sr-users May 2007

sr-users@lists.kamailio.org

164 participants
295 discussions

by Stefano Blasco

Dear all, i'm trying to have call pickup with SER (i use ser 0.9.6), i mean i'd like to have the capability to pick up a call that is directed to another user while the other user telephone is ringing. Is there someone that suceeded in this features? If it is not possible to get this feature by configuriation of ser (i mean using already present ser functions), is there someone that can suggest how to implemente such features? (for istance with external scripts,or with avpops, or using fifo, or writing a new ser module). Maybe the use of fifo could help, in particular i saw the ctd.sh example that is able to iniziate a call for two users using t_uac_dlg fifo function. I tryed to use t_uac_dlg to send a CANCEL to a transaction in progress (in the 180 ringing status) (taking by hand parameters like Call-ID, tag, branch etc.), but it seems that t_uac_dlg is not able to interact with the transaction in progress because it creates one different transaction. Is it real, or am i wrong? This is some log for this attempt: ################################################################ 1(17187) INFO: fifo_server: command empty 1(17187) fifo_get_method: method: 'CANCEL' 1(17187) DEBUG: fifo_get_ruri: 'sip:801@x.x.x' 1(17187) DEBUG: fifo_get_nexthop: next hop empty 1(17187) fifo_get_headers: headers: To: <sip:801@x.x.x> From: Stefano<sip:803@x.x.x>;tag=237abc61 Call-ID: 262a660822777d26 CSeq: 2 CANCEL 1(17187) parse_headers: flags=-1 1(17187) end of header reached, state=9 1(17187) DEBUG: get_hdr_field: <To> [28]; uri=[sip:801@x.x.x] 1(17187) DEBUG: to body [<sip:801@x.x.x> ] 1(17187) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 1(17187) DEBUG: fifo_uac: parse_headers succeeded 1(17187) fifo_get_body: body: 1(17187) DEBUG: add_param: tag=237abc61 1(17187) end of header reached, state=29 1(17187) DEBUG: fifo_uac: EoL -- proceeding to transaction creation 1(17187) DEBUG:tm:t_uac: next_hop=<sip:801@x.x.x> 1(17187) DEBUG: mk_proxy: doing DNS lookup... 1(17187) get_record: lookup(_sip._udp.x.x.x, 33) failed 1(17187) sip_resolvehost: no SRV record found for x.x.x, trying 'normal' lookup... 1(17187) DEBUG:destroy_avp_list: destroying list (nil) 1(17187) DEBUG: dlg2hash: 55633 1(17187) DEBUG: add_to_tail_of_timer[4]: 0xb5ba0ac0 1(17187) DEBUG: add_to_tail_of_timer[0]: 0xb5ba0ad0 1(17187) **** done consume 2(17188) SIP Request: 2(17188) method: <CANCEL> 2(17188) uri: <sip:801@x.x.x> 2(17188) version: <SIP/2.0> 2(17188) parse_headers: flags=1 2(17188) Found param type 232, <branch> = <z9hG4bK159d.9169633.0>; state=16 2(17188) end of header reached, state=5 2(17188) parse_headers: Via found, flags=1 2(17188) parse_headers: this is the first via 2(17188) After parse_msg... 2(17188) preparing to run routing scripts... 2(17188) parse_headers: flags=128 2(17188) end of header reached, state=9 2(17188) DEBUG: get_hdr_field: <To> [28]; uri=[sip:801@x.x.x] 2(17188) DEBUG: to body [<sip:801@x.x.x> ] 2(17188) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 2(17188) DEBUG: get_hdr_body : content_length=0 2(17188) found end of header 2(17188) DEBUG: is_maxfwd_present: max_forwards header not found! 2(17188) DEBUG: add_param: tag=237abc61 2(17188) end of header reached, state=29 2(17188) parse_headers: flags=256 2(17188) find_first_route: No Route headers found 2(17188) loose_route: There is no Route HF 2(17188) is_local(): Realm 'x.x.x' is local 2(17188) DEBUG: t_newtran: msg id=9 , global msg id=8 , T on entrance=0xffffffff 2(17188) parse_headers: flags=-1 2(17188) parse_headers: flags=60 2(17188) t_lookup_request: start searching: hash=55633, isACK=0 2(17188) DEBUG: RFC3261 transaction matching failed 2(17188) DEBUG: t_lookup_request: no transaction found 2(17188) DBG: trans=0xb5ba1488, callback type 1, id 0 entered 2(17188) parse_headers: flags=44 2(17188) DEBUG: t_lookupOriginalT: searching on hash entry 55633 2(17188) DEBUG: RFC3261 transaction matching failed 2(17188) DEBUG: t_lookupOriginalT: no CANCEL matching found! 2(17188) DEBUG: t_lookupOriginalT completed 2(17188) DEBUG: mk_proxy: doing DNS lookup... 2(17188) get_record: lookup(_sip._udp.x.x.x, 33) failed 2(17188) sip_resolvehost: no SRV record found for x.x.x, trying 'normal' lookup... 2(17188) check_via_address(x.x.x.x, x.x.x.x, 0) ################################################################ Another thing, i saw that t_uac_cancel, instead, is able to interact with an existing transaction, by sending CANCEL to called and 487 to caller (using t_lookup_callid function to get transaction from callid and cseq). Does someone think that it is possible to modify t_uac_cancel to handle basic pickup functions? (that should be CANCEL for the called side and redirect for caller towards pickup user) This is some log for this case: ################################################################## 1(16115) INFO: fifo_server: command empty 1(16115) DEBUG: fifo_uac_cancel: ############### begin ############## 1(16115) DEBUG: fifo_uac_cancel: callid="0b29c5266716aa67" 1(16115) DEBUG: fifo_uac_cancel: cseq="2" 1(16115) created comparable call_id header field: >Call-ID: 0b29c5266716aa67 < 1(16115) created comparable cseq header field: >CSeq: 2 INVITE< 1(16115) just locked hash index 37328, looking for transactions there: 1(16115) we have a match: callid=>>Call-ID: 0b29c5266716aa67 << cseq=>>CSeq: 2<< 1(16115) DEBUG: t_lookup_callid: transaction found. 1(16115) DEBUG: fifo_uac_cancel: now calling cancel_uacs 1(16115) DEBUG: cancel_branch: sending cancel... 1(16115) DEBUG: add_to_tail_of_timer[4]: 0xb5b763f0 1(16115) DEBUG: add_to_tail_of_timer[0]: 0xb5b76400 1(16115) DEBUG: fifo_uac_cancel: ################ end ############## ################################################################## Thank you very much for your help Stefano Blasco

17 years, 6 months

[Serusers] behind 2 routers

by root

Hi, I have a DSL router, which has only one port. To connect multiple computers I have another 4 ports router behind this DSL router. So SER is behind two routers. How can I configure these two routers so that I can use SER in this situation? Thanks,

17 years, 6 months

[Serusers] SIP Security and Monitoring tool running on iptel.org

by Tudor Golubenco

Hi SER users and devs, During the past few months we, at Iptego, have been working on a security and monitoring application for SIP sites, one of the recurring subjects of the SER user meeting in Prague ;-). To be short, VSP is a network intrusion detection system, which can sniff the VoIP traffic, parse SIP messages and create alerts when something seems wrong. As the project evolved, we have presented our work to the iptel.org staff and, together with them, decided to make the monitoring extension of VSP available to all public. So, if you're interested in real time statistics of the iptel.org SIP site, or in the repartition of User-Agents, you can create your own graphs here: http://fly.iptel.org/vsp/ We have already received great feedback from the iptel.org guys and I hope to get even more from the rest of community. Disclaimer: The project is BETA for now and the graphs haven't been verified yet. I wouldn't use them for anything important. Cheers, Tudor Golubenco -- Dipl. Ing. Tudor Golubenco Software Engineer iptego GmbH Am Borsigturm 40, 13507 Berlin, Germany http://www.iptego.de

17 years, 6 months

[Users] x-lite and nat_uac_test(16)

by Alexander Bergolth

Hi! I've noticed that (at least on my boxes) x-lite uses a different source-port for the sip-connection than the one that is announced in the Via-header. (See the example below.) This behavior results in nat_uac_test(16) always reporting that the client is behind a NAT. Can anyone confirm this behavior? Is it a bug in X-Lite or is there any good reason for it? Cheers, --leo -------------------- 8< -------------------- No. Time Source Destination Protocol Info 4 0.000808 137.208.90.164 137.208.89.45 SIP Request: REGISTER sip:wu-wien.ac.at Frame 4 (640 bytes on wire, 640 bytes captured) Ethernet II, Src: DellEsgP_85:83:ea (00:0b:db:85:83:ea), Dst: Cisco_f3:2e:80 (00:0d:29:f3:2e:80) Internet Protocol, Src: 137.208.90.164 (137.208.90.164), Dst: 137.208.89.45 (137.208.89.45) Transmission Control Protocol, Src Port: 1831 (1831), Dst Port: 5060 (5060), Seq: 1, Ack: 1, Len: 586 Source port: 1831 (1831) Destination port: 5060 (5060) Sequence number: 1 (relative sequence number) [Next sequence number: 587 (relative sequence number)] Acknowledgement number: 1 (relative ack number) Header length: 20 bytes Flags: 0x18 (PSH, ACK) Window size: 1028096 (scaled) Checksum: 0x491e [correct] Session Initiation Protocol Request-Line: REGISTER sip:wu-wien.ac.at SIP/2.0 Message Header Via: SIP/2.0/TCP 137.208.90.164:1676;branch=z9hG4bK-d87543-bd24cf102007467f-1--d87543-;rport Max-Forwards: 70 Contact: <sip:30001@137.208.90.164:1676;rinstance=7ce462f932219953;transport=TCP> To: "Alexander Bergolth"<sip:30001@wu-wien.ac.at> From: "Alexander Bergolth"<sip:30001@wu-wien.ac.at>;tag=b904b134 Call-ID: N2VjNGQxYzZlOWNiYWU5OTMwNzQ4OTZkMWU3NmRkYTE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 -------------------- 8< -------------------- -- e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

17 years, 6 months

[Users] WeSIP: Click2Call Linefeed

by Helmut Kuper

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello WeSIPer, I found a little but annoying Error in WeSIP's C2C application. I send back a 483 reply on a C2C's first caller-INVITE to WeSIP via openser's sl-send_reply(). I expected that C2C terminates the call function, cause C2C has an error handling method for 4xx replies, but nothing happens. openser as wesip's frontend received definately the reply. By chance I found, that the reason for this was my reply message. This doesn't work with WeSIP's C2C application: sl_send_reply("483", "My message\n"); But this works: sl_send_reply("483", "My message"); It seems, that C2C's or WeSIPs reply parser has some problems with linefeeds at the end of reply's message text. I use openser 1.2.0, WeSIP 0.1.4 and click2call-08_11_06 regards Helmut PS: Is there already a fix for WeSIP to reconnect to openser's seas socket, when openser restarts? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGUa8R4tZeNddg3dwRAscvAKDC49yJRkA4ALOCoC5naJzr9gavMwCff7vX oEc4vGa8tEP5r0IVMhvC+jA= =dMZ6 -----END PGP SIGNATURE-----

17 years, 6 months

[Users] Openser and Mediaproxy behind NAT

by Maciek Kowalski

Hi All, I am new to Openser and propably answer to my question is very simple.. I,ve been trying to run Openser with Mediaroxy behind NAT (they both work on a same machine with a private IP - 192.168.1.x). I am connected to internet through ADSL router (dynamic IP). I ve tried few configurations and I cant get it to work - I have "no audio problem" with nated users who register from outside of my LAN. I ve read that mediaproxy must listen on a public IP address, but first I thought that this is just a matter of port forwarding. And here is my question: Is it possible for this to work with described configuration? (Openser with mediaproxy both behind NAT) I ve used similar solution with Asterisk and Xtunnels and it worked fine. Thanks in advance and sorry for my English:). Best regards, Maciek Kowalski

17 years, 6 months

[Serusers] sip uri transform

by Ҷ��

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hi. I have a question. I use x_lite do client,and ser server do route. if user1 is 1001 and user2 is 1002, assumes i used 1001 to dial 999991002, I thinking in the ser.conf, first to remove the prefix(99999),then route to 1002. if used asterisk the idea is ok but I don't know ser whether to achieve? [...] Content analysis details: (9.8 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_HELO_PASS SPF: HELO matches SPF record 4.1 FROM_ILLEGAL_CHARS From: has too many raw illegal characters -0.0 SPF_PASS SPF: sender matches SPF record 0.5 HTML_40_50 BODY: Message is 40% to 50% HTML 0.0 HTML_MESSAGE BODY: HTML included in message 0.0 BAYES_50 BODY: Bayesian spam probability is 40 to 60% [score: 0.5035] 0.0 MIME_HTML_ONLY BODY: Message only has text/html MIME parts 0.2 DNS_FROM_RFC_ABUSE RBL: Envelope sender in abuse.rfc-ignorant.org 2.2 RCVD_IN_WHOIS_INVALID RBL: CompleteWhois: sender on invalid IP block [202.105.45.47 listed in combined-HIB.dnsiplists.completewhois.com] 1.7 DNS_FROM_RFC_POST RBL: Envelope sender in postmaster.rfc-ignorant.org 1.1 HTML_MIME_NO_HTML_TAG HTML-only message, but there is no HTML tag The original message was not completely plain text, and may be unsafe to open with some email clients; in particular, it may contain a virus, or confirm that your address can receive spam. If you wish to view it, it may be safer to save it to a file and open it with an editor.

17 years, 6 months

[Users] Presence Problem - Publish Offline

by Hildebrando Trannin

Hi all, I was testing OpenSER presence functionality using eyebeam and a softphone developed by my team. The OFFLINE state with eyebeam worked perfectly, but with my softphone it didn't work. I set force_active = 1. XML sent by PUBLISH and OpenSER log are below. Can anyone help me? Thanks, Hildebrando PIDF XML: <?xml xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:hildebrando@voip.maritaca"> <presence> <tuple id=""> <status> <basic>closed</basic> </status> <note>offline</note> </tuple> </presence> The basic difference between my application and eyebeam is that eyebeam uses RPID XML and I use PIDF XML. OPENSER LOG: 3(26039) PRESENCE: update_presentity: The presentity status is offline; do not send notify

17 years, 6 months

[Serusers] Does SER's MYSQL work in SUSE Linux?

by Cyprian Clement

Hello Community, Does anyone know whether it's possible to install the current recommended MYSQL on SUSE Linux. I have a Machine with a SUSE installed on it. I have successfully installed SER 0.9.6 but now I would like to have MYSQL for authentication support. Does the SER's recommended MYSQL work with SUSE?. I only see the talk about Redhat and Debian so unsure if it works with SUSE. Many thanks for in advance. Cyprian. ---------- ___________________________________________________________ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk

17 years, 6 months

[Users] Dropped calls with OpenSER 1.1.x and Asterisk >= 1.2.14

by Edoardo Serra

Hi guys, I'm having a problem about asterisk dropping calls received from OpenSER The problems was spotted at Asterisk users mailing lists http://lists.digium.com/pipermail/asterisk-users/2007-April/184875.html Asterisk complains about an ACK not received and drops the calls. It expects OpenSER to send an ACK back to its '200 OK' when the call is set up Is that a problem OpenSER should solve ??? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

17 years, 6 months

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sr-users May 2007