Unfortunately, this wasn't too successful.
I am now on a different way, trying to make a workaround on client side. There are mainly two solutions - both require UDP and can't handle TCP properly ( siproxd and the new linux module ip_conntrack_sip in netfilter's patch-o-matic ).
Does anyone know a way to enforce UDP on Snom 360 phones/UAs in general as transport mechanism?
Thanks in advance, Bernd Froemel
On Sat, 2005-06-25 at 18:35 +0200, Bernd Froemel wrote:
Ah, thanks - Would have been too easy to blame Snom for everything :)
You've been a great help, I will do some tests with your suggestions and will report back about the success.
Maybe someone should update the example etc/ser.conf.m4 and use a different approach to detect if an uri is really a private or just a nated target. I know, I know I could be the someone :)
Thank you again, Bernd
On Sat, 2005-06-25 at 18:27 +0200, Nils Ohlmeier wrote:
No it is not a Snom issue! A UA has to the option to implement the Outbound Proxy either by adding a Route header to the initial INVITE, or by just sending the INVITE without any Route header directly to the IP address of the Outbound Proxy. Snom, as well a almost all other vendors AFAIK, do implement the later. So it is completely normal that the initial INVITE's do not have a Route header.
Nils
On Saturday 25 June 2005 18:22, Bernd Froemel wrote:
So it is indeed also a Snom issue then (because I use the outbound proxy feature but there is no Route: added, thus because of the private IP in the URI I always get the "479 forward request to private ip address denied" message if I send anything witch goes route[1]) ?
Bernd
On Sat, 2005-06-25 at 18:14 +0200, Nils Ohlmeier wrote:
On Saturday 25 June 2005 17:59, Bernd Froemel wrote:
Thanks a lot! Didn't thought it could be the Snom phones - finally a new pointer! It got quite boring to watch ngrep and ser debug output while don't seeing anything really wrong.
But one thing is still a kind of riddle for me: are the Snoms not working properly on the private ip and Route: header too, or do I have something wrong in my cfg?
If you expect a Route header in the initial INVITE then please read this: http://www.iptel.org/ser/sipintro.html Normaly you wont see a Route header in initial INVITE's (except the UA uses the Route header for the outbound proxy).
Nils
There is another problem with my INVITEs - for some reason they don't
> get a 'Route:' header added; as it maybe is supposed to be. > > > #if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" > > && ! search("^Route:")) > > #{ > > # sl_send_reply("479", "forward request to private ip address > > denied"); # break; > > #};
Regards, Bernd
On Sat, 2005-06-25 at 16:42 +0200, Nils Ohlmeier wrote:
Hi Bernd,
I guess you use snom UA's with more or less recent firmwares. These UA's do not accept incoming TCP connections any more. SER tries to open new TCP connections to the UA, which is rejected. Thus it ends up with an transport error. I'm not very familiar with it, but you could have more luck if you play around with the tcp_accept_aliases and other tcp_*_aliases parameters on your SER.
Regards Nils
On Saturday 25 June 2005 11:12, Bernd Froemel wrote: > I could fix some of my problems by using a proper conntrack kernel > module in my router - now the phones behind the same NAT remain > reachable; at least as long as there is no server restart or one of > the phones goes offline and online again without doing a > unregister. > > Still a lot of > > er[27588]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection > > > refused > > ser[27588]: ERROR: tcpconn_connect: tcp_blocking_connect failed > > ser[27588]: ERROR: tcp_send: connect failed > > ser[27588]: msg_send: ERROR: tcp_send failed > > ser[27588]: ERROR: t_forward_nonack: sending request failed > > errors (especially on bye/cancel/acks which results in failure to > detect call terminations) while t_relay is issued. > > Anyone has an idea/pointer whats wrong here? > > There is another problem with my INVITEs - for some reason they > don't get a 'Route:' header added; as it maybe is supposed to be. > > > #if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" > > && ! search("^Route:")) > > #{ > > # sl_send_reply("479", "forward request to private ip address > > denied"); # break; > > #}; > > Could anybody please post me a correct INVITE to CANCEL/BYE session > with two phones behind the same nat? > > Thanks in advance, > Bernd > > On Thu, 2005-06-23 at 13:15 +0200, Bernd Froemel wrote: > > Dear list, > > > > I have some wired problem between users behind the same NAT > > connected to a multihomed ser server (public&private IP). Ser is > > listening only on the public interface (ser.ip.address). The > > NATed clients have official.ip.address as their official IP > > address and 192.168.10.1 and 192.168.10.2 are their IP addresses > > behind NAT. > > > > Now after a register I get for the NATed clients the following > > entries into the location table: > > > > UA1: > > contact: sip:test@192.168.10.1:2050;transport=tcp;line=1hzwxw3z > > received: sip:official.ip.address:2050;transport=TCP > > flags: 1 > > > > UA2: > > contact: sip:test2@192.168.10.2:2050;transport=tcp;line=1hzwxw3z > > received: sip:official.ip.address:2050;transport=TCP > > flags: 1 > > > > which looks quite good (why is the flag 1 and not 6?), but then > > on INVITE/SUBSCRIBE/ (everthing which issues a t_relay) my logs > > get full of: > > > > ser[27588]: ERROR: tcp_blocking_connect: SO_ERROR (111) > > Connection refused > > ser[27588]: ERROR: tcpconn_connect: tcp_blocking_connect failed > > ser[27588]: ERROR: tcp_send: connect failed > > ser[27588]: msg_send: ERROR: tcp_send failed > > ser[27588]: ERROR: t_forward_nonack: sending request failed > > > > and the UA which issued the command receives a: 477 Unfortunately > > error on sending to next hop occurred message. > > > > I think it has something to do with the lookup of the target UA - > > at least I always get the uri back which contains the private IP. > > (debugging output: > > > > befor lookup 80.123.216.181 - - sip:test@domain.com > > after lookup 80.123.216.181 - - > > sip:test@192.168.10.1:2050;transport=tcp;line=lhzwxw3z > > ) > > > > Also I guess that the commented part which is uncommented in the > > default cfg shouldn't prevent all NAT calls, but only calls to > > real private IPs. > > > > ( found in route[1] beginning, the nat route) > > > > #if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" > > && ! search("^Route:")) > > #{ > > # sl_send_reply("479", "forward request to private ip address > > denied"); # break; > > #}; > > ) > > > > But to due the lookup returning me a private IP uri this would > > result in a 479 error. > > > > My current test config is basically the one found in > > cvs/etc/ser.cfg.m4 with inserted values. I already tried other > > cfgs - even the example in modules/nathelper -- no success. Yes > > rtpproxy is running, I can even call the other UA and audio is > > working full duplex, but only at the very beginning. After a few > > minutes idleing or a server restart, without clearing the sql > > location table a call results always in the 477 error. > > > > > > Please help me - what am I missing here? > > > > Thanks in advance, > > Bernd > > > > ----- > > Here my cfg and some SIP messages during REGISTER and INVITE > > (I have removed IPs and unnecessary parts, Asterisk gw is > > 192.168.xx.xx): > > > > # > > # ----------- global configuration parameters > > ------------------------ > > > > [...] > > > > check_via=no # (cmd. line: -v) > > dns=yes # (cmd. line: -r) > > rev_dns=no # (cmd. line: -R) > > mhomed=1 > > > > [...] > > > > > > # ------------------ module loading > > ---------------------------------- > > > > [...] > > > > # ----------------- setting module-specific parameters > > --------------- > > > > [...] > > > > > > modparam("nathelper", "natping_interval", 15) > > modparam("nathelper", "ping_nated_only", 1) > > modparam("nathelper", "rtpproxy_sock", > > "unix:/var/run/rtpproxy.sock") > > > > modparam("registrar", "nat_flag", 6) > > modparam("registrar", "use_domain", 1) > > > > modparam("acc", "report_ack", 1) > > modparam("acc", "log_level", 1) > > #if BYE fails (telephone is dead, record-routing broken, etc.), > > generate #a report nevertheless -- otherwise we would have no > > STOP event; => 1 modparam("acc", "failed_transactions", 1) > > > > modparam("acc", "log_flag", 1) > > modparam("acc", "db_flag", 1) > > modparam("acc", "log_missed_flag", 3) > > modparam("acc", "db_missed_flag", 3) > > > > > > > > modparam("usrloc", "db_mode", 0) > > modparam("usrloc", "db_mode", 2) > > modparam("usrloc", "use_domain", 1) > > modparam("usrloc", "timer_interval", 10) > > > > modparam("auth_db", "calculate_ha1", yes) > > modparam("auth_db", "password_column", "password") > > modparam("auth", "nonce_expire", 300) > > > > modparam("rr", "enable_full_lr", 1) > > > > modparam("tm", "fr_timer", 20) > > modparam("tm", "fr_inv_timer", 90) > > modparam("tm", "wt_timer", 20) > > > > #modparam("enum", "domain_suffix", "e164.arpa.") > > > > modparam("msilo", "registrar", "sip:registrar@xxxxxxxx") > > > > alias=domain1.com > > alias=domain2.com > > > > /* > > flags: > > 1 ... ACCOUNT > > 3 ... MISSED CALLS > > 4 ... VOICEMAIL > > 6 ... NAT > > */ > > > > # ------------------------- request routing logic > > ------------------- > > > > # main routing logic > > > > route > > { > > # initial sanity checks -- messages with > > # max_forwards==0, or excessively long requests > > if (!mf_process_maxfwd_header("10")) > > { > > sl_send_reply("483","too many hops (loop?)"); > > break; > > }; > > if (msg:len >= 2048 ) > > { > > sl_send_reply("513", "message too large"); > > break; > > }; > > > > if (nat_uac_test("3")) > > { > > if (method=="REGISTER" || !search("^Record-Route:")) > > { > > if (method=="REGISTER") > > { > > fix_nated_register(); > > } else > > { > > fix_nated_contact(); > > }; > > if (method=="INVITE") > > { > > log(1,"natted caller sent invite!\n"); > > fix_nated_sdp("1"); > > }; > > force_rport(); > > setflag(6); > > log(1, "natted caller detected\n"); > > append_to_reply("P-NATed-Caller: Yes\r\n"); > > } else > > { > > log(1, "something wrong here..\n"); > > }; > > }; > > > > > > #antispam > > > > if ( > > search("(From|F):.*@(domain1.com)|(domain2.com)|(192.168.xx > > .xx)") ) > > { > > if ( (method=="INVITE" || method=="SUBSCRIBE") && !(src_ip == > > SER_IP > > > > || src_ip == 192.168.xx.xx) ) > > > > { > > if (!(proxy_authorize("","subscriber"))) > > { > > proxy_challenge("","0"); > > break; > > }; > > if (!check_from()) > > { > > log("LOG: From Cheating attempt in INVITE!\n"); > > sl_send_reply("403", "use From=id"); > > break; > > }; > > #consume_credentials(); > > }; #non-REGISTER from other domain > > } else if ((method=="INVITE" || method=="SUBSCRIBE" || > > method=="REGISTER") && !(uri==myself || uri=~"(@(192.168.xx > > .xx)([;:].*)*)") ) > > { > > sl_send_reply("403", "no relaying"); > > break; > > }; > > > > if (!(method=="REGISTER")) > > { > > record_route(); > > }; > > > > #if (method=="BYE" || method=="CANCEL") > > #{ > > # unforce_rtp_proxy(); > > #}; > > > > > > > > # subsequent messages withing a dialog should take the > > # path determined by record-routing > > if (loose_route()) { > > if ((method=="INVITE" || method=="ACK" || method=="CANCEL") && > > uri=~"(@(192.168.xx.xx)([;:].*)*)") > > { > > route(4); # to asterisk > > } > > else > > { > > append_hf("P-hint: rr-enforced\r\n"); > > if (method=="BYE") > > { > > setflag(1); > > }; > > log(1, "and directly to nat..\n"); > > route(1); # to nat > > }; > > break; > > }; > > > > if (!(uri==myself || uri=~"(@(192.168.xx.xx)([;:].*)*)")) > > { > > # mark routing logic in request > > append_hf("P-hint: outbound\r\n"); > > log(1, "outbound\n"); > > route(1); # to nat > > break; > > }; > > > > # ---->request is for our domains!<---- # > > > > if (method=="REGISTER") > > { > > if (!www_authorize("","subscriber")) > > { > > www_challenge("","0"); > > break; > > }; > > if (!check_to()) > > { > > log("LOG: To Cheating attempt\n"); > > sl_send_reply("403", "use From=id"); > > break; > > }; > > log(1, "(un)register successful\n"); > > if (!save("location")) > > { > > sl_reply_error(); > > }; > > m_dump(); > > break; > > }; > > > > if (uri=~"sip:daemon@") > > { > > sl_send_reply("410", "daemon is gone"); > > break; > > }; > > > > lookup("aliases"); > > > > if (!(uri==myself || uri=~"(@(192.168.xx.xx)([;:].*)*)")) > > { > > append_hf("P-hint: ALIASED-OUTBOUND\r\n"); > > route(1); #to nat > > break; > > }; > > > > if (uri=~"^[a-zA-Z]+:+[0-9]+@") > > { > > strip(1); > > prefix("00"); > > }; > > > > if (!does_uri_exist()) > > { > > if(uri=~"^[a-zA-Z]+:[0-9]+@") > > { > > route(4); #to pstn > > } > > else > > { > > sl_send_reply("604", "does not exist anywhere"); > > }; > > break; > > }; > > > > if (is_user_in("Request-URI", "voicemail")) > > { > > setflag(4); > > }; > > > > exec_msg("echo befor lookup $SIP_SRCIP - $SIP_ORURI - > > $SIP_RURI > > > > >> /tmp/ser.log"); > > > > if (!lookup("location")) > > { > > log(1, "lookup failed\n"); > > route(6); > > break; > > } else > > { > > log(1, "lookup successful\n"); > > }; > > exec_msg("echo after lookup $SIP_SRCIP - $SIP_ORURI - > > $SIP_RURI > > > > >> /tmp/ser.log"); > > > > if (uri=~"(@(192.168.xx.xx)([;:].*)*)") > > { > > log(1, "LOG: Gateway address in UsrLoc!\n"); > > route(4); # to PSTN > > break; > > }; > > > > if (method=="INVITE" && isflagset(4)) > > { > > t_on_failure("1"); > > }; > > > > setflag(3); > > > > append_hf("P-hint: USRLOC\r\n"); > > log(1, "nearly at end and going to nat..\n"); > > exec_msg("echo $SIP_SRCIP - $SIP_ORURI - $SIP_RURI >> > > /tmp/ser.log"); route(1); # to nat > > } > > > > route[1] > > { > > > > #if (uri=~"[@:](192.168.|10.|172.(1[6-9]|2[0-9]|3[0-1]).)" > > && ! search("^Route:")) > > #{ > > # sl_send_reply("479", "forward request to private ip address > > denied"); # break; > > #}; > > > > if (isflagset(6)) > > { > > if(!is_present_hf("P-RTP-Proxy")) > > { > > force_rtp_proxy(); > > append_hf("P-RTP-Proxy: Yes\r\n"); > > log(1, "rtp proxied\n"); > > }; > > log(1,"natted calee\n"); > > append_hf("P=NATed-Calee: Yes\r\n"); > > }; > > exec_msg("echo :: $SIP_SRCIP - $SIP_ORURI - $SIP_RURI > > > > >> /tmp/ser.log"); > > > > log(1, "1\n"); > > t_on_reply("1"); > > log(1, "2\n"); > > if (!t_relay()) { > > sl_reply_error(); > > break; > > }; > > log(1, "3\n"); > > } > > > > > > route[4] > > { > > [...] > > } > > > > onreply_route[1] > > { > > log(1, "taking onreply route\n"); > > if(isflagset(6) && status=~"(183)|2[0-9][0-9]" && > > !search("^Content-Length:\ +0")) > > { > > log(1,"onreply fixing nat\n"); > > fix_nated_contact(); > > force_rtp_proxy(); > > } else if (nat_uac_test("1")) > > { > > log (1, "onreply fixing nat alternate\n"); > > fix_nated_contact(); > > }; > > } > > > > route[4] > > { > > [...] > > } > > > > route[6] > > { > > [...] > > } > > > > > > ----------- > > REGISTER > > ----------- > > T official.ip.address:2062 -> ser.ip.address:5060 [A] > > REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/TCP > > 192.168.10.1:2062;br anch=z9hG4bK-wekq229vr3vn;rport..From: > > "Test" sip:test@domain.com;tag=3q2blh64wf..To: "Test" > > sip:test@domain.com..Call-ID: > > 3c26818830d4-ugwagbaz5vkz@snom360..CSeq: 22 REGISTER..Max- > > Forwards: 70..Contact: > > <sip:test@192.168.10.1:2062;transport=tcp;line= lhzwxw3z>;q=1.0; > > +sip.instance="urn:uuid:0c541696-09aa-4d0c-b8ca-fb9889cc61ad" > > ;audio;mobility="fixed";duplex="full";description="snom360";actor > >="pr inci pal"; > > events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTI > >FY ,SUBSCRIBE,PRACK,MESSAGE,INFO"..User-Agent: > > snom360/3.60k..Supported: gruu. > > .Allow-Events: dialog..X-Real-IP: 192.168.0.191..WWW-Contact: > > http://192.168.10.1:80..WWW-Contact: > > https://192.168.10.1:443..Expires: 3600..Content-Length: 0 > > > > T ser.ip.address:5060 -> official.ip.address:2062 [AP] > > SIP/2.0 401 Unauthorized..Via: SIP/2.0/TCP > > 192.168.10.1:2062;branch=z9hG4b > > K-wekq229vr3vn;rport=2062;received=official.ip.address..From: > > "Test" sip:test@domain.com;tag=3q2blh64wf..To: "Test" sip: > > > test@domain.com>;tag=5431d75005d8ed216f7c100a44746400.19d5..Call- > >ID > > > > : 3c26818830d4-ugwagbaz5vkz@snom360..CSeq: 22 > > > > REGISTER..P-NATed-Caller: Yes > > ..WWW-Authenticate: Digest realm="domain.com", > > nonce="42baac51779c17ebe > > ec20a5ee2f9492821bd723e"..Server: Sip EXpress router (0.9.3 > > (i386/linux)).. > > Content-Length: 0..Warning: ser.ip.address:5060 "Noisy feedback > > tells: p > > id=27776 req_src_ip=official.ip.address req_src_port=2062 > > in_uri=sip:domain.com out_uri=sip:domain.com via_cnt==1" > > > > > > T official.ip.address:2062 -> ser.ip.address:5060 [A] > > REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/TCP > > 192.168.10.1:2062;br anch=z9hG4bK-6s4mq8dda681;rport..From: > > "Test" sip:test@domain.com;tag=3q2blh64wf..To: "Test" > > sip:test@domain.com..Call-ID: > > 3c26818830d4-ugwagbaz5vkz@snom360..CSeq: 23 REGISTER..Max- > > Forwards: 70..Contact: > > <sip:test@domain.com:2062;transport=tcp;line= lhzwxw3z>;q=1.0; > > +sip.instance="urn:uuid:0c541696-09aa-4d0c-b8ca-fb9889cc61 > > > ad>";audio;mobility="fixed";duplex="full";description="snom360"; > > actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE, > >REFE R,OP TIONS,NOTIFY ,SUBSCRIBE,PRACK,MESSAGE,INFO"..User-Agent: > > snom360/3.60k..Supported: gruu. > > .Allow-Events: dialog..X-Real-IP: 192.168.10.1..WWW-Contact: > > http://192.1 > 68.10.1:80>..WWW-Contact: > > https://192.168.10.1:443..Authorization: > > Digestusername="test",realm="domain.com",nonce="42baac51779c17ebe > >ec20 a > > > > 5ee2f9492821bd723e",uri="sip:domain.com",response="d2415424805014 > >aac5 04 b643ea489795",algorithm=md5..Expires: > > 3600..Content-Length: 0.... > > > > T ser.ip.address:5060 -> official.ip.address:2062 [AP] > > SIP/2.0 200 OK..Via: SIP/2.0/TCP > > 192.168.10.1:2062;branch=z9hG4bK-6s4mq8dd > > a681;rport=2062;received=official.ip.address..From: "Test" > > sip:test@domain.com;tag=3q2blh64wf..To: "Test" > > sip:test@domain.com;tag=5431d75005d8ed216f7c100a44746400.d02e.. > >Call -ID: 3c268188 30d4-ugwagbaz5vkz@snom360..CSeq: 23 > > REGISTER..P-NATed-Caller: Yes..Contact: > > > > sip:test@192.168.10.1:2056;transport=tcp;line=lhzwxw3z;q=1;expi > >res= 2204;received="sip:official.ip.address:2056;transport=TCP", > > sip:test@192.168.10.1:2055;transport=tcp;line=lhzwxw3z;q=1;expi > >res= 1551 ; received="sip:official.ip.address:2055;transport=TCP", > > sip:test@192.168.10.1:2062;transport=tcp;line=lhzwxw3z;q=1;expi > >res= 3600 ; > > received="sip:official.ip.address:2062;transport=TCP"..Server: > > Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0.. > > Warning: 392 ser.ip.address:5060 "Noisy feedback tells: > > pid=27776 req_ src_ip=official.ip.address req_src_port=2062 > > in_uri=sip:domain.com out_uri=sip:domain.com via_cnt==1" > > > > ----------- > > INVITE > > ----------- > > T official.ip.address:33255 -> ser.ip.address:5060 [AP] > > INVITE sip:test@domain.com SIP/2.0..Via: SIP/2.0/TCP > > 192.168.10.2; branch=z9hG4bK3AA2CBD1;alias..CSeq: 2681 > > INVITE..To: > > <sip:test@domain .com>..Content-Type: application/sdp..From: > > "test2" sip:test2@domain.com > > > >;tag=67807379..Call-ID: 1766830276@192.168.10.2..Subject: > > > sip:test > > > > @domain.com..Content-Length: 230..User-Agent: > > kphone/4.1.1..Contac t: "test" > > sip:test@192.168.10.2;transport=tcp....v=0..o=username 0 0 IN > > IP4 192.168.10.2..s=The Funky Flow..c=IN IP4 192.168.10.2..t=0 > > 0..m=audio > > 32874 RTP/AVP 0 97 8 3..a=rtpmap:0 PCMU/8000..a=rtpmap:3 > > GSM/8000..a=rtpma > > p:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=30.. > > ## > > T ser.ip.address:5060 -> official.ip.address:33255 [AP] > > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/TCP > > 192.168.10.2;b > > > > ranch=z9hG4bK3AA2CBD1;alias;rport=33255;received=official.ip.addr > >ess. .CSe q: 2681 INVITE..To: > > sip:test@domain.com;tag=5431d75005d8ed216f7c100a4474 > > 6400.5110..From: "test" > > sip:test2@domain.com;tag=67807379..Call-I D: > > 1766830276@192.168.10.2..P-NATed-Caller: Yes..Proxy-Authenticate: > > Diges t realm="domain.com", > > nonce="42baaf784cd57486fa11fe10929ade10b8fc4 3ec"..Server: Sip > > EXpress router (0.9.3 (i386/linux))..Content-Length: 0..W > > arning: 392 ser.ip.address:5060 "Noisy feedback tells: > > pid=27776 req_src_ip= > > official.ip.address req_src_port=33255 > > in_uri=sip:test@domain.com out_ur > > i=sip:test@domain.com via_cnt==1".... > > ## > > T official.ip.address:33255 -> ser.ip.address:5060 [AP] > > ACK sip:test@domain.com SIP/2.0..Via: SIP/2.0/TCP > > 192.168.10.2;bra nch=z9hG4bK3AA2CBD1;alias..CSeq: 2681 ACK..To: > > sip:test@domain.com > > > >;tag=5431d75005d8ed216f7c100a44746400.5110..From: "test2" > > > sip:test2@ > > > domain.com>;tag=67807379..Call-ID: > > 1766830276@192.168.10.2..Content-Leng > > th: 0..User-Agent: kphone/4.1.1..Contact: "test" > > sip:test@192.168.10.2;tr > ansport=tcp>.... > > ## > > T official.ip.address:33255 -> ser.ip.address:5060 [AP] > > INVITE sip:test@domain.com SIP/2.0..Via: SIP/2.0/TCP > > 192.168.10.2; branch=z9hG4bK36EBCF00;alias..CSeq: 2682 > > INVITE..To: > > sip:test@domain.com > > > >..Proxy-Authorization: Digest username="test2@domain.com", r > > > > ealm="domain.com", > > nonce="42baaf784cd57486fa11fe10929ade10b8fc43ec ", > > uri="sip:test@domain.com", cnonce="abcdefghi", nc=00000001, resp > > onse="2c088d59cb24c70c61d890245fe0f5ca", opaque="", > > algorithm="MD5"..Conten > > t-Type: application/sdp..From: "test2" > > sip:test2@domain.com;tag=67 807379..Call-ID: > > 1766830276@192.168.10.2..Subject: > > sip:test2@domain.com > > ..Content-Length: 230..User-Agent: kphone/4.1.1..Contact: "test" > > <sip: test@192.168.10.2;transport=tcp>....v=0..o=username 0 0 IN > > IP4 192.168.0.1 > > 03..s=The Funky Flow..c=IN IP4 192.168.10.2..t=0 0..m=audio > > 32874 RTP/AVP > > 0 97 8 3..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 > > PCMA/8000.. > > a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=30.. > > ## > > T ser.ip.address:5060 -> official.ip.address:33255 [AP] > > SIP/2.0 100 trying -- your call is important to us..Via: > > SIP/2.0/TCP 192.16 > > > > 8.10.2;branch=z9hG4bK36EBCF00;alias;rport=33255;received=official > >.ip. addr ess..C Seq: 2682 INVITE..To: > > sip:test@domain.com..From: "test" sip:test > @domain.com>;tag=67807379..Call-ID: > > 1766830276@192.168.10.2..P-NA Ted-Caller: Yes..Server: Sip > > EXpress router (0.9.3 > > (i386/linux))..Content-L > > ength: 0..Warning: 392 ser.ip.address:5060 "Noisy feedback > > tells: pid=27776 > > req_src_ip=official.ip.address req_src_port=33255 > > in_uri=sip:test@domain.com > > out_uri=sip:test@192.168.10.1:2056;transport=tcp;line=lhzwxw3z > > vi a_cnt==1".... > > ## > > T ser.ip.address:5060 -> official.ip.address:33255 [AP] > > SIP/2.0 477 Unfortunately error on sending to next hop occurred > > (477/TM)..V > > ia: SIP/2.0/TCP > > 192.168.10.2;branch=z9hG4bK36EBCF00;alias;rport=33255;rece > > ived=official.ip.address..CSeq: 2682 INVITE..To: > > sip:test@domain.com;t > > ag=a0de3507a8823f96a254cc0a187acbf0-2573..From: "test2" > > sip:test2@domain.com > > > >;tag=67807379..Call-ID: > > > 1766830276@192.168.10.2..P-NATed-Caller: > > > > Yes..Server: Sip EXpress router (0.9.3 > > (i386/linux))..Content-Length: 0..W > > arning: 392 ser.ip.address:5060 "Noisy feedback tells: > > pid=27776 req_src_ip= > > official.ip.address req_src_port=33255 > > in_uri=sip:test@domain.com out_ur > > i=sip:test@192.168.10.1:2056;transport=tcp;line=lhzwxw3z > > via_cnt==1".. .. > > ## > > T official.ip.address:33255 -> ser.ip.address:5060 [AP] > > ACK sip:test@domain.com SIP/2.0..Via: SIP/2.0/TCP > > 192.168.10.2;bra nch=z9hG4bK36EBCF00;alias..CSeq: 2682 ACK..To: > > sip:test@domain.com > > > >;tag=a0de3507a8823f96a254cc0a187acbf0-2573..From: "test2" > > > > sip:test2@domain > .com>;tag=67807379..Call-ID: > > 1766830276@192.168.10.2..Content-Leng th: 0..User-Agent: > > kphone/4.1.1..Contact: "test" > > sip:test@192.168.10.2;tr > ansport=tcp>.... > > # > > _______________________________________________ > Serusers mailing list > serusers@lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers
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