sr-users January 2006

sr-users@lists.kamailio.org

134 participants
259 discussions

RE: [Users] direct manipulation of DB
by Douglas Garstang 09 Jan '06

09 Jan '06

Isn't there a script called 'sc' somewhere that does the same thing as openserctl? I looked at the sc script and was able to determine how it generates those fields. I think they're gen_ha and something else... installed by openser. -----Original Message----- From: unplug [mailto:maillisting@gmail.com] Sent: Sun 1/8/2006 7:50 PM To: Norman Brandinger Cc: users(a)openser.org Subject: Re: [Users] direct manipulation of DB There are 2 parts. One is server and one is client. How to setup the server as openser has already a fifo interfece? On 12/23/05, Norman Brandinger <norm(a)goes.com> wrote: > I posted a simple FIFO Client that can be used to test sending commands > and receiving responses from OpenSER. > > The link is here: http://openser.org/dokuwiki/doku.php?id=utilities > > Regards, > Norm > > p.s. Comments or suggestions for improvements are welcome. > > p.s.s. I believe that the OpenSER roadmap has an API project that will > depreciate the FIFO interface. Perhaps one of the developers can > elaborate more on this. > > Klaus Darilion wrote: > > > unplug wrote: > > > >> I am using openser with mysql as db. There is a tool call openserctl > >> to manage the data in the openser. I wonder whether I can direct > >> manipulate the db directly without using openserctl. I have tried to > >> insert a record in the aliases table directly using mysql client but > >> it doesn't work. Can I conclude direct manipulate the DB is > >> impossible? > > > > > > It depends on the table! There are some tables which will be cached > > inside openser, e.g: > > location > > aliases > > domain > > gw (lcr module) > > > > If you make changes to location/aliases you have to: > > - restart openser to reload the tables or > > - use openserctl (inserts via FIFO to openser which wirtes back to > > the db or > > - in case of aliases use the aliases_DB module > > > > for the domain and gw table, you can use the respectife FIfo commands > > to reload the tables. > > > > regards > > klaus > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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RE: [Users] Re: OpenSER and transport selection (SRV and NAPTR)
by Douglas Garstang 09 Jan '06

09 Jan '06

Oh yes! That would be AMAZING if that functionality could be added! I think I'm gonna poo my pants! -----Original Message----- From: Greg Fausak [mailto:lgfausak@gmail.com] Sent: Fri 1/6/2006 11:06 AM To: users(a)openser.org Cc: Subject: [Users] Re: OpenSER and transport selection (SRV and NAPTR) Howdy guys, I completely missed this thread. How is this going? I just read the roadmap which indicates NAPTR lookup is on the list. I've been maintaining NAPTR and SRV records looking forward to the day when t_relay() forwards using NAPTR (and SRV). Does that mean that the t_relay() will maintain the state, slogging through the NAPTR/SRV records appropriately, without me having to do additional logic??? Specifically, I'm keen on the SRV serial forking. -g Hi Bogdan, > I meant t_reply() will keep its behaviour as looking into URI for the > destination - but it will incorporate the NAPTR lookup. Great - this answers my question. > to response also to on earlier question, regarding the timing > - I say 3 months are more than enough ;). Excellent. Many thanks for your help (and patience ;) --Joachim > Joachim Fabini wrote: > > >Hi Bogdan, > > > > > > > >>indeed, there was a misunderstanding :) .t_relay() with no > >>param will be kept with the current functionality. > >>Or you suggest to be able to specify only a different proto > >>or port from the RURI? > >> > >> > > > >I did not yet find the primitive which is supposed to > >statefully relay and do the following: > >1. NAPTR query on the transport to use PLUS > >2. _exactly_ what t_relay() does now for the > > transport returned by the NAPTR query. > > > >You say that t_relay() is kept with the current functionality. > >Does this mean no NAPTR, or will t_relay() be extended to use > >NAPTR before SRV/A query? If the latter then everything is OK. > > > >Otherwise we have the alternatives: > > > >t_relay() -> Stateful relaying according to destination > > address using the incoming transport (no NAPTR). > >t_relay_to() -> Stateful relaying to a specific host (you said > > that <host> is mandatory here) using NAPTR to > > determine the transport. > > > > > >To summarize: > >My point is that none of these two primitives covers the case > >when the message is to be relayed to the next hop based only > >on the destination address _and_ using the transport selected > >by the destination proxy (determined via NAPTR query). > > > >Except if either a) t_relay() does NAPTR or b) the host > >parameter in t_relay_to() is optional. > > > >--Joachim > > > > > > > > > -- Greg Fausak greg(a)thursday.com _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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Re: [Users] deactivate an account
by Helge Waastad 09 Jan '06

09 Jan '06

Hi, In the openserctl script there are acl options static configured (free-pstn etc). Just add inactive on the same line, and it should work br hw There is an error when I issue acl grant openserctl acl grant myname(a)mydomain.com inactive Invalid privilege: inactive ignored -- Mvh/Br Helge Waastad Senior Engineer Systemavdelingen Smartnet

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[Users] direct manipulation of DB
by unplug 09 Jan '06

09 Jan '06

I am using openser with mysql as db. There is a tool call openserctl to manage the data in the openser. I wonder whether I can direct manipulate the db directly without using openserctl. I have tried to insert a record in the aliases table directly using mysql client but it doesn't work. Can I conclude direct manipulate the DB is impossible?

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[Users] Re: Users Digest, Vol 8, Issue 14
by Kenny Yeh 09 Jan '06

09 Jan '06

Hi,Philipp Thanks for your help !!!!!! You show me what I need .thanks!!! ======= 2006-01-07 03:15:05 ��д��======= >Send Users mailing list submissions to > users(a)openser.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > >You can reach the person managing the list at > users-owner(a)openser.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. uac_replace_from function (unplug) > 2. Re: NAT+uac_replace_from (Daniel-Constantin Mierla) > 3. Re: uac_replace_from function (Daniel-Constantin Mierla) > 4. Re: OpenSER and Solaris (Daniel-Constantin Mierla) > 5. Re: OpenSER and transport selection (SRV and NAPTR) (Greg Fausak) > 6. Re: help: what's tls_ca tls_privat-key tls_calist > (Alexander Philipp Lintenhofer) > 7. contact field port problem when behind nats (Jack Wei) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Fri, 6 Jan 2006 19:08:34 +0800 >From: unplug <maillisting(a)gmail.com> >Subject: [Users] uac_replace_from function >To: users(a)openser.org >Message-ID: > <6fbb529e0601060308p5a0eab32wafce0fcc5f04b286(a)mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 > >I am using uac_replace_from function as below. > ># replace only display and do not touch uri >uac_replace_from("35418474",""); > >After running uac_replace_from function, I get the following message. > >U 203.193.4.242:5060 -> 203.193.4.226:5060 >SIP/2.0 183 Session Progress. >Via: SIP/2.0/UDP >203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP >10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. >From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. >To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. >Date: Fri, 06 Jan 2006 10:41:17 GMT. >Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. >Server: Cisco-SIPGateway/IOS-12.x. > >In the normal message, sip message, there is a space between display >name and uri. >ie. From: "35418474" <sip:882754853589@ow01.owtel.com> >However, after I apply uac_replace_from function, there is a space >there. How can I add a space such that it looks like a normal sip >message? Does the space will affect the caller display? > > > >------------------------------ > >Message: 2 >Date: Fri, 06 Jan 2006 13:28:52 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] NAT+uac_replace_from >To: unplug <maillisting(a)gmail.com> >Cc: users(a)openser.org >Message-ID: <43BE5474.4010400(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Send me the network traces for this case to see what is wrong. The uac >module has a mode when it should do automatic replacement in subsequent >requests. > >Daniel > > >On 01/06/06 04:37, unplug wrote: >> Thanks for your reply. According to your reply, I add the following >> code for the PRACK. >> if (method=="PRACK") { >> if (avp_db_load("$from/uri","s:alias")) { >> xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); >> uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); >> } else { >> xlog("L_INFO","sip411: no alias\n"); >> }; >> } >> >> The result is the same as before. It seems there is a conflict in the >> from header and it causes the 481 error. To be more precise, below is >> the protocol flow. >> >> UA1 server1 server2 >> fromtag:taga -----from username:9000 INVITE-----> >> fromtag:taga <----from username:9000 100 trying--- >> after uac_replace_from("1234","sip:1234@$si") >> fromtag:taga ------from username:1234 INVITE ----> >> (callee rings but caller drops) >> fromtag:taga <-----from username:1234 100 trying---- >> (callee still rings until answers the call) >> fromtag:taga <----from username:1234 session progress 183--- >> fromtag:taga <-----from username:9000 session prgress 183---- >> fromtag:taga -----from username:9000 PRACK-----------> >> fromtag:taga ----------from username:9000 PRACK---> >> fromtag:taga <----from username:9000 481 call leg ----- >> fromtag:taga <----from username:9000 481 call leg ----- >> >> >> On 1/5/06, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> >>> Hello, >>> >>> "SIP/2.0 481 Call Leg/Transaction Does Not Exist" is for PRACK, because you do not change the From header for it. You have to do the same translation for all requests within the dialog. See the documentation of uac module: . >>> >>> http://openser.org/docs/modules/1.1.x/uac.html >>> >>> Cheers, >>> Daniel >>> >>> >>> On 01/04/06 12:21, unplug wrote: >>> >>>> Actually, I am replacing the username of the uri with the alias that >>>> stored in the database. In my configuration file, it is using >>>> mediaproxy for NAT function (features-callfwd.5.0.cfg from getting >>>> started). I also add the following codes in the very first of the >>>> route routine for alias replacing purpose. >>>> >>>> route { >>>> ... >>>> if (!has_totag() && method=="INVITE") { >>>> if (avp_db_load("$from/uri","s:alias")) { >>>> xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); >>>> uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); >>>> } else { >>>> xlog("L_INFO","sip411: no alias\n"); >>>> }; >>>> }; >>>> ... >>>> } >>>> >>>> When I make a call from a phone to the PSTN phone, the caller drops >>>> the ring when the callee rings. Callee hangs up and callee rings >>>> again few seconds after. You can find an error message "SIP/2.0 481 >>>> Call Leg/Transaction Does Not Exist" in line 180. I wonder if there >>>> is any thing wrong with the above code or my concept is wrong. Please >>>> help. Below is the sip message. >>>> [...] >>>> >>>> >> >> > > > >------------------------------ > >Message: 3 >Date: Fri, 06 Jan 2006 13:33:32 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] uac_replace_from function >To: unplug <maillisting(a)gmail.com> >Cc: users(a)openser.org >Message-ID: <43BE558C.70404(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >It might be a bug in the module, please send the original SIP reply as >well, to be able to trace the problem quicker. > >Daniel > > >On 01/06/06 13:08, unplug wrote: >> I am using uac_replace_from function as below. >> >> # replace only display and do not touch uri >> uac_replace_from("35418474",""); >> >> After running uac_replace_from function, I get the following message. >> >> U 203.193.4.242:5060 -> 203.193.4.226:5060 >> SIP/2.0 183 Session Progress. >> Via: SIP/2.0/UDP >> 203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP >> 10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. >> From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. >> To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. >> Date: Fri, 06 Jan 2006 10:41:17 GMT. >> Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. >> Server: Cisco-SIPGateway/IOS-12.x. >> >> In the normal message, sip message, there is a space between display >> name and uri. >> ie. From: "35418474" <sip:882754853589@ow01.owtel.com> >> However, after I apply uac_replace_from function, there is a space >> there. How can I add a space such that it looks like a normal sip >> message? Does the space will affect the caller display? >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > >------------------------------ > >Message: 4 >Date: Fri, 06 Jan 2006 19:41:19 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] OpenSER and Solaris >To: martin(a)campus-merseburg.de >Cc: users(a)openser.org >Message-ID: <43BEABBF.3090302(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >thanks for reporting, I have renamed the internal structure meminfo to >mem_info to avoid this conflict. Please take the latest 1.0.0 from cvs >and let me know if it works. > >Here are the commands to get the stable version from cvs: > ># export CVSROOT=:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser ># cvs login - for password simply press the Enter key. ># cvs -z3 checkout -r rel_1_0_0 sip-server > > >Cheers, >Daniel > > >On 01/06/06 12:45, martin(a)campus-merseburg.de wrote: >> Hi, >> >> does anyone use OpenSER under Solaris? >> >> I get this error when compiling: >> >> gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc >> -DNAME='"openser"' -DVERSION='"1.0.0"' -DARCH='"sparc64"' >> -DOS='"solaris"' -DCOMPILER='"gcc 3.4"' -D__CPU_sparc64 >> -D__OS_solaris -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC >> -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP >> -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK >> -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME >> -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -c action.c -o >> action.o >> In file included from mem/f_malloc.h:40, >> from mem/mem.h:54, >> from action.c:50: >> mem/meminfo.h:33: error: redefinition of `struct meminfo' >> gmake: *** [action.o] Error 1 >> >> any idea for fixing this? >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > >------------------------------ > >Message: 5 >Date: Fri, 6 Jan 2006 12:06:42 -0600 >From: Greg Fausak <lgfausak(a)gmail.com> >Subject: [Users] Re: OpenSER and transport selection (SRV and NAPTR) >To: users(a)openser.org >Message-ID: > <8c7b802f0601061006p373d8949we5e48c9953eb8dc9(a)mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 > >Howdy guys, > >I completely missed this thread. >How is this going? > >I just read the roadmap which indicates NAPTR lookup >is on the list. I've been maintaining NAPTR and SRV records >looking forward to the day when t_relay() forwards using NAPTR (and >SRV). Does that mean that the t_relay() will maintain the state, slogging >through the NAPTR/SRV records appropriately, without me having >to do additional logic??? Specifically, I'm keen on the SRV serial forking. > >-g > > > > >Hi Bogdan, > >> I meant t_reply() will keep its behaviour as looking into URI for the >> destination - but it will incorporate the NAPTR lookup. > >Great - this answers my question. > > >> to response also to on earlier question, regarding the timing >> - I say 3 months are more than enough ;). > >Excellent. > >Many thanks for your help (and patience ;) >--Joachim > > > >> Joachim Fabini wrote: >> >> >Hi Bogdan, >> > >> > >> > >> >>indeed, there was a misunderstanding :) .t_relay() with no >> >>param will be kept with the current functionality. >> >>Or you suggest to be able to specify only a different proto >> >>or port from the RURI? >> >> >> >> >> > >> >I did not yet find the primitive which is supposed to >> >statefully relay and do the following: >> >1. NAPTR query on the transport to use PLUS >> >2. _exactly_ what t_relay() does now for the >> > transport returned by the NAPTR query. >> > >> >You say that t_relay() is kept with the current functionality. >> >Does this mean no NAPTR, or will t_relay() be extended to use >> >NAPTR before SRV/A query? If the latter then everything is OK. >> > >> >Otherwise we have the alternatives: >> > >> >t_relay() -> Stateful relaying according to destination >> > address using the incoming transport (no NAPTR). >> >t_relay_to() -> Stateful relaying to a specific host (you said >> > that <host> is mandatory here) using NAPTR to >> > determine the transport. >> > >> > >> >To summarize: >> >My point is that none of these two primitives covers the case >> >when the message is to be relayed to the next hop based only >> >on the destination address _and_ using the transport selected >> >by the destination proxy (determined via NAPTR query). >> > >> >Except if either a) t_relay() does NAPTR or b) the host >> >parameter in t_relay_to() is optional. >> > >> >--Joachim >> > >> > >> > >> > >> >-- >Greg Fausak >greg(a)thursday.com > > > >------------------------------ > >Message: 6 >Date: Fri, 06 Jan 2006 19:47:10 +0100 >From: Alexander Philipp Lintenhofer <lintenhofer(a)aon.at> >Subject: Re: [Users] help: what's tls_ca tls_privat-key tls_calist >To: users(a)openser.org >Message-ID: <20060106194710.x2agnokfltq8ok4s(a)liho.dyndns.org> >Content-Type: text/plain; charset=UTF-8; format="flowed" > >Hi Kenny, > >It would be a big deal first to acquire knowledge about the theoretical >background before installing this service. > >So please read following (and related) sites: >http://www.tech-invite.com/Ti-sip-security.html >http://www.tech-invite.com/Ti-SSL.html > >The definitive documentation about OpenSER/TLS >(http://openser.org/docs/tls.html) is very good and therefore easy to >understand in succession! > >regards, >Philipp > >> Hello all, >> >> I have tested tls with openser by SIPp,but I donn't understant >> the flows of TLS ctreated,who can show me the flow about TLS >> handshake. >> >> When I test tls by SIPp,I put the ca,private-key to it's >> DIR,that's fils created by openssl ,and add these files path to cfg >> of openser.the same files used.I want to know what's mean is tls_ca >> tls_privatekey tls_calist,and how there files work. Client need >> what's ca and private-key for creating tls connection? Is it the same >> with openssl? to create ssl connection? >> >> >> >> >> >> ======= 2006-01-05 19:01:27 您在来信中写道：======= >> >>> Send Users mailing list submissions to >>> users(a)openser.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> or, via email, send a message with subject or body 'help' to >>> users-request(a)openser.org >>> >>> You can reach the person managing the list at >>> users-owner(a)openser.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of Users digest..." >>> >>> >>> Today's Topics: >>> >>> 1. Re: deactivate an account (Helge Waastad) >>> >>> >>> ---------------------------------------------------------------------- >>> >>> Message: 1 >>> Date: Thu, 05 Jan 2006 11:23:30 +0100 >>> From: Helge Waastad <helge(a)smartnet.no> >>> Subject: Re: [Users] deactivate an account >>> To: maillisting(a)gmail.com >>> Cc: users(a)openser.org >>> Message-ID: <1136456610.11157.10.camel(a)localhost.localdomain> >>> Content-Type: text/plain >>> >>> Hi, >>> In the openserctl script there are acl options static configured >>> (free-pstn etc). >>> Just add inactive on the same line, and it should work >>> >>> br hw >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> There is an error when I issue acl grant >>> openserctl acl grant myname(a)mydomain.com inactive >>> Invalid privilege: inactive ignored >>> >>> >>> >>> -- >>> Mvh/Br >>> Helge Waastad >>> Senior Engineer >>> Systemavdelingen >>> Smartnet >>> >>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> End of Users Digest, Vol 8, Issue 11 >>> ************************************ >> >> = = = = = = = = = = = = = = = = = = = = >> >> >> 　　　　　　　　�? >> 礼！ >> >> >> 　　　　　　　　Kenny Yeh >> 　　　　　　　　kenny(a)artdio.com.tw >> 　　　　　　　　　　2006-01-06 >> >> > > > > > > >------------------------------ > >Message: 7 >Date: Fri, 6 Jan 2006 11:14:14 -0800 (PST) >From: Jack Wei <cowlemon(a)yahoo.com> >Subject: [Users] contact field port problem when behind nats >To: users(a)openser.org >Message-ID: <20060106191415.56473.qmail(a)web50008.mail.yahoo.com> >Content-Type: text/plain; charset=iso-8859-1 > >Hi, > >I'm having a problem when an UAC is behind NAT and has a public port other than >5060. The public port isn't present in the contact field so when an ACK for an >INVITE is sent to the UAC, it's directed to the default port of 5060. I'm >currently using v1.0.0 w/ the NATHELPER module. The UAC is SJphone. Any help >would be greatly appreciated. > > > > > >NAT part of my openser.cfg : > >if (nat_uac_test("19")) >{ > if ( method == "REGISTER" ) > { > fix_nated_register(); > force_rport(); > } > else > { > force_rport(); > fix_nated_contact(); > > if (method == "INVITE") > { > fix_nated_sdp("1"); > }; > }; > > append_hf( "P-hint: NAT traversal\r\n" ); >} > > > > >the sip proxy is voip.somewhere.com and 1.1.1.1 >the pstn gateway is 2.2.2.2 > >the caller is from pstn and the callee is the uac. > > >Here's a log of SIP msgs: > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >INVITE sip:2139435286@voip.somewhere.com:5060;user=phone SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >Remote-Party-Id: ><sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >m: <sip:2139961998@2.2.2.2:5060;user=phone>. >k: replaces. >c: application/sdp. >Accept: application/sdp. >Accept-Encoding: . >Accept-Language: en. >User-Agent: Lucent-Universal-Gateway. >l: 244. >. >v=0. >o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. >s=Session SDP. >c=IN IP4 2.2.2.2. >t=0 0. >m=audio 41788 RTP/AVP 0 101. >a=silenceSupp:off. >a=ecan:b on g168. >a=ptime:20. >a=rtpmap:101 telephone-event/8000. >a=rtpmap:0 PCMU/8000. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 100 trying -- your call is important to us. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >v: SIP/2.0/UDP 2.2.2.2:5060. >Server: OpenSer (1.0.0-tls (i386/linux)). >Content-Length: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:32883 >INVITE sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >Remote-Party-Id: ><sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK8325.35c96c63.0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >m: <sip:2139961998@2.2.2.2:5060;user=phone>. >k: replaces. >c: application/sdp. >Accept: application/sdp. >Accept-Encoding: . >Accept-Language: en. >User-Agent: Lucent-Universal-Gateway. >l: 244. >P-hint: VOIP OUTBOUND. >. >v=0. >o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. >s=Session SDP. >c=IN IP4 2.2.2.2. >t=0 0. >m=audio 41788 RTP/AVP 0 101. >a=silenceSupp:off. >a=ecan:b on g168. >a=ptime:20. >a=rtpmap:101 telephone-event/8000. >a=rtpmap:0 PCMU/8000. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 100 Trying. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: ""<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 0. >Server: SJphone/1.61.306c (SJ Labs). >. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP 2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on >SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >Route: <sip:2139435286@64.77.236.227>. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:5060 >ACK sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >Via: SIP/2.0/UDP voip.somewhere.com;branch=0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >User-Agent: Lucent-Universal-Gateway. >l: 0. >P-hint: rr-enforced. >. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP 2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on >SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >Route: <sip:2139435286@64.77.236.227>. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:5060 >ACK sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >Via: SIP/2.0/UDP voip.somewhere.com;branch=0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >User-Agent: Lucent-Universal-Gateway. >l: 0. >P-hint: rr-enforced. >. > > > > > > >................<REPEAT OKs and ACKs>........................ > > > > > > > > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. >v: SIP/2.0/UDP >192.168.0.100;rport;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >f: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Max-Forwards: 70. >User-Agent: SJphone/1.61.306c (SJ Labs). >l: 0. >Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. >Record-Route: <sip:2139961998@voip.somewhere.com;ftag=71451719939;lr=on>. >Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >f: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Max-Forwards: 69. >User-Agent: SJphone/1.61.306c (SJ Labs). >l: 0. >P-hint: NAT traversal. >P-hint: rr-enforced. >. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >f: "Jack Wei" ><sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Server: Lucent-Universal-Gateway. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:32883 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >f: "Jack Wei" ><sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Server: Lucent-Universal-Gateway. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > > >Jack Wei > > > >__________________________________________ >Yahoo! DSL ?Something to write home about. >Just $16.99/mo. or less. >dsl.yahoo.com > > > > >------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >End of Users Digest, Vol 8, Issue 14 >************************************ = = = = = = = = = = = = = = = = = = = = �� Kenny Yeh ��kenny(a)artdio.com.tw ��2006-01-09

1 0

[Users] contact field port problem when behind nats
by Jack Wei 07 Jan '06

07 Jan '06

Hi, I'm having a problem when an UAC is behind NAT and has a public port other than 5060. The public port isn't present in the contact field so when an ACK for an INVITE is sent to the UAC, it's directed to the default port of 5060. I'm currently using v1.0.0 w/ the NATHELPER module. The UAC is SJphone. Any help would be greatly appreciated. NAT part of my openser.cfg : if (nat_uac_test("19")) { if ( method == "REGISTER" ) { fix_nated_register(); force_rport(); } else { force_rport(); fix_nated_contact(); if (method == "INVITE") { fix_nated_sdp("1"); }; }; append_hf( "P-hint: NAT traversal\r\n" ); } the sip proxy is voip.somewhere.com and 1.1.1.1 the pstn gateway is 2.2.2.2 the caller is from pstn and the callee is the uac. Here's a log of SIP msgs: # U 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:2139435286@voip.somewhere.com:5060;user=phone SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. Remote-Party-Id: <sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. m: <sip:2139961998@2.2.2.2:5060;user=phone>. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: Lucent-Universal-Gateway. l: 244. . v=0. o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. s=Session SDP. c=IN IP4 2.2.2.2. t=0 0. m=audio 41788 RTP/AVP 0 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 trying -- your call is important to us. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. v: SIP/2.0/UDP 2.2.2.2:5060. Server: OpenSer (1.0.0-tls (i386/linux)). Content-Length: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:32883 INVITE sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. Remote-Party-Id: <sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK8325.35c96c63.0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. m: <sip:2139961998@2.2.2.2:5060;user=phone>. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: Lucent-Universal-Gateway. l: 244. P-hint: VOIP OUTBOUND. . v=0. o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. s=Session SDP. c=IN IP4 2.2.2.2. t=0 0. m=audio 41788 RTP/AVP 0 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 100 Trying. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: ""<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 0. Server: SJphone/1.61.306c (SJ Labs). . # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. Route: <sip:2139435286@64.77.236.227>. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:5060 ACK sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. Via: SIP/2.0/UDP voip.somewhere.com;branch=0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. User-Agent: Lucent-Universal-Gateway. l: 0. P-hint: rr-enforced. . # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. Route: <sip:2139435286@64.77.236.227>. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:5060 ACK sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. Via: SIP/2.0/UDP voip.somewhere.com;branch=0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. User-Agent: Lucent-Universal-Gateway. l: 0. P-hint: rr-enforced. . ................<REPEAT OKs and ACKs>........................ # U 64.77.236.227:32883 -> 1.1.1.1:5060 BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. v: SIP/2.0/UDP 192.168.0.100;rport;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. f: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Max-Forwards: 70. User-Agent: SJphone/1.61.306c (SJ Labs). l: 0. Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . # U 1.1.1.1:5060 -> 2.2.2.2:5060 BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. Record-Route: <sip:2139961998@voip.somewhere.com;ftag=71451719939;lr=on>. Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. f: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Max-Forwards: 69. User-Agent: SJphone/1.61.306c (SJ Labs). l: 0. P-hint: NAT traversal. P-hint: rr-enforced. . # U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. f: "Jack Wei" <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Server: Lucent-Universal-Gateway. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:32883 SIP/2.0 200 OK. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. f: "Jack Wei" <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Server: Lucent-Universal-Gateway. User-Agent: Lucent-Universal-Gateway. l: 0. . Jack Wei __________________________________________ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com

2 1

[Users] help: what's tls_ca tls_privat-key tls_calist
by Kenny Yeh 06 Jan '06

06 Jan '06

Hello all, I have tested tls with openser by SIPp,but I donn't understant the flows of TLS ctreated,who can show me the flow about TLS handshake. When I test tls by SIPp,I put the ca,private-key to it's DIR,that's fils created by openssl ,and add these files path to cfg of openser.the same files used.I want to know what's mean is tls_ca tls_privatekey tls_calist,and how there files work. Client need what's ca and private-key for creating tls connection? Is it the same with openssl? to create ssl connection? ======= 2006-01-05 19:01:27 您在来信中写道：======= >Send Users mailing list submissions to > users(a)openser.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > >You can reach the person managing the list at > users-owner(a)openser.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. Re: deactivate an account (Helge Waastad) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 05 Jan 2006 11:23:30 +0100 >From: Helge Waastad <helge(a)smartnet.no> >Subject: Re: [Users] deactivate an account >To: maillisting(a)gmail.com >Cc: users(a)openser.org >Message-ID: <1136456610.11157.10.camel(a)localhost.localdomain> >Content-Type: text/plain > >Hi, >In the openserctl script there are acl options static configured >(free-pstn etc). >Just add inactive on the same line, and it should work > >br hw > > > > > > > > > > > >There is an error when I issue acl grant >openserctl acl grant myname(a)mydomain.com inactive >Invalid privilege: inactive ignored > > > >-- >Mvh/Br >Helge Waastad >Senior Engineer >Systemavdelingen >Smartnet > > > >------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >End of Users Digest, Vol 8, Issue 11 >************************************ = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-06

2 1

[Users] OpenSER and Solaris
by martin＠campus-merseburg.de 06 Jan '06

06 Jan '06

Hi, does anyone use OpenSER under Solaris? I get this error when compiling: gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.0.0"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4"' -D__CPU_sparc64 -D__OS_solaris -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -c action.c -o action.o In file included from mem/f_malloc.h:40, from mem/mem.h:54, from action.c:50: mem/meminfo.h:33: error: redefinition of `struct meminfo' gmake: *** [action.o] Error 1 any idea for fixing this?

2 1

[Users] 500 command 'domain_dump' not available
by Bobber Cheng 06 Jan '06

06 Jan '06

I use openser-1.0.0. #/usr/local/sbin/openserctl domain show 500 command 'domain_dump' not available i load domain.so in my openser.cf like loadmodule "/usr/local/lib/openser/modules/domain.so"

3 4

Re: Re: [Users] what's about srtp on openser?
by Kenny Yeh 05 Jan '06

05 Jan '06

Klaus Darilion,hi You mean is if I add SRTP to re-place RTP of client,Client can create srtp to communication?? So I just re-build my client code is ok?it's changed by udp or SIP lays or both? ======= 2006-01-05 17:53:26 您在来信中写道：======= >srtp is independent from SIP. There is no need to support srtp on the >SIP proxy. If you use mediaproxy or rtpproxy (rtp relay), it should work >too as they do not care about the payload, just forward the packets. > >regards >klaus > >Kenny Yeh wrote: >> Bogdan-Andrei Iancu,hello >> >> Ask for you again. :) >> >> I am succesful on openser for TLS connected by SIPp,but not minisip :( >> thanks for your more help! >> Now I step to srtp! Can I use SRTP to build peer to peer communication from openser? if outbound via openser.or do you learn more about srtp of SIP. >> >> >> Best Regard! >> >> Kenny from China Shanghai >> 2006-1-5 >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-05

2 1

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sr-users January 2006