Hello all,
I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose
is to have several interconnections with PSTN.
I configured it like this :
Audiocodes registers as a gateway to the Kamailio, using a dedicated port
(5062).
Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the
proxy.
I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm.
But the audiocodes returns some errors about SIP headers sent by Kamailio :
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
Here you have the example of an INVITE from a SIP phone to the PSTN :
** audiocodes debug **
4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from
77.246.81.132:5060 ----
INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0
Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes>
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0
Via: SIP/2.0/UDP
192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96
From: "Sam" <sip:0123451010@sip.720.fr
<sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0
To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>>
Call-ID: 944d8aec-27503ee6(a)192.168.0.113
CSeq: 102 INVITE
Max-Forwards: 49
Contact: "Sam" <sip:0123451010@77.246.81.162:15170>
Expires: 240
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 281
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp
P-Asserted-Identity: <0123451010>
Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes
v=0
o=- 26933860 26933860 IN IP4 192.168.0.113
s=-
c=IN IP4 77.246.81.133
t=0 0
m=audio 35038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26]
( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26]
( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26]
The outgoing INVITE from Kamailio is exactly the same received by the
AudioCodes.
When I searched over Google, I just found 2 answers about Asterisk /
Audiocodes unsolved problem, but no more informations.
I supposed that the problem is as indicated : " s=- " where source is empty
in place of "NULL" / "0" or something like this ...
Someone can confirm or already met the problem ?
Many thanks all :)
.Sam.
high.all!
i'm wondering if there is any support of uaCSTA in openser (planned)?
i'm just working on the integration of asterisk (*) environment to OCS 2007
environment, having openSER in the middle (mainly for TCP/UDP translation
and smoothing out the protocol deficienes on both sides). in this setup the
* having the openSER in front is talking to the OCS (and vice versa) via the
OCS mediation server, which is moreorless sending standard SIP messages,
which enables normal softphone (integration to *) of the office
communicator. this configuration is already working...
now i'm planning to go for the CTI integration, where there is no OCS
mediation server in between OCS and openSER, doing the translation of
SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why
i'm looking for a CSTA module or perl programm, which is capable of this
functionality.
afaik for the CTI communication there isn't the full complexity of CSTA
needed, just a subset mainly for call setup and call clearing.
anyone having experience on this topic?
thx & cheers
-hugo
Great Ideas for Small Devices
Hugo Koblmueller
Senior Staff Engineer Software Development COMNEON electronic
technology GmbH & Co. OHG
Freistaedter Strasse 400
4040 Linz
Austria
hugo.koblmueller(a)comneon.com
tel:
fax:
mobile:
Skype ID: +43 (5) 1777 - 15730
<http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2
9+1777+%2D+15730&Email=hugo(a)koblmueller.com>
+43 (5) 1777 - 15810
+43 (676) 82051280
<http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676
%29+82051280&Email=hugo(a)koblmueller.com>
drhookson
Want to always have my latest info?
<https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join
&invite=1&lang=en> Want a signature like
<http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en>
this?
Hello,
I'm using the set_dlg_profile function from the dialog module (kamailio 1.4).
I use it to control limit of simultaneous calls to clients.
I'm noticing that after this function is called, even if the call
terminates immediately, it takes some 3 to 4 seconds for the profile
to be cleared. Is there any reason for this?
I can see even if I call unset_dlg_profile on failure_route, the
profile will take that time span to be cleared.
regards,
mayama
Hi,
I am a new user, just installed Kamailio 1.4.3. The Siremis interface will be of great help
but I am getting this error when I try to login.
[2009-02-13 18:22:53 (GMT)] An exception occurred while executing this script:
Error message: #256, No Database information found in the config file.
Script name and line number of error: /var/www/siremis-0.9.1/openbiz/bin/Configuration.php:42
I created the ob_users Table, inserted a user (admin,admin), but I can't seem to figure out what
I did wrong. Maybe its not finding the Config.xml ???
Thanks in advance for any help,
Alberto Furtado
No virus found in this outgoing message.
Checked by AVG - http://www.avg.com
Version: 8.0.176 / Virus Database: 270.10.23/1951 - Release Date: 13/2/2009 06:51
I'm running into a problem with rtpproxy on this point,
quoting from the README:
- - - - - - - - - - -
- after the session has been created, the proxy listens on the port it has
allocated for that session and waits for receiving at least one UDP
packet from each of two parties participating in the call. Once such
packet is received, the proxy fills one of two ip:port structures
associated with each call with source ip:port of that packet. When both
structures are filled in, the proxy starts relaying UDP packets between
parties;
- - - - - - - - - - -
However, a number of clients frequently fail to emit any audio
when originating a call until they hear something from the
TDM gateway, such as ring-back or the called party answering.
So although rtpproxy is receiving a stream of audio, such as
a voice mail menu robot, the calling party can't hear any of
it unless they happen to make some noise or randomly and blindly
press a DTMF key. This seems to be made worse on links with
silence suppression, so there is no background noise to
trigger two-way audio. This is being encountered between Class 4
carriers, so we don't have the option to get someone to
adjust their phone/PBX settings or have them breathe heavier.
Is there a setting adjustment to get rtpproxy to just pass
the RTP packets from directed calling and called sources
even if one party hasn't happened to make noise yet?
I personally don't understand why this requirement for
seeing audio from both sides before starting the flow in
either direction if audio starts coming in even exists.
It seems to have no benefit but is bound to cause this
deadly embrace problem in many situations that may be
beyond the control of the owners of the equipment
passing traffic along to the site where rtpproxy is in
use.
Suggestions? Fix? I have looked at the latest snapshot
of rtpproxy and the README is unchanged since 1.1 so
apparently this behavior is still the same.
Thanks in advance!
HI,
Ive tested the LDAP authentication using Kamailio 1.4.2 and it works
great. However, I'd like prevent the passwords being sent back "in the
clear" from the LDAP server. Is there a way to use the Kamailio ldap
module with TLS/SSL? I already have openssl and openldap-client on the
kamailio box and can successfully use the ldapsearch command line tool
via TLS.
Sven Schulz
Penn State University
Hi All,
I have situation where I need to change the request uri with To header
value.
Scenario:
OpenServ will receive INVITE as follows
INVITE sip:2000@open-ser-ip SIP/2.0.
To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>.
Now I would like to change the request uri as follows:
INVITE sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com> SIP/2.0.
To: "2000"<sip:2000@sip.mydomain.com <sip%3A2000(a)sip.mydomain.com>>.
and will do a lookup location.
Ramu
I have seen lots of default config files where in the reply route only
after checking the message (client_nat_test(1)) fix_nated is called.
Why is not called when the NAT flag is set upong lookup_XX?
In ser-oob I think the reply_route should include the case of a user
called behind a NAT and the reply is not fixed due to some router in
the middle. Will it hurt including fix_nated_contact in the case of
checking the flag?
Friday thoughts...
Samuel
hello,
I have a list of servers in my database.
I need that openser send OPTIONS to servers.
for example, I could have a col 'active' to know if each server is active or
not.
In fact, I need to do an other style of dispatch, and, I cant use the
dispatcher module!
is there a solution for that ?
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
Hi
When i dial #31# 011111with a sip phone i have an INVITE in OPENSER with a number
%2331%23 011111.
Who know how can i send an INVITE whith prefix #31# using a phone or softphone.
thanks,