Re: [OpenSIPS-Users] access to password as variable?
2009/10/16 Steven E. Ames sa...@officescape.com: Thanks. I'll start down the radius path. My problem with avp_db_query is I don't actually have a password. The idea here is to let a few select devices that we use internally to REGISTER in a dynamic fashion because the password they are presenting can be generated by sending their username through a hashing algorithm. So they don't actually have passwords or even usernames in the database. My original plan was just to exec_avp sending username and password and getting back a GO/NO_GO. Now it looks like it'll be send username/password to radius which execs script and sends back go/no_go. More complicated but you do what you gotta to make the powers that be happy. What is their hashing algorithm? Opensips can do md5 and other string operations itself from the script, so maybe you don't need any external programs at all? Can you show the algorithm? If you can get repeat it in opensips, then you can repeat the Digest Hash Algorithm too and check the password that way. -- KTHXBYE, Stanisław Pitucha, Gradwell Voip Engineer ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Using RTPProxy to proxy all non-NAT'ed calls as well
I have a working set up where OpenSIPS is my registrar and SIP Proxy. NAT'ed UA calls are handled with RTPProxy. I want RTPProxy to handle non-NAT'ed UA's (for example, UA's on a public internet IP). I tried a quick test of running the force_rtp_proxy() function on non-NAT'ed INVITE packets but I get this error: Oct 18 00:18:17 sip1 /sbin/opensips[18749]: ERROR:nathelper:extract_body: message body has length zero Oct 18 00:18:17 sip1 /sbin/opensips[18749]: ERROR:nathelper:force_rtp_proxy2_f: can't extract body from the message Is it possible to run RTPProxy in this fashion? I'm looking for the easiest fix. Thanks, Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using RTPProxy to proxy all non-NAT'ed calls as well
You are attempting to invoke rtpproxy on a request or reply which does not contain an SDP body, and therefore force_rtp_proxy() cannot operate on it. Julian Yap wrote: I have a working set up where OpenSIPS is my registrar and SIP Proxy. NAT'ed UA calls are handled with RTPProxy. I want RTPProxy to handle non-NAT'ed UA's (for example, UA's on a public internet IP). I tried a quick test of running the force_rtp_proxy() function on non-NAT'ed INVITE packets but I get this error: Oct 18 00:18:17 sip1 /sbin/opensips[18749]: ERROR:nathelper:extract_body: message body has length zero Oct 18 00:18:17 sip1 /sbin/opensips[18749]: ERROR:nathelper:force_rtp_proxy2_f: can't extract body from the message Is it possible to run RTPProxy in this fashion? I'm looking for the easiest fix. Thanks, Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Using RTPProxy to proxy all non-NAT'ed calls as well
Hmm, so in theory RTPProxy should work in my case? I just need to check the SDP? My NAT section has something like this: if(!isflagset(22) !search(^Content-Length:[ ]*0)) That Content-Length check should suffice? On Sun, Oct 18, 2009 at 12:34 AM, Alex Balashov abalas...@evaristesys.com wrote: You are attempting to invoke rtpproxy on a request or reply which does not contain an SDP body, and therefore force_rtp_proxy() cannot operate on it. Julian Yap wrote: I have a working set up where OpenSIPS is my registrar and SIP Proxy. NAT'ed UA calls are handled with RTPProxy. I want RTPProxy to handle non-NAT'ed UA's (for example, UA's on a public internet IP). I tried a quick test of running the force_rtp_proxy() function on non-NAT'ed INVITE packets but I get this error: Oct 18 00:18:17 sip1 /sbin/opensips[18749]: ERROR:nathelper:extract_body: message body has length zero Oct 18 00:18:17 sip1 /sbin/opensips[18749]: ERROR:nathelper:force_rtp_proxy2_f: can't extract body from the message Is it possible to run RTPProxy in this fashion? I'm looking for the easiest fix. Thanks, Julian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ 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] Load balancer and Access control list
Hi Peter, Interesting - the LB was not designed to work via 3xx redirect model, even so, with some changes, it is possible to do it. On the other side, can you trick FS to look for the IP (used for ACLs) somewhere else than the net level (src IP) ? maybe you can configure LB to put the original SRC IP into a SIP header into the request. Regards, Bogdan Peter P GMX wrote: Hello, I am using OpenSIPS as a load balancer in front of Freeswitch by using the load balancer module. Scenario: All phones are registered at Freeswitch. Some gateways provide calls via registered accounts and some external gateways are accepted by their IPs (access control list, ACL). When using OpenSIPS in front of Freeswitch I am losing the ACL feature for some external gateways, as from Freeswitch's perpective all calls are now coming from OpenSIPS. Does anybody know how to solve this ACL problem? Is there a way load balance by redirecting invites (302 Moved temporarily) to Freeswitch? Then the gateway will contact Freeswitch directly and ACLs will still apply. So is there any load balancing feature in OpenSIPS which uses redirects or do I have to implement it by myself? E.g. by a perl script which changes the redirect IP on every request (e.g. round robin). Best regards Peter ___ 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
Hi Sebastian, So use: Dpid = 1 Pr = 0 match_op = 1 match_exp = ^4224+ match_len = 0 subst_exp = ^(*4224*)(.+) repl_exp = \2 Regards, Bogdan Sebastian Sastre wrote: Agreed, I followed a previous post from Bogdan but I cant replicate it. In short I have some numbers which I need to strip the first 4 numbers from. How would you do this? I've been playing aroung with the different regexp expressions I know of, but I cant get the result. Any help appreciated. Thanks The previous post was Jul 17, 2009; 07:36am Re: 2 dialplan issues: simple strip\prefix cases and execution with drouting use_next_gw function Hi Ricardo, for stripping some digits, try something like: match: ^55+ subst: ^(55)(.+) repl: \2 This will strip the 55 prefix from the number For item 2) - take care as the do_routing() function is computing all the next branches in advances, based on the current URI...The use_next_gw() function will just upload the precomputed RURIs in the new branch. If you want some per branch changes via dialplan, do it after the do_routing() and use_next_gw() functions.. Regards, Bogdan Ricardo Martins wrote: Hi all! I've been working a quite time with dialplan module and need some orientation from more experienced users. On one case, I need to strip some digits depending on the dial prefix and on the other case I need to subst some digits with 0 or 00. I could make the subst case to work, replacing the starting 55 with 0, using the following rule: match: ^55+ subst: ^(55)(.+) Sebastian Sastre Next IP sebast...@next-ip.com + 305-507-8722- Main + 305-507-8728- Direct The information transmitted herein is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. La información contenida en este mensaje electrónico tiene carácter CONFIDENCIAL, está dirigida únicamente al destinatario de la misma y sólo podrá ser usada por éste. Si el lector de este mensaje no es el destinatario del mismo, se le notifica que cualquier copia o distribución de éste se encuentra totalmente prohibida. Si usted ha recibido este mensaje por error, por favor notifique inmediatamente al remitente por este mismo medio y borre el mensaje de su sistema. -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Brett Nemeroff Sent: Friday, October 16, 2009 12:31 AM To: OpenSIPS users mailling list Subject: 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org
Re: [OpenSIPS-Users] Issue with incoming calls.
Peter, check for ICMP errors - maybe the requests from opensips do not get to the external party due net issues. Regards, Bogdan Leon Li wrote: 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? ___ 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 Larry, Kemp, Larry wrote: 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. what you mean by crash ? does not start? do you get an error and it stops? Regards, Bogdan 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
Re: [OpenSIPS-Users] OpenSips as SMS-GW
OpenSIPS has not SMS capabilities. How the SMS in encapsulated via SIP (in a 3GPP case) ? Regards, Bogdan Ajay Pratap Singh Pundhir wrote: yes i am talking about SIP Client which support SMS. 3GPP has recently released standard for SMS over IP, so i am talking about the Softphone which support SMS service, i.e. which can register as a SMS capable device, and can Send/Recieve SMS in the format described in Stsndards. One idea which came to my mind is to add one more tab in UCTIMSClient for SMS. On Wed, Oct 14, 2009 at 2:37 AM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: what is a client for SMS services ??? everything is SIP here, so maybe you need a SIP client with IM support, right? Regards, Bogdan Ajay Pratap Singh Pundhir wrote: Thanks, is there any open source client available for SMS service in IMS. or i have to make one from scratch. On Tue, Oct 13, 2009 at 7:32 PM, Bogdan-Andrei Iancu bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro mailto:bog...@voice-system.ro wrote: Hi, OpenSIPS has a builtin SMS module, but for SMS over AT-modems - see the SMS module (http://www.opensips.org/html/docs/modules/1.5.x/sms.html). The module can control an AT-modem to send and receive SMS directly from the GSM network. Another options is to simply integrate (via DB for example) opensips with kannel (SMS gateway). Regards, Bogdan Ajay Pratap Singh Pundhir wrote: 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
