I am using LCR module and I have two set of gateways (group id 1 and group
id 2). I have defined a list of prefixes and the corresponding gateway
group id's.
All is working fine vis-à-vis the prefixes and the choice of the gateway
group. My problem is that if a RURI is supposed to be handled by gateways in
group id 1, and all the gateways in that group are busy, the call then gets
forwarded to the gateways in group id 2 although the prefix does not match
to the one associated with group id 2.
Is there any way to disable this so that if all gateways are busy in group
id 1, SER does not forward calls to the group id 2?
Thanks in advance
I am trying to change/re-write From URI username and From display name (*$fU
and **$fn respectively) to avposp values*I tried: $fU =
$avp(s:somevariable);
avp_write("$fU","$avp(s:somevariable)");
append_hf("From-username: $avp(s:somevariable)\r\n");
but none of them works, any ideas on how to re-write those variables in SIP
messages? Thank you*
*
Hi, I use MediaProxy 1.9.1 with OpenSer.
idleTimeout=50 seconds
I've received an INVITE and a 183 with early media during 52 seconds and after
that the 200 was received. But the RTP session in MediaProxy was terminated
after 50 seconds (idleTimeout). Why?
Is a still no answered call considered "idle"?
This is the syslog:
-----------------------------------------------
# INVITE arrives:
Jul 25 15:21:17 server-sip proxydispatcher[21258]: request
9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667
77.14.6.152:57102:audio,77.14.6.152:41292:image 77.14.0.144 77.14.0.144
remote 192.168.10.2 remote CS2000_NGSS/9.0
info=from:882534800@77.14.0.144:5060,to:887332031@88.99.3.10:5060,fromtag:e3e-13c4-aa99fd-62c9f59e-aa99fd,totag:
Jul 25 15:21:17 server-sip mediaproxy[21253]: request
9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667
77.14.6.152:57102:audio,77.14.6.152:41292:image 77.14.0.144 77.14.0.144
remote 192.168.10.2 remote CS2000_NGSS/9.0
info=totag:,to:887332031@88.99.3.10:5060,from:882534800@77.14.0.144:5060,fromtag:e3e-13c4-aa99fd-62c9f59e-aa99fd
Jul 25 15:21:17 server-sip mediaproxy[21253]: session
9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667: started.
listening on 88.99.3.10:61820,61822
Jul 25 15:21:17 server-sip mediaproxy[21253]: execution time: 0.92 ms
Jul 25 15:21:17 server-sip proxydispatcher[21258]: forwarding to mediaproxy
on /var/run/mediaproxy.sock: got: '88.99.3.10 61820 61822'
# After 50 seconds the session is ended by MediaProxy:
Jul 25 15:22:06 server-sip mediaproxy[21253]: session
9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667: 0/0/0 packets,
0/0/0 bytes (caller/called/relayed)
Jul 25 15:22:06 server-sip mediaproxy[21253]: session
9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667: ended (did
timeout).
# After 2 seconds the 200 OK arrives so media fails.
Jul 25 15:22:08 server-sip proxydispatcher[21258]: lookup
9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667
192.168.10.2:19084:audio 88.99.1.213 sip.domain.net local unknown unknown
Asterisk=20testDesign
info=from:882534800@sip.domain.net,to:887332031@88.99.3.10:5060,fromtag:e3e-13c4-aa99fd-62c9f59e-aa99fd,totag:as6156af6c
Jul 25 15:22:08 server-sip mediaproxy[21253]: lookup
9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667
192.168.10.2:19084:audio 88.99.1.213 sip.domain.net local unknown unknown
Asterisk=20testDesign
info=totag:as6156af6c,to:887332031@88.99.3.10:5060,from:882534800@sip.domain.net,fromtag:e3e-13c4-aa99fd-62c9f59e-aa99fd
Jul 25 15:22:08 server-sip mediaproxy[21253]: session
9b984e5890000e3e13c4aa99fd9a69852d69d32b2614fe1cc8-0228-6667: started.
listening on 88.99.3.10:61824
Jul 25 15:22:08 server-sip mediaproxy[21253]: execution time: 0.60 ms
Jul 25 15:22:08 server-sip proxydispatcher[21258]: forwarding to mediaproxy
on /var/run/mediaproxy.sock: got: '88.99.3.10 61824'
Jul 25 15:22:08 server-sip proxydispatcher[21258]: execution time: 1.09 ms
-----------------------------------------------
--
Iñaki Baz Castillo
Hello,
I have a problem with parallel forking and TCP when one connection isn't available and the inv_timeout expires.
I have two clients registered with the same username and transport=TCP.
If I make a call to this number when one of them is not reachable, the INVITE goes to the phone reachable, if the call timeout expires (fr_inv_timer_avp) Openser sends the CANCEL, but doesn't send anything to the caller.
The failure_route doesn't hit.
There is also a delay between the t_relay and the forwarding of the packet TCP on the net (around 3 seconds). The TCP packet is forwarded to the available connection only after Openser detects the failure for the closed TCP connection (the INVITE goes on the net only after the message ERROR:tm:t_forward_nonack: sending request failed).
This is the trace (debug level 3, Openser version 1.3.2):
Jun 25 16:30:01 SAM-IP ser[23888]: R1 t_relay R-uri: <sip:10240@10.45.10.63:2054;transport=tcp;line=zdhv6xi4>
Jun 25 16:30:01 SAM-IP ser[23888]: Branch route <sip:10240@10.45.10.63:2054;transport=tcp;line=zdhv6xi4> 1
Jun 25 16:30:01 SAM-IP ser[23888]: Branch route <sip:10240@10.45.10.36:5064;transport=tcp> 2
Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:core:tcp_blocking_connect: poll error: flags 18
Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (113) No route to host
Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed
Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:core:tcp_send: connect failed
Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:tm:msg_send: tcp_send failed
Jun 25 16:30:04 SAM-IP ser[23888]: ERROR:tm:t_forward_nonack: sending request failed
Jun 25 16:30:04 SAM-IP ser[23890]: Reply Route: provision 180 - Ringing ip <10.45.10.36:33109> From <sip:10242@10.45.8.249> To <sip:10240@10.45.8.249>
Jun 25 16:30:04 SAM-IP ser[23890]: ERROR:core:io_wait_loop_sigio_rt: ignoring event 41 on fd 24 (fm->fd=-1, fm->data=0x406e5488)
Jun 25 16:30:04 SAM-IP ser[23900]: ERROR:core:io_wait_loop_sigio_rt: ignoring event 41 on fd 25 (fm->fd=-1, fm->data=0x406c5258)
Jun 25 16:30:11 SAM-IP ser[23889]: Reply Route: 487 - <Request Terminated> ip <10.45.10.36:33110> From <sip:10242@10.45.8.249> To <sip:10240@10.45.8.249>
Jun 25 16:30:11 SAM-IP ser[23890]: ERROR:core:io_wait_loop_sigio_rt: ignoring event 41 on fd 24 (fm->fd=-1, fm->data=0x406d5370)
Jun 25 16:30:11 SAM-IP ser[23891]: ERROR:core:tcp_read: error reading: Connection reset by peer
Jun 25 16:30:11 SAM-IP ser[23891]: ERROR:core:tcp_read_req: failed to read
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Hi All,
I am currently working on OpenSER-1.3.2. I want to exclude the RLS module
but when I did getting some errors.
Could you please send me the openser.cfg with RLS working.
Regards,
Mahesh Peddi
Infospectrum India Pvt. Ltd.
Cell: +91 9765775176
IP-Phone Ext. - 764
Hi, I am using the presence_xml module to provide presence across multiple clients. X-Lite, Bria works well however Snom phones produce the following error: ------------- ERROR ----------------namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Lunch</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Busy</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^namespace error : Namespace prefix hs on detected-state is not defined <hs:detected-state>Lunch</hs:detected-state> ^namespace error : Namespace prefix hs on device is not definedtact priority="0.000">sip:--- ADDRESS ----</contact> <hs:device ^namespace error : Namespace prefix hs on caption is not defined <hs:caption>snom370</hs:caption> ^------------- END ERROR ----------------It looks like a validation issue with the XML, is there anyway you can switch off validation or update the DTD? Thanks, Ross
_________________________________________________________________
Play and win great prizes with Live Search and Kung Fu Panda
http://clk.atdmt.com/UKM/go/101719966/direct/01/
This is a feature of OpenXCAP, to retrieve the list of active watchers
from OpenSER. It does not require any support in OpenSER.
Adrian
> Hi All
> is openseer is supporting watcher application (NDO)
>
> ~Suresh..
>
>
what do you mean by watcher application?
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
El Viernes, 20 de Junio de 2008, Valerio Di Marino escribió:
> Thanks for your answer.
Ok, but next time continue the thread in the maillist ;)
> 2008/6/20 Iñaki Baz Castillo ibc(a)aliax.net:
> > AFAIK if a UAC is redirected it will not include the already used
> > credentials
> > in the new request to the new destination (maybe I'm wrong).
>
> Do you think is possible to force using the pseudovariables ?
No, it's not possible. When a UAAC receives a 302 it generates a completely
new request with destination the URI of the "Contact" in the 302.
> > If you need solving NAT in your OpenSer then you need to forward the
> > request
> > in it after make fix NAT.
>
> How ? Can explain better ?
If a UAC receives a 302 it will generate a new INVITE. If the UAC has not
configured your OpenSer as outbound proxy, the new INVITE it generates will
go directly to the SIP provider, so it's not possible your OpenSer makes fix
the NAT (fix "Contact" and so).
But maybe your SIP provider makes fix the NAT by itself, or you can try STUN
in your UAC's.
> > If not you could use forwarding or redirect.
>
> With redirect, openser will send an address and the uer will contact the
> new address: is it correct ? (I am not shure).
Yes, OpenSer replies a 302 with the new URI in the "Contact" header.
> > In OpenSer to get a redirection you must set the RURI ($ru = ...) and
> > reply a
> > 302 to the user.
>
> Can you send ma a routine syntax ?
> For example, I would redirect to sip.voipstunt.com
Something like:
$ru = "sip.voipstunt.com";
send_reply("302", "Redirect to SIP provider");
exit;
--
Iñaki Baz Castillo