh Nambiar wrote: Hello all, I have setup an opensips with call and presence capabilities with xcap. Now there are some Endpoints that does not support presence capabilities for which I use PUA module. So whenever those endpoints are registered the watchers for those endpoints will see them as online. And when they Un-register the watchers should see them as offline. The question here is how to enable the xcap functionality for such endpoints. How does an endpoint which does not have presence capabilites allow or disallow someone to check their presence status which is generated by the PUA module? Is it something possible to do from the web-interface? For eg. someone wants to check the presence status of endpoint A which does not have presence capabilites. User A logs into the web-interface, allows the watcher to view its presence status which actually creates an xcap doc for that user. And only then the PUBLISH from PUA gets notified to those users based upon the xcap rules of that user. Any proper direction in this context will be really helpful. Thanks, You may want to have a look at SOAP SIMPLE Proxy (http://openxcap.org/wiki/soap-simple-proxy) it allows you to interact with OpenXCAP and manage presence policy with a SOAP interface, which you can use from a web page. Regards, Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips = SEMS(voicemail)
Hi, The sems example uses variables which are setup at serweb. (email/language/...) These should be set in opensips config (maybe taken from database/ whatever...) There are variables in opensips containing some of the values you could use like the request domain / from uri / ... http://www.opensips.org/Resources/DocsCoreVar16 here is a simple real life example: avp_db_query(select email_address FROM subscriber WHERE username = '$rU',$avp(s:677)); append_hf(P-App-Param: eml=$avp(s:677);mod=voicemail;lng=english;snd=$fU;dom=$Ri\r\n); Best Regards Max M. Am 18.05.2011 19:19, schrieb Steven Pokrandt: SEMS docs make it sound easy to integrate with SER (opensips?) but opensips complains about the 2nd append line. I'm new to all of this and as you know it's a steep learning curve. where do i get the $t.*, @from.*, @ruri.* variables set from? is there a module? do I need to look them up and assign them from the DB in opensips? append_hf(P-App-Name: voicemail\r\n); append_hf(P-App-Param: mod=%$t.voicemail%|;eml='%$t.email%|';usr=%@ruri.user%|;snd='%@from.uri%|';dom=%@ruri.host%|;uid=%$t.uid%|;did=%$t.did%|;); -- Steven Pokrandt (425) 686 WABO http://steven.pokrandt.me/Steven_Pokrandt.vcf ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] adding 1 alias to more than one subscriber
Hi Dave, i've added the line modparam(alias_db, append_branches, 1), but now i'm trying to call the alias 314001 and the INVITE is going to both users 1000 and 1001 as expected: that's nice... # U 2011/05/18 13:55:52.918314 *192.168.131.129:5060 - 192.168.131.1:61052 *INVITE *sip:vipexamk@192.168.131.1:61052* SIP/2.0 Record-Route: sip:192.168.131.129;lr=on Via: SIP/2.0/UDP 192.168.131.129;branch=z9hG4bKec48.66d63466.0 Via: SIP/2.0/UDP 192.168.131.1:5060 ;branch=z9hG4bK-d8754z-801d2084170f1b21-1---d8754z- Max-Forwards: 69 Contact: sip:1000@192.168.131.1:5060;transport=UDP To: sip:314001@192.168.131.129;transport=UDP From: 1000sip:1000@192.168.131.129;transport=UDP;tag=ff69d424 Call-ID: N2JiMTRhODJhNTRiNTgzNjc3NjdkZTI1NThiOWQ1MTU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.5324 Content-Length: 329 v=0 o=Zoiper_user 0 0 IN IP4 192.168.131.1 s=Zoiper_session c=IN IP4 192.168.131.1 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2011/05/18 13:55:52.918371 *192.168.131.129:5060 - 192.168.131.129:5060 *INVITE *sip:1000@192.168.131.129* SIP/2.0 Record-Route: sip:192.168.131.129;lr=on Via: SIP/2.0/UDP 192.168.131.129;branch=z9hG4bKec48.66d63466.1 Via: SIP/2.0/UDP 192.168.131.1:5060 ;branch=z9hG4bK-d8754z-801d2084170f1b21-1---d8754z- Max-Forwards: 69 Contact: sip:1000@192.168.131.1:5060;transport=UDP To: sip:314001@192.168.131.129;transport=UDP From: 1000sip:1000@192.168.131.129;transport=UDP;tag=ff69d424 Call-ID: N2JiMTRhODJhNTRiNTgzNjc3NjdkZTI1NThiOWQ1MTU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.5324 Content-Length: 329 v=0 o=Zoiper_user 0 0 IN IP4 192.168.131.1 s=Zoiper_session c=IN IP4 192.168.131.1 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv As you can see, the R-URI of the first invite is ok, but not the second, as you can see the domain part is the ip address of the proxy and not the ip of the location user 1000, what's happening? seems to be a loop on the second branch* (192.168.131.129:5060 - 192.168.131.129:5060*)...as i can see, the DB alias is correctly fetching both users from DB, but only the first one is resolved via lookup(location) (registration search). any suggest? Many thanks! 2011/5/18 Dave Singer dave.dorasin...@gmail.com Toyima, check: http://www.opensips.org/html/docs/modules/1.6.x/alias_db.html#id250030 Dave On Tue, May 17, 2011 at 6:44 AM, Toyima Dias toyim...@gmail.com wrote: Is it possible to add the same alias to more than one subscriber? for example, if an incoming call to the number 14882736524 is assigned to subscriber 1000, is it possible to assign this DID to subscriber 1001? the idea is to create a group of ringing or something like that, is that possible? i've tried to insert the data on phpmyadmin and also on opensips-panel, it doesn't work: Error consulta SQL: INSERT INTO `opensips`.`dbaliases` ( `id` , `alias_username` , `alias_domain` , `username` , `domain` ) VALUES ( NULL , '314001', '192.168.131.129', '1000', '192.168.131.129' ) MySQL ha dicho: #1062 - Duplicate entry '314001-192.168.131.129' for key 'alias_idx' Thanks! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- David Singer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Reroute B2B call after failure
Do you have some code example of how to do this? I am not sure what variables need to be overwritten. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Reroute-B2B-call-after-failure-tp6371369p6383969.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Rewrite Username in URI when using userloc
Hello all,
I'm using Opensips in between several Freeswitch and Asterisk boxes (each with
multiple DIDs) and my ITSP's proxies. The local IP PBXs register to Opensips
and inbound routing is done via the userloc module. The problem I have is that
I need to send in incoming INVITE to the original username, not the registered
contact so the PBXs can perform the proper routing. Currently I set $rU=$Ou in
the request route which works perfectly if there is only one contact
registered. If there is more than one contact registered, the INVITE is sent
to the registered contact and then the PBX doesn't know how to route the call
since it hits a default contact. Is there a better way to do this? I'm fairly
new to Opensips and would appreciate any help. :-)
A snippet from my script:
route {
...
if (!lookup(location,m)) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply(404, Not Found);
exit;
case -2:
sl_send_reply(405, Method Not Allowed);
exit;
}
}
$rU=$oU;
route(1);
}
route[1] {
if (is_method(INVITE)) {
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
branch_route[2] {
xlog(new branch at $ru\n);
}
onreply_route[2] {
xlog(incoming reply\n);
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
if (t_check_status(3[0-9][0-9])) {
t_reply(404,Not found);
exit;
}
}
Thanks,
Spencer
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Reroute B2B call after failure
On Thu, May 19, 2011 at 6:16 PM, osiris123d duane.lar...@gmail.com wrote: Do you have some code example of how to do this? I am not sure what variables need to be overwritten. destination URI and request URI:: http://www.opensips.org/Resources/DocsCoreVar#toc29 http://www.opensips.org/Resources/DocsCoreVar#toc67 Regards, Ovidiu Sas ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Reroute B2B call after failure
Thanks Ovidiu. I was using $rd before to send it to the Proxy, but $du looks to be better suited for that. And then $ru works perfect for representing the Primary or Backup SIP Gateway. Appreciate it!!! On , Ovidiu Sas o...@voipembedded.com wrote: On Thu, May 19, 2011 at 6:16 PM, osiris123d duane.lar...@gmail.com wrote: Do you have some code example of how to do this? I am not sure what variables need to be overwritten. destination URI and request URI:: http://www.opensips.org/Resources/DocsCoreVar#toc29 http://www.opensips.org/Resources/DocsCoreVar#toc67 Regards, Ovidiu Sas ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] adding 1 alias to more than one subscriber
Toyima, Not sure how much you know about branches, so... With that switch branches are added for each alias. By default any test or action in the rest of the main route script only applies to the first branch. That is why only the first branch has the lookup location done to it. In order to test/act on other branches, you need to use t_on_branch() and define a route block. When t_on_branch() is set to that route block it will be run for all branches that were created. Also of note is that in failure route, the vars like $ru and headers are all set to the way they were before any branch route is run (may have an effect on how you failover to voicemail). So consider that when choosing to apply changes in main routing or branch routing. A good example is uac_replace_from() which can only be run once on a message so it is good to put it as one of the last things and if there are branches, do it in the branch block. What I've found to work good is to define a named route block to handle last minute changes and test in main route block if there are multiple branches. If only main branch, just call the last_minute_changes from main route before exit; other wise set t_on_branch() and call last_minute_changes from the branch route block. Hope that helps. Dave On Thu, May 19, 2011 at 2:58 AM, Toyima Dias toyim...@gmail.com wrote: Hi Dave, i've added the line modparam(alias_db, append_branches, 1), but now i'm trying to call the alias 314001 and the INVITE is going to both users 1000 and 1001 as expected: that's nice... # U 2011/05/18 13:55:52.918314 192.168.131.129:5060 - 192.168.131.1:61052 INVITE sip:vipexamk@192.168.131.1:61052 SIP/2.0 Record-Route: sip:192.168.131.129;lr=on Via: SIP/2.0/UDP 192.168.131.129;branch=z9hG4bKec48.66d63466.0 Via: SIP/2.0/UDP 192.168.131.1:5060;branch=z9hG4bK-d8754z-801d2084170f1b21-1---d8754z- Max-Forwards: 69 Contact: sip:1000@192.168.131.1:5060;transport=UDP To: sip:314001@192.168.131.129;transport=UDP From: 1000sip:1000@192.168.131.129;transport=UDP;tag=ff69d424 Call-ID: N2JiMTRhODJhNTRiNTgzNjc3NjdkZTI1NThiOWQ1MTU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.5324 Content-Length: 329 v=0 o=Zoiper_user 0 0 IN IP4 192.168.131.1 s=Zoiper_session c=IN IP4 192.168.131.1 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv # U 2011/05/18 13:55:52.918371 192.168.131.129:5060 - 192.168.131.129:5060 INVITE sip:1000@192.168.131.129 SIP/2.0 Record-Route: sip:192.168.131.129;lr=on Via: SIP/2.0/UDP 192.168.131.129;branch=z9hG4bKec48.66d63466.1 Via: SIP/2.0/UDP 192.168.131.1:5060;branch=z9hG4bK-d8754z-801d2084170f1b21-1---d8754z- Max-Forwards: 69 Contact: sip:1000@192.168.131.1:5060;transport=UDP To: sip:314001@192.168.131.129;transport=UDP From: 1000sip:1000@192.168.131.129;transport=UDP;tag=ff69d424 Call-ID: N2JiMTRhODJhNTRiNTgzNjc3NjdkZTI1NThiOWQ1MTU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.5324 Content-Length: 329 v=0 o=Zoiper_user 0 0 IN IP4 192.168.131.1 s=Zoiper_session c=IN IP4 192.168.131.1 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv As you can see, the R-URI of the first invite is ok, but not the second, as you can see the domain part is the ip address of the proxy and not the ip of the location user 1000, what's happening? seems to be a loop on the second branch (192.168.131.129:5060 - 192.168.131.129:5060)...as i can see, the DB alias is correctly fetching both users from DB, but only the first one is resolved via lookup(location) (registration search). any suggest? Many thanks! 2011/5/18 Dave Singer dave.dorasin...@gmail.com Toyima, check: http://www.opensips.org/html/docs/modules/1.6.x/alias_db.html#id250030 Dave On Tue, May 17, 2011 at 6:44 AM, Toyima Dias toyim...@gmail.com wrote: Is it possible to add the same alias to more than one subscriber? for example, if an incoming call to the number 14882736524 is assigned to subscriber 1000, is it possible to assign this DID to subscriber 1001? the idea is to create a group of ringing or something like that, is that possible? i've tried to insert the data on phpmyadmin and also on opensips-panel, it doesn't work: Error consulta SQL: INSERT INTO `opensips`.`dbaliases` ( `id` , `alias_username` , `alias_domain` , `username` , `domain` ) VALUES ( NULL , '314001', '192.168.131.129', '1000', '192.168.131.129' ) MySQL ha
Re: [OpenSIPS-Users] Rewrite Username in URI when using userloc
Spencer,
Looks like you just need to move $rU=$oU; into branch_route[2]
Note that in failure route it will be reset to what it was before
changes done in branch route so you may want to do it in main route
and in branch route.
Dave
On Thu, May 19, 2011 at 4:02 PM, Spencer Thomason
spen...@5ninesolutions.com wrote:
Hello all,
I'm using Opensips in between several Freeswitch and Asterisk boxes (each
with multiple DIDs) and my ITSP's proxies. The local IP PBXs register to
Opensips and inbound routing is done via the userloc module. The problem I
have is that I need to send in incoming INVITE to the original username, not
the registered contact so the PBXs can perform the proper routing. Currently
I set $rU=$Ou in the request route which works perfectly if there is only one
contact registered. If there is more than one contact registered, the INVITE
is sent to the registered contact and then the PBX doesn't know how to route
the call since it hits a default contact. Is there a better way to do this?
I'm fairly new to Opensips and would appreciate any help. :-)
A snippet from my script:
route {
...
if (!lookup(location,m)) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply(404, Not Found);
exit;
case -2:
sl_send_reply(405, Method Not Allowed);
exit;
}
}
$rU=$oU;
route(1);
}
route[1] {
if (is_method(INVITE)) {
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
branch_route[2] {
xlog(new branch at $ru\n);
}
onreply_route[2] {
xlog(incoming reply\n);
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
if (t_check_status(3[0-9][0-9])) {
t_reply(404,Not found);
exit;
}
}
Thanks,
Spencer
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
