[OpenSIPS-Users] Getting Error when using STUN
Hi, I'm getting and error when I configure STUN in OpenSIPs. As following documentation of OpenSIPs, I enabled stun as listed below; listen=udp:198.65.166.165:5060 listen=udp:75.101.138.128:5060 loadmodule stun.so # Stun modparam(stun,primary_ip,198.65.166.165) modparam(stun,primary_port,5060) modparam(stun,alternate_ip,75.101.138.128) modparam(stun,alternate_port,5060) Where the IP of OpenSIPs which is hosted on public IP i.e. 11.22.33.44. And the error I'm getting after restarting the OpenSIPs is listed below; May 17 05:37:13 newtest /usr/local/sbin/opensips[31199]: ERROR:core:udp_init: bind(5, 0x81b7374, 16) on 77.66.16.35: Cannot assign requested address When I commented out the listen=udp:IP:5060 the error I'm getting is; May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if host==us: 11==9 [77.66.16.35] == [127.0.0.1] May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if host==us: 11==11 [77.66.16.35] == [77.66.2.137] May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:stun:stun_mod_init: grep_sock_in()1 failed May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: ERROR:core:init_mod: failed to initialize module stun May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: ERROR:core:main: error while initializing modules Kindly assist me how can I resolve this problem. -- Regards, Ahmed Munir ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips running on 2 NIC's
Hi Everyone, I had two network interfaces eth0 and eth1. I configured eth0 to listen public IP and eth1 to listen on private IP. My question is how to configure opensips in order to listen on both interfaces. I configured in opensips cfg as follows listen=udp:public IP [eth0] alias=udp:private IP [eth1], but in vain. Can any one suggests how to configure opensips in order to listen on both interfaces. Regards, Indiver. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/opensips-running-on-2-NIC-s-tp5063824p5063824.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] Music on Hold without RTPProxy
Hello everyone,
I feel that lately I've been a pain in the ass for this mailing list but hey, I
need to ask someone :)
Ok here goes, I've went through the documentation and seen that it is possible
to inject music on hold on the On-Hold Re-Invite, via the RTPproxy. My case is
however, that I am not using rtpproxy for nat traversal, since I am having
asterisks behind OpenSIPS, always forcing the media through them.
So I was wondering, is there any way to intercept the On-Hold Re-Invite and
push it back to asterisk? Or any other intelligent solution?
I also noticed this:
if (method == INVITE) {
rtpproxy_offer();
if (detect_hold()) {
rtpproxy_stream2uas(/var/rtpproxy/prompts/music_on_hold, -1);
} else {
rtpproxy_stop_stream2uas();
};
};
However, and I wonder how I didn't find this mentioned anywhere else, the
detect_hold() method does not exist? I am on 1.6.1.
Regards,
Paris Stamatopoulos
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Re: [OpenSIPS-Users] Music on Hold without RTPProxy
Hello Paris, On Monday 17 May, 2010, at 9:41 AM, Paris Stamatopoulos wrote: Hello everyone, I feel that lately I've been a pain in the ass for this mailing list but hey, I need to ask someone :) Ok here goes, I've went through the documentation and seen that it is possible to inject music on hold on the On-Hold Re-Invite, via the RTPproxy. My case is however, that I am not using rtpproxy for nat traversal, since I am having asterisks behind OpenSIPS, always forcing the media through them. If your call is already terminated by asterisk, then you're done and you don't need anything else: the in-dialog re-invite is directely forwarded to asterisk which knows how to handle it... (you just need to define the music on hold for the given context in asterisk) Regards, - vma . ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips running on 2 NIC's
Hello Indiver, On Monday 17 May, 2010, at 8:18 AM, Indiver wrote: Hi Everyone, I had two network interfaces eth0 and eth1. I configured eth0 to listen public IP and eth1 to listen on private IP. My question is how to configure opensips in order to listen on both interfaces. I configured in opensips cfg as follows listen=udp:public IP [eth0] alias=udp:private IP [eth1], but in vain. Can any one suggests how to You need a second 'listen' line for your private IP... Regards, -vma . ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Music on Hold without RTPProxy
Hello! Well I have to say that I got so mixed up that I didn't even check to see if the Re-INVITE was sent to the asterisk. I had musiconhold classes configured wrong, and I was not getting the On-Hold message in the cli! Sorry about that and thanks for making me look the right way :) Regards, Paris Stamatopoulos -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of vallimamod abdullah Sent: Monday, May 17, 2010 11:37 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Music on Hold without RTPProxy Hello Paris, On Monday 17 May, 2010, at 9:41 AM, Paris Stamatopoulos wrote: Hello everyone, I feel that lately I've been a pain in the ass for this mailing list but hey, I need to ask someone :) Ok here goes, I've went through the documentation and seen that it is possible to inject music on hold on the On-Hold Re-Invite, via the RTPproxy. My case is however, that I am not using rtpproxy for nat traversal, since I am having asterisks behind OpenSIPS, always forcing the media through them. If your call is already terminated by asterisk, then you're done and you don't need anything else: the in-dialog re-invite is directely forwarded to asterisk which knows how to handle it... (you just need to define the music on hold for the given context in asterisk) Regards, - vma . ___ 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] How to reset load in load_balancer module
Hi, I'm doing some basic tests with load_balancer module and, when a gateway fails, the failover (mode 2) works well, but the problem is that when the gateway is working again, the load value is the same than before the gateway failure occurs, but now there isn't any ongoing call. I know that this is because Opensips hasn't received the corresponding BYEs to close de dialogs, but there is any way to work around this behavior? Can we reset de load value from de load_balancer module? Thanks, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips running on 2 NIC's
Hi Abdullah, Thanks for your response. I tried the solution mentioned in ur post ,but no luck. I even tried to listen the two interfaces on starting opensips by using -l option. But nothing went right for me. Regards, Indiver -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/opensips-running-on-2-NIC-s-tp5063824p5064479.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] how to use DB_HTTP modue ?
Hi everyone, I already posted a message about of how to use db but I did not have concrete answers, all those who responded to me told me to see the documentation of this module but after reading the documentation several time I can't manage to understand how it works, if someone already used the module db_http it would very nice of him to show me how he did. In my opensips.cfg I did this : modparam(avpops,db_url,http://localhost/db_http;) modparam(avpops,avp_table,script.php) I make a query to the http server : avp_db_query(1); __ if I change 1 by a string I will have it on ::1 - - [17/May/2010:12:31:21 +0200] GET /db_http/script.php/?q=string __ in the log of apache i have this : ::1 - - [17/May/2010:12:31:21 +0200] GET /db_http/script.php/?q=1query_type=custom HTTP/1.1 200 362 - - 127.0.0.1 - - [17/May/2010:12:31:21 +0200] POST / HTTP/1.1 200 45 - - My script.php which is in db_http directory on the http server get the variable by the url with $_GET['q'] but I don't know how change the name of variable q AND how send a lot of variable on the http request ??? I would like to send the avp variable to the http sever, after analyzing it by my script I wont to retrieve result on my opensips.cfg. Someone can help me ?? Best regards. Sam. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-use-DB-HTTP-modue-tp5064518p5064518.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] how to use DB_HTTP module ?
Hi everyone, I already posted a message about of how to use db but I did not have concrete answers, all those who responded to me told me to see the documentation of this module but after reading the documentation several time I can't manage to understand how it works, if someone already used the module db_http it would very nice of him to show me how he did. In my opensips.cfg I did this : modparam(avpops,db_url,http://localhost/db_http;) modparam(avpops,avp_table,script.php) I make a query to the http server : avp_db_query(1); __ if I change 1 by a string I will have it on ::1 - - [17/May/2010:12:31:21 +0200] GET /db_http/script.php/?q=string __ in the log of apache i have this : ::1 - - [17/May/2010:12:31:21 +0200] GET /db_http/script.php/?q=1query_type=custom HTTP/1.1 200 362 - - 127.0.0.1 - - [17/May/2010:12:31:21 +0200] POST / HTTP/1.1 200 45 - - My script.php which is in db_http directory on the http server get the variable by the url with $_GET['q'] but I don't know how change the name of variable q AND how send a lot of variable on the http request ??? I would like to send the avp variable to the http sever, after analyzing it by my script I wont to retrieve result on my opensips.cfg. Someone can help me ?? Best regards. Sam. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-use-DB-HTTP-module-tp5064520p5064520.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] how to use DB_HTTP module ?
Hi, This module was designed to act as a database module, if you want to use it to exchange data with a server you have to work around this. The q variable represents a raw query. Its name cannot be changed, but you can give it any value. You can use the avpops module like this to retrieve data: avp_db_query(whatever you want to put in the q variable,$avp(i:678)); The server MUST reply with a HTTP message that has a body similar to: string whatever answer you want in the avp Opensips will take this reply, see that it has only one column and place the result in the given avp. If you want to exchange several variables you have to invent a way to put all the variables in a single string. Andrei. samoh wrote: Hi everyone, I already posted a message about of how to use db but I did not have concrete answers, all those who responded to me told me to see the documentation of this module but after reading the documentation several time I can't manage to understand how it works, if someone already used the module db_http it would very nice of him to show me how he did. In my opensips.cfg I did this : modparam(avpops,db_url,http://localhost/db_http;) modparam(avpops,avp_table,script.php) I make a query to the http server : avp_db_query(1); __ if I change 1 by a string I will have it on ::1 - - [17/May/2010:12:31:21 +0200] GET /db_http/script.php/?q=string __ in the log of apache i have this : ::1 - - [17/May/2010:12:31:21 +0200] GET /db_http/script.php/?q=1query_type=custom HTTP/1.1 200 362 - - 127.0.0.1 - - [17/May/2010:12:31:21 +0200] POST / HTTP/1.1 200 45 - - My script.php which is in db_http directory on the http server get the variable by the url with $_GET['q'] but I don't know how change the name of variable q AND how send a lot of variable on the http request ??? I would like to send the avp variable to the http sever, after analyzing it by my script I wont to retrieve result on my opensips.cfg. Someone can help me ?? Best regards. Sam. -- Andrei Dragus www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] how to use DB_HTTP module ?
Hi Andrei, Thank you for your fast reply. now I understand better. In the documentation of this module, I see : k= Describes the keys (columns) that will be used for comparison.Can have multiple values. op= Describes the operators that will be used for comparison.Can have multiple values. v= Describes the values that columns will be compaired against. Can have multiple values. How can I use this variables ?. Best regards. Sam. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-use-DB-HTTP-module-tp5064520p5064772.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
Re: [OpenSIPS-Users] how to use DB_HTTP module ?
Hi, You cannot use these variables directly from script. Each module will use these variables if it needs to, they are only accessible from C-code directly. If you want, you can use the avp_db_load and avp_db_store functions and see what variables these use. I have the feeling that you are trying to use the module for something it was not designed for, so you will need to workaround its design. Andrei | | samoh wrote: Hi Andrei, Thank you for your fast reply. now I understand better. In the documentation of this module, I see : k= Describes the keys (columns) that will be used for comparison.Can have multiple values. op= Describes the operators that will be used for comparison.Can have multiple values. v= Describes the values that columns will be compaired against. Can have multiple values. How can I use this variables ?. Best regards. Sam. -- Andrei Dragus www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] New SIP SIMPLE client SDK release 0.14.2
Hello, SIP SIMPLE client SDK is now available as Debian package for the following Linux distributions: Debian Unstable (Sid) # AG Projects software deb http://ag-projects.com/debian unstable main deb-src http://ag-projects.com/debian unstable main Debian Stable (Lenny) # AG Projects software deb http://ag-projects.com/debian stable main deb-src http://ag-projects.com/debian stable main Ubuntu Karmic (9.10) # AG Projects software deb http://ag-projects.com/ubuntu karmic main deb-src http://ag-projects.com/ubuntu karmic main Ubuntu Lucid (10.04) # AG Projects software deb http://ag-projects.com/ubuntu lucid main deb-src http://ag-projects.com/ubuntu lucid main Update the list of available packages: sudo apt-get update Install SIP SIMPLE client SDK: sudo apt-get install python-sipsimple Install Command Line Tools: sudo apt-get install sipclients Kind regards, Adrian Georgescu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] how to use DB_HTTP module ?
Hi, I understand that this module isn't intended to be used to share variables with PHP? I want to use this module to use a php script to process each call If I do this: avp_db_query(select something from table, $avp(i:986)); can I retrieve the results on my $avp(i:986) ? What I must have In my script.php to treat this request and make the exchange ? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-use-DB-HTTP-module-tp5064520p5065134.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
Re: [OpenSIPS-Users] How to reset load in load_balancer module
Hi Santi, You cannot reset the load of the LB module in a direct way - the only way to do it is to force termination of the ongoing calls (known to LB module) - you can do this via the dlg_end_dlg MI command from dialog module. See: http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id272967 http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id272866 Regards, Bogdan Santi Anton wrote: Hi, I'm doing some basic tests with load_balancer module and, when a gateway fails, the failover (mode 2) works well, but the problem is that when the gateway is working again, the load value is the same than before the gateway failure occurs, but now there isn't any ongoing call. I know that this is because Opensips hasn't received the corresponding BYEs to close de dialogs, but there is any way to work around this behavior? Can we reset de load value from de load_balancer module? Thanks, ___ 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] opensips running on 2 NIC's
Hi, are you saying that: listen=udp:pub_ip:5060 listen=udp:priv_ip:5060 does not work for you ?? have you enabled forking mode (fork=yes param) ? Regards, Bogdan Indiver wrote: Hi Abdullah, Thanks for your response. I tried the solution mentioned in ur post ,but no luck. I even tried to listen the two interfaces on starting opensips by using -l option. But nothing went right for me. Regards, Indiver -- 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] how to use DB_HTTP module ?
I understand that this module isn't intended to be used to share variables with PHP? No, this was not the purpose of this module. I want to use this module to use a php script to process each call If I do this: avp_db_query(select something from table, $avp(i:986)); can I retrieve the results on my $avp(i:986) ? What I must have In my script.php to treat this request and make the exchange ? I don't know PHP but what you have to do is build a 200 OK HTTP reply containing the body: string answer After this $avp(i:986) will contain answer. -- Andrei Dragus www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP Presence Aggregation Issue
Hi Iñaki, No worry, no rudeness felt ;) I guess everybody feels a bit (or more) frustrated about the current SIMPLE situation. I'm really interested to see the outcome of your specs - my question is (assuming that from tech point of view, it is a good approach), how do you foresee the wide spreading of the specs ? Regards, Bogdan PS: if you need help on the specs or testing, let me know ;) Iñaki Baz Castillo wrote: 2010/4/12 Bogdan-Andrei Iancu bog...@voice-system.ro: Hi Iñaki, Hi Bogdan, replies inline: well, right now there is a kind of pressure coming from the providers level - providers do want to offer presence with SIP ; also presence comes in a natural way of doing dome enhanced services (more complex than simple BLA, BLF, etc). -- please note I said presence, not SIMPLE. So, a natural demand for it there, and we, as developers, are looking for solutions to make it happened. and for implementations you need some specs. Now, if you see the SIMPLE specs are wrong - it might be - I'm not directly involved in the depth of SIMPLE to be able to say yes or no. This aggregation problem is the first we encountered during some projects - not only once, but several times, different contexts ; and I'm trying with Anca to see how to get over it. So, overall, there are 2 options (according to your perception): - use SIMPLE and get a poor result (a crippled presence) SIMPLE is not just poor, but also inneficient at server level (a single change in a XCAP document requires the presence server to reload all the permissions for that user). Even in case of solving it, the result owuld be poor, sure. - come up with a new spec Yes. I'm doing a presence spec for SIP from scratch, by learning about XMPP and so. I've already defined the concept of resources, different status priority, global status. And best of all, there is no HTTP/XCAP, but just SIP. Well, I have to spent lot of hours yet :) - do feedback to IETF to make SIMPLE simpler and working IMHO this is not possible at this point, as IETF already chose XCAP for buddies and permissions management (along with others). IMHO there is no way to improve/fix current SIMPLE specs. For SIMPLE, looking at the basics (exchanging the info), the aggregation is the biggest issue I see. Whatever is on top (RLS, XCAP, buddy lists, etc) is another story and it might need a second look and thought. OMA tries to define an aggregation mechanism (like rules). I've read it, and it's a pain, a dirty hack over IETF *incomplete* specifications. Sorry for sounding so rude :) -- 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 announcement pstn
Hi guys, an idea will be to use the b2bua module in OpenSIPS to send first the call to an announcement server (asterisk, yate, freeswitch, sems) and when getting a BYE to create a new call leg to the final destination (the PSTN GW). That is a simple scenario with a b2bua script - http://www.opensips.org/Resources/B2buaTutorial Regards, Bogdan Andreas Sikkema wrote: On May 15, 2010, at 3:35 AM, Albert Paijmans wrote: Thanks for the reply. The reason we do not want to use Asterisk, but SEMS, is because SEMS offers the possibility to play a different announcement (could be from database) to every extension. This ofcourse makes it more attractive to our sponsors. We want to do both sponsor messages for outgoing calls and we will have some discreet advertisement on our website. We think we can offer free phonecalls to most international destinations thanks to Open Source and we are all volunteers :) So forwarding calls to Asterisk and using Asterisk as a media server for voicemail or busy tones I understand that part. But how could I send outgoing (pstn) calls to SEMS first and then to Asterisk? Is there something like a service route for this? The dialplan in Asterisk is much much much more flexible than a lot of people seem to realize. It's in some ways quite a powerful programming language, although it does have weaknesses (some larger than others). And since you already have an Asterisk in the callpath it seems to me to be superfluous to add another element, that will just make things a lot less reliable. You can do conditional branching and database queries from the dialplan, that's all the power required to create a variable experience for each call. It just takes a little lateral thinking and some tinkering. If you want to you could use an AGI script, but I always feel like that being a cop-out, it's more fun to do it from the dialplan. Now, let's get back to OpenSIPS ;-) -- 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 alwaysreturning 404
Hi, try doing it step by step - first call between 2 sip users - register them to your proxy (use opensipsctl ul show to see the registered accounts) - if you get a 404 when dialing from user A to user B, it means user B is not registered with the system. Regards, Bogdan Aditya Kumar wrote: Hi All, Thanks for the responses for the previous questions. I have gone thru all suggestions and made some progress. This is my new issue: 1) For any call I make to opensips, I am getting 404. All that I want to do is opensips to act as a relay proxy between by UA and a proxyserver. I don't need Autentication. I did not enable/use the database. I am using the default opensips.conf file. As I am making the opensips as a relay proxy. I am not using the register message. All I need is a invite coming to the opensips to be forwarded to the UA on the other side . Please let me know what all I should make in the UA and opensips. -- If you feel that I should use the db, so that incoming From UA1 to opensips (to u...@opensips) there must be a db updated with UA2 and its corresponding IP address pl let me know how to use the db. (any url for this is also ok. I searched the opensips I did not get info directly that I am looking for). waiting f ur suggestions/help on this,. ___ 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] Getting Error when using STUN
Hi Ahmed, in stun module, you need to configure IPs that are local to your server - some of these IP can be also configured as listeners for SIP traffic in OpenSIPS. the error you get points to the fact you are using an IP that is not locally configured on the machine. Regards, Bogdan Ahmed Munir wrote: Hi, I'm getting and error when I configure STUN in OpenSIPs. As following documentation of OpenSIPs, I enabled stun as listed below; listen=udp:198.65.166.165:5060 http://198.65.166.165:5060 listen=udp:75.101.138.128:5060 http://75.101.138.128:5060 loadmodule stun.so # Stun modparam(stun,primary_ip,198.65.166.165) modparam(stun,primary_port,5060) modparam(stun,alternate_ip,75.101.138.128) modparam(stun,alternate_port,5060) Where the IP of OpenSIPs which is hosted on public IP i.e. 11.22.33.44. And the error I'm getting after restarting the OpenSIPs is listed below; May 17 05:37:13 newtest /usr/local/sbin/opensips[31199]: ERROR:core:udp_init: bind(5, 0x81b7374, 16) on 77.66.16.35 http://77.66.16.35: Cannot assign requested address When I commented out the listen=udp:IP:5060 the error I'm getting is; May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if host==us: 11==9 [77.66.16.35] == [127.0.0.1] May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if host==us: 11==11 [77.66.16.35] == [77.66.2.137] May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:core:grep_sock_info: checking if port 5060 matches port 5060 May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: DBG:stun:stun_mod_init: grep_sock_in()1 failed May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: ERROR:core:init_mod: failed to initialize module stun May 17 05:59:48 newtest /usr/local/sbin/opensips[31303]: ERROR:core:main: error while initializing modules Kindly assist me how can I resolve this problem. -- Regards, Ahmed Munir ___ 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] SIP Presence Aggregation Issue
The specs may lack but the real problem is adoption in the real world. Most of the SIMPLE implementations are happening behind closed doors and are driven by IMS like deployments that have not come to term yet. The number of these implementations grown steadily if you get the statistics from SIPITs and meet the vendors in person. Commercial clients are available, Zoiper now implements full support for XCAP and Publish method. CounterPath latest build is working finally against OpenXCAP. Latest Snom software release 8 stores the contacts in XCAP server. Open source related, Ekiga and SIP Communicator are currently developing support for XCAP and SIP communicator is adding MSRP support including the Relay extension. Blink will tip this balance soon. Related to Inaki's efforts to make a change, I am sure he can make a great contribution. SIP however as it is deployed today cannot be used reliably for any form of remote storage of data as long as TLS is not mandatory. It is an endurance game, it will not happen over the night. Adrian On May 17, 2010, at 6:41 PM, Bogdan-Andrei Iancu wrote: Hi Iñaki, No worry, no rudeness felt ;) I guess everybody feels a bit (or more) frustrated about the current SIMPLE situation. I'm really interested to see the outcome of your specs - my question is (assuming that from tech point of view, it is a good approach), how do you foresee the wide spreading of the specs ? Regards, Bogdan PS: if you need help on the specs or testing, let me know ;) Iñaki Baz Castillo wrote: 2010/4/12 Bogdan-Andrei Iancu bog...@voice-system.ro: Hi Iñaki, Hi Bogdan, replies inline: well, right now there is a kind of pressure coming from the providers level - providers do want to offer presence with SIP ; also presence comes in a natural way of doing dome enhanced services (more complex than simple BLA, BLF, etc). -- please note I said presence, not SIMPLE. So, a natural demand for it there, and we, as developers, are looking for solutions to make it happened. and for implementations you need some specs. Now, if you see the SIMPLE specs are wrong - it might be - I'm not directly involved in the depth of SIMPLE to be able to say yes or no. This aggregation problem is the first we encountered during some projects - not only once, but several times, different contexts ; and I'm trying with Anca to see how to get over it. So, overall, there are 2 options (according to your perception): - use SIMPLE and get a poor result (a crippled presence) SIMPLE is not just poor, but also inneficient at server level (a single change in a XCAP document requires the presence server to reload all the permissions for that user). Even in case of solving it, the result owuld be poor, sure. - come up with a new spec Yes. I'm doing a presence spec for SIP from scratch, by learning about XMPP and so. I've already defined the concept of resources, different status priority, global status. And best of all, there is no HTTP/XCAP, but just SIP. Well, I have to spent lot of hours yet :) - do feedback to IETF to make SIMPLE simpler and working IMHO this is not possible at this point, as IETF already chose XCAP for buddies and permissions management (along with others). IMHO there is no way to improve/fix current SIMPLE specs. For SIMPLE, looking at the basics (exchanging the info), the aggregation is the biggest issue I see. Whatever is on top (RLS, XCAP, buddy lists, etc) is another story and it might need a second look and thought. OMA tries to define an aggregation mechanism (like rules). I've read it, and it's a pain, a dirty hack over IETF *incomplete* specifications. Sorry for sounding so rude :) -- Bogdan-Andrei Iancu www.voice-system.ro ___ 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] re : opensips as a SIP AS..
Thanks for pointing me in the right direction. Appreciate it. I will give it a try with B2BUA .. thanks, --Jignesh From: Bogdan-Andrei Iancu bog...@voice-system.ro To: OpenSIPS users mailling list users@lists.opensips.org; jhgan...@yahoo.com Sent: Mon, May 17, 2010 10:07:20 AM Subject: Re: [OpenSIPS-Users] re : opensips as a SIP AS.. Hi, A simple proxy may not generate a re-INVITE, but the b2bua part of opensips can; see the local REFER handling using a b2bua scenario: http://www.opensips.org/Resources/B2buaTutorial#toc15 Regards, Bogdan Iñaki Baz Castillo wrote: 2010/5/15 jignesh gandhi jhgan...@yahoo.com: Hello, Reading the features list it shows that opensips can be used as a SIP AS. Not clear if I can do the following ... 1} Can opnesips process REFER and generate a RE-INVITE and then response back to the REFER with appropriate NOTIFY ? Hi, you already asked the same in kamailio maillist and several people explained why you are wrong. A proxy MUST NOT generate a RE-INVITE nad NOTIFY upon receipt of a REFER as a proxy is... a proxy, and not a UA (phone) or PBX (B2BUA). Please, don't ask it again, and at least reply to the people already tried to help you. -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] re : OpenSIPs
Hello, I have put your reply below... Here are my comments... I was not attempting to use OpenSIPS as a PROXY as I am well aware of what a PROXY can and can't do, but was looking to perform one of the functionalities of a SIP AS ( in an IMS ). No where in my question did I mention PROXY, I wish you would have read the question carefully before replying, thanks any way.. I will reply back to people that tried to help me in Kamailio mailing list in that mailing list. thanks, --Jignesh Message: 4 Date: Sat, 15 May 2010 22:57:27 +0200 From: I?aki Baz Castillo i...@aliax.net Subject: Re: [OpenSIPS-Users] re : opensips as a SIP AS.. To: OpenSIPS users mailling list users@lists.opensips.org Message-ID: aanlktilo9o4l3h2yeanxwytvzuah5eseait_qu0uz...@mail.gmail.com Content-Type: text/plain; charset=UTF-8 2010/5/15 jignesh gandhi jhgan...@yahoo.com: Hello, Reading the features list it shows that opensips can be used as a SIP AS. Not clear if I can do the following ... 1} Can opnesips process REFER and generate a RE-INVITE and then response back to the REFER with appropriate NOTIFY ? Hi, you already asked the same in kamailio maillist and several people explained why you are wrong. A proxy MUST NOT generate a RE-INVITE nad NOTIFY upon receipt of a REFER as a proxy is... a proxy, and not a UA (phone) or PBX (B2BUA). Please, don't ask it again, and at least reply to the people already tried to help you. -- I?aki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
