sr-users October 2005

sr-users@lists.kamailio.org

214 participants
457 discussions

[Serusers] fee-for-service SER server - clarified

by Michael Kendall

Greger, Thanks so much for your reply to my question, and thanks also to Olivier, Bravo, Iqbal, and Phil, who each sent me some great SER server recommendations or requested clarification to my confusing question. You are right that I did not provide enough info in my question. I do not need any of the example services that you mentioned in a, b, c, d. I don't want to control or customize any SER server, I only want to be a normal customer of an existing SER server. I do not need any access to regular telephones. What I want to do is have a call center with 10 tutors, with each tutor connected to a different remote student, The 10 teachers and 10 students would each be using the Windows (SIP) Messenger softphone and would find and call each other using the SER server. The SER server would need to allow media-proxy service so that we can traverse NATs and firewalls. I have selected Windows (SIP) Messenger as my softphone because it is cost-free (quite important if I have 300 students) and because it has a whiteboard (required for the tutoring). If someone can suggest a better softphone which has a whiteboard I would be very happy, especially if the softphone supports STUN. If the suggested softphone is free, that would be perfect. I can't really afford to use a softphone which charges me a fee for each named user, but if I need to pay a reasonanable price for each SIMULTANEOUS user, that would be ok. Most of the SER server websites that I have visited seem to recommend the Xten softphone, but since it has no whiteboard, I cannot use it. Windows (SIP) Messenger does not appear to support STUN, so media-proxy is the only way I know to traverse NATs. (Please correct me if I am wrong about Messenger not supporting STUN.) I know that the newest versions of Windows (SIP) Messenger do not allow connecting to 3rd party servers, so I plan to use the older versions that do allow it. Some of the SER server websites ( like fonosip ) claim (after a search) that they support Windows (SIP) Messenger, but they do not offer any configuration instructions. The fonosip site also does not say whether they will offer media-proxy services, which I need for NAT traversal. It is strange that these SER service provider websites provide very little info about whether they support media-proxy or how to configure a softphone for STUN. I know that media-proxy places a heavy load on the SER server, and that some SER servers are therefore reluctant to consume their expensive bandwidth by offering media-proxy services. But since there must be many other customers who are willing to pay for media-proxy service, there should be a group of SER servers who are offering this service for a fee. I just need to find one. Can someone please suggest a SER server which might satisfy the Messenger and media-proxy requirements? And can someone recommend a better softphone with whiteboard? thanks, Michael Greger wrote: Dear Michael, I know that many on this list operate own services they sell in the market. Others sell software packages and services, both to corporations (ex. Asterisk + ser in a combo setup) and to service providers (some sort of white-label service). There are also hosted services for enterprises that are not based on SER. I would suspect that lack of answer may have something to do with lack of information. You are not saying what you need. Here are some examples: a. A hosted SER server where you can log in and do changes to ser.cfg (for fun or commercial) b. A hosted corporate PBX-type solution c. A hosted SIP service ? la university setups with ENUM lookup and authentication against your own user database d. A white-label service that you will brand as your own and resell in a given market And BTW, you are not saying anything about your requirements (just a vanilla SER server?!) nor the size of the subscriber base you want to support etc etc. If you are looking for b, you are probably better off looking in your local market for a telephony provider who can give you IP access. g-)

19 years

[Serusers] need a help!!!

by deviator

Hello! I'm using openser 0.9.5 with rtpproxy, i'm found a good config for this on onsip.org site. All works fine with public ips ua's, but i have a problem with nat :( server have a public ip, ua's have a private ips, symmetric nat. I have an acknowledgement error (ACK Timeout) and calls failure after 30-32 sec. What should i change or add in openser.cfg (nat-rtpproxy.5.0.cfg version) pls help!!!

19 years

[Serusers] When to use MediaProxy

by tcchan

Hi All, I have SER work together with MediaProxy and two UAs; UA1 is NATED whereas UA2 has public IP. I have something like this to process the INVITE if (client_nat_test("3")) { setflag(4); }; if (isflagset(4)) { force_rport(); fix_contact(); use_media_proxy(); }; t_on_reply("1"); and on the reply_route, I have if (client_nat_test("1")) { setflag(4); }; if (isflagset(4) && status=~"(180) || (183) || 2[0-9][0-9]") { if(search("^Content-Type: .*$")) { fix_contact(); use_media_proxy(); } }; Now, if UA1 (nated) calls UA2, it is OK ( mediaproxy is properly setup and both side can hear each other. The client_nat_test() is met and flag4 is set which caused both the invite and the on_reply to use_media_proxy). If UA2(public) calls UA1 (nated), problem arises, UA2 can hear UA1 but not the reverse. The client_nat_test("3") is not met and thus flag4 not set. On the reply_route, although client_nat_test("1") is met, flag4 set and use_media_proxy() is executed, it return error: use_media_proxy(): empty response from mediaproxy ERROR: on_reply processing failed To temporary work around this problem, I forcefully set flag4 on and thus all calls route through the mediaproxy ( which is expensive in bandwidth usage). Is there any better way to solve this? Any suggestion? Has anybody already looks into this and has a solution? Thanks for any help available. Regards, TC Chan

19 years

[Serusers] Inserting SER into my voice network

by Ray Van Dolson

First off, here is my voice network layout currently: http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png We're using Asterisk for voicemail, call routing (for long-distance, LNP, etc) and SER/rtpproxy at the other end which handles NAT onto private networks where customer's exist. This setup works fairly well for the most part, except that Asterisk does not have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy) for their jitter buffer on both ends of our voice links here. To me, that would mean shoving a SER/rtpproxy combo between Asterisk and our provider network. Possibly on the same server. I could easily throw up a SER installation, but I'm trying to figure out if there's any way to leave Asterisk in the SIP path but remove it from the media path (have RTP just go straight from SER/rtpproxy to my provider's RTP proxy). Have any of you set up a scenario somewhat like this? Any recommendations? Asterisk CVS-HEAD does let you apply a patch and get some very alpha jitter buffer support. But CVS HEAD doesn't work reliably for me at all currently so I'm sticking with the latest stable release of 1.0.x series. I'm running SER 0.9.3 FYI on the SER proxies I have set up currently. Thanks. -- Ray Van Dolson Linux/Unix Systems Administrator Digital Path, Inc.

19 years

[Serusers] Please advise...

by Stanly Levithan

Dear SER-users, We are looking to establish an educational business, namely a virtual learning center,using a conference service providing capabilities with the following calls flow: --Caller dial a local number -- Media gateway/media server (MG/MS)accepts the call and automatically mutes it so all the callers become passive listeners if number is registered in the Data Base; if the number is not registered the call is discarded --MG/MS plays Welcome message --MG/MS asks for a virtual conference room number --Caller enters virtual conference room number --Caller is entered into virtual conference room associated with the virtual conference room number. All callers entered virtual conference room number are now in associated virtual conference room --MG/MS accepts any callers decision to switch to a different virtual conference room by entering appropriate virtual conference room number without making another call (without leaving the system) --MG/MS permits the caller to terminate the call at will --MG/MS of our dream has to be able to process up to 50, 000 callers simultaneously --MG/MS provides billing. Kindly let us know if SER is capable of supporting such a calls flow. Please let us know if you need any additional information in order to advise us on a most efficient configuration. Regards, Stan *** This email was scanned by eSafe Content Inspection Server. *** __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com

19 years

[Serusers] Please advise...

by Stanly Levithan

Dear SER-users, We are looking to establish an educational business, namely a virtual learning center, using a conference service providing capabilities. Kindly take a look at our business requirements described in the attachment and let us know if SER is capable of supporting such a calls flow. Please let us know if you need any additional information in order to advise us on a most efficient configuration. Regards, Stan *** This email was scanned by eSafe Content Inspection Server. *** __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

19 years

[Serusers] SER installation procedure for Redhat linux 9.0: Please note this

by Dhiman D Chowdhury

Dear friends, If you are installing SER for the first time in REDHAT LINUX 9.0, please do the followings: 1. download ser-0.9.3_linux_i386_tar.gz 2. save it to a user directory 3. login as root and go to / directory 4.from "/" as root run the following command: tar -zxvf /home/user/ser-0.9.3_linux_i386_tar.gz [replace the /home/user directory with the directory name where you have initially downloaded the ser file. Please note ser-0.9.4_linux_i386.tar.gz will not work if you have downloaded it. For ser 0.9.4, you need to make binary by yourself. 5. Now run /usr/local/sbin/ser 6. Your SER server is running. 7. At this point you can download xten pro sofphone client and configure it to connect to your serv server. 8. Just to check out whether or not xten is registering with the server use admin as user and heslo as password. Now you are ready to connect mySQL server and follow the procedure for that. Thanks Dhiman Deb Chowdhury

19 years

[Users] Making OpenSER give up being man-in-the-middle?

by Juan Carlos Castro y Castro

Hello, I'm using OpenSER to make load balancing among a bunch of Asterisk boxes. I whipped up a very simple script that works: # module stuff goes before here route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; route(1); break; } route[1] { exec_dset("/usr/local/bin/selectserver.sh"); if (!t_relay()) { sl_reply_error(); }; } selectserver.sh alters the server in the destination URI to choose from a dynamic pool of servers that's always changing according to a funky proprietary logic of ours. This works just peachy. The problem is OpenSER stays as man-in-the-middle all the time. I'd like it to do a REFER to a random server of our choosing and stay out of the loop from there on. How could I do that?

19 years

[Serusers] Ser v0.9.4 error message " /lib/tls/libc.so.6: version `GLIBC_2.3.4' not found (required by ser)

by Dhiman D Chowdhury

Hi, I am using redhat linux 9.0 and installed ser v0.9.4. While I am trying to start ser, I get the following message: /lib/tls/libc.so.6: version `GLIBC_2.3.4' not found (required by ser) I will appricate your kind help in this regard. Thanks Dhiman Chowdhury

19 years

[Users] usrloc-cl with openser

by Daryl Sanders

Greetings Everyone, Has anyone tried the Andreas' cacheless usrloc module with openser. Will it run on openser? I am thinking of testing the new openser release and I like to have this functionality if possible. Thanks! - Daryl

19 years

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sr-users October 2005