Hrmmm... I consistently get a Rails error when I get to the
config_create stage. This is on both a Linux and Windows machine (a
dedicated T1 and a proxied DS3, respectively for connection) using
Firefox.
Any ideas?
- Brad
> -----Original Message-----
> From: users-bounces(a)openser.org
> [mailto:users-bounces@openser.org] On Behalf Of Andreas Granig
> Sent: Thursday, February 22, 2007 9:46 AM
> To: users(a)openser.org
> Subject: [Users] OpenSER Configuration Generator
>
> All,
>
> I've finished the first version of sip:wizard, an online
> SER/OpenSER configuration generator: http://www.sipwise.com/wizard
>
> Currently, support for OpenSER 1.1 is available. In the
> future, also new OpenSER releases as well as SER will be supported.
>
> At the moment, we have config features for basic routing,
> far-end NAT traversal, ENUM and PSTN in place. This will also
> be extended, depending on suggestions from you, the users.
>
> I hope you'll find it useful. Feedback is of course very much
> appreciated.
>
> Cheers,
> Andreas
>
>
> This e-mail is confidential and may well also be legally
> privileged. If you have received it in error, you are on
> notice of its status. Please notify us immediately by reply
> e-mail and then delete this message from your system. Please
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> contents to any other person: to do so could be a breach of
> confidence. Thank you for your cooperation.
>
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The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.
All,
I've finished the first version of sip:wizard, an online SER/OpenSER
configuration generator: http://www.sipwise.com/wizard
Currently, support for OpenSER 1.1 is available. In the future, also new
OpenSER releases as well as SER will be supported.
At the moment, we have config features for basic routing, far-end NAT
traversal, ENUM and PSTN in place. This will also be extended, depending
on suggestions from you, the users.
I hope you'll find it useful. Feedback is of course very much appreciated.
Cheers,
Andreas
This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation.
Running openser 1.2
failure_route[1]
xlog("L_INFO", "status codes are [$rs] [$rr]\n");
gives "status codes are [<null>] [<null>]
any ideas?
Ibrahim Hamouda
Sphinx Information Technologies Inc.
IT Canada International
Tel: 403-668-6880
Fax: 403-229-0407
N.A.: 1-877-500-7664
Mobile: 403-714-3336
Today I discovered that SER does not respond well to international chars in contact header of at least 200 OK.
The 200 OK message had the below contact header. The two dots are the swedish chars ä and ö.
Contact: Hampus H.ggl.f <sip:330039@217.168.167.xyz:5061>.
The error logged from SER was:
Feb 22 09:22:27 sip /usr/local/sbin/ser[40800]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: mk_proxy: could not resolve hostname: "Hampus"
Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: uri2proxy: bad host name in URI <sip:Hampus>
Feb 22 09:22:30 sip /usr/local/sbin/ser[40798]: ERROR: t_forward_nonack: failure to add branches
The UA is a PAP2-EU device running latest firmware. The text Hampus H.ggl.f was entered in the display name field of PAP2.
Is this an error with SER or is it illegal to use international characters in Contact header?
Kind regards
Roger
Hello
I need to write a CAC (Call Admission Control) module for an 802.11 AP
(Access Point).
The idea is to use a SIP Proxy to monitoring bandwith utilization
according to codec, and allow or disallow new sessions, depending on
resources.
On a first stage I just want to limit the session to a fix number. For
that, for each new session request (INVITE), I need to know how many
active session there is already.
I think using the module dialog but it is experimental.
What is exactly its status? Is it usable for now?
Is there another way to get information about the number of active sessions?
Thanks.
Regards,
Michel
Hello!
I'm having a issue with a 302 moved temporarily from a UA i want to
attach to my proxy. The UA replies to a INVITE from the proxy with a 302
MOVED TEMPORARILY and i'm using "get_redirects" from the
uac_redirect-module to extract the contact. However, i don't know how
to continue from there as the domain part in the redirect-contact is
the hostname of the proxy but the port is 5065 and transport=TCP (see
debug for details) and if i simply relay it, it would be relayed back
to the proxy on the wrong port 5065.
I guess i could use the request branch pseudo variable($br) and extract
the port from it(or change the domain-part to the actual domain/ip of
the UA and relay it), still i don't really know how to do it.
This is the sip-trace of the relevant dialog part:
2xx.xxx.xxx.x06 is the openser-host, 6x.xxx.xxx.x93 is the useragent
######################################
####
T 2xx.xxx.xxx.x06:56669 -> 6x.xxx.xxx.x93:5060 [AP]
INVITE sip:4989111111@proxydomain.org SIP/2.0.
Record-Route:
<sip:2xx.xxx.xxx.x06;transport=tcp;r2=on;lr;ftag=as784d1ec2>.
Record-Route:
<sip:2xx.xxx.xxx.x06;r2=on;lr;ftag=as784d1ec2>. Via:
SIP/2.0/TCP 2xx.xxx.xxx.x06;branch=z9hG4bK96db.2c3c0006.0. Via:
SIP/2.0/UDP 2xx.xxx.xxx.97:5060;branch=z9hG4bK52fef574;rport=5060.
From: "SIPPHONE1"
<sip:089222222@otherproxydomain.org>;tag=as784d1ec2. To:
<sip:4989111111@proxydomain.org>.
Contact:<sip:089222222@21x.xxx.xxx.97>.
Call-ID:7fc122590665124f347fb2d956ba4c07@sip.at.telgo.cc.
CSeq: 102 INVITE.
User-Agent: MyMediaGW1. Max-Forwards: 69. Date: Wed, 21 Feb 2007
13:02:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY. Content-Type: application/sdp.
Content-Length: 492.
.
v=0.
o=root 14334 14334 IN IP4 2xx.xxx.xxx.97.
s=session.
c=IN IP4 2xx.xxx.xxx.97.
t=0 0.
m=audio 10380 RTP/AVP 0 8 18 4 3 97 111 5 10 7 110 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:97 iLBC/8000.
a=rtpmap:111 G726-32/8000.
a=rtpmap:5 DVI4/8000.
a=rtpmap:10 L16/8000.
a=rtpmap:7 LPC/8000.
a=rtpmap:110 speex/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -
.
#
T 6x.xxx.xxx.x93:5060 -> 2xx.xxx.xxx.x06:56669 [AP]
SIP/2.0 100 Trying.
FROM: "SIPPHONE1"<sip:089222222@otherproxydomain.org>;tag=as784d1ec2.
TO: <sip:4989111111@proxydomain.org>. CSEQ: 102 INVITE.
CALL-ID: 7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org.
MAX-FORWARDS: 70.
VIA: SIP/2.0/TCP
2xx.xxx.xxx.x6;branch=z9hG4bK96db.2c3c0006.0,SIP/2.0/UDP
2xx.xxx.xxx.97:5060;branch=z9hG4bK52fef574;rport=5060.
CONTENT-LENGTH: 0.
.
##
T 6x.xxx.xxx.x93:5060 -> 2xx.xxx.xxx.x06:56669 [AP]
SIP/2.0 302 Moved Temporarily.
FROM: "SIPPHONE1"<sip:089222222@otherproxydomain.org>;tag=as784d1ec2.
TO: <sip:4989111111@proxydomain.org>;tag=b3addacf4f. CSEQ: 102
INVITE. CALL-ID: 7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org.
MAX-FORWARDS: 70.
VIA: SIP/2.0/TCP
2xx.xxx.xxx.x06;branch=z9hG4bK96db.2c3c0006.0,SIP/2.0/UDP
213.208.4.97:5060;branch=z9hG4bK52fef574;rport=5060.
CONTENT-LENGTH: 0.
SERVER: RTCC/2.0.6017.0.
CONTACT: <sip:4989111111@proxydomain.org:5065;transport=TCP>.
.
##
T 2xx.xxx.xxx.x06:56669 -> 6x.xxx.xxx.x93:5060 [AP]
ACK sip:4989111111@proxydomain.org SIP/2.0.
Via: SIP/2.0/TCP 213.208.4.106;branch=z9hG4bK96db.2c3c0006.0.
From: "SIPPHONE1"
<sip:089222222@otherproxydomain.org>;tag=as784d1ec2. Call-ID:
7fc122590665124f347fb2d956ba4c07(a)otherproxydomain.org.
TO:<sip:4989111111@proxydomain.org>;tag=b3addacf4f. CSeq: 102
ACK. User-Agent: OpenSer (1.1.0-tls (i386/linux)).
Content-Length: 0.
.
######################################
Either i'm lacking understanding of the openser functionalities and sip
redirect in general and simply don't see how this could be accomplished
easier or i guess i need to grab
"sip:4989111111@proxydomain.org:5065;transport=TCP" and extract the
right port and protocol from it to relay it correctly(don't know how
to do that though), i'm not sure. Currently i'm using a simple
t_relay("tcp:6x.xxx.xxx.x93:5065");
to get it working, but i'd like to implement it in a more generic way.
thx for all answers!
best regards
christian
Hi
I am trying to find a way to detect if the Invite URI starts with a *.
How do I match this as ser normally sees * as a wildcard.
Regards
Jon
--
Jon Farmer
Telford, Shropshire, UK
Hello all,
I've just installed openSER on OpenBSD and I'm now trying to get openSER to start and am having trouble. I try openserctl start and here's what i get:
# openserctl start
database engine 'MYSQL' loaded
Control engine 'FIFO' loaded
Starting OpenSER :
\E[37;31mERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed
#
After a bunch of reading it seems as if openserctl is now obsolete (???) and that you should use the init script. I can't find reference for anyone running it like this on OpenBSD. I have found openser.init file in /<src directory>/packaging/gentoo and /<src directory>/packaging/rpm, but i am not sure how to use them on OpenBSD (I'm rather new to OpenBSD... I'm used to the rpm style of things.)
I've searched and searched but can't find any info... So. anybody out there using openSER on OpenBSD? Or anybody know howto (or whereto find the info to) set up the start script for openSER on OpenBSD?
Any help is greatly appreciated!
Thanks,
Ben
On 2/22/07, "Ibrahim Hamouda" <ihamouda(a)itcanint.net> wrote:
>
> There is already a xlog entry in the beginning of the failure route, and no
> it doesn't reach it.
>
>
> Any other ideas?
Have you tried to put the 't_on_failure("1")' statement to the main route
(instead of the invite route) ?
--
Regards,
-vma
.
Hi !
Does someone know the maximum length of an Instant Message I can send with SER ?
When I send an IM wich is longer, I get the message by SER :
2(9833) ERROR: read_line: request line too long
2(9833) ERROR: fifo_server: line expected
2(9833) ERROR: fifo_uac_error: body expected
How can I solve this problem ?
Can I set the maximum length of an IM in any file or something like that ?
Best regards
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