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[Serusers] Re: [Serdev] Better NAT support in registrar
by Klaus Darilion 01 Mar '05

01 Mar '05

Wow! A perfect description. Thanks Jan. Wouldn't it be a good idea to put such detailed descriptions from the mailinglist into the README files or any other repository, e.g. FAQomatic? Sometimes there are really good descriptions on the mailinglist (e.g. http://lists.iptel.org/pipermail/serdev/2005-February/003877.html) which should also be put into the README files. regards, klaus Jan Janak wrote: > Hello, > > The development version of SER now includes enhanced NAT support in > registrar and user location database. A small example: > > Let's suppose that you have a single SER instance listening on two > ports -- 5060 and 5090. Using a different port seems to be often > necessary because of broken SIP ALGs (Application Level Gateways) that > are often built into DSL routers or other devices. Such ALGs would > mangle SIP requests and responses coming to and from port 5060 and the > only option how to avoid such mangling is using a different port number. > > Let's suppose that we have two UAs beind NAT, one of them is configured > to reach SER on port 5060, the other one is configured to use port 5090 > (due to the reasons described above): > > +---------+ > UA1 ---- NAT1 ----| 5060 | > | SER| > UA2 ---- NAT2 ----| 5090 | > +---------+ > > The previous version of registrar would store the public IP of NAT with > each registered contact, thus it would know how to reach both user > agents. > > But, it would not store the port number of SER the UAs used to register > themselves and later, when an INVITE comes, it would not know which > port number (5060 or 5090) it should use as the source port number of > the outgoing INVITE. The previous version would use the default one, > which would be the first one configured in the configuration file > (usually 5060). > > When an INVITE for UA1 comes, everything would work because UA1 used > port 5060 when registering and that is also the destination port in the > pinhole being kept open in NAT1: > > SER > INVITE UA1 +--------+ INVITE UA1 > UA1 ---- NAT1 <------------- | 5060 | <---------------- > | | > UA2 ---- NAT2 | 5090 | > +--------+ > > When an INVITE for UA2 comes, SER would again use port 5060 as the > default outgoing source port number, but this time the message will be > dropped by NAT2, because the pinhole opened during the registration has > 5060 as the destination port number: > > SER > INVITE UA2 +--------+ INVITE UA2 > UA1 ---- NAT1 +------ | 5060 | <--------------- > | | | > UA2 ---- NAT2 X <----+ | 5090 | > +--------+ > > The new version of registrar would also store the destination IP and > port number used on SER side when performing registration and that > information would be used later by SER when forwarding INVITEs: > > SER > +--------+ INVITE UA2 > UA1 ---- NAT1 | 5060 | <--------------- > INVITE UA2 | | > UA2 ---- NAT2 <------------- | 5090 | > +--------+ > > (Note that the X next to NAT2 has disappeared in this picture which > means that the message will be lucky enough to make it through :-). > > SER would encode this information into a SIP URI when saving contacts in > database and later, after restart of SER, this information will be > restored. The URI looks like: > > sip:1.2.3.4:5060;dstip=5.6.7.8;dstport=5090 > > Where the hostname part is the public IP of the NAT, the port (5060) is > the port allocated by the NAT, "dstip" parameter is the IP used on SER > side and "dstport" parameter is the port number used on SER side during > registration. This information is later used to find the correct outgoing > socket in SER. If no such socket can be found (it can happen if you > reconfigure SER to use different ports or IPs), it will use the default > IP/port again. > > Daredevils are encouraged to give it a try and send feedback to the list :-) > > Jan. > > _______________________________________________ > Serdev mailing list > serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev > >

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[Serusers] Turn 3xx into new Invite
by Martin Koenig 01 Mar '05

01 Mar '05

Hello, is it possible to take 3xx-responses in a reply_route and turn them into a new INVITE on Ser using append_branch or simliar? Regards, Martin

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[Serusers] syn_branch directive
by Java Rockx 01 Mar '05

01 Mar '05

Hi All. Can anyone point me to information on what the syn_branch directive in ser.cfg does? Regards, Paul

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[Serusers] HELP: SER-0.9 Record Route Problem Between SER and Asterisk
by Java Rockx 01 Mar '05

01 Mar '05

Hi All. I'm using ser-0.9 Can anyone take a quick look at this short SIP conversation and tell me if they think the problem is with my ser.cfg or a bug in Asterisk 1.0.2. We use a 3rd party for PSTN gateway functionality. This 3rd party uses a Sonus box behind a SIP proxy. Our SER proxy talks directly to their SIP proxy as needed to complete PSTN calls. The problem is that when a PSTN caller dials a SIP phone and gets sent to voice mail (Asterisk) because of a no answer or busy condition, Asterisk hangs up after the caller leaves a message. When Asterisk hangs up, the BYE from Asterisk is sent to SER, however, SER incorrectly forwards the BYE directly to their Sonus gateway, rather than the their SIP proxy. This causes our PSTN gateway provider to have "open" billing records in their system. If you look at the BYE message from Asterisk to SER you can see that route headers are missing (I think). The final BYE should have been sent to 68.86.100.20, but it was sent to 68.86.100.30 instead. I am record_route()ing all messages except for REGISTER and I have the mhomed=1 parameter set. Can anyone help me put the blame on either my ser.cfg or Asterisk? Regards, Paul IP LEGEND ----------- 68.86.100.30 - 3rd Party Sonus PSTN Gateway 68.86.100.20 - 3rd Party SIP Proxy 24.11.12.24 - Sip Express Router (eth0) 10.255.255.1 - Sip Express Router (eth1) 10.255.255.2 - Asterisk PBX NOTE: I have Asterisk connected to the SER server with a crossover cable. U 2005/02/23 22:24:18.848582 68.86.100.20:5060 -> 24.11.12.24:5060 INVITE sip:4075551212@24.11.12.24:5060 SIP/2.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Max-Forwards: 4. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Type: application/sdp. Content-Length: 312. . v=0. o=NexTone-MSW 1234 187 IN IP4 66.236.245.98. s=sip call. c=IN IP4 66.236.245.98. t=0 0. m=audio 16814 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. # U 2005/02/23 22:24:18.860022 24.11.12.24:5060 -> 68.86.100.20:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Content-Length: 0. . # U 2005/02/23 22:24:18.860259 10.255.255.1:1033 -> 10.255.255.2:5060 INVITE sip:699@10.255.255.2:5060 SIP/2.0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. To: 4075551212 <sip:4075551212@68.86.100.30:5060>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. Max-Forwards: 3. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Type: application/sdp. Content-Length: 312. . v=0. o=NexTone-MSW 1234 187 IN IP4 66.236.245.98. s=sip call. c=IN IP4 24.11.12.24. t=0 0. m=audio 36574 RTP/AVP 18 0 4 8 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=fmtp:4 annexa=yes. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. # U 2005/02/23 22:24:18.871131 10.255.255.2:5060 -> 10.255.255.1:1033 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Length: 0. . # U 2005/02/23 22:24:18.879160 10.255.255.2:5060 -> 10.255.255.1:1033 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:68.86.100.20:5060;lr>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Type: application/sdp. Content-Length: 362. . v=0. o=root 550 550 IN IP4 10.255.255.2. s=session. c=IN IP4 10.255.255.2. t=0 0. m=audio 17900 RTP/AVP 97 18 3 4 2 0 8 101. a=rtpmap:97 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 2005/02/23 22:24:18.883882 24.11.12.24:5060 -> 68.86.100.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:68.86.100.20:5060;lr>. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:699@10.255.255.2>. Content-Type: application/sdp. Content-Length: 363. . v=0. o=root 550 550 IN IP4 10.255.255.2. s=session. c=IN IP4 24.11.12.24. t=0 0. m=audio 36574 RTP/AVP 97 18 3 4 2 0 8 101. a=rtpmap:97 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:4 G723/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 2005/02/23 22:24:19.097436 68.86.100.20:5060 -> 24.11.12.24:5060 ACK sip:699@10.255.255.2 SIP/2.0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 ACK. Max-Forwards: 4. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Content-Length: 0. . # U 2005/02/23 22:24:19.098087 10.255.255.1:1033 -> 10.255.255.2:5060 ACK sip:699@10.255.255.2 SIP/2.0. Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>. Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>. Via: SIP/2.0/UDP 10.255.255.1;branch=0. Via: SIP/2.0/UDP 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045. Via: SIP/2.0/UDP 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045. To: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. From: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 1 ACK. Max-Forwards: 3. Contact: sip:3211231234@68.86.100.30:5060. Record-Route: <sip:68.86.100.20:5060;lr>. Content-Length: 0. . ### U 2005/02/23 22:24:25.104860 10.255.255.2:5060 -> 10.255.255.1:1033 BYE sip:3211231234@68.86.100.30:5060 SIP/2.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport. Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>,<sip:68.86.100.20:5060;lr>,<sip:3211231234@68.86.100.30:5060>. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Contact: <sip:699@10.255.255.2>. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. User-Agent: Asterisk PBX. Content-Length: 0. . # U 2005/02/23 22:24:25.108961 24.11.12.24:5060 -> 68.86.100.30:5060 BYE sip:3211231234@68.86.100.30:5060 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:24.11.12.24;r2=on;ftag=as588114d9;lr=on>. Record-Route: <sip:10.255.255.1;r2=on;ftag=as588114d9;lr=on>. Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Contact: <sip:699@10.255.255.2>. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. User-Agent: Asterisk PBX. Content-Length: 0. Route: <sip:3211231234@68.86.100.30:5060>. . # U 2005/02/23 22:24:25.175832 68.86.100.30:5060 -> 24.11.12.24:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. Content-Length: 0. . # U 2005/02/23 22:24:25.176182 10.255.255.1:1033 -> 10.255.255.2:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060. From: 4075551212 <sip:4075551212@68.86.100.30:5060>;tag=as588114d9. To: sip:3211231234@66.236.245.98;tag=27DECB5C-17BD. Call-ID: 9028535-3318204258-749010(a)68.86.100.30. CSeq: 102 BYE. Content-Length: 0. .

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[Serusers] Which SIP messages types use lookup("location")?
by Java Rockx 01 Mar '05

01 Mar '05

Hi All, Can anyone tell me for which message types I should call lookup("location") before t_relay()? ACK BYE CANCEL INVITE NOTIFY REFER SUBSCRIBE

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[Serusers] sip_msg->buf does not contain modifications
by José Luis Peñalver Paret 01 Mar '05

01 Mar '05

Hello, according to the guide, changes done on a message while server processing are kept in sip_msg->buf. But if I do something like (in ser.cfg): [...] append_hf("Anyheader: hello"); showrequest(); [...] being 'showrequest' a function in a module that just does: static int showrequest_f(struct sip_msg* msg, char *foo, char *foo2) { parse_headers(msg, HDR_EOH, 0); fprintf(stderr, "MESSAGE:\n"); fprintf(stderr, msg->buf); return 1; } then I can't see the new header! What am I doing wrong? Thank you.

1 0

[Serusers] kphone
by johnny Laura 01 Mar '05

01 Mar '05

Hi again, I'm trying to install Kphone in Mandrake 10.0 but I have the next error: checking for Qt... configure: error: Qt (>= Qt 2.2.2) (libraries) not found. Please check your installation! For more details about this problem, look at the end of config.log. But I have instaled the libqt3 and libqt3-devel; and when I compile ; Icompile this way ./configure --with-qt-dir=/usr/lib/qt3 --with-qt-includes=/usr/lib/qt3/include --with-qt-raries=/usr/lib/qt3/lib And even so Ihave tha same error ..Can anyone help me???

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[Serusers] Can ser Presence Agent support windows messenger 5.0 or above?
by Jeffrey Zhou 01 Mar '05

01 Mar '05

Hi, Can ser Presence Agent support windows messenger 5.0 or above? It seems i can not see the on line contact with windows messenger 5.0 with ser

2 1

[Serusers] parse error
by Tina 01 Mar '05

01 Mar '05

Hello, I added custom module (mailbox.so) to SER to add some DB queries (just like in DB example). The module exports the function mbx_lookupmailbox. The function is available for other modules; unfortunately I obtain the following when trying to call it from ser.cfg: >ser -c >0(20596) parse error (137,24-25): unknown command, missing loadmodule? (Though the module is loaded, the function is registered and found by other functions). Have I missed something? Thank you, Tina ============================================================= # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/mailbox.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication #mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) # ------------------------- 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("localhost", "subscriber")) { www_challenge("localhost", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; # if we do not get a positive reply, continue at reply_route[1] # forward the request to all destinations in destination set now t_on_failure("1"); t_relay(); }; } failure_route[1]{ log(1, "failure_route[1]\n"); if (t_check_status("408|486|487")){ log(1, "t_check_status(408|486|487)\n"); mbx_lookupmailbox(); #here will be some processing t_relay(); break; } } --------------------------------- Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.

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[Serusers] fifo db
by AJ Grinnell 01 Mar '05

01 Mar '05

Is the command fifo_db_url meant to enable access to all tables in the ser database or just certaian tables such as grp? Any changes that I make to the grp table take effect immediately, but if I make a change to aliases or domain, SER requires a restart.

2 1

[Serusers] Can I modify a transaction already in progress?
by Paul Belanger 01 Mar '05

01 Mar '05

I need some help understanding transactions, I know how to start transactions, are create a simple stateful proxy. But I'm having some troubles understanding how I can modify a current transaction. Everytime I try I always get the error: 0(10257) ERROR: t_newtran: transaction already in process Which leads me to believe once the transaction is created, I can't modify it. Is this true? ------------------------------- Paul Belanger (mailto:paul.belanger@pronexus.com) Technical Support Specialist Cisco Certified Network Associate Pronexus Inc. - A Powerful Voice in Communication Solutions ------------------------------- Tel: 613.271.8989 ext. 516 Fax: 613.271.8388 http://support.pronexus.com -------------------------------

2 1

[Serusers] Logic in Mediaproxy (ser.cfg).
by Amos Nungu 01 Mar '05

01 Mar '05

I'm trying hard to succeed with NAT without STUN. I have seen the default ser.cfg provided in the mediaproxy module (0.8.14 as well as 0.9). To me it seems as if the use_media_proxy() function is called for all calls and not for NATed only (media is relayed for all calls) because the setflag(2) set during registration is never checked anywhere.. Am I right in thinking so? Please help.

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Re: [Serusers] kphone
by Andrea Federico 01 Mar '05

01 Mar '05

try this: 1) make distclean 2) CPPFLAGS='-I/usr/lib/qt3/include -L/usr/lib/qt3/lib' ./configure --prefix=/otp/kphone-4.1.0 3) make 4) make install bye Andrea ----------------------------------------------------------------------------- Some influential environment variables: CC C compiler command CFLAGS C compiler flags LDFLAGS linker flags, e.g. -L<lib dir> if you have libraries in a nonstandard directory <lib dir> CPPFLAGS C/C++ preprocessor flags, e.g. -I<include dir> if you have headers in a nonstandard directory <include dir> CPP C preprocessor CXX C++ compiler command CXXFLAGS C++ compiler flags CXXCPP C++ preprocessor ----------------------------------------------------------------------------- -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

1 0

[Serusers] SER Message waiting indicator light Cisco 7940/7960
by Patrick Baker 01 Mar '05

01 Mar '05

Hello, I'm sure this may have been asked but, I just wanted clarify how this would be accomplished. I'm looking to have the MWI light on my cisco phone turn on when a message is left in VM. Here is my current situation: sip user -> ser -> asterisk Best regards, Patrick b.

2 1

[Serusers] corrupted address in sip message
by thibault jouannic 01 Mar '05

01 Mar '05

hello ser users; first, please don't pay attention to my english, because i'm french :-)... I'm using the pbx asterisk and ser as a sip proxy, to pass over nat problems. Here's my configuration : ipphone (192.168.0.108) ---> asterisk (192.168.0.4:5061) SER (192.168.0.4:5060) ---> nat router... During the call, a sip message comes from asterisk to SER, but when it arrives at the router, it seems to be corrupted. Here's the sip/sdp message before SER : Session Initiation Protocol Request line: INVITE sip:0467751647@212.94.190.153;user=phone SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.0.4:5061;branch=z9hG4bK653c2038 Route: <sip:22222@212.94.190.153;user=phone> From: "11111" <sip:11111@192.168.0.4:5061>;tag=as5ea95a2d To: <sip:22222@192.168.0.4>;tag=1c11534 Contact: <sip:11111@192.168.0.4:5061> Call-ID: 5de9a6a3391006b40343cfc8370a4aa0(a)192.168.0.4 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 264 Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 23042 23043 IN IP4 192.168.0.108 Owner Username: root Session ID: 23042 Session Version: 23043 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 192.168.0.108 Session Name (s): session Connection Information (c): IN IP4 192.168.0.108 ... the same after being routed by SER : Session Initiation Protocol Request line: INVITE sip:22222@212.94.190.153;user=phone SIP/2.0 Message Header Max-Forwards: 10 Via: SIP/2.0/UDP 192.168.0.4;branch=z9hG4bK5fe.bf341295.0 Via: SIP/2.0/UDP 192.168.0.4:5061;rport=5061;branch=z9hG4bK653c2038 From: "11111" <sip:11111@192.168.0.4:5061>;tag=as5ea95a2d To: <sip:22222@192.168.0.4>;tag=1c11534 Contact: <sip:11111@192.168.0.4:5061> Call-ID: 5de9a6a3391006b40343cfc8370a4aa0(a)192.168.0.4 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 353 Route: <sip:22222@212.94.190.153;user=phone> Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 23042 23043 IN IP4 192.168.0.108 Owner Username: root Session ID: 23042 Session Version: 23043 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 192.168.0.108 Session Name (s): session Connection Information (c): IN IP4 192.168.0.482.231.33.XXX192.168.0.4 ... ...as tou can see, the connection information in the sdp message is a bit strange... My ser.cfg : ------------------------- check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/mangler.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") # ------------------------- 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"); break; }; if (msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; log(1, "nouvelle trame reçue\n"); if(method=="INVITE"){ log(1, "invite route\n"); add_rport(); fix_nated_sdp("3"); fix_nated_contact(); sdp_mangle_ip("192.0.0.0/255.0.0.0", "82.231.33.XXX"); force_rtp_proxy(); t_on_reply("1"); } if (loose_route()) { log(1, "loose_route\n"); t_relay(); break; } if(method=="INVITE"){ #log(1, "trame INVITE\n"); record_route(); rewritehostport("212.94.190.153:5060"); } log(1, "relai de la trame\n\n"); if (!t_relay()) { log(1, "erreur de relai\n\n"); sl_reply_error(); } } onreply_route[1]{ log(1, "on reply\n"); if(status=~"[12][0-9][0-9]") force_rtp_proxy(); } ---------------------------- I have the same problem with ser 0.8.14 and 0.9.0. So my question is : is this a bug in ser, or something strange in my ser.cfg ? Did anybody ever see something like this ??? thanks. thibault.

1 0

[Serusers] (no subject)
by Ozan Blotter 01 Mar '05

01 Mar '05

Conditions as follows : * SER runs on a Public IP * SER works without auth & database modules, * Nearly all user behind NAT (but routers configured to do port forwarding for TCP/UDP 5060) to help SER in some cases, * Users numbers in format of 833XXXXXXX 834XXXXXXX and they should call each P2P-SIP-Calls (if not behind NAT), * If a user need to call PSTN end point (SIP Gateway located at 212.154.32.154) the call traffic should flow over SER to SIP Gateway via T1 connection already located between that systems so SER handles all voice traffic by help of RTP Proxy. * UA's registers on SER (Zyxel Prestige 2000, Zyxel Prestige 200W, Cisco ATA186 etc.) Problem is users cannot call each other (if i comment lines for nathelper they can call) It's clear i think, and below is my ser.cfg, what do i need extra or erase. <-<-<-<-< MY SER.CFG STARTS HERE >->->->-> # # $Id nathelper.cfg,v 1.1.2.1 20050301 by Ozan Blotter Exp $ # # simple quick-start config script including nathelper support # This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------ # debug=3 # debug level (cmd line -dddddddddd) # fork=yes # log_stderror=no # (cmd line -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line -v) dns=no # (cmd. line -r) rev_dns=no # (cmd. line -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 10) # Ping interval 10 seconds modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind 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"); break; }; if (msg:len > max_len ) { sl_send_reply("513", "Message Too Big"); break; }; # if ((method=="NOTIFY")&& search("^Event: Keep-Alive")) { # ls_send_reply("200","OK"); # break; # }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method=="REGISTER" || ! search("^Record-Route:")) { # log("LOG: Someone trying to register from private IP, rewriting\$ # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it $ # called symmetric media and symmetric signalling. fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { if (!(uri=~"sip:(833)|(834)")) { t_relay_to_udp("212.154.32.154","5060"); save("location"); break; }; # lookup(aliases); # if (!uri==myself) { # append_hf("P-hint: outbound alias\r\n"); # route(1); # break; # }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !searc$ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); if (!search("^Content-Length:\0")){ force_rtp_proxy(); }; # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); if (!search("^Content-Length:\0")){ force_rtp_proxy(); }; # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } <-<-<-<-< MY SER.CFG ENDS HERE >->->->->

1 0

[Serusers] MySQL problem
by sandra.garcia.espada＠dmr-consulting.com 01 Mar '05

01 Mar '05

I have the following error while restarting the SER: connect_db(): Client does not support authentication protocol requested by server. Consider upgrading MySQL client. I have upgraded the server, the client and the librarys, but it still does not work. Can anybody help me please? Thank you. Sandra García Espada DMR Consulting

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FW: [Serusers] SIP user authentication
by Lakshmi Sharma 01 Mar '05

01 Mar '05

PLEASE HELP ME... ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Lakshmi Sharma Sent: Thursday, February 17, 2005 6:04 PM To: serusers(a)lists.iptel.org Subject: [Serusers] SIP user authentication Hello, I've installed the latest version of SER on SUSE linux 9.0. Setup MYSQL as well to test sutehntication/challenge. When SER receives Register request, I'm seeing these msgs on console: "Received SIP_msg. No mem for the SIP_msg". It would be great if someone could help me with this. Thanks & Regards Lakshmi _______________________________________________ Serusers mailing list Serusers(a)iptel.org http://mail.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] More call transfer questions
by Steve Blair 28 Feb '05

28 Feb '05

I'm a bit stuck and wonder if someone on this list can help. When a call arrives at a Cisco IP phone registered to our SER proxy from the PSTN it is answered. The called party then tries to transfer the call. In either case a new invite is generated. I'm under the impression a refer message should be generated but given the way attended transfer works on the 7960s it doesn't seem possible to have a refer generated. Has anyone else seen this? If so any thoughts on what is happening? Thanks,Steve

2 3

[Serusers] "3xx Redirection" - How Should This Work?
by Java Rockx 28 Feb '05

28 Feb '05

Hi all. I'm trying to get my call working working and I have a question. Suppose a SIP subscriber has voicemail and call forwarding is enabled. When an incoming call is destined for this SIP user, SER will reply with a 302 redirect message. Next the caller's phone should construct a brand new INVITE to the redirected destination. My question is this: If the forwarded destination is busy, then the call should be sent to the voicemail of the SIP subscriber that was originally called. This is the way the RBOCs call forwarding works. If I'm using sl_send_reply("302", "Temporarily Moved") back to the caller, how can I make the voicemail piece work? Regards, Paul

2 3

[Serusers] lookup not finding local users
by Glenn Dalgliesh 28 Feb '05

28 Feb '05

I dont seem to be able to get a match when I send a call from my gw to ser registered user. Debug indicates that user Not found in usrloc even though the database shows the users registered. I am able to make out bound calls the gw just fine. If anyone can shed some light in it it would be great. Part 1 of 2 Part to follow with ser.cfg Thanks ============================================================================ ================================= mysql> select * from location; +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ | username | domain | contact | received | expires | q | callid | cseq | last_modified | replicate | state | flags | user_agent | +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ | 9901 | | sip:9901@69.250.205.31 | NULL | 2005-02-26 03:31:09 | -1.00 | e5201b3819008718(a)192.168.199.102 | 107 | 20050226033016 | 0 | 0 | 0 | Grandstream BT100 1.0.5.11 | +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ 1 row in set (0.01 sec) ====debugt output====================================================================== ========================= 20(18672) parse_headers: flags=256 20(18672) find_first_route: No Route headers found 20(18672) loose_route: There is no Route HF 20(18672) lookup(): '9901' Not found in usrloc ============================================================================ ================================= U 69.19.88.16:5060 -> 69.19.88.19:5060 INVITE sip:9901@69.19.88.19 SIP/2.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <sip:4103532264 @69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Contact: <sip:4103532264@69.19.88.16>..Call-ID: 745bcf587074e77d4a8118dc212 ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Sat, 26 Feb 2005 08:41:01 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS , BYE, REFER..Content-Type: application/sdp..Content-Length: 240....v=0..o=root 11581 11581 IN IP4 69.19.88.16..s=session..c=IN IP4 66. 159.88.16..t=0 0..m=audio 14934 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-16..a=silenceSupp:off - - - -.. # U 69.19.88.19:5060 -> 69.19.88.16:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" < sip:4103532264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Call-ID: 745bcf587074e77d4a8118dc212ccba7@69.19.88.16..CSeq: 102 INVITE..Server: Sip EXpress router (0.9.0 (i386/linux))..Content-Length: 0..Warning: 392 69.19.88.19:5060 "Noisy feedback tells: p id=18682 req_src_ip=69.19.88.16 req_src_port=5060 in_uri=sip:9901@69.19.88.19 out_uri=sip:9901@69.19.88.16:5060 via_cnt==1".... # U 69.19.88.19:5060 -> 69.19.88.16:5060 INVITE sip:9901@69.19.88.16:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:69.19.88.19;ftag=as149fbe93;lr=on>..Via: SIP/2.0/UDP 66 .159.88.19;branch=z9hG4bK96c9.80b8ddd7.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <sip:410353 2264(a)69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Contact: <sip:4103532264@69.19.88.16>..Call-ID: 745bcf587074e77d4a8118d c212ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Sat, 26 Feb 2005 08:41:01 GMT..Allow: INVITE, ACK, CANCEL, OPT IONS, BYE, REFER..Content-Type: application/sdp..Content-Length: 240..P-hint: GATEWAY....v=0..o=root 11581 11581 IN IP4 69.19.88.16..s= session..c=IN IP4 69.19.88.16..t=0 0..m=audio 14934 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone -event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -.. # U 69.19.88.16:5060 -> 69.19.88.19:5060 SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 69.19.88.19;branch=z9hG4bK96c9.80b8ddd7.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4b K2fb13d17..From: "Cell Phone MD" <sip:4103532264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>;tag=as149fbe93..Call-ID: 74 5bcf587074e77d4a8118dc212ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER ..Contact: <sip:4103532264@69.19.88.16>..Content-Length: 0....

2 2

[Serusers] SER MySQL + NAT problems
by Paulo Vicentini 28 Feb '05

28 Feb '05

Hi Would mind sending your ser.cfg file, i am facing problems with NAT traversal I have already read > <http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/Co> okbook_D2/ but it still does not working This is my ser.cfg file: Is there anything wrong whit it? # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # We will you flag 6 to mark NATed contacts modparam("registrar", "nat_flag", 6) # Enable NAT pinging modparam("nathelper", "natping_interval", 60) # Ping only contacts that are known to be # behind NAT modparam("nathelper", "ping_nated_only", 1) # ------------------------- 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # special handling for NATed clients; first, nat test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding used); also, # the received test should, if complete, should check all # vias for presence of received if (nat_uac_test("3")) { # allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("200.151.192.13", "subscriber")) { www_challenge("200.151.192.13", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } # # Forcing media relay if necessary # route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(6)) { force_rtp_proxy(); t_on_reply("1"); append_hf("P-Behind-NAT: Yes\r\n"); }; if (!t_relay()) { sl_reply_error(); break; }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); }; } Thanks Paulo _________________________________________________________________ MSN Messenger: converse online com seus amigos . http://messenger.msn.com.br

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[Serusers] ERROR: tcp_blocking_connect !!!
by Mohammad Khan 28 Feb '05

28 Feb '05

Hello List: I am getting this error message: ERROR: tcp_blocking_connect: timeout (10) By doing a search in Archive, I have found, it tcp connection take long time, this error raise. Is it right? I am not aware about any network problem in my network. Other issue, If any UA register, my locatoin table keeps record very good. But if UA re-register, location table have double entry of that user. Any idea, why its happening or how to fix it. BTW, I am running ser-0.9.0 with cvs version of rtpproxy. Thanks, MOhammad

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[Serusers] SIP encrypted calls
by Stefano 28 Feb '05

28 Feb '05

Hi all, i have a question: -do you know how it's possible to encrypt the comunication between two SIP client? -does the proxy implement some particular tasks or actions, or is it a client-side problem? -if it is so, do you know any SIP client that implements encription? Thank you very much stefano __________________________________________________________________ Tiscali Adsl 3 Mega Flat, 3 MESI GRATIS! Con Tiscali Adsl 3 Mega Flat navighi in Rete alla supervelocita' a soli 29.95 euro al mese senza limiti di tempo. Attivati entro il 28 Febbraio 2005, 3 MESI sono GRATIS Scopri come http://abbonati.tiscali.it/adsl/sa/2flat_tc/

5 11

[Serusers] part 2 lookup not finding local users
by Glenn Dalgliesh 28 Feb '05

28 Feb '05

Part 2 of 2 ==ser.cfg=================================================================== ======================================== # # $Id: serconf.sh,v 1.3 2003/11/03 13:03:58 janakj Exp $ # # autogenerated SER configuration # # user: uid=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wheel) # system: Linux baltodb.popdial.com 2.6.10-1.760_FC3 #1 Wed Feb 2 00:14:23 EST 2005 i686 i686 i386 GNU/Linux # date: Sat Feb 26 02:57:50 EST 2005 # # ----------- global configuration parameters ------------------------ /* debug=4 fork=yes port=5060 log_stderror=no memlog=5 */ mhomed=yes fifo="/tmp/ser_fifo" alias=baltodb.popdial.com # uncomment to override config values for test # /* debug=4 # debug level (cmd line: -ddd) fork=yes port=5060 log_stderror=yes # (cmd line: -E) fifo="/tmp/ser_fifox" # */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=16 # if changing fifo mode to a more restrictive value, put # decimal value in there, e.g. dec(rw|rw|rw)=dec(666)=438 #fifo_mode=438 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/enum.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- # all DB urls here #modparam("usrloc|acc|auth_db|group|msilo|uri", "db_url","mysql://ser:heslo@localhost/ser") modparam("usrloc|acc|auth_db|group|msilo|uri_db", "db_url","mysql://ser:heslo@localhost/ser") # -- usrloc params -- /* 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "user_column", "user_id") modparam("auth_db", "password_column", "password") modparam("auth", "nonce_expire", 300) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # that is the flag for which we will account -- don't forget to modparam("acc", "db_flag", 1 ) modparam("acc", "db_missed_flag", 3 ) # -- tm params -- modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 90 ) modparam("tm", "wt_timer", 20 ) # -- msilo params modparam("msilo", "registrar", "sip:registrar@baltodb.popdial.com") # -- enum params -- # modparam("enum", "domain_suffix", "e164.arpa.") # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Alas Too Many Hops"); break; }; #if (len_gt( max_len )) { if ( msg:len > max_len ) { sl_send_reply("513", "Message too large sorry"); break; }; # Make sure that requests dont advertise addresses # from private IP space (RFC1918) in Contact HF # (note: does not match with folded lines) if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { # allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if ((method=="REGISTER" || ! search("^Record-Route:")) && !( src_ip==192.168.0.0/16 || src_ip==10.0.0.0/8 || src_ip==172.16.0.0/12 )) { log("LOG: Someone trying to register from private IP again\n"); sl_send_reply("479", "We dont accept private IP contacts" ); break; }; }; # anti-spam -- if somene claims to belong to our domain in From, # challenge him (skip REGISTERs -- we will chalenge them later) if (search("(From|F):.*(baltodb\.popdial\.com|66\.159\.88\.19)")) { # invites forwarded to other domains, like FWD may cause subsequent # request to come from there but have iptel in From -> verify # only INVITEs (ignore FIFO/UAC's requests, i.e. src_ip==myself) if (method=="INVITE" & !(src_ip==69.19.88.19)) { if (!(proxy_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */ ))) { proxy_challenge("baltodb.popdial.com" /* realm */, "0" /* no-qop */); break; }; # to maintain outside credibility of our proxy, we enforce # username in From to equal digest username; user with # "john.doe" id could advertise "bill.gates" in From otherwise; if (!check_from()) { log("LOG: From Cheating attempt in INVITE\n"); sl_send_reply("403", "That is ugly -- use From=id next time (OB)"); break; }; # we better don't consume credentials -- some requests may be # spiraled through our server (sfo@iptel->7141@iptel) and the # subsequent iteration may challenge too, for example because of # iptel claim in From; UACs then give up because they # already submitted credentials for the given realm #consume_credentials(); }; # INVITEs claiming to come from our domain } else if (method=="INVITE" && !(uri=~"[@:\.](baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" # ... and we serve our gateway too if present | uri=~"@66\.159\.88\.16([;:].*)*" )) { #the INVITE neither claims to come from our domain nor is it targeted to it # -> junk it sl_send_reply("403", "No relaying"); break; }; /* ********* RR ********************************** */ # to be safe, record route everything; UAs may use different # transport protocols and need to have SER in path record_route(); # if route forces us to forward to some explicit destination, # do so; check however first that a cheater didn't preload # a gateway destination to bypass PSTN ACLs if (loose_route()) { if (uri=~"sip:[+0-9]+@66\.159\.88\.16") { # it is gateway -- proceed to ACLs route(3); break; }; # route HF determined next hop; forward there append_hf("P-hint: rr-enforced\r\n"); t_relay(); break; }; /* ********* check for requests targeted out of our domain... ******* */ # sign of our domain: there is '@' (username) or : (nothing) in # front of our domain name ; ('.' is not there -- we handle all # xxx.iptel.org as outbound hosts);if none of these cases matches, # proceed with processing of outbound requests in route[2] if (!(uri=~"[@:](sip[\.)?(baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" | uri=~"@66\.159\.88\.16([;:].*)*")) { route(2); break; }; /* ************ requests for our domain ********** */ /* now, the request is for sure for our domain */ # registers always MUST be authenticated to # avoid stealing incoming calls if (method=="REGISTER") { # Make sure that user's dont register infinite loops # (note: does not match with folded lines) if (search("^(Contact|m): .*(a)(baltodb\.popdial\.com|66\.159\.88\.19)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; if (search("^(Contact|m): .*66\.159\.88\.16")) { log(1, "LOG: alert: protected contacts\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; if (!www_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */ )) { # challenge if none or invalid credentials www_challenge( "baltodb.popdial.com" /* realm */, "0" /* no qop -- some phones can't deal with it */); break; }; # prohibit attempts to grab someone else's To address # using valid credentials; if (!check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use To=id in REGISTERs"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; m_dump(); break; }; # some UACs might be fooled by Contacts our UACs generate to make MSN # happy (web-im, e.g.) -- tell its urneachable if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon is gone"); break; }; # is this an ENUM destination (leading +?)? give it a try, if the lookup # doesn't change URI, just continue if (uri=~"sip:\+[0-9]+@") { if (!enum_query("voice")) { # if parameter empty, it defaults to "e2u+sip" enum_query(""); # E2U+sip }; } else { # aliases (take precedences over PSTN number; provisioning interface # is set up to assinge aliases beginning with 8) lookup("aliases"); }; # check again, if it is still for our domain after aliases are resolved if (!(uri=~"[@:](sip[\.)?(baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" | uri=~"@66\.159\.88\.16([;:].*)*")) { route(5); break; }; # now check if it's about PSTN destinations through our gateway; # note that 8.... is exempted for numerical non-gw destinations if (uri=~"sip:\+?[0-79][0-9]*@.*") { route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found ... route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to UsrLoc to bypass our authorization logic if (uri=~"sip:[+0-9]+@66\.159\.88\.16") { # it is gateway -- proceed to ACLs route(3); break; }; /* ... and also report on missed calls ... */ setflag(3); # we now know we may, we know where, let it go out now! append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------------------- OUTBOUND ---------------------------------------- # routing logic for outbound requests targeted out of our domain # (keep in mind messages to our users can end up here too: for example, # an INVITE may be UsrLoc-ed, then the other party uses outbound # proxy with r-uri=the usr_loced addredd (typically IP)) route[2] { append_hf("P-hint: OUTBOUND\r\n"); t_relay(); } #------- ALIASED OUTBOUND -------------------------------------------- # routing logic for inbound requests aliased outbound; unlike # with real outbound requests we do not force authentication # as these calls are server by our server and we do not want # to disqualify unathenticated request originatiors from other # domains route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); t_relay(); } #----------------- PSTN ---------------------------------------------- # logic for calls to the PSTN route[3] { # turn accounting on setflag(1); /* require all who call PSTN to be members of the "int" group; apply ACLs only to INVITEs -- we don't need to protect other requests, as they don't imply charges; also it could cause troubles when a call comes in via PSTN and goes to a party that can't authenticate (voicemail, other domain) -- BYEs would fail then; exempt Cisco gateway from authentication by IP address -- it does not support digest */ if (method=="INVITE" && (!src_ip==69.19.88.16)) { if (!proxy_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "baltodb.popdial.com" /* realm */, "0" /* no qop */ ); break; }; # let's check from=id ... avoids accounting confusion if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; if(!is_user_in("credentials", "int")) { sl_send_reply("403", "NO PSTN Privileges..."); break; }; consume_credentials(); }; # INVITE to authorized PSTN # if you have passed through all the checks, let your call go to GW! rewritehostport("69.19.88.16:5060"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("69.19.88.16","5060"); } /* *********** handling of unavailable user ******************* */ route[4] { /**/ # message store if (method=="MESSAGE") { t_newtran(); if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); } else { t_reply("503", "Service Unavailable"); }; break; }; /**/ # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # voicemail subscribers ... t_newtran(); t_reply("404", "Not Found"); # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted if (method=="INVITE") { acc_db_request("404 missed call", "missed_calls"); }; }

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