sr-users October 2006

sr-users@lists.kamailio.org

189 participants
345 discussions

by Legos Engineering

HI, The log of xlog are generated in messages. So i would like to know, what i must do if i want that xlog send its information in an other file for only one request L_WARN(...) I want to use this request in order to do stats of connection of UA Thx for your help.

17 years, 6 months

[Users] doubt in Loose route section

by raviprakash sunkara

Hello Users, When I working account part and Nat with OpenSER, The Transaction of route record entered into the Loose route Section. if(loose_route()) { acc_db_request(" ",""); t_relay(); exit; }; # call type transaction taking place here......... .............. ..................... ................... ...................... ..................... .................. All the Accounting and voice part are working ... but i'm doubt in loose route taking place all the call transcation.... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

17 years, 6 months

[Serusers] Ser_acct_authentication using ser

by rahman atikur

Dear all: I am new in ser. I install ser 0.9.6. And make a call using halaphone.. Now i want to use radius server for authenticate. I already install Radius server freeradius-0.9.0 and radiusclient-ng-0.5.3 .. Now the problem is auth_radius module is not installed... >From the scratch ... how i solve the problem. Regards John __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

17 years, 6 months

[Users] How to leave offline-voice message?

by cv aik

Hi all, I'd like to ask some questions about MSILO module.. I have setup server and it works well, then I add msilo configuration into openser.cfg For addition, my clients are using MS Windows. I used 2 clients to test MSILO feature. I try to call the offline client, but my x-lite display that user not found. Please tell me what's the problem. Thanx Regards, Aldi --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.

17 years, 6 months

[Serusers] SER connecting to SIP VoIP providers

by Voipers Portugal

Dear all, I am am using SER to provide SIP service to my clients, but I would like to allow them to make outbound calls. This is where a SIP VoIP provider comes in. As far as I know, I have 2 possibilities: - using IP authentication - using client authentication As for the first one, I only have to forward the INVITE messages, right? What I would like to know, is how to use "user authentication". In this case, it would be transparent to the VoIP provider since it should be treated as a regular SIP user agent. Can anyone help me out? Jose Simoes

17 years, 6 months

[Users] how to handle "488 Not Acceptable Here"

by Richard Bennett

hi, When calling from Sipura to softphone (either SJphone or the sip client in my e61 nokia phone), with the Sipura set to Prefered codec: g723 Use prefered codec only: no and the softphone only supporting g711, I get this reply from the softphone: U xxxx:11474 -> xxxxx:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP xxxx;branch=z9hG4bK23e.248c6cd1.1,SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. To: <sip:1@test.com>;tag=ujk6mpqhbhhc6kj68irr. From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0. Call-ID: e5052808-575f7746(a)192.168.1.52. CSeq: 102 INVITE. Warning: 304 192.168.1.3 Media type not available. Content-Length: 0. the invite was: U xxxx:5060 -> xxxx:11474 INVITE sip:1@test.co SIP/2.0. Record-Route: <sip:xxxx;lr=on;ftag=d0404120874d710eo0>. Via: SIP/2.0/UDP xxxx:branch=z9hG4bK23e.248c6cd1.1. Via: SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0. To: <sip:1@test.com>. Call-ID: e5052808-575f7746(a)192.168.1.52. CSeq: 102 INVITE. Max-Forwards: 69. Contact: sipura line1 <sip:2@xxx:10001>. Expires: 240. User-Agent: Sipura/SPA2002-3.1.5. Content-Length: 419. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 36143 36143 IN IP4 xxxx. s=-. c=IN IP4 xxxxxx. t=0 0. m=audio 62238 RTP/AVP 4 0 2 8 18 96 97 98 100 101. a=rtpmap:4 G723/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. If I set prefered codec to g711 on the Sipura it works normally. What is the best way to handle this on the sip proxy? Thanks, Richard

17 years, 6 months

[Serusers] Accounting problem with radius server

by Caxton

Dear all: I setup SER and use radius server to accounting. However, the record after I test are not complete. A(Caller) <------> B(Callee) Suppose A is caller and B is callee. The record will be only one and correct if A hang up first. Unfortunately, if B hang up faster than A, the record will be divide into two records like table1(for example.). Table 1. |-----------------------------------------------------------------------------------------------------------------| |AccSessionID AccUniqueId ...<skip>... Start Time Stop Time ...<skip>... | |-----------------------------------------------------------------------------------------------------------------| |AHandUpFirst 001 2006-10-02 2006-10-02 | | 12:40:03 12:45:32 | |-----------------------------------------------------------------------------------------------------------------| |BHandUpFirst 002 2006-10-02 0000-00-00 | | 12:58:40 00:00:00 | |-----------------------------------------------------------------------------------------------------------------| |BHandUpFirst 003 2006-10-02 2006-10-02 | | 13:05:03 13:05:03 | |-----------------------------------------------------------------------------------------------------------------| How can I solve this problem? Please give me a hand. Test User Agent: Window messenger 5.1 and X-lite Regards, Caxton

17 years, 6 months

[Users] OT: searching for a softclient with SDP feature

by Thomas Balsfulland

hello group, i am searching for a sip-softclient (windows or linux - opensource, freeware preferred), which is able to send an INVITE completely without any SDP information (no message body). does someone of you know something like this? regards thomas -- thomas balsfulland tbals(a)ctrl-c.de zwischen mut und dummheit liegt nur ein grat der sich durch das vorbereitetsein unterscheidet

17 years, 6 months

[Serusers] SER for inbound calls

by Voipers Portugal

Dear all, Another question that I have to do is: Imagine that I have a connection to a VoIP provider using a username and a password (so that SER register in the VoIP provider as a normal user). Once it is registered, do you think it possible to receive calls? That is, make it ring in one or all of the SIP users configured in SER? Any help would be appreciated. Jose Simoes

17 years, 6 months

[Serusers] DNS SRV support ?

by Kostas Marneris

Hello, I run ser.0.9.6 and I can use either rewritehostport("1.2.3.4:5060"); or rewritehostport("pstngw.foo.com:5060"); with t_relay() command on INVITE/PSTN handler in order to route calls to a Cisco PSTN GW. This works fine. Is "DNS SRV queries" supported on ser.0.9.6 and is there a specific conf ? thanx Kostas

17 years, 6 months

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sr-users October 2006