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37513 discussions

[Users] how to handle "488 Not Acceptable Here"
by Richard Bennett 02 Oct '06

02 Oct '06

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

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[Serusers] Accounting problem with radius server
by Caxton 02 Oct '06

02 Oct '06

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

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[Users] OT: searching for a softclient with SDP feature
by Thomas Balsfulland 02 Oct '06

02 Oct '06

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

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[Serusers] SER for inbound calls
by Voipers Portugal 02 Oct '06

02 Oct '06

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

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[Serusers] DNS SRV support ?
by Kostas Marneris 02 Oct '06

02 Oct '06

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

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[Serusers] problem
by vijay tiwari 02 Oct '06

02 Oct '06

i have install the ser and it installed but i have some problem 1. first i am not able to run the ser command 2. i want to make the scripts for ser and mediaproxy but its show me one error for running this command i am in /etc/init.d when i am running this command chkconfig --add ser error reading information on service ser: No such file or directory please help me out with regards vijay --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

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[Users] complete openser.cfg available?
by Ronald Wiplinger 02 Oct '06

02 Oct '06

I would like to see beside the sample and the "advices" to add to the cfg file a complete cfg file. Is such a more complese sample available? I am sure most of the users will use more than just the basics, and most of them will use the same modules and set-up. As new user, I face often the problem where to put the snip set from the documentation. I think it would be better to comment out all not necessary parts for the basic, or mention the spot to put in other configuration parts (anyway only the route() part needs that) bye Ronald

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[Serusers] UAC multiple Registration
by Abdul Qadir 02 Oct '06

02 Oct '06

Hi, Is it possible in SER to implement login concept like yahoo messenger, when you login from another pc your prevois login is signedoff by server. i.e unregister a previously registerd client when same client tries to register from another location. Thanks in advance, Abdul Qadir --------------------------------- Get your email and more, right on the new Yahoo.com

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[Users] Duplicating SIP Messages
by Adi Simon 01 Oct '06

01 Oct '06

Hi, I'm trying to figure out if it is possible to duplicate SIP Messages. In a setup of a SER proxy and two asterisk boxes I want to send all the Register messages to both of the asterisks, I tried doing the following in the openser.cfg: if (method=="REGISTER") { route(2); } else { route(1); }; route[2] { rewritehostport("10.0.0.1:5060"); if (!t_relay()) { sl_reply_error(); }; rewritehostport("10.0.0.2:5060"); if (!t_relay()) { sl_reply_error(); }; exit; } But I keep getting : 0(22183) ERROR: t_newtran: transaction already in process 0xb6136318 0(22183) DEBUG:sl:sl_reply_error: error text is I'm terribly sorry, server error occurred (1/SL) Is it possible at all? Thanks, Adi.

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[Users] Please help for tasting TLS with minisip
by Ferianto siregar 01 Oct '06

01 Oct '06

Dear Jeevan, How are you? I do hope your are fine. I would like to say thanks to you for your help before. Thank you. I am so sorry for disturb you. Sorry Jeevan, I still have problem with configuring minisip like I asked you a week ago. I still can not make a call with minisip. There is always error message. Can I ask you again, Please? Here is my problem: 1. I run ./gen_rooCA.sh and running this script, I got rootCA directory. This directory contains: a.cacert.pem b.index c.serial d.private directory which contains cakey.pem e. cert directory which contain 01.pem 2. I run ./gen_usercert.sh user and running this script,I got user directory. This directory contains: a.user-calist.pem b.user-cert.pem c.user-privkey.pem d.user-cert_req.pem 3. When running minisip (suppported TLS), I done these tasks: a.In "Preferences settings", I use TLS port in "enable TLS Local Port" and "Transport Method" is TLS. My Questions: 1. In "Personal Settings", what file should I put? I mean, in menu "choose a certificate", what should I browse to fill this? 2. In " choosen a private key", what should I browse to fill this? 3. In "CA database", what sholud I browse to fill this? Because when I fill "chosee a certificate" with "cacert.pem", the "choosen a private key" with "cakey.pem" ,and the " CA databese" with "user-cert.pem","cacert.pem","user-calist.pem","user-privkey.pem", I got the error message: The error message said that the "SSL : connect failed" Here is : SipMessageTransport: sendMessage: creating new socket IP4Address(string): sip.pcr.ac.id (202.95.149.251) Creating new SSL_CTX Enter PEM pass phrase:openser Unknown command: o Packets will be displayed to the screen Unknown command: e Unknown command: n Unknown command: s Unknown command: e SSL: connect failed SipMessageTransport: sendMessage: exception thrown! Registering all identities to their registrar: Registration request sent (username = aldi(a)pcr.ac.id) SipMessageTransport: sendMessage: creating new socket IP4Address(string): sip.pcr.ac.id (202.95.149.251) What should I do, Jeevan? I have tried it many times, but I still can not make a call with minisip. I realize that my skill is too weak. Would you mind guide me again to solve this problem? Please... Thank you very much.. I am sorry for waste your time, Thanks. Regards :) Ferianto Note: 1. I can not run openssl x509 -in cakey.pem -out cakey.crt. It says that it can not converted. TRUSTED CERTIFICATE 2. I copy all the file ("cacert.crt",cakey.pem","user-calist.pem",etc) from my linux server to windows. Because windows can not download with the extension .pem. Is it correct? --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

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