Can someone please explain to me how RPID works under openser? I have
some experience with SIP and Asterisk, but I am having some trouble
manipulating the RPID variables under openser. How does RPID relate
to Callerid or simple services like private calling, or anonymous
call block. How do I tell openser to send all calls from a certain
user with the privacy=full flag, I'm assuming this tells the sip
proxy or UA on the other end not to present the callerid? Any help
greatly appreciated.
-Brandon Price-
Where is the T1 timer set in Ser? The RFC suggests a default of 500 ms for
round-trip time.
I have an ethereal trace showing Ser re-transmitting after less than 300 ms.
Anyway, because
of our environment I'd like to set T1 to be 1000 ms. I'm using 0.8.14.
thanks
I have openser set up to remove previously authorized credentials
from the message being processed by the server, using
consume_credentials. Yet the authorization credentials are still
being passed. Any help is as always appreciated. Thank you. The
related openser.cfg portions are posted below...
if (!www_authorize("", "subscriber")) {
www_challenge("", "1");
break;
};
consume_credentials();
save("location");
break;
if I then restart openser, make a call and start tethereal I get....
tethereal -V port 5060 | grep username
Authorization: Digest
username="theusernamewashere",realm="ser1.manhattan.whatever.net",nonce=
"43457703fa69c4dd5321a7229ea41102f2200390",uri="sip:
13143212222(a)ser1.manhattan.whatever.net",algorithm=MD5,response="fc57c2b
4897d3d7b8ad887a788bf07f6",qop=aut
Hi all,
is there a possibility to use variables in ser.cfg (ser0.9.0) like
...
set myip=192.168.1.11
set myaccount=testuser
set mypasswd=testpasswd
...
modparam("uac","credential","$myaccount:$myip:$mypasswd");
...
rewritehost($myip);
...
Perhaps a solution would be with avpops, but I didn't get it to work in conjunction with dbtext.
Has someone an example how to read variables with avpops out of a dbtext table which are not connected to a SIP-message?
(I read the README for avpops!!)
Any other solutions would be a help!!
Would this be possible with openser's pseudo variables or another ser-version?
Thanks
Thorsten
Hi Guys,
Is there a way for SER to block calls coming from a specific
user-agent? I want to block calls so that uniform useragaent will be
used by my users. Any modules that can do this?
Thanks,
Ryan
Hi,
I'm using SER with Sipura and VoIP GW.
When there is no line available from the VoIP GW, the VoIP GW replies with
486 Busy Here.
The Sipura receives this message and sends an ACK.
The ACK has the IP addr. Of the SER. If it is ok, the ACK does not include
the "Route" parameter, is it a problem from the Sipura?
This cause the SIP to loop the ACK because there is no Route and with the
following code, the packet becomes too big!!!
............
if (!method=="REGISTER") record_route();
# ------------------------------------------------------------------
# Loose Route Section
# ------------------------------------------------------------------
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
t_relay();
break;
};
......................
Same configuration and same trace
192.168.20.155 is the SIP Proxy
192.168.20.105 is the VoIP GW
192.168.20.90 is the Sipura
312 54.858597 192.168.20.90 192.168.20.155 SIP/SDP
Request: INVITE sip:3353@192.168.20.155, with session description
313 54.860243 192.168.20.155 192.168.20.90 SIP
Status: 407 Proxy Authentication Required
314 54.877006 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
315 54.886801 192.168.20.90 192.168.20.155 SIP/SDP
Request: INVITE sip:3353@192.168.20.155, with session description
316 54.889720 192.168.20.155 192.168.20.105 SIP/SDP
Request: INVITE sip:3353@192.168.20.105:5060, with session description
318 54.924666 192.168.20.105 192.168.20.155 SIP
Status: 100 Trying
319 54.925866 192.168.20.155 192.168.20.90 SIP
Status: 100 Trying
320 54.950257 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
321 54.951442 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
323 54.968942 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
440 55.527290 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
441 55.528528 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
442 55.547237 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
627 56.624649 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
639 56.626924 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
649 56.647321 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
912 58.719683 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
913 58.720967 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
916 58.746851 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
1192 62.806510 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
1193 62.807741 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
1194 62.827148 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
1417 66.892115 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
1418 66.893350 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
1419 66.907704 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
1628 68.143481 192.168.20.101 192.168.20.55 SIP
Thanks.
Micheline
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Please help.
----- Original Message -----
From: "Nicky" <nicky(a)caliber.com.sg>
To: <serusers(a)lists.iptel.org>; <sems(a)lists.iptel.org>
Sent: Wednesday, September 28, 2005 10:38 PM
Subject: Sems IVR Module
> Hi all,
>
> I am testing the sems version from the Head and when i uses a SIP Phone to
> call into
> the IVR module for recording when I am in the middle of recording, the
> program detected DTMF
> signal and jump out of the script. Why is it so? as I never pressed on any
> key? How can I resolve this
> problem? Why is the voice signal become the DTMF signal? Is there any
> adjustment that
> i can do to solve this problem , as my voice goes louder the more DTMF
> signal is being detected.
>
> Please help.
>
> regards,
> nicky
>
Using the last cfg from onsip.org, i just see that now Uas are registered
with their private Ip when using mediaproxy.
Mediaproxy seems the best solution for scalability, that's why I want to
test it.
Any Idea?
Hi,
I'm using SER with Sipura and VoIP GW.
When there is no line available from the VoIP GW, the VoIP GW replies with
486 Busy Here.
The Sipura receives this message and sends an ACK.
The ACK has the IP addr. Of the SER. Is it ok, or it should be the IP
address of the VoIP GW. When SER receives the ACK with its IP address
192.168.20.155, there is a loop of ACK in SER.........
I attached a trace and this is my configuration.
192.168.20.155 is the SIP Proxy
192.168.20.105 is the VoIP GW
192.168.20.90 is the Sipura
312 54.858597 192.168.20.90 192.168.20.155 SIP/SDP
Request: INVITE sip:3353@192.168.20.155, with session description
313 54.860243 192.168.20.155 192.168.20.90 SIP
Status: 407 Proxy Authentication Required
314 54.877006 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
315 54.886801 192.168.20.90 192.168.20.155 SIP/SDP
Request: INVITE sip:3353@192.168.20.155, with session description
316 54.889720 192.168.20.155 192.168.20.105 SIP/SDP
Request: INVITE sip:3353@192.168.20.105:5060, with session description
318 54.924666 192.168.20.105 192.168.20.155 SIP
Status: 100 Trying
319 54.925866 192.168.20.155 192.168.20.90 SIP
Status: 100 Trying
320 54.950257 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
321 54.951442 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
323 54.968942 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
440 55.527290 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
441 55.528528 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
442 55.547237 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
627 56.624649 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
639 56.626924 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
649 56.647321 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
912 58.719683 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
913 58.720967 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
916 58.746851 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
1192 62.806510 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
1193 62.807741 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
1194 62.827148 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
1417 66.892115 192.168.20.105 192.168.20.155 SIP
Status: 486 Busy Here
1418 66.893350 192.168.20.155 192.168.20.90 SIP
Status: 486 Busy Here
1419 66.907704 192.168.20.90 192.168.20.155 SIP
Request: ACK sip:3353@192.168.20.155
1628 68.143481 192.168.20.101 192.168.20.55 SIP
Request: BYE sip:06@192.168.20.105:5060;maddr=192.168.20.105
1630 68.147489 192.168.20.55 192.168.20.105 SIP
Request: BYE sip:06@192.168.20.105:5060;maddr=192.168.20.105
1631 68.174246 192.168.20.105 192.168.20.55 SIP
Status: 200 OK
1632 68.176946 192.168.20.55 192.168.20.101 SIP
Status: 200 OK
Micheline
Hello,
latest openser version from cvs includes a set of new functions (core
and avpops) and pseudo-variable which allow to access and manage the
value of dst_uri.
The dst_uri field contains the address of the next hop, when the routing
does not follow R-URI address. Such cases are Route header driven
routing (dst_uri is set after loose_route() to the value of next Route
header), contacts behind nat (dst_uri is set to the address of nat after
lookup("location")) or dispatcher usage. The dst_uri has higher priority
in routing over R-URI but less than explicit parameters in relaying
functions (e.g., t_relay_to*()).
The field was invisible from the script but it proved in the discussions
on the mailing list to be important to access it. For example, to detect
whether the caller and callee are behind same nat. Also, it can be
checked in case of preloaded Route header to secure relaying to special
resources (e.g., pstn gateway). With the new branch_route the value of
dst_uri can be checked for each branch.
A short summary of what was added:
* core functions:
- setdsturi("uri") - set the value of dst_uri
- resetdsturi() - reset the value of dst_uri
- isdsturiset() - test if the value of dst_uri is null
* pseudo-variables
- $du - value of dst_uri (added long time ago)
- $dd - domain of dst_uri
- $dp - port of dst_uri
- $dP - transport protocol of dst_uri
* avpops changes
- avp_write() can read the value of dst_uri and write it in an avp
- avp_pushto() can write the value of an avp in dst_uri
Tutorials and dokuwiki will be updated soon.
Daniel