Re: [OpenSIPS-Users] OpenSIPS - OpenXCAP integration
Tried to get around this by adding IP address individually to trusted_peer list. Seems to do the trick. A config setting like 10.0.0.0/24 does not seem to work. Kindly check this ticket. http://openxcap.org/ticket/90 Regards Sanjeev -Original Message- From: Sanjeev BA [mailto:as290...@samsung.com] Sent: Thursday, October 15, 2009 2:56 PM To: 'OpenSIPS users mailling list' Subject: RE: [OpenSIPS-Users] OpenSIPS - OpenXCAP integration Has anyonbe been able to make OpenXCAP work with OpenSIPS using digest authentication?I am seeing an error in twisted/web2 python script, but I do not know how to fix it. I cannot proceed any further due to the error detailed below. Any help regarding this issue would be highly appreciated. Thanks in advance, Regards Sanjeev -Original Message- From: Sanjeev BA [mailto:as290...@samsung.com] Sent: Thursday, October 15, 2009 8:50 AM To: 'OpenSIPS users mailling list' Subject: RE: [OpenSIPS-Users] OpenSIPS - OpenXCAP integration Yes. Here's the sequence. After changing to digest, I can observe a 500 Internal Server Error in the OpenXCAP server. Logs are provided below. Client Request T 2009/10/15 08:42:04.997913 10.254.140.240:43525 - 10.89.10.235:80 [AP] GET /xcap-root/org.openmobilealliance.xcap-directory/users/sip%3Atester%40imsdem o.com/directory.xml HTTP/1.1. Host: 10.89.10.235:80. Content-length: 0. Connection: keep-alive. . 401 Response ## T 2009/10/15 08:42:05.106696 10.89.10.235:80 - 10.254.140.240:43525 [AP] HTTP/1.1 401 Unauthorized. Date: Wed, 14 Oct 2009 23:42:05 GMT. Content-Length: 141. Content-Type: text/html. WWW-Authenticate: digest nonce=62026475982401434245613, opaque=23870f726b78fb5d872eced92ca0c1dd-NjI0NDQ0MDI2NDc1OTgyNDAxNDM0MjQ1NjE zLDEwLjI1NC4xNDAuMjQwLDEyNTU1NjM3MjU=, realm=imsdemo.com, algorithm=MD5, qop=auth. Server: OpenXCAP/1.1.2.. Date: Wed, 14 Oct 2009 12:17:24 GMT. Content-Length: 141. Content-Type: text/html. WWW-Authenticate: basic realm=imsdemo.com. Server: OpenXCAP/1.1.2. . htmlheadtitleUnauthorized/title/headbodyh1Unauthorized/h1p You are not authorized to access this resource./p/body/html Second GET from client # T 2009/10/15 08:42:05.378179 10.254.140.240:43527 - 10.89.10.235:80 [AP] GET /xcap-root/org.openmobilealliance.xcap-directory/users/sip%3Atester%40imsdem o.com/directory.xml HTTP/1.1. Host: 10.89.10.235:80. Authorization: Digest username=tester, realm=imsdemo.com, qop=auth, nonce=62026475982401434245613, algorithm=MD5, uri=/xcap-root/org.openmobilealliance.xcap-directory/users/sip%3Atester%40i msdemo.com/directory.xml, response=, cnonce=poc123test, opaque=23870f726b78fb5d872eced92ca0c1dd-NjI0NDQ0MDI2NDc1OTgyNDAxNDM0MjQ1NjE zLDEwLjI1NC4xNDAuMjQwLDEyNTU1NjM., nc=0001. Content-length: 0. Connection: keep-alive. 500 Internal Server Error ## T 2009/10/15 08:42:05.387184 10.89.10.235:80 - 10.254.140.240:43527 [AP] HTTP/1.1 500 Internal Server Error. Date: Wed, 14 Oct 2009 23:42:05 GMT. Content-Length: 96. Content-Type: text/plain. Server: OpenXCAP/1.1.2. . An error occurred while processing the request. More information is available in the server log. XCAP Server Error Log. 10.254.140.240 'GET /xcap-root/org.openmobilealliance.xcap-directory/users/sip%3Atester%40imsdem o.com/directory.xml HTTP/1.1' 401 0 141 - - REQUEST headers: Host: 10.89.10.235:80 NjI0NDQ0MDI2NDc1OTgyNDAxNDM0MjQ1NjEzLDEwLjI1NC4xNDAuMjQwLDEyNTU1NjM error: 10.254.140.240 'GET /xcap-root/org.openmobilealliance.xcap-directory/users/sip%3Atester%40imsdem o.com/directory.xml HTTP/1.1' 500 0 96 - - error: REQUEST headers: error: Host: 10.89.10.235:80 error: Authorization: Digest username=tester, realm=imsdemo.com, qop=auth, nonce=62026475982401434245613, algorithm=MD5, uri=/xcap-root/org.openmobilealliance.xcap-directory/users/sip%3Atester%40i msdemo.com/directory.xml, response=, cnonce=poc123test, opaque=23870f726b78fb5d872eced92ca0c1dd-NjI0NDQ0MDI2NDc1OTgyNDAxNDM0MjQ1NjE zLDEwLjI1NC4xNDAuMjQwLDEyNTU1NjM, nc=0001 error: RESPONSE headers: error: Content-Type: text/plain error: RESPONSE: Content-Type: text/plain error: An error occurred while processing the request. More information is available in the server log. error: TRACEBACK (most recent call last): error:File /usr/lib/python2.5/site-packages/twisted/web2/server.py, line 268, in lambda error: d.addCallback(lambda res, req: res.renderHTTP(req), self) error:File /usr/lib/python2.5/site-packages/xcap/authentication.py, line 284, in renderHTTP error: d = self.authenticate(request) error:File /usr/lib/python2.5/site-packages/xcap/authentication.py, line 212, in authenticate error: (request,), None) error:File /usr/lib/python2.5/site-packages/twisted/internet/defer.py, line 186, in addCallbacks error: self._runCallbacks() error: --- exception caught here --- error:File /usr/lib/python2.5/site-packages/twisted/internet/defer.py, line 328, in _runCallbacks error: self.result = callback(self.result, *args,
Re: [OpenSIPS-Users] OpenSIPs behind a firewall
On Jueves, 15 de Octubre de 2009 00:31:10 Daniel Goepp escribió: Route: sip:75.101.136.125;lr Have you declared as a local domain?, if not ... OpenSIP will try to route it, so thats were you have the loop of the ACK. -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Issue with incoming calls.
Hello, I've placed a new testing opensips server inside my network. It has a private modem + router, connected to the sip trunk. When i call outside, it goes great, i see the route goes to the sip trunk and then my mobile phone rings. But when i call inside, something goes wrong. The signal does reach my server, here you can see the ngrep: http://dl.getdropbox.com/u/1382962/log.txt As you can see, (in my eyes) a lot of the same messages to the same server! I've opened in my router the udp port 5060 and let it forward directly to my server. If i close that, nothing reaches my opensips server. Any ideas? -- View this message in context: http://n2.nabble.com/Issue-with-incoming-calls-tp3828026p3828026.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] parallel forking and CANCEL/BYE
Hi, I am using opensips to fork calls to UAs which are registrered from different IPs/Ports. If one UA accepts the INVITE the other UAs will get a CANCEL. Now I have one subscriber with 2 asterisk server which asked me to send a BYE after the CANCEL. Otherwise he wants me to send an BYE which could not be processed correctly on the opensips. I am pretty sure, that this kind of handling would not be RFC conform and so its not possible to handle this inside the routing script. Or did I missed something? BR Uwe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
On Jueves, 15 de Octubre de 2009 10:52:12 Uwe Kastens escribió: Hi, I am using opensips to fork calls to UAs which are registrered from different IPs/Ports. If one UA accepts the INVITE the other UAs will get a CANCEL. Now I have one subscriber with 2 asterisk server which asked me to send a BYE after the CANCEL. Otherwise he wants me to send an BYE which could not be processed correctly on the opensips. I am pretty sure, that this kind of handling would not be RFC conform and so its not possible to handle this inside the routing script. Or did I missed something? You are wright, that will be non-RFC conform ... moreover .. I don't undestand why your subscriber needs that ... because Asterisk is non-RFC conform on lot of things .. but not on that one. -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenXCAP stops with this error.
Hi, OpenXCAP receives a PUT request to add a new contact and shows an error as below. error: Unhandled error in Deferred: Traceback (most recent call last): File /usr/lib/python2.5/site-packages/twisted/protocols/basic.py, line 239, in dataReceived return self.rawDataReceived(data) File /usr/lib/python2.5/site-packages/twisted/web/http.py, line 467, in rawDataReceived self.handleResponseEnd() File /usr/lib/python2.5/site-packages/twisted/web/http.py, line 430, in handleResponseEnd self.handleResponse(b) File /usr/lib/python2.5/site-packages/twisted/web/xmlrpc.py, line 279, in handleResponse self.factory.parseResponse(contents) --- exception caught here --- File /usr/lib/python2.5/site-packages/twisted/web/xmlrpc.py, line 307, in parseResponse response = xmlrpclib.loads(contents)[0][0] File /usr/lib/python2.5/xmlrpclib.py, line 1132, in loads return u.close(), u.getmethodname() File /usr/lib/python2.5/xmlrpclib.py, line 787, in close raise Fault(**self._stack[0]) xmlrpclib.Fault: Fault -506: 'Requested command (pua_publish) is not available!' Stopping factory twisted.web.xmlrpc._QueryFactory instance at 0x891f70c How can I resolve this error? Please help. Regards, Sanjeev ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Shared Registers Accross Servers - Unable to call
Hi, I have two OpenSips servers setup and each server replicates its registrations over to the other server. At present I am not able to call between servers, i.e. a soft phone registers to server A and another to server B, but A can not call server B even though the registrations are present in both databases. Any advice would be much appreciated. Thanks, Ross _ Stay in touch with your friends through Messenger on your mobile http://clk.atdmt.com/UKM/go/174426567/direct/01/___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSips as SMS-GW
Hi , Any one have idea about can OpenSips/Openser be used as SMS-GW for the SMS over IP application ( I have OpenIMSCore Network Configured) . Is there any opensourse implimentaion of SMS over IP ?? Thanks -- --With Regards-- Ajay Pratap Singh Pundhir M.Tech International Institute of Information Technology, Bangalore. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] check_address() causes crash
Hello Jeff,
I managed to get a core dump only when the second parameter of the
check_address is empty.
I added a check for that (rev. 6272), so it shouldn't crash anymore.
Also, you can use $rd as the second parameter only if the domain name is
an ip address, otherwise it won't work.
Thanks!
Irina Stanescu
On Wed, Oct 14, 2009 at 10:43 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hello,
I have the following:
if (check_address(10, $rd, 0, $proto)) {
setflag(7);
}
In many cases, and I can't seem to determine what those cases are, this
causes the system to run very slowly for about 30 seconds, and then
Opensips
exits.
I need to know if the source or destination IP addresses fall into one of
the blocks included in group 10 of the address table.
check_source_address() works great with Irina's fix; this is the
destination
half. It tanks the system.
On the doc page it says:
Transport protocol is either ANY or any valid transport protocol value:
UDP, TCP, TLS, and SCTP.
Is case relevant? Is lowercase just as valid as the uppercase examples?
- Jeff
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[OpenSIPS-Users] rewrite user part in all branches from location lookup
Hi, How can I rewrite the user part of all branches I get back from lookup(location)? Do I need to serialize 1st? BR Uwe -- kiste lat: 54.322684, lon: 10.13586 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] check_address() causes crash
Irina,
In this case $rd was an IP address, freshly loaded from a
lookup(location). But, going forward, you're right, this isn't a great
way to do it. Any suggestions on how to get the IP of the where the message
will go if it were to hit a t_relay(), for example?
I'll update and see if it still crashes.
Thanks,
Jeff
On 10/15/09 9:21 AM, Irina Stanescu ironmi...@gmail.com wrote:
Hello Jeff,
I managed to get a core dump only when the second parameter of the
check_address is empty.
I added a check for that (rev. 6272), so it shouldn't crash anymore.
Also, you can use $rd as the second parameter only if the domain name is an
ip address, otherwise it won't work.
Thanks!
Irina Stanescu
On Wed, Oct 14, 2009 at 10:43 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
Hello,
I have the following:
if (check_address(10, $rd, 0, $proto)) {
setflag(7);
}
In many cases, and I can't seem to determine what those cases are, this
causes the system to run very slowly for about 30 seconds, and then Opensips
exits.
I need to know if the source or destination IP addresses fall into one of
the blocks included in group 10 of the address table.
check_source_address() works great with Irina's fix; this is the destination
half. It tanks the system.
On the doc page it says:
Transport protocol is either ANY or any valid transport protocol value:
UDP, TCP, TLS, and SCTP.
Is case relevant? Is lowercase just as valid as the uppercase examples?
- Jeff
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Re: [OpenSIPS-Users] Shared Registers Accross Servers - Unable to call
What error message are you getting? On Thu, Oct 15, 2009 at 1:41 PM, Ross Beer ross_b...@hotmail.com wrote: Hi, I have two OpenSips servers setup and each server replicates its registrations over to the other server. At present I am not able to call between servers, i.e. a soft phone registers to server A and another to server B, but A can not call server B even though the registrations are present in both databases. Any advice would be much appreciated. Thanks, Ross Stay in touch with your friends through Messenger on your mobile. Learn more. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 1.6 - ready to be released today evening UTC-7
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Re: [OpenSIPS-Users] SIP trace and OpenSIPS-CP 2.0
Hi Bogdan,
I tried values of both 22 (as per the link you sent) and 13 (from google
search results) both still only see incoming requests. No outgoing requests
or reponses.
#-- siptrace db url
modparam(siptrace, db_url,mysql://root:abc...@localhost/opensips)
modparam(siptrace, trace_flag, 22)
### Routing Logic
# main request routing logic
route{
sip_trace();
.
.
.
Is there a description anywhere as to what the value of the flag mean?
Any ideas?
Cheers,
Marc
Bogdan-Andrei Iancu wrote:
Hi Marc,
sip_trace() traces only the current request - that's it; to trace the
whole transaction (the replies also), you need to set the tracing flag
(http://www.opensips.org/html/docs/modules/1.5.x/siptrace.html#id227228)
also:
seflag(NN);
sip_trace();
Regards,
Bogdan
marcher wrote:
Hi Bogdan,
Again, thanks for the help.
I want to capture all ingress and egress traffic for now. So I put
sip_trace(); at the very start of the main request routing logic. This
worked great for incoming methods REGISTER, INVITE, OPTIONS, ACK and BYE.
When tracing is enabled via opensips-cp I see the methods, and can click
on
Call to expand to see all the messages.
However, I don't see any egress messages, either outgoing requests or
1xx,
2xx, 4xx responses from opensips in the opensips-cp table.
I tried adding sip_trace() to opensips.cfg as follows but no dice. I
guess I
didn't understand what your suggestion just before sending the request
out
actually means in terms of opensips.cfg
route[1] {
# for INVITEs enable some additional helper routes
if (is_method(INVITE)) {
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
sip_trace();
if (!t_relay()) {
sl_reply_error();
};
exit;
}
I completely understand about not having a standard config that covers
all
the possibilities opensips-cp can control.
I do wish to use drouting such that I can conveniently add new phone
number
prefixes and have them route to gateways via opensips-cp. But I also wish
this functionality to work with calls to endpoints directly registered
with
opensips.
If I add the do_routing logic before the usrloc lookup logic, I get a 503
for a call to a valid registered endpoint.
xlog(-Doing routing\n);
if (!do_routing(1)) {
sl_send_reply(503,No destination available);
exit;
}
xlog(-gw attr is $avp(s:dr_attrs)\n);
xlog(-ruri is $ru\n);
if (!lookup(location)) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply(404, Not Found);
exit;
case -2:
sl_send_reply(405, Method Not
Allowed);
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
If I add the do_routing logic after the usrloc lookup logic, I get a 404
from case -3 for a call destined for a gateway.
How can I setup my opensips.cfg such that calls to registered endpoints
and
calls to gateway hosted numbers work in conjunction?
Cheers,
Marc
Bogdan-Andrei Iancu wrote:
Hi Marc,
marcher wrote:
Hi Bogdan,
I appreciate you taking the time to answer my basic questions in
getting
opensips-cp functional with my opensips implementation.
I had read the link you included, but its still not clear to me where
the
sip_trace function should be called within the opensips config file.
there is not special place for it - you need to call the sip_trance()
and set the trace flag when you process the SIP requests - you can do
this in the very beginning of the script or just before sending the
request out - it is up to you and up to what kind of traffic you want to
trace.
For example, if you want to trace only calls to your local subscribers,
you can add the sip_trace() in the if (ruri==myself) {} block.
My opensips config file is very straightforward, based heavily on the
distribution sample, but adding piecemeal the config necessary to
integrate
opensips-cp (mi_xmlrpc, dialplan, drouting and siptrace modules to
date)
I
also wish to add in SIP trunking gateways using drouting.
you do not need to put in the opensips script all the functionalities
required by opensips-cp. You can select in opensips-cp only the tools
you find useful in your opensips script. If you do not need drouting in
opensips cfg, simply remove the drouting tool from CP.
To that end, I am also struggling with the correct opensips config to
implement drouting such that it works in tandem with the
Re: [OpenSIPS-Users] OpenSIPS - OpenXCAP integration
2009/10/15 Sanjeev BA as290...@samsung.com:
/usr/lib/python2.5/site-packages/twisted/web2/auth/digest.py, line 257, in
verifyOpaque
error: key = opaqueParts[1].decode('base64')
error: File /usr/lib/python2.5/encodings/base64_codec.py, line 43, in
base64_decode
error: output = base64.decodestring(input)
error: File base64.py, line 321, in decodestring
error: return binascii.a2b_base64(s)
error: binascii.Error: Incorrect padding
Interesting error. I suggest you to open a ticket with it in OpenXCAP tracker.
--
Iñaki Baz Castillo
i...@aliax.net
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[OpenSIPS-Users] Shared Registers Accross Servers - Unable to call
I have successfully replicated the actual registrations, however it is not possible to call a soft phone that is registered to a different server. For example a phone that registers to server 'A' can not call server 'B'. I have tried setting a header with where the registration was received using 'add_sock_hdr' however this produces an error on the other server saying that the socket is not local. Any advice on the correct way of doing this would be much appreciated. The ideal solution would allow 'Messages' and 'Calls' to pass to the soft phone no matter what server the phone registers too. Thanks, Ross What error message are you getting? On Thu, Oct 15, 2009 at 1:41 PM, Ross Beer ross_b...@hotmail.com wrote: Hi, I have two OpenSips servers setup and each server replicates its registrations over to the other server. At present I am not able to call between servers, i.e. a soft phone registers to server A and another to server B, but A can not call server B even though the registrations are present in both databases. Any advice would be much appreciated. Thanks, Ross _ Use Windows Live Messenger for free on selected mobiles http://clk.atdmt.com/UKM/go/174426567/direct/01/___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs behind a firewall
Thanks Raul and Jeff, I now have successfully got signaling working...now to battle with the media more :) -dg 2009/10/15 Raúl Alexis Betancor Santana r...@dimension-virtual.com On Jueves, 15 de Octubre de 2009 00:31:10 Daniel Goepp escribió: Route: sip:75.101.136.125;lr Have you declared as a local domain?, if not ... OpenSIP will try to route it, so thats were you have the loop of the ACK. -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ 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] OpenSIPS - OpenXCAP integration
El Martes, 13 de Octubre de 2009, Sanjeev BA escribió: I use Pytho-twisted 8.2.0-1ubuntu1on Ubuntu 8.04.3 Have you checked if your issue is the same as the report?: http://openxcap.org/ticket/121 Thanks. -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] FW: An Old OpenSER Error For A New OpenSIPS User
No, it is not in RC file, but in opensips.cfg - there you need to configure that domains should be considered as local. Regards, Bogdan Kemp, Larry wrote: Bogdan, Assuming you are talking about the file /etc/openser/openserctlrc; at the top of that script it shows: ## your SIP domain SIP_DOMAIN=usmetrotel.com Would I still need to place: alias=usmetrotel.com or ALIAS=USMEROTEL.COM in there and restart the daemon? All the variables look to be all CAPS. Not certain if the /sbin/openserctlrc cares or not. Right underneath the SIP_DOMAIN=usmetrotel.com line in /etc/openser/openser.cfg I added ALIAS=usmetrotel.com and restarted the service. It restarted okay so this did not apparently break the OpenSER service. I'll try registering the X-Lite phones and let you know what happens. Larry Kemp Network Engineer U.S. Metropolitan Telecom, LLC Toll Free: (877) 244-0242 (239) 333-4150 Desk: (239) 325-4105 Ext 263 Email: larry.k...@usmetrotel.com Address: 24017 Production Circle Bonita Springs, FL 34135 Certified Adtran ASP/ATSA Internetworking, ASP/ATSA IP Telephony, ASP/ATSA Wireless -Original Message- From: Kemp, Larry Sent: Thursday, October 15, 2009 10:57 AM To: 'Bogdan-Andrei Iancu' Subject: RE: [OpenSIPS-Users] FW: An Old OpenSER Error For A New OpenSIPS User Bogdan, Thanks always for your very much appreciated help! Is that in the /etc/openser/openser.cfg in /etc/openser/openserctl or some other file? In /etc/openser/openser.cfg a search for the word alias shows results at Line 151 at Col 25 as: lookup(aliases); And then again at line 153 Column 53 as: append_hf (P-hint: outbound alias\r\n); In /etc/openser/openserctlrc the word alias appears several times commented out at Line 31 and then two lines down as a variable: ALIASES+TYPE=DB It seems more like it would go in this file but I have no idea, I am not a Kung Fu Master with C or OpenSER either. I am having a tough go of it. Larry Kemp Network Engineer U.S. Metropolitan Telecom, LLC Toll Free: (877) 244-0242 (239) 333-4150 Desk: (239) 325-4105 Ext 263 Email: larry.k...@usmetrotel.com Address: 24017 Production Circle Bonita Springs, FL 34135 Certified Adtran ASP/ATSA Internetworking, ASP/ATSA IP Telephony, ASP/ATSA Wireless -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, October 14, 2009 11:53 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] FW: An Old OpenSER Error For A New OpenSIPS User Hi Larry, most probably your opensips does not recognize the usmetrotel.com as a local domain, so it is keep forwarding the request to itself. adding something like: alias=usmetrotel.com in your script should solve the problem. Regards, Bogdan Kemp, Larry wrote: After Bogdan helped me to correct my errors I was able to manually register extension 1000 on my OpenSER 1000 by entering: /sbin/openstlrc add 1000 password 1...@mydomain.com Then when I went to register my soft-phone to the OpenSER as extension 1000. On my soft-phone (X-Lite 3.0 running on WinXP) I got the message 503 - Message Too Big. I corrected this by editing the file on the OpenSER /etc/openser/openser.cfg (at line 96 Column 24) increasing the default value from 2048 to 8192. Then I got the message 483 - Too Many Hops on my soft phone. So back on the OpenSER I edited /etc/openser/openser.cfg (at Line 91 Column 40) and increased the default value from 10 to 100. I still got the 488 - Too Many Hops message on the client. So I ran a SIP capture on the OpenSER by using the command: ngrep -p -q -W byline port 5060 test.txt Here is the output of that capture: interface: eth0 (10.100.100.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) U 208.76.137.2:31215 - 10.100.100.199:5060 REGISTER sip:usmetrotel.com SIP/2.0. Via: SIP/2.0/UDP 208.76.137.2:31215;branch=z9hG4bK-d8754z-1271d5783f053154-1---d8754z-;rport. Max-Forwards: 70. Contact: sip:1...@208.76.137.2:31215;rinstance=705236a41a2fcc6d. To: Larry-Kempsip:1...@usmetrotel.com. From: Larry-Kempsip:1...@usmetrotel.com;tag=863d2266. Call-ID: OTljYmExZTU2ZjI0ODU2NDlhYzYwYmFmMmE3OTkxYjU.. CSeq: 1 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1103k stamp 53621. Content-Length: 0. . U 10.100.100.199:5060 - 208.76.137.2:31215 SIP/2.0 483 Too Many Hops. Via: SIP/2.0/UDP 208.76.137.2:31215;branch=z9hG4bK-d8754z-1271d5783f053154-1---d8754z-;rport=31215. To: Larry-Kempsip:1...@usmetrotel.com;tag=329cfeaa6ded039da25ff8cbb8668bd2.cb9e. From: Larry-Kempsip:1...@usmetrotel.com;tag=863d2266. Call-ID:
Re: [OpenSIPS-Users] OpenSIPs behind a firewall
How do I declare a domain as local? I was told this was done in the /etc/openser/openser.cfg. Larry Kemp Network Engineer U.S. Metropolitan Telecom, LLC From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Daniel Goepp Sent: Thursday, October 15, 2009 12:36 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPs behind a firewall Thanks Raul and Jeff, I now have successfully got signaling working...now to battle with the media more :) -dg 2009/10/15 Raúl Alexis Betancor Santana r...@dimension-virtual.commailto:r...@dimension-virtual.com On Jueves, 15 de Octubre de 2009 00:31:10 Daniel Goepp escribió: Route: sip:75.101.136.125;lr Have you declared as a local domain?, if not ... OpenSIP will try to route it, so thats were you have the loop of the ACK. -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ Users mailing list Users@lists.opensips.orgmailto: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] OpenSIPs behind a firewall
I fixed the problem on my server by doing: opensipsctl domain add 75.101.136.125 -dg On Thu, Oct 15, 2009 at 11:16 AM, Kemp, Larry larry.k...@usmetrotel.comwrote: How do I declare a domain as local? I was told this was done in the /etc/openser/openser.cfg. Larry Kemp Network Engineer U.S. Metropolitan Telecom, LLC *From:* users-boun...@lists.opensips.org [mailto: users-boun...@lists.opensips.org] *On Behalf Of *Daniel Goepp *Sent:* Thursday, October 15, 2009 12:36 PM *To:* OpenSIPS users mailling list *Subject:* Re: [OpenSIPS-Users] OpenSIPs behind a firewall Thanks Raul and Jeff, I now have successfully got signaling working...now to battle with the media more :) -dg 2009/10/15 Raúl Alexis Betancor Santana r...@dimension-virtual.com On Jueves, 15 de Octubre de 2009 00:31:10 Daniel Goepp escribió: Route: sip:75.101.136.125;lr Have you declared as a local domain?, if not ... OpenSIP will try to route it, so thats were you have the loop of the ACK. -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] FW: An Old OpenSER Error For A New OpenSIPS User
Hi Bogdan, Thanks for your help very very much. I edited /etc/openser/openser.cfg adding alias=usmetrotel.com and restarted OpenSER. It crashes the openser daemon. I cannot place calls, but I can see the SIP messages from the softphones when I do the ngrep on the OpenSER. I statically entered the extensions 1000 and 1001 in the OpenSER via the: /sbin/openserctl add 1000 password 1...@usmetrotel.com /sbin/openserctl add 1001 password 1...@usmetrotel.com Commands, as the book states to do. So when I perform an openserctl ul show I expected to see my two statically entered users (1000 and 1001) show up. But here is what comes back: database engine MYSQL loaded Control engine FIFO loaded entering fifo_cmd ul_dump 200 OK Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=0 max_slot=0 FIFO command was: :ul_dump:openser_receiver_31972 If I try to run: /sbin/openserctl add 1000 password 1...@usmetrotel.com or /sbin/openserctl add 1001 password 1...@usmetrotel.com again, the system tells me: database engine MYSQL loaded Control engine 'FIFO' loaded is_user: user counter=1 ERROR: user 1000 already exists. Yet if I try to remove it using: /sbin/openserctl rm 1000 the command hangs with this output: database engine MYSQL loaded Control engine FIFO loaded is_user: user counter=1 Also, when I run the command openserctl domain showdb it displays: +++-+ | id | domain | last_modified | +++-+ | 1 | usmetrotel.com | -00-00 00:00:00 | +++-+ So I am not sure if the server knows that it hosts the domain usmetrotel.com and the extensions 1000 and 1001 for that domain or not. Not sure how to tell the openser.cfg that usmetrotel.com is a local domain and make my soft phones communicate with the OpenSER. According to everything I have read in the book, it states it should be working. Perhaps the next section on page 101 Enhancing The Script and page 102 Managing Multiple Domains will make thing s more clear. I guess I was expecting the calls to work as the book states they should be working and able to call each other several times by this point. Again I appreciate the help, very much. Thanks LK -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Thursday, October 15, 2009 1:40 PM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] FW: An Old OpenSER Error For A New OpenSIPS User No, it is not in RC file, but in opensips.cfg - there you need to configure that domains should be considered as local. Regards, Bogdan Kemp, Larry wrote: Bogdan, Assuming you are talking about the file /etc/openser/openserctlrc; at the top of that script it shows: ## your SIP domain SIP_DOMAIN=usmetrotel.com Would I still need to place: alias=usmetrotel.com or ALIAS=USMEROTEL.COM in there and restart the daemon? All the variables look to be all CAPS. Not certain if the /sbin/openserctlrc cares or not. Right underneath the SIP_DOMAIN=usmetrotel.com line in /etc/openser/openser.cfg I added ALIAS=usmetrotel.com and restarted the service. It restarted okay so this did not apparently break the OpenSER service. I'll try registering the X-Lite phones and let you know what happens. -Original Message- From: Kemp, Larry Sent: Thursday, October 15, 2009 10:57 AM To: 'Bogdan-Andrei Iancu' Subject: RE: [OpenSIPS-Users] FW: An Old OpenSER Error For A New OpenSIPS User Bogdan, Thanks always for your very much appreciated help! Is that in the /etc/openser/openser.cfg in /etc/openser/openserctl or some other file? In /etc/openser/openser.cfg a search for the word alias shows results at Line 151 at Col 25 as: lookup(aliases); And then again at line 153 Column 53 as: append_hf (P-hint: outbound alias\r\n); In /etc/openser/openserctlrc the word alias appears several times commented out at Line 31 and then two lines down as a variable: ALIASES+TYPE=DB It seems more like it would go in this file but I have no idea, I am not a Kung Fu Master with C or OpenSER either. I am having a tough go of it. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Wednesday, October 14, 2009 11:53 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] FW: An Old OpenSER Error For A New OpenSIPS User Hi Larry, most probably your opensips does not recognize the usmetrotel.com as a local domain, so it is keep forwarding the request to itself. adding something like: alias=usmetrotel.com in your script should solve the problem. Regards, Bogdan Kemp, Larry wrote: After Bogdan helped me to correct my errors I was able to manually register extension 1000 on my OpenSER 1000 by entering: /sbin/openstlrc add
[OpenSIPS-Users] critical qm_debug_frag error
Hello, On 1.6 rev 6274 I see this error occur on the process that handles a reinvite from an Asterisk box after the call goes 200 OK. Odd, because I've t_relayed calls to this box before without a problem. But that was several revisions ago, and several script versions ago as well. CRITICAL:core:qm_debug_frag: qm_*: fragm. 0x82abf04 (address 0x82abf1c) end overwritten(c0c0c020, abcdefed)! A more complete log from debug=4 is attached. There was nothing more after the CRITICAL error from this process before the whole thing crashed. The error seems to be 100% reproducible. - Jeff sip1-crash3.log Description: sip1-crash3.log ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] critical qm_debug_frag error
Never mind... Turns out I had complied revision 6274 but never installed it. Installing it fixed the crashes. - Jeff On 10/15/09 3:29 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Hello, On 1.6 rev 6274 I see this error occur on the process that handles a reinvite from an Asterisk box after the call goes 200 OK. Odd, because I've t_relayed calls to this box before without a problem. But that was several revisions ago, and several script versions ago as well. CRITICAL:core:qm_debug_frag: qm_*: fragm. 0x82abf04 (address 0x82abf1c) end overwritten(c0c0c020, abcdefed)! A more complete log from debug=4 is attached. There was nothing more after the CRITICAL error from this process before the whole thing crashed. The error seems to be 100% reproducible. - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] critical qm_debug_frag error
Jeff, do not do that again! :) you scared me ! I though - auch a new bug in hours before the release !! :) Regards, Bogdan Jeff Pyle wrote: Never mind... Turns out I had complied revision 6274 but never installed it. Installing it fixed the crashes. - Jeff On 10/15/09 3:29 PM, Jeff Pyle jp...@fidelityvoice.com wrote: Hello, On 1.6 rev 6274 I see this error occur on the process that handles a reinvite from an Asterisk box after the call goes 200 OK. Odd, because I've t_relayed calls to this box before without a problem. But that was several revisions ago, and several script versions ago as well. CRITICAL:core:qm_debug_frag: qm_*: fragm. 0x82abf04 (address 0x82abf1c) end overwritten(c0c0c020, abcdefed)! A more complete log from debug=4 is attached. There was nothing more after the CRITICAL error from this process before the whole thing crashed. The error seems to be 100% reproducible. - Jeff ___ 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] Shared Registers Accross Servers - Unable to call
Hi Ross, Is the B user behind a NAT ? Check the following steps (when calling from A to B) 1) Invites gets to the A registrar 2) Invite is sent to the B registrar 3) B registrar sends the the Invite to B Also check the reply code you get on A. Regards, Bogdan Ross Beer wrote: I have successfully replicated the actual registrations, however it is not possible to call a soft phone that is registered to a different server. For example a phone that registers to server 'A' can not call server 'B'. I have tried setting a header with where the registration was received using 'add_/sock_/hdr' however this produces an error on the other server saying that the socket is not local. Any advice on the correct way of doing this would be much appreciated. The ideal solution would allow 'Messages' and 'Calls' to pass to the soft phone no matter what server the phone registers too. Thanks, Ross What error message are you getting? On Thu, Oct 15, 2009 at 1:41 PM, Ross Beer ross_b...@hotmail.com wrote: Hi, I have two OpenSips servers setup and each server replicates its registrations over to the other server. At present I am not able to call between servers, i.e. a soft phone registers to server A and another to server B, but A can not call server B even though the registrations are present in both databases. Any advice would be much appreciated. Thanks, Ross Use Windows Live Messenger for free on selected mobiles. Learn more. http://clk.atdmt.com/UKM/go/174426567/direct/01/ ___ 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
[OpenSIPS-Users] Registration problem?
Hello all, I'm having a random registration problem I haven't had a chance to fight yet. Right now, 100% of my users have their SIP gateways on static, public IP addresses. We use static entries in the location table to route calls to these locations presently. We're wanting to deploy Linksys ATAs to smaller locations, these units would be behind NAT with dynamic addresses. I have MediaProxy all setup and happy. Outbound calls work great. However, I'm a little confused about how dynamic registrations work in OpenSIPS. My Linksys devices re-register every 5 minutes or so. When this happens, I end up with multiple entries in the subscriber table. I'd really only like to keep the most up-to-date registration. Any thoughts? My second and much larger problem is that I'm using multiple servers, with db_mode set to 3 on the usrloc module (performance is not a concern at the moment), and my servers randomly can't seem to locate the most up-to-date registration. It feels almost as if the server that received the registration knows about it (it's in the DB table immediately, as well) but the other boxes don't pick up on this registration for a while. Shouldn't a value of 3 mean always check the database? It doesn't seem to work as expected. Of note, stopping/starting opensips makes the new registration available immediately. Any advice that would put me in the right direction would be appreciated. Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Additional info on potential registration issue
I'm getting this over and over in my syslog: WARNING:usrloc:get_all_db_ucontacts: non-local socket udp:HI.DDE.N.12:5060...ignoring Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Issue with incoming calls.
Peter, I am new to OpenSIPs, but from your ngrep, there seems to be a loop of INVITE msgs. U 10.0.100.99:5060 - 90.145.5.83:5060 INVITE sip:0031851110...@90.145.5.83 SIP/2.0. U 90.145.5.83:5060 - 10.0.100.99:5060 INVITE sip:0031851110...@90.145.5.83 SIP/2.0. So it looks like something wrong in script handling outbound -- inbound call. If you can paste your config, someone should be able to check it. :) Regards, Leon -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Peter den Hartog Sent: Thursday, 15 October 2009 7:47 PM To: users@lists.opensips.org Subject: [OpenSIPS-Users] Issue with incoming calls. Hello, I've placed a new testing opensips server inside my network. It has a private modem + router, connected to the sip trunk. When i call outside, it goes great, i see the route goes to the sip trunk and then my mobile phone rings. But when i call inside, something goes wrong. The signal does reach my server, here you can see the ngrep: http://dl.getdropbox.com/u/1382962/log.txt As you can see, (in my eyes) a lot of the same messages to the same server! I've opened in my router the udp port 5060 and let it forward directly to my server. If i close that, nothing reaches my opensips server. Any ideas? -- View this message in context: http://n2.nabble.com/Issue-with-incoming-calls-tp3828026p3828026.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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs behind a firewall
On Thu, Oct 15, 2009 at 8:16 PM, Kemp, Larry larry.k...@usmetrotel.com wrote: How do I declare a domain as local? I was told this was done in the /etc/openser/openser.cfg. If you use the domain module add it to the domain table, or just add alias=mydomain.com to your cfg file. -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Registration problem?
My Linksys devices re-register every 5 minutes or so. When this happens, I end up with multiple entries in the subscriber table. I'd really only like to keep the most up-to-date registration. Any thoughts? Subscriber? Do you mean the location table? You should only have one location if you sent a re-REGISTER,, right after the original register expired... so do you have any SIP trace on that? Regards, -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP trace and OpenSIPS-CP 2.0
Hi Marc,
Do not use random values :) - the flag you configure must also be used:
#-- siptrace db url
modparam(siptrace, db_url,mysql://root:abc...@localhost/opensips)
modparam(siptrace, trace_flag, 22)
### Routing Logic
# main request routing logic
route{
setfalg(22);
sip_trace();
.
.
.
}
regards,
bogdan
marcher wrote:
Hi Bogdan,
I tried values of both 22 (as per the link you sent) and 13 (from google
search results) both still only see incoming requests. No outgoing requests
or reponses.
#-- siptrace db url
modparam(siptrace, db_url,mysql://root:abc...@localhost/opensips)
modparam(siptrace, trace_flag, 22)
### Routing Logic
# main request routing logic
route{
sip_trace();
.
.
.
Is there a description anywhere as to what the value of the flag mean?
Any ideas?
Cheers,
Marc
Bogdan-Andrei Iancu wrote:
Hi Marc,
sip_trace() traces only the current request - that's it; to trace the
whole transaction (the replies also), you need to set the tracing flag
(http://www.opensips.org/html/docs/modules/1.5.x/siptrace.html#id227228)
also:
seflag(NN);
sip_trace();
Regards,
Bogdan
marcher wrote:
Hi Bogdan,
Again, thanks for the help.
I want to capture all ingress and egress traffic for now. So I put
sip_trace(); at the very start of the main request routing logic. This
worked great for incoming methods REGISTER, INVITE, OPTIONS, ACK and BYE.
When tracing is enabled via opensips-cp I see the methods, and can click
on
Call to expand to see all the messages.
However, I don't see any egress messages, either outgoing requests or
1xx,
2xx, 4xx responses from opensips in the opensips-cp table.
I tried adding sip_trace() to opensips.cfg as follows but no dice. I
guess I
didn't understand what your suggestion just before sending the request
out
actually means in terms of opensips.cfg
route[1] {
# for INVITEs enable some additional helper routes
if (is_method(INVITE)) {
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
}
sip_trace();
if (!t_relay()) {
sl_reply_error();
};
exit;
}
I completely understand about not having a standard config that covers
all
the possibilities opensips-cp can control.
I do wish to use drouting such that I can conveniently add new phone
number
prefixes and have them route to gateways via opensips-cp. But I also wish
this functionality to work with calls to endpoints directly registered
with
opensips.
If I add the do_routing logic before the usrloc lookup logic, I get a 503
for a call to a valid registered endpoint.
xlog(-Doing routing\n);
if (!do_routing(1)) {
sl_send_reply(503,No destination available);
exit;
}
xlog(-gw attr is $avp(s:dr_attrs)\n);
xlog(-ruri is $ru\n);
if (!lookup(location)) {
switch ($retcode) {
case -1:
case -3:
t_newtran();
t_reply(404, Not Found);
exit;
case -2:
sl_send_reply(405, Method Not
Allowed);
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(1);
If I add the do_routing logic after the usrloc lookup logic, I get a 404
from case -3 for a call destined for a gateway.
How can I setup my opensips.cfg such that calls to registered endpoints
and
calls to gateway hosted numbers work in conjunction?
Cheers,
Marc
Bogdan-Andrei Iancu wrote:
Hi Marc,
marcher wrote:
Hi Bogdan,
I appreciate you taking the time to answer my basic questions in
getting
opensips-cp functional with my opensips implementation.
I had read the link you included, but its still not clear to me where
the
sip_trace function should be called within the opensips config file.
there is not special place for it - you need to call the sip_trance()
and set the trace flag when you process the SIP requests - you can do
this in the very beginning of the script or just before sending the
request out - it is up to you and up to what kind of traffic you want to
trace.
For example, if you want to trace only calls to your local subscribers,
you can add the sip_trace() in the if (ruri==myself) {} block.
My opensips config file is very straightforward, based heavily on the
distribution sample, but adding piecemeal the config necessary to
integrate
opensips-cp (mi_xmlrpc, dialplan, drouting and siptrace modules to
date)
I
also wish to add in SIP trunking gateways using
Re: [OpenSIPS-Users] Registration problem?
Yes, location table, sorry. I appear to have solved my problem, albeit entirely from guessing. I had usrloc's matching_mode set to 1. I'm not 100% confident I understand the difference between using the call ID to match on registration, but I do know I went from perhaps a 50% chance to receive a call every few minutes to not having a single failure the rest of the day. I'd love to know what I fixed. Advice? Jeff -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Saúl Ibarra Sent: Thursday, October 15, 2009 6:37 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Registration problem? My Linksys devices re-register every 5 minutes or so. When this happens, I end up with multiple entries in the subscriber table. I'd really only like to keep the most up-to-date registration. Any thoughts? Subscriber? Do you mean the location table? You should only have one location if you sent a re-REGISTER,, right after the original register expired... so do you have any SIP trace on that? Regards, -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ 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] Registration problem?
It sems like re-REGISTERS where seen as new ones. Have a look at the contact matching doc: http://www.opensips.org/html/docs/modules/devel/usrloc.html#contact-matching-algs It would also be nice to see the first REGISTER and compare it to the re-REGISTER so we could undertand why this happened. -- /Saúl http://www.saghul.net | http://www.sipdoc.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS 1.6.0 - a newmajor release is out
After almost 7 months from the last major release (1.5.0), OpenSIPS evolves with a new major release, 1.6.0. *OpenSIPS 1.6.0* comes with several critical improvements (DB area, dialog support), but also with new functionalities (like memchaced support, B2Bua implementation, virtual DB URLs, STUN server, JSON support, new AAA/RADIUS interface, etc). A complete compilation with all the additions and improvements for *OpenSIPS 1.6.0 *release is available under: http://www.opensips.org/Main/Ver160 Many thanks to all the people who got involved in this release (and in the overall OpenSIPS project) and contributed with code, with testing and debuging, with patches or reports, with support on the lists, help with packaging and documentation. I will avoid listing names, not because they do not deserve it, but simply because it will impossible to list list everybody here and I do not what to be unfair with some of them (because I simply forgot a name or because of the limited space). But nevertheless, I want to thanks you all for out great job and not in my behalf, but in the behalf of people who will find this piece of software a useful tool. The full Changelog is available here: http://opensips.org/pub/opensips/1.6.0/src/ChangeLog To get the *OpenSIPS 1.6.0* version, see : * website page: http://www.opensips.org/Resources/Downloads * SF project: https://sourceforge.net/project/showfiles.php?group_id=232389package_id=281827release_id=670379 Note that for the moment only the source tarballs are available. The packages (debs,rpms, etc) will be generated starting now. If anybody can help in generating packages for different distros or architectures, please let me know and I will upload them on the website. Best regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] multiple opensips_radius instances in CDRTool
Adrian, I added the new data source (also with opensips_radius but a new name). I go to configure the login account in question and I see the new data sources. I make sure everything is selected. On the CDR query screen, I see only the second (new) data source available in the dropdown. The first one is gone. If I comment out the second data source, I see only the first once again. ? - Jeff On 10/13/09 2:57 PM, Adrian Georgescu a...@ag-projects.com wrote: You add the new data source to the allowed data sources for the login account in question. -- Adrian On Oct 13, 2009, at 11:55 AM, Jeff Pyle wrote: Hello, I have a working CDRTool 6.8.0 configuration. It looks at one Opensips radius data source, and one Asterisk data source. I'm attempting to add a second Opensips radius data source. I'm having trouble. If I add a second opensips_radius instance, the second one seems to overwrite the first. It works, but the first is gone. If I add it with a different name, say, opensips_radius_sun, it doesn't show up in the available Data source drop-down box. It does, however, appear as an available data source in the permissions section when editing a user. What is the proper way to do this? Thanks, Jeff ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] multiple opensips_radius instances in CDRTool
Did you select both data sources? Be careful with that field, is a multiple selection box, if you just simple click one, the others get de-selected. Use Control or Option key depending on your computer to add to the selection. -- Adrian On Oct 15, 2009, at 8:24 PM, Jeff Pyle wrote: Adrian, I added the new data source (also with opensips_radius but a new name). I go to configure the login account in question and I see the new data sources. I make sure everything is selected. On the CDR query screen, I see only the second (new) data source available in the dropdown. The first one is gone. If I comment out the second data source, I see only the first once again. ? - Jeff On 10/13/09 2:57 PM, Adrian Georgescu a...@ag-projects.com wrote: You add the new data source to the allowed data sources for the login account in question. -- Adrian On Oct 13, 2009, at 11:55 AM, Jeff Pyle wrote: Hello, I have a working CDRTool 6.8.0 configuration. It looks at one Opensips radius data source, and one Asterisk data source. I'm attempting to add a second Opensips radius data source. I'm having trouble. If I add a second opensips_radius instance, the second one seems to overwrite the first. It works, but the first is gone. If I add it with a different name, say, opensips_radius_sun, it doesn't show up in the available Data source drop-down box. It does, however, appear as an available data source in the permissions section when editing a user. What is the proper way to do this? Thanks, Jeff ___ 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 ___ 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] Dialplan issues: simple strip\prefix cases
I don't mean to point out the obvious here, but \4 doesn't exist. that'd refer to the 4th set of parens in your expression. See your exp, you only have 2. As a completely separate issue, I also suspect that ^4224+ is wrong. I'm not an expert in regex, but I think that means starts with 4224 and then continues with 1 or more '4' . You probably mean ^4224.+ -Brett On Thu, Oct 15, 2009 at 4:38 PM, Sebastian Sastre sebast...@next-ip.com wrote: Hello, I’m trying to use the Dialplan module to strip the prefix of my calls. On the database I have. Dpid = 1 Pr = 0 match_op = 1 match_exp = ^4224+ match_len = 0 subst_exp = ^(4224)(.+) repl_exp = \4 Oct 15 17:25:32 RouteMe /usr/sbin/opensips[30516]: INFO:dialog:mod_init: Dialog module - initializing Oct 15 17:25:32 RouteMe /usr/sbin/opensips[30516]: INFO:dialplan:mod_init: initializing module... Oct 15 17:25:32 RouteMe /usr/sbin/opensips[30516]: ERROR:dialplan:build_rule: repl_exp uses a non existing subexpression Oct 15 17:25:32 RouteMe /usr/sbin/opensips[30516]: ERROR:dialplan:init_db_data: failed to load database data Oct 15 17:25:32 RouteMe /usr/sbin/opensips[30516]: ERROR:dialplan:mod_init: could not initialize data Oct 15 17:25:32 RouteMe /usr/sbin/opensips[30516]: ERROR:core:init_mod: failed to initialize module dialplan Oct 15 17:25:32 RouteMe /usr/sbin/opensips[30516]: ERROR:core:main: error while initializing modules Thanks ___ 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
