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37541 discussions

[Serusers] SER replies to CANCEL too early
by Esteban Monturus 15 Apr '05

15 Apr '05

I am installing a SIP proxy in a Linux machine in the local network of my office. I have implemented a SIP proxy with SER. The function of the SIP proxy is to let the users talk each other without knowing the IP where each one is reachable. My problem appears when the caller wants to cancel the call because the callee does not answer. Since I want the calle's phone to stop ringing, and I haven't received a final response, the caller's phone has to send a CANCEL request. When the proxy server receives the CANCEL, it replies with a 487 "Request cancelled". SER does this before forwarding the CANCEL to the callee's phone. These 487 response generated by SER makes the INVITE client transaction of the caller finish. When the INVITE server transaction finishes in the callee's phone, it sends another 487 response, which is forwarded by SER, but the caller's transaction is already finished, so it doesn't receive the corresponding ACK. I would like to know how to solve this problem, since SER should wait for the callee's phone to send the 487 and then forward this response to the caller's phone. Best regards, Esteban. ------------------------------------------------------------------------ --------------- Esteban Monturus de Carandini (mailto:emonturus@teltronic.es) Enhanced Wireless Solutions (EWS Project) TELTRONIC, S.A.U. Parque Tecnologico Walqa Edificio Uno Ctra Zaragoza N-330a, Km 556 22197 Cuarte (Huesca) - Spain Phone/Fax: +34 974 215018 http://www.teltronic.es http://ptwalqa.com ------------------------------------------------------------------------ --------------- ***** AVISO LEGAL ***** Este mensaje es solamente para la persona a la que va dirigido. Puede contener informacion confidencial o legalmente protegida. La transmision erronea de este mensaje no supone renuncia a su confidencialidad o a cualquier privilegio. Si usted ha recibido este mensaje por error, le rogamos que borre de su sistema inmediatamente el mensaje asi como todas sus copias y que notifique al remitente. No debe, directa o indirectamente, usar, revelar, distribuir, imprimir o copiar ninguna de las partes de este mensaje si no es usted el destinatario. Cualquier opinion expresada en este mensaje proviene del remitente, excepto cuando el mensaje establezca lo contrario y el remitente este autorizado para establecer que dichas opiniones provienen de TELTRONIC. En el caso de que el destinatario de este mensaje no consienta la utilizacion del correo electronico via Internet, rogamos lo ponga en nuestro conocimiento de manera inmediata. ***** DISCLAIMER ***** This message is intended exclusively for the named person. It may contain confidential, propietary or legally privileged information. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this message in error, please immediately delete it and all copies of it from your system, destroy any hard copies of it and notify the sender. Your must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message if you are not the intended recipient. Any views expressed in this message are those of the individual sender, except where the message states otherwise and the sender is authorised to state them to be the views of TELTRONIC. If the addressee of this message does not consent to the use of internet e-mail, please communicate it to us immediately. ------------------------------------------------------------------------ ---------------

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[Serusers] parallel fork woes
by Daniel Poulsen 15 Apr '05

15 Apr '05

Hi- I am having a problem where if I parallel fork a call, my failure_route gets called after the caller hangs up. I thought adding t_check_status("487") to the failure_route would take care of that, but it still t_relays when I don't want it to. This only seems to be a problem if it parallel forks and one of the SIP targets is not available. If I dial an alias with a single [not forked] target the t_check_status("487") works fine. Here is my code.. any thoughts? Thanks much. Dan route { <<<< Registration code snipped for brevity >>>> if(lookup("aliases")) { xlog("L_NOTICE", "%ci: alias lookup changed uri to %ru\n"); # If alias points outbound, forward it without question. if (!(uri==myself)) { xlog("L_NOTICE", "%ci: outbound alias\n"); route(4); # relay with hunt on failure break; }; }; if ( (uri=~"^sip:911@.*") | (uri=~"^sip:011[0-9]+@.*") | (uri=~"^sip:1[0-9]{10}@.*") ) { route(3); break; }; # Local location route(4); # relay with hunt on failure } /* end of initial routing logic */ route[4] { xlog("L_INFO", "r4: Relay with Hunt\n"); # If an invitation, we want to hunt on failure if(method == "INVITE") { t_on_failure("1"); # first hunt }; if (!lookup("location")) { xlog("L_NOTICE", "%ci: no location for %ru\n"); }; append_hf("P-hint: relay\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } failure_route[1] { xlog("L_INFO", "failure+route+1 %ru\n"); revert_uri(); setflag(9); # Check to see if the call was cancelled if( t_check_status("487") | method == "CANCEL") { xlog("L_INFO", "t_check_status is 487\n"); break; } prefix("h1-"); if(lookup("hunt")) { # Hunt table is an exact copy of aliases xlog("L_NOTICE", "%ci: fail_r_1: hunt changed URI to %ru, relaying\n"); append_branch(); append_hf("P-hint: pt-hunt\r\n"); t_relay(); break; } else { xlog("L_NOTICE", "%ci: r8: no further hunts, giving up\n"); break; }; }

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[Serusers] Rewrite To Heaader
by Eric Dean 15 Apr '05

15 Apr '05

Our VoIP service provider routes calls based upon the To field within the SIP header. While SER can easily rewrite the URI, I need to actually prepend our account code to every outgoing call with a prefix. does anyone know how to prefix the To field

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[Serusers] ERROR: No Digest-Nonce Cannot perform Digest authentication
by Alex 15 Apr '05

15 Apr '05

I have installed freeradius 1.0.2 and radiusclient-ng.0.5.0 after the installation i am trying to run the test : radtest test test localhost 1649 testing123 User-Name = "test" User-Password = "test" NAS-IP-Address = MAINSIP NAS-Port = 1649 rad_recv: Access-Reject packet from host 127.0.0.1:1812, id=134, length=45 Reply-Message = "Hello, test with digest" I have file digest: --------------------------------------- User-Name = "test", Digest-Response = "631d6d73147add2f9e437f59bbc3aeb7", Digest-Realm = "testrealm", Digest-Nonce = "1234abcd" , Digest-Method = "INVITE", Digest-URI = "sip:5555551212@example.com", Digest-Algorithm = "MD5", Digest-User-Name = "test" in the debug of the radius server i receive that error: ---------------------------------------------------------------------- rad_recv: Access-Request packet from host 127.0.0.1:32842, id=115, length=56 User-Name = "test" User-Password = "test" NAS-IP-Address = 255.255.255.255 NAS-Port = 1649 modcall: entering group authorize modcall[authorize]: module "preprocess" returns ok modcall[authorize]: module "chap" returns noop rlm_eap: EAP-Message not found modcall[authorize]: module "eap" returns noop modcall[authorize]: module "digest" returns noop rlm_realm: No '@' in User-Name = "test", looking up realm NULL rlm_realm: No such realm "NULL" modcall[authorize]: module "suffix" returns noop users: Matched DEFAULT at 152 users: Matched test at 215 modcall[authorize]: module "files" returns ok modcall[authorize]: module "mschap" returns noop modcall: group authorize returns ok rad_check_password: Found Auth-Type Digest auth: type "digest" modcall: entering group authenticate ******ERROR: No Digest-Nonce: Cannot perform Digest authentication****** modcall[authenticate]: module "digest" returns invalid modcall: group authenticate returns invalid auth: Failed to validate the user. Thanks for the help

1 0

[Serusers] Another Looping Question
by Java Rockx 15 Apr '05

15 Apr '05

Hi All. Below is a snippet of an INVITE and you can see the usual "trying - your call is important to us" in the dialog. Anyhow, for some reason SER occasionally attempts to sent the ACK to itself, which causes a looping condition. The last ACK shown is the start of the looping condition. It only happens on occasion. SER should have consumed the ACK rather than attempted to forward it. Can anyone explain this? Regards, Paul IP Legend ------------------- 10.3.0.221 <http://10.3.0.221> - SER (behind a Cisco 3600 so ALG rewrote the IPs) 69.36.46.178 <http://69.36.46.178> - NATed SIP UA U 2005/04/15 02:19:16.252614 69.35.46.178:5060 <http://69.35.46.178:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> INVITE sip:8187768080@sipdev.mycompany.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3915905414. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>. Call-ID: 2671219156(a)192.168.1.102. CSeq: 101 INVITE. Contact: <sip:3215591439@192.168.1.102:5060>. Proxy-Authorization: Digest username="3215591439", realm=" sipdev.mycompany.net <http://sipdev.mycompany.net>", nonce="425f5e0de25efdb3f5a6125dcf2147bf30fe9177", uri=" sip:8187768080@sipdev.mycompany.net", response="61996c3bdc10990fec7e245751777516", algorithm=MD5, cnonce="ae02e47a3355bada49ce344c92f48587", qop=auth, nc=00000002. max-forwards: 70. Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY. Content-Type: application/sdp. Content-Length: 180. . v=0. o=- 32014 3005 IN IP4 192.168.1.102 <http://192.168.1.102>. s=-. c=IN IP4 192.168.1.102 <http://192.168.1.102>. t=0 0. m=audio 13456 RTP/AVP 18 0 8 2 4 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. . # U 2005/04/15 02:19:16.276216 10.3.0.221:5060 <http://10.3.0.221:5060> -> 69.35.46.178:5060 <http://69.35.46.178:5060> SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3915905414;rport=5060;received=69.35.46.178<http://69.35.46.178> . From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>. Call-ID: 2671219156(a)192.168.1.102. CSeq: 101 INVITE. Content-Length: 0. . # U 2005/04/15 02:19:16.792795 69.35.46.178:5060 <http://69.35.46.178:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> ACK sip:8187768080@sipdev.mycompany.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3499714875. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>;tag= cdbfe46960d7566fcbe09f199f2b328d.c2bb. Call-ID: 2671219156(a)192.168.1.102. CSeq: 100 ACK. Content-Length: 0. . # U 2005/04/15 02:19:16.912146 10.3.0.221:5060 <http://10.3.0.221:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> ACK sip:8187768080@sipdev.mycompany.net SIP/2.0. Max-Forwards: 10. Record-Route: <sip:10.3.0.221 <http://10.3.0.221>;ftag=1465261089;lr>. Via: SIP/2.0/UDP 10.3.0.221 <http://10.3.0.221>;branch=0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060>;received= 69.35.46.178 <http://69.35.46.178>;branch=z9hG4bK3499714875. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>;tag= cdbfe46960d7566fcbe09f199f2b328d.c2bb. Call-ID: 2671219156(a)192.168.1.102. CSeq: 100 ACK. Content-Length: 0. P-hint: Local Destination.

1 0

[Serusers] DNS Lookup and other issue
by info＠beeplove.com 15 Apr '05

15 Apr '05

If I setup an UA with URL sip:beeplove@mydomain.org my client is going to find its sip server from DNS SRV record, right? If I have these in nameserver _sip._udp SRV 0 0 5060 sip.mydomain.org. _sip._tcp SRV 0 0 5060 sip.mydomain.org. is that mean, sip.mydomain.org is my registar server? or proxy server or both? And If my UA have an option to specify proxy and registar server, UA is not going to lookup SRV record, right? Please correct me someone, if I am understading wrong. Thanks Mohammad -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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[Serusers] Not ANOTHER silly NAT question...
by Rod Bacon 15 Apr '05

15 Apr '05

Well.. it probably is. I have had mixed success with SER 0.8.14 and RTPPROXY. I have RADIUS auth, accounting, MYSQL, blah, blah, blah, all working.... My question, however, surrounds NAThelper configuration. If I leave all clients behind NATs to attempt registration with private IPs, fix the IPs with NAThelper and proxy the streams, then everything works OK. If I inroduce a client that is on a LIVE IP address, I only get one-way audio to a natted client (the natted client can't hear the live client). I assume that what is happening is that the live client is attempting to send it's audio direct to the natted client, instead of via the proxy. I figure that I've missed something silly somewhere in the config, but I'm darned if I know where. I won't bother posting my entire config (unless specifically requested), rather I'd like a general pointer as to where I should be looking. Thanks in advance.

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[Serusers] Ser Redirect Server !!!
by Felipe Martins 15 Apr '05

15 Apr '05

Hi everybody, I finally managed to configure and make 2 Ser Servers talk to each other, each one with a registered soft phone or hard phone, they are talking with no delay and nearly no noise. I've made a redirect configuration at the routing logics and for me it worked. I found a similar configuration at the link: (http://www.voip-info.org/wiki-SER+example+redirect), but it's totally diferent from what I've done. This is what i've done. -------------------------------------------------------------------------- if (method=="REGISTER") { log(1, "REGISTER message received\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("192.168.4.11", "subscriber")) { www_challenge("192.168.4.11", "0"); break; }; save("location"); break; }; # Repassando Chamadas Internacionais para Asterisk if (uri=~"^sip:[2][0-9].*@") { log(1, "Forwarding to Another Gateway\n"); rewritehost("192.168.4.10"); forward(192.168.4.10,5060); t_relay(); break; } ------------------------------------------------------------------------- Is there some error in this configuration ? -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

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[Serusers] Strange problems
by Alex 15 Apr '05

15 Apr '05

Hi guys thanks for any help, i got a lot of useful information from this list. I will try to explain my problem. i have 2 linux servers : 1- server Red Hat Linux release 9 ser + freeradius 0.9.3 + radius client 2-server Red Hat Enterprise Linux AS release 3 ser + freeradius 1.2 + radius client 4.8 I trying to register my ip phone through ser + radius . On the 1 server it's working fine i receiving register requests, i can see that in the logs, and the request is going through the authentication process. On the 2 server i receiving register requests, but it's not going to the radius authentication, i don't see anything inside the radius logs. Here is the ngrep on port 5060: from the 2 server ------------------------------------------------------------------------------------------------- U telephoneip:10739 -> xxx.xxx.xxx.xxx:5060 REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" < sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>..Contact: <sip :Phonenumber@telephoneip:10000;user=phone>..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..Expires: 3600..User-Agent : Grandstream HT286 1.0.5.18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt h: 0.... # U xxx.xxx.xxx.xxx:5060 -> telephoneip:10000 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=b27e1a1d33761e85846fc9 8f5f3a7e58.f894..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonc e="425e3ac34dc9509392435c11fb260f41420049c7"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=1912 req_src_ip=telephoneip req_src_port=10739 in_uri=sip:xxx.xxx.xxx.xxx out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1".... # U telephoneip:10740 -> xxx.xxx.xxx.xxx:5060 REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" < sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>..Contact: <sip :Phonenumber@telephoneip:10000;user=phone>..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..Expires: 3600..User-Agent : Grandstream HT286 1.0.5.18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt h: 0.... # U xxx.xxx.xxx.xxx:5060 -> telephoneip:10000 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=b27e1a1d33761e85846fc9 8f5f3a7e58.f894..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonc e="425e3acb812b5b2e8aa023e3fcffc618dc4cf661"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=1885 req_src_ip=telephoneip req_src_port=10740 in_uri=sip:xxx.xxx.xxx.xxx out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1".... # here is the debug from ser : --------------------------------------------------------------------------------------------- 14(1036) parse_headers: flags=-1 14(1036) check_via_address(62.219.158.191, 62.219.158.191, 1) 14(1036) DEBUG:destroy_avp_list: destroing list (nil) 14(1036) receive_msg: cleaning up 9(1012) SIP Request: 9(1012) method: <REGISTER> 9(1012) uri: <sip:xxx.xxx.xxx.xxx> 9(1012) version: <SIP/2.0> 9(1012) parse_headers: flags=1 9(1012) Found param type 232, <branch> = <z9hG4bKfc5751413c832e6d>; state=16 9(1012) end of header reached, state=5 9(1012) parse_headers: Via found, flags=1 9(1012) parse_headers: this is the first via 9(1012) After parse_msg... 9(1012) preparing to run routing scripts... 9(1012) REGISTER: Authenticating user 9(1012) parse_headers: flags=4 9(1012) end of header reached, state=9 9(1012) DEBUG: get_hdr_field: <To> [45]; uri=[sip:phonenumber@xxx.xxx.xxx.xxx;user=phone] 9(1012) DEBUG: to body [<sip:phonenumber@xxx.xxx.xxx.xxx;user=phone> ] 9(1012) parse_headers: flags=4096 9(1012) get_hdr_field: cseq <CSeq>: <103> <REGISTER> 9(1012) DEBUG: get_hdr_body : content_length=0 9(1012) found end of header 9(1012) pre_auth(): Credentials with given realm not found 9(1012) REGISTER: challenging user 9(1012) build_auth_hf(): 'WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonce="425e063022afc1142ed6730d46da41692ff3ed57" Daniel and Alex send some explanations i really appreciate that, but i don't know where to start to fix that problem, i am trying to understand why on the first server it's working , but on the second it's not working . both of them have the same configuration only the versions are different. I tried to connect different ip phone and it's sending the packets to the 2 server and it's working, but the 1 ip phone not working with the 2 server, but with the 1 one it's working fine, i just changing the sip proxy ip. Thanks for any help.

2 3

[Serusers] B2BUA and NAT.
by Amos Nungu 15 Apr '05

15 Apr '05

Hi. I can't find enough info regarding B2BUA, and how it works. At least I know that it is not scalling and it breaks SIP peer2peer functionality. BUT, it is said that it can handle NAT, HOW? 1.Does it solve NAT because it stays in the middle and proxy everything (+ media) or 2.can it be involved in the call only when NAT is detected like rtp proxy ? 3. iF 2, THEN how does it work in corporation with SER or any SIP Server? 4. Which situation will rtp proxy (rtpproxy/mediaproxy) fail to solve NAT issues NB. I just need to expand my knowledge regarding options. I'm using rtpproxy, it works fine in every respect only if I start it in foreground Regards.

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