[OpenSIPS-Users] Prepaid_lock problem in CDRTool
Hi all,
I have installed CDRtool version 6.9.9. *I have set the Prepaid_lock to
1* to disable multiple calls from a single prepaid account. But the parallel
call gets established. How can I solve this problem
$RatingEngine=array(socketIP = xxx.xxx.xxx.xxx,
socketPort = 9024,
cdr_source = opensips_radius,
allow = array
('10.','xxx.xxx.xxx.'),
prepaid_lock = 1,
priceDenominator= 1, // Rates units (global
setting)
--
Thanking You,
Ashwini BR Naidu
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Re: [OpenSIPS-Users] Opensips 1.6.1 dos not start on Ubuntu 9.1 64bits
Bogdan, I started the Opensips with debug=6 and there weren't changes on the initialization log, I've used the strace (follow trace attached), I'm not an expert in the strace, but the only thing that i could see was a message 14752 read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83 after that the shutdown process started. I already double checked the user and password of the mysql, and it's right in the script, and in the opensipsctlrc, the opensipsctl is working properly. Could somebody take a look in the stracelog to help me? Antonio Anderson M. Souza Voice Technology http://www.antonioams.com On Wed, Dec 30, 2009 at 5:17 PM, Antonio Anderson M. de Souza antonio...@gmail.com wrote: Bogdan, I started the Opensips with debug=6 and there weren't changes on the initialization log, I've used the strace (follow trace attached), I'm not an expert in the strace, but the only thing that i could see was a message 14752 read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83 after that the shutdown process started. I already double checked the user and password of the mysql, and it's right in the script, and in the opensipsctlrc, the opensipsctl is working properly. Could somebody take a look in the stracelog to help me? Best regards, Antonio Anderson M. Souza Voice Technology http://www.antonioams.com On Tue, Dec 29, 2009 at 12:38 PM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Antonio, try the highest debug level (6) to see if you get something more - also try to start opensips under strace . Regards, Bogdan Antonio Anderson M. de Souza wrote: Hi Bogdan, Yes it disappears without any log message, the last log message was loading Dialplan module, after that it disappear. Best Regards, Antonio Anderson M. Souza Voice Technology http://www.antonioams.com On Sun, Dec 27, 2009 at 7:31 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi Antonio, so your opensips is not blocking but completely failing to start - your opensips simply disappears without any log, message? Regards, Bogdan Antonio Anderson M. de Souza wrote: Bogdan, I executed the opensips with debug=4 (follows the logs bellow [1]), and i confirmed that is not running executing the ps -ax | grep opensips [2] and I've sent SIP Requests to test it and the OS returned the ICMP port unreachable. [1] - Logs Dec 26 18:44:26 asouza-laptop opensips[19549]: NOTICE:core:main: version: opensips 1.6.0-notls (x86_64/linux) Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:core:main: using 32 Mb shared memory Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:core:main: using 1 Mb private memory per process Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:sl:mod_init: Initializing StateLess engine Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:tm:mod_init: TM - initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:maxfwd:mod_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:usrloc:ul_init_locks: locks array size 512 Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:textops:mod_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:xlog:mod_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:acc:mod_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:auth:mod_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:auth_db:mod_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:alias_db:mod_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:uac:mod_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:avpops:avpops_init: initializing... Dec 26 18:44:26 asouza-laptop opensips[19549]: WARNING:permissions:mod_init: default allow file (/usr/local/etc/opensips/permissions.allow) not found = empty rule set Dec 26 18:44:26 asouza-laptop opensips[19549]: WARNING:permissions:mod_init: default deny file (/usr/local/etc/opensips/permissions.deny) not found = empty rule set Dec 26 18:44:26 asouza-laptop opensips[19549]: WARNING:core:mk_net: invalid network address/netmask combination fixed... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:dialog:mod_init: Dialog module - initializing Dec 26 18:44:26 asouza-laptop opensips[19549]: NOTICE:signaling:mod_init: initializing module ... Dec 26 18:44:26 asouza-laptop opensips[19549]: INFO:drouting:dr_init:
[OpenSIPS-Users] Can OpenSips support SBC?
nbsp;Hi, I want to know whether the opensips support SBC with dual NICs, If yes, whether opensips can be used to construct a sbc cluster with the function of load balancing, and is there any performance test data as a sbc? ths, yinlin___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Need help for Call to another network
Ahmed, One way to avoid the problem of Asterisk returning 491 Request Pending would be to authenticate the call between the UA and Opensips (see proxy_authorize and proxy_challenge functions in modules auth and auth_db) then relay it to Asterisk with IP address authentication only. i.e. so Asterisk trusts all INVITE requests sent to it by Opensips. The peer definition in Asterisk would then look like this: [opensips] type=peer insecure=invite context=whatever host=yy.yy.179.54 For this to work, the UAC credentials must be known to Opensips. They would usually be stored in the subscriber table. If the UA registers, then it registers with Opensips, not Asterisk. John Quick Smartvox Limited ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA not passing ACKs
Hi Brian, the error you posted has nothing to do with blocking / non-blocking - it simply shows that opensips fails to open a new TCP connection because there was nobody listening on the other side. Regards, Bogdan opensipsl...@encambio.com wrote: Hello list, It seems that in August 2008 as OpenSER forked, Bogdan announced a priority list of next steps. It included: TCP+TLS rework of: blocking, scalability, nat traversal, performance Does anybody know if work proceeded on the TLS blocking problems? error opensips[7950]: ERROR:core:tcp_blocking_connect: timeout 10 s elapsed from 10 s error opensips[7950]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed error opensips[7950]: ERROR:core:tcp_send: connect failed error opensips[7950]: ERROR:tm:msg_send: tcp_send failed error opensips[7950]: ERROR:tm:t_forward_nonack: sending request failed Do these errors (in 1.6.0) reflect the priority as described by Bogdan? Regards, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] need advice on B2b
Hi Ha`,
There is a very simple example in the documentation:
route[b2b_request] {
xlog(b2b_request ($ci)\n);
}
route[b2b_reply] {
xlog(b2b_reply ($ci)\n);
}
You can call in these routes any function that you call in a request route.
Regards,
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www.voice-system.ro
ha do wrote:
Hi all
modparam(b2b_entities, script_req_route, b2b_request)
modparam(b2b_entities, script_reply_route, b2b_reply)
Could you please let me know how it work
is there the example of b2b_request and b2b_reply
the initial request can be seen as b2b_request?
the onreply_route is as same as b2b_reply?
Thank you
Ha`
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Re: [OpenSIPS-Users] B2BUA not passing ACKs
Hi Brian, There is a misunderstanding from your side in what the b2b scenario documents are concerned ( please read carefully the documentation - http://www.opensips.org/Resources/B2buaTutorial ). The important thing is that there should only be rules in the scenario for requests that need a special handling. In the prepaid scenario - when the BYE from the media server is received the caller must be connected to a human operator, so we have a rule for this. All the other requests need only simple pass forward - so if an ACK is received from one side it only need to be forwarded to the other. 'pass forward' is the implicit action and it will be applied to all requests that don't match a rule. I see that you say that the prepaid scenario does not work for you. What version are you testing with? Regards, -- Anca Vamanu www.voice-system.ro opensipsl...@encambio.com wrote: Hello list, According to the example: http://www.opensips.org/Resources/B2buaTutorial#toc13 ...implementing the prepaid service should handle INVITE, ACK, and BYE. Strangely the XML file includes only the BYE scenario: rules request bye I don't know enough about the B2BUA modules to say for sure, but it would seem that the online example is missing 'ACK' at least. When I copy the XML code and config code 1:1, my B2BUA implementation of the prepaid service sends the initial INVITE, receives the OK, but as the UAC (caller in the diagram) sends the ACK the B2B (green line in the diagram) does not pass it onto the media server. Is there a bug in the online XML code for the prepaid scenario? Regards, Brian ___ 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] compile deb files 1.6.1
Hi Jan, are you sure you have the opensips-mysql-module_1.6.1.0_amd64.deb file in the current dir (or name is correct) ?? I just tried the install of the same packages (freshly generated from svn tree) and I had no issue. Regards, Bogdan Jan D. wrote: Bogdan, Thanks for the quick response, I ran the 'make deb' again, now the version number of the deb file is OK, but there is still seems to be a problem with a directory (install on a clean system): dpkg -i opensips_1.6.1-0_amd64.deb opensips-mysql-module_1.6.1.0_amd64.deb Unpacking opensips (from opensips_1.6.1-0_amd64.deb) ... dpkg: error processing opensips-mysql-module_1.6.1.0_amd64.deb (--install): cannot access archive: No such file or directory Setting up opensips (1.6.1-0) ... OpenSIPS not yet configured. Edit /etc/default/opensips first. Processing triggers for man-db ... Errors were encountered while processing: opensips-mysql-module_1.6.1.0_amd64.deb Any clue? Jan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips 1.6.1 dos not start on Ubuntu 9.1 64bits
Hi Antonio,
The relevant part is:
14752 socket(PF_FILE, SOCK_STREAM, 0) = 5
14752 fcntl(5, F_SETFL, O_RDONLY) = 0
14752 fcntl(5, F_GETFL) = 0x2 (flags O_RDWR)
14752 connect(5, {sa_family=AF_FILE,
path=/var/run/mysqld/mysqld.sock}, 110) = 0
14752 setsockopt(5, SOL_SOCKET, SO_RCVTIMEO,
\2003\341\1\0\0\0\0\0\0\0\0\0\0\0\0, 16) = 0
14752 setsockopt(5, SOL_SOCKET, SO_SNDTIMEO,
\2003\341\1\0\0\0\0\0\0\0\0\0\0\0\0, 16) = 0
14752 setsockopt(5, SOL_IP, IP_TOS, [8], 4) = -1 EOPNOTSUPP (Operation
not supported)
14752 setsockopt(5, SOL_SOCKET, SO_KEEPALIVE, [1], 4) = 0
14752 read(5, =\0\0\0\n5.1.37-1ubuntu5\0\223\1\0\00076JP4Q|..., 16384)
= 65
14752 write(5,
i\0\0\1\215\242\3\200\0\...@\10\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0...,
77) = 77
14752 read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83
Seams mysql refuses the a connection from opensips. Are you sure you
properly configured the db_url for all the module that requires DB
connection ?
Regards,
Bogdan
Antonio Anderson M. de Souza wrote:
Bogdan,
I started the Opensips with debug=6 and there weren't changes on the
initialization log, I've used the strace [1], I'm not an expert in the
strace, but the only thing that i could see was a message 14752
read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83
after that the shutdown process started.
I already double checked the user and password of the mysql, and it's
right in the script, and in the opensipsctlrc, the opensipsctl is
working properly.
Could somebody take a look in the stracelog to help me?
[1] -
http://dl.dropbox.com/u/2134454/strace-opensips-crashs-ubuntu9.10-64.log.tar.gz
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
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Re: [OpenSIPS-Users] Opensips 1.6.1 dos not start on Ubuntu 9.1 64bits
Bogdan,
Thank you very much, the problem was because I forgot to add the module
dialplan in the modparam db_url:
Before:
modparam(domain|alias_db|auth_db|usrloc|drouting,db_url,
mysql://user:u...@localhost/db)
After:
modparam(domain|alias_db|auth_db|usrloc|drouting*|dialplan*,db_url,
mysql://user:u...@localhost/db)
The biggest problem that take longer time to discover the problem was
because in the CentOS the same configuration script works properly, do you
have some explanation to this behavior in other OS?
An improvement in the Script Compiler (opensips -c) could be to check if
the db_url is properly set, what do you think?
Best regards,
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
On Mon, Jan 4, 2010 at 9:22 AM, Bogdan-Andrei Iancu
bog...@voice-system.rowrote:
Hi Antonio,
The relevant part is:
14752 socket(PF_FILE, SOCK_STREAM, 0) = 5
14752 fcntl(5, F_SETFL, O_RDONLY) = 0
14752 fcntl(5, F_GETFL) = 0x2 (flags O_RDWR)
14752 connect(5, {sa_family=AF_FILE,
path=/var/run/mysqld/mysqld.sock}, 110) = 0
14752 setsockopt(5, SOL_SOCKET, SO_RCVTIMEO,
\2003\341\1\0\0\0\0\0\0\0\0\0\0\0\0, 16) = 0
14752 setsockopt(5, SOL_SOCKET, SO_SNDTIMEO,
\2003\341\1\0\0\0\0\0\0\0\0\0\0\0\0, 16) = 0
14752 setsockopt(5, SOL_IP, IP_TOS, [8], 4) = -1 EOPNOTSUPP (Operation
not supported)
14752 setsockopt(5, SOL_SOCKET, SO_KEEPALIVE, [1], 4) = 0
14752 read(5, =\0\0\0\n5.1.37-1ubuntu5\0\223\1\0\00076JP4Q|..., 16384)
= 65
14752 write(5,
I\0\0\1\215\242\3\200\0\0\0@
\10\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0...,
77) = 77
14752 read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83
Seams mysql refuses the a connection from opensips. Are you sure you
properly configured the db_url for all the module that requires DB
connection ?
Regards,
Bogdan
Antonio Anderson M. de Souza wrote:
Bogdan,
I started the Opensips with debug=6 and there weren't changes on the
initialization log, I've used the strace [1], I'm not an expert in the
strace, but the only thing that i could see was a message 14752
read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83
after that the shutdown process started.
I already double checked the user and password of the mysql, and it's
right in the script, and in the opensipsctlrc, the opensipsctl is
working properly.
Could somebody take a look in the stracelog to help me?
[1] -
http://dl.dropbox.com/u/2134454/strace-opensips-crashs-ubuntu9.10-64.log.tar.gz
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
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Re: [OpenSIPS-Users] Opensips 1.6.1 dos not start on Ubuntu 9.1 64bits
Hi Antonio,
It should not work on any OSmaybe your mysql setups are different on
the OSs you tested. NOTE that db_url has a default value which may work
if the DB was created with default access users.
Regards,
Bogdan
Antonio Anderson M. de Souza wrote:
Bogdan,
Thank you very much, the problem was because I forgot to add the
module dialplan in the modparam db_url:
Before:
modparam(domain|alias_db|auth_db|usrloc|drouting,db_url,
mysql://user:u...@localhost/db)
After:
modparam(domain|alias_db|auth_db|usrloc|drouting*|dialplan*,db_url,
mysql://user:u...@localhost/db)
The biggest problem that take longer time to discover the problem was
because in the CentOS the same configuration script works properly, do
you have some explanation to this behavior in other OS?
An improvement in the Script Compiler (opensips -c) could be to
check if the db_url is properly set, what do you think?
Best regards,
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
On Mon, Jan 4, 2010 at 9:22 AM, Bogdan-Andrei Iancu
bog...@voice-system.ro mailto:bog...@voice-system.ro wrote:
Hi Antonio,
The relevant part is:
14752 socket(PF_FILE, SOCK_STREAM, 0) = 5
14752 fcntl(5, F_SETFL, O_RDONLY) = 0
14752 fcntl(5, F_GETFL) = 0x2 (flags O_RDWR)
14752 connect(5, {sa_family=AF_FILE,
path=/var/run/mysqld/mysqld.sock}, 110) = 0
14752 setsockopt(5, SOL_SOCKET, SO_RCVTIMEO,
\2003\341\1\0\0\0\0\0\0\0\0\0\0\0\0, 16) = 0
14752 setsockopt(5, SOL_SOCKET, SO_SNDTIMEO,
\2003\341\1\0\0\0\0\0\0\0\0\0\0\0\0, 16) = 0
14752 setsockopt(5, SOL_IP, IP_TOS, [8], 4) = -1 EOPNOTSUPP (Operation
not supported)
14752 setsockopt(5, SOL_SOCKET, SO_KEEPALIVE, [1], 4) = 0
14752 read(5, =\0\0\0\n5.1.37-1ubuntu5\0\223\1\0\00076JP4Q|...,
16384)
= 65
14752 write(5,
i\0\0\1\215\242\3\200\0\...@\10\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0...,
77) = 77
14752 read(5, O\0\0\2\377\24\4#42000Access denied for u...,
16384) = 83
Seams mysql refuses the a connection from opensips. Are you sure you
properly configured the db_url for all the module that requires DB
connection ?
Regards,
Bogdan
Antonio Anderson M. de Souza wrote:
Bogdan,
I started the Opensips with debug=6 and there weren't changes on the
initialization log, I've used the strace [1], I'm not an expert
in the
strace, but the only thing that i could see was a message 14752
read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83
after that the shutdown process started.
I already double checked the user and password of the mysql, and
it's
right in the script, and in the opensipsctlrc, the opensipsctl is
working properly.
Could somebody take a look in the stracelog to help me?
--
Bogdan-Andrei Iancu
www.voice-system.ro
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Re: [OpenSIPS-Users] NAT per call leg
Hi Daniel, Daniel Goepp wrote: I'm going to look the complete fool here for my lack of understanding how OpenSIPS would handle this via it's route branching, but I am just banging my head on this one. Here is the problem: For simplification I'm using the first octet of the IP to identify the system, and the connect line of the SDP for both INVITE and OK How we have it setup today: OpenSIPS B doesn't see the call as NATd which it isn't between proxies so sends sdp w/204 to UAC B. UAC A (192) - INVITE (192) - OpenSIPS A (204) - INVITE (204) - OpenSIPS B (76) - INVITE (204) - UAC B (192) UAC A (192) - OK (204) - OpenSIPS A (204) - OK (192) - OpenSIPS B (76) - OK (192) - UAC B (192) OpenSIPS B should see that the callee is behind NAT (after doig lookup(location), the nat bflag will be on) and use the rtpp. Regards, Bogdan What we would like it to do: Split the handling of NAT, so the sdp on the call leg between proxies is not touched, but the call leg to the UAC B is rewritten for NAT, and the SDP in the OK back to Proxy A is rewritten. UAC A (192) - INVITE (192) - OpenSIPS A (204) - INVITE (204) - OpenSIPS B (76) - INVITE (76) - UAC B (192) UAC A (192) - OK (204) - OpenSIPS A (204) - OK (76) - OpenSIPS B (76) - OK (192) - UAC B (192) Does this make sense. I'm trying to dig through the archives to see if there is something on this, but I'm not finding much yet. Any help is MUCH appreciatedI know, it's Xmas even, and I'm messing around with OpenSIPS...what a life ;) -dg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] .Can OpenSips support SBC?
Hi YinLin, Well, for opensips, the cps (calls per second) is the most relevant param. And the value is higly depending on the complexity of the configuration/routing script. You can do a simple SBC for NAT, but also you can do (as have recently implemented) and SBC with TLS, NAT traversal, disaptching and failover between servers, topology hiding, redirect handling, etc So, it is hard to give a number without having the script itself. Regards, Bogdan 尹林 wrote: Hi Bogdan, Thank you for your reply, Our SBC runs in the Internet not IMS,so we need SBC for NAT transfer,media stream transfer and security, I have used OpenSBC,However, the performance is not meet our requirement (Single OpenSBC only serves 100-250 concurrent call sessions). so,if the opensips can act as the SBC with the functions that I have mentioned above,I really want to know the performance,like maximum allowable number of concurrent connections,the Maximum Allowable Call Rate per Second,and the testing envirionment. Thank you! Regards, YinLin ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] re-REGISTER with To tag gets 404
Hi Jeff,
should be ok this way.
Regards,
Bogdan
Jeff Pyle wrote:
Is there any danger in adapting the default config from:
if (has_totag()) {
...
}
to:
if (has_totag() !is_method(REGISTER)) {
...
}
- Jeff
On 12/31/09 7:46 AM, Olle E. Johansson o...@edvina.net wrote:
31 dec 2009 kl. 12.12 skrev Victor Pascual Avila:
Please, see RFC 3261 - Section 10.2: A REGISTER request does not
establish a dialog
Right, but there are many servers that request that you reuse the same dialog
identifiers as the challenged transaction when you authenticate.
/O
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Re: [OpenSIPS-Users] something on stun documment
Hi Ha, the alternate port is whatever port you want to use as secondary port by the STUN server. 3479 is the default alternate port in STUN, so it is ok. Regards, Bogdan ha do wrote: Hi admin http://www.opensips.org/html/docs/modules/devel/stun.html#id227269 1.3.3. |alternate_ip| (str) Another ip from another interface. *Example 1.3. Set |alternate_ip| parameter * ... modparam(stun,alternate_port,3479) -- this is right? Thank you Ha` ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Send simultaneous commands with exec module
Hi Jennifer, A Happy New Year to you too! for CMD1 and CMD2, try from command line (CMD1 CMD2) or (CMD1 CMD2) Regards, Bogdan Jennifer-4 wrote: Hi everybody and Happy New Year!! I´m working with Opensips 1.4. I need to send two commands simultaneously from Opensips, specifically I want to send two flows with ITG. If I write two sentences with exec_avp or exec_dset, the commands are sent in order, not simultaneously. So I´m trying to send the commands in the same exec sentence and launching the first command in background: exec_dset(ssh -l root 192.168.0.56 ./D-ITG-2.6.1d/bin/ITGSend -a 192.168.4.38 -rp 12003 -t 16000 -x lograror VoIP -x G.729.2 -h RTP /dev/null ssh -l root 192.168.0.38 ./D-ITG-2.6.1d/bin/ITGSend -a 192.168.2.56 -rp 11003 -t 16000 -x lograror2 VoIP -x G.729.2 -h RTP); But with this sentence, only the first command is executed, and the second is lost. Can anybody help me? Thanks! -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Users Digest, Vol 18, Issue 2
Hi Ahmed, Asterisk is confused (by sending the request pending reply to second CALL - csesq 2) because it never received an ACK for the reply it sent for the first call (cseq 1). What is strange is that I do not see any ACK neither from OpenSIPS to Asterisk, nor from UAC to OpenSIPSCould you verify this? What version of opensips are you using and how are you sending the call from OpenSIPS to Asterisk (what function are you using)? Regards, Bogdan Ahmed Munir wrote: Hi, Thanks for reply. I'm sorry for not adding further information previously, well there is no firewall involve in it because iptables service is stop and about the traces, I'm listing down the traces below; Where xx.xx.2.137 is OpenSIPs IP, yy.yy.179.54 is Asterisk IP and yy.yy.176.22 is UAC IP U yy.yy.176.22:9782 - xx.xx.2.137:5060 INVITE sip:3214426...@xx.xx.2.137 SIP/2.0. To: sip:3214426...@xx.xx.2.137. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. Via: SIP/2.0/UDP yy.yy.176.22:9782;branch=z9hG4bK-d87543-388987748-1--d87543-;rport. Call-ID: b95db141291c3838. CSeq: 1 INVITE. Contact: sip:322...@yy.yy.176.22:9782. Max-Forwards: 70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3007n stamp 17816. Content-Length: 286. . v=0. o=- 1386156 1386539 IN IP4 yy.yy.176.22. s=eyeBeam. c=IN IP4 yy.yy.176.22. t=0 0. m=audio 9784 RTP/AVP 0 8 101. a=alt:1 1 : F2F04F20 00D5 yy.yy.176.22 9784. a=alt:2 3 : F2C6E608 001F 192.168.0.168 9784. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. U xx.xx.2.137:5060 - yy.yy.179.54:5060 INVITE sip:3214426...@yy.yy.179.54:5060 SIP/2.0. To: sip:3214426...@xx.xx.2.137. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. Via: SIP/2.0/UDP xx.xx.2.137;branch=z9hG4bK-d87543-388987748-1--d87543-. Via: SIP/2.0/UDP yy.yy.176.22:9782;received=yy.yy.176.22;branch=z9hG4bK-d87543-388987748-1--d87543-;rport=9782. Call-ID: b95db141291c3838. CSeq: 1 INVITE. Contact: sip:322...@yy.yy.176.22:9782. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3007n stamp 17816. Content-Length: 286. . v=0. o=- 1386156 1386539 IN IP4 yy.yy.176.22. s=eyeBeam. c=IN IP4 yy.yy.176.22. t=0 0. m=audio 9784 RTP/AVP 0 8 101. a=alt:1 1 : F2F04F20 00D5 yy.yy.176.22 9784. a=alt:2 3 : F2C6E608 001F 192.168.0.168 9784. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. U xx.xx.2.137:5060 - yy.yy.176.22:9782 SIP/2.0 100 Giving a try. To: sip:3214426...@xx.xx.2.137. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. Via: SIP/2.0/UDP yy.yy.176.22:9782;branch=z9hG4bK-d87543-388987748-1--d87543-;rport=9782. Call-ID: b95db141291c3838. CSeq: 1 INVITE. Server: OpenSIPS (1.6.0-notls (i386/linux)). Content-Length: 0. . U xx.xx.2.137:5060 - yy.yy.179.54:5060 INVITE sip:3214426...@yy.yy.179.54:5060 SIP/2.0. To: sip:3214426...@xx.xx.2.137. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. Via: SIP/2.0/UDP xx.xx.2.137;branch=z9hG4bK2d48.760e5071.0. Via: SIP/2.0/UDP yy.yy.176.22:9782;received=yy.yy.176.22;branch=z9hG4bK-d87543-388987748-1--d87543-;rport=9782. Call-ID: b95db141291c3838. CSeq: 1 INVITE. Contact: sip:322...@yy.yy.176.22:9782. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 3007n stamp 17816. Content-Length: 286. . v=0. o=- 1386156 1386539 IN IP4 yy.yy.176.22. s=eyeBeam. c=IN IP4 yy.yy.176.22. t=0 0. m=audio 9784 RTP/AVP 0 8 101. a=alt:1 1 : F2F04F20 00D5 yy.yy.176.22 9784. a=alt:2 3 : F2C6E608 001F 192.168.0.168 9784. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. U yy.yy.179.54:5060 - xx.xx.2.137:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP xx.xx.2.137;branch=z9hG4bK-d87543-388987748-1--d87543-;received=xx.xx.2.137. Via: SIP/2.0/UDP yy.yy.176.22:9782;received=yy.yy.176.22;branch=z9hG4bK-d87543-388987748-1--d87543-;rport=9782. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. To: sip:3214426...@xx.xx.2.137;tag=as70d2441c. Call-ID: b95db141291c3838. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Proxy-Authenticate: Digest algorithm=MD5, realm=sip.vopium.com http://sip.vopium.com, nonce=2db539d6. Content-Length: 0. . U xx.xx.2.137:5060 - yy.yy.176.22:9782 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP yy.yy.176.22:9782;received=yy.yy.176.22;branch=z9hG4bK-d87543-388987748-1--d87543-;rport=9782. From: 322025sip:322...@xx.xx.2.137;tag=ff6e2a00. To: sip:3214426...@xx.xx.2.137;tag=as70d2441c. Call-ID: b95db141291c3838. CSeq: 1 INVITE. User-Agent: Asterisk PBX.
Re: [OpenSIPS-Users] Load balancer probe _mode=1 bug?
Hey Bogdan, It looks like that fixed it. Thanks so much! Bill Bogdan-Andrei Iancu wrote: Hi Bill, Could you please try the attached patch? It seams that there was an issue with the the probing values in the code. Let me know if the patch does solves your problem and I will upload it on SVN. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load balancer probe _mode=1 bug?
Super! I uploaded the fix on SVN. Thanks and regards, Bogdan Bill W wrote: Hey Bogdan, It looks like that fixed it. Thanks so much! Bill Bogdan-Andrei Iancu wrote: Hi Bill, Could you please try the attached patch? It seams that there was an issue with the the probing values in the code. Let me know if the patch does solves your problem and I will upload it on SVN. Regards, Bogdan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Media-dispatcher
Interesting. I think you may have mentioned this before Dan but I didn't catch it for some reason. That makes installing media proxy on CentOS / RHEL 5.x easier. I've found that creating an RPM to install python 2.5 (along with the 2.4 rather than upgrading it) and then using the virtualenv package to use it in a sandbox for non-distribution applications made things a lot easier too. So, looking at the INSTALL file it seems that the two changes needed would be easy_install python-application1.2.0 and easy_install sqlobject==0.9 I'll try that out this afternoon. Will that 1.2.0 work for easy_install or do I need to go find a specific version that I want to use? Richard On Jan 3, 2010, at 8:53 AM, Dan Pascu wrote: On 31 Dec 2009, at 10:52, Chandrakant Solanki wrote: Hello I have installed media-proxy 2.3.8 using http://www.smartvox.co.uk/serfaq_install_mediaproxy2.htm But when i start media-dispatcher/media-relay it gives following error [r...@users ~]# media-dispatcher Traceback (most recent call last): File /usr/bin/media-dispatcher, line 12, in ? from application.process import process, ProcessError File build/bdist.linux-i686/egg/application/process.py, line 12, in ? File build/bdist.linux-i686/egg/application/log/__init__.py, line 12, in ? File /usr/lib/python2.4/site-packages/python_application-1.2.1- py2.4.egg/application/log/extensions/twisted/__init__.py, line 4 SyntaxError: future feature absolute_import is not defined Either use python2.5 or use a python-application package 1.2.0 -- Dan ___ 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] 3rd party Client sending 5xx
Please see: http://www.apps.ietf.org/rfc/rfc3261.html#sec-21.5 On Jan 4, 2010, at 9:53 PM, Aditya Kumar wrote: Hi all, I am inter working with a 3rd party SIP UA and I see they are sending 503/500. Can any one tell me what are the cases at which a SIP UAS will sent 503 or 500 ___ 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] 3rd party Client sending 5xx
Thanks Jeff for the link. I did see 3261. Question is : RFC will tel high level about response codes. I am trying to understand in particular when a End User (UAS) will sent. I am clear from a proxy perspective. Was looking some one can share your experiences when a UAS will send 5xx From: Jeff Pyle jp...@fidelityvoice.com To: OpenSIPS users mailling list users@lists.opensips.org Sent: Mon, January 4, 2010 7:01:34 PM Subject: Re: [OpenSIPS-Users] 3rd party Client sending 5xx Please see: http://www.apps.ietf.org/rfc/rfc3261.html#sec-21.5 On Jan 4, 2010, at 9:53 PM, Aditya Kumar wrote: Hi all, I am inter working with a 3rd party SIP UA and I see they are sending 503/500. Can any one tell me what are the cases at which a SIP UAS will sent 503 or 500 ___ 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] 3rd party Client sending 5xx
Aditya, That all depends on the switch. Asterisk sends a 503 if you look at it the wrong way, that is, under many conditions. In my experience most carriers will send a 503 when their upstream paths are full (or your concurrent calls to them is full) and they expect you to route advance to another carrier. A 503 could cause a proxy or a switch to blacklist the host that sent it for a period, so proxies (Opensips included) will convert a 503 to a 500 before relaying. When a switch has trouble processing the request because of some internal issue, say, a database error, it will reply with a 500. This list is by no means all inclusive. Hopefully it will point you in the right direction. - Jeff On Jan 4, 2010, at 10:13 PM, Aditya Kumar wrote: Thanks Jeff for the link. I did see 3261. Question is : RFC will tel high level about response codes. I am trying to understand in particular when a End User (UAS) will sent. I am clear from a proxy perspective. Was looking some one can share your experiences when a UAS will send 5xx From: Jeff Pyle jp...@fidelityvoice.commailto:jp...@fidelityvoice.com To: OpenSIPS users mailling list users@lists.opensips.orgmailto:users@lists.opensips.org Sent: Mon, January 4, 2010 7:01:34 PM Subject: Re: [OpenSIPS-Users] 3rd party Client sending 5xx Please see: http://www.apps.ietf.org/rfc/rfc3261.html#sec-21.5 On Jan 4, 2010, at 9:53 PM, Aditya Kumar wrote: Hi all, I am inter working with a 3rd party SIP UA and I see they are sending 503/500. Can any one tell me what are the cases at which a SIP UAS will sent 503 or 500 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] How to implement access control list on opensips
Hi, I want to implement ACL on OpenSIPs to accept the call on behalf of source URI + IP address. Can anyone tell me which modules and functions are required for it? Also kindly share some example template with it. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
