sr-users November 2007

sr-users@lists.kamailio.org

138 participants
248 discussions

[Serusers] Segmentation Fault (SER Not Starting)

by Alexandr Volkov

Hi All, I am having trouble when trying to start SER. I am using Ser 0.9.6 on centos4. When I am trying to start it gives: Segmentation Fault; but only if in 'location' table there is a row with callId starting by number: <field name="callid">49c72d7ed685500e7d844d1cf1357dbd</field> if I change this field in database (simple add any character), ser will start. <field name="callid">a49c72d7ed685500e7d844d1cf1357dbd</field> seems like a bug

16 years, 5 months

[Serusers] ERROR: forward_reply: no 2nd via found in reply - via parsing error

by Tomasz Zieleniewski

Hi, I have a chcek via parameter set in the ser.cfg file When ser receives reply with such via header it returns ERROR: forward_reply: no 2nd via found in reply This Via seems to be ok? Via: SIP/2.0/UDP 192.168.0.165:7160;branch=z9hG4bK1589.3bc310b3.1,SIP/2.0/UDP 192.168.0.165:7060;branch=z9hG4bK1589.e1306596.0,SIP/2.0/UDP 192.168.129.74 ;rport=5060;branch=z9hG4bK1589.606a3e61.0,SIP/2.0/UDP 192.168.129.74:5160 ;branch=z9hG4bK1589.a5bd9fb4.0,SIP/2.0/UDP 192.168.129.74;branch= z9hG4bK1589.506a3e61.0,SIP/2.0/UDP 192.168.0.102:7060 ;rport=7060;branch=z9hG4bK-d87543-a228576fa32c0b22-1--d87543- What is wrong? Cheers tomasz

16 years, 5 months

[Serusers] Unexpected disconnect - lost SIP packets?

by Jeremy A

Hello, I'm need some help with SIP protocol / SER configuration to resolve a problem. I have a system with a SER instance and a SIP client application running on the same box. SER is 10.50.32.10:5060 APP is 10.50.32.10:5070 Remote SIP server 10.1.39.10 Remote phone is 10.1.39.202 The user agent is registered with the SER instance as extension 1009. It has aliases 1001-1008 and is actually called as 231x which maps onto 100x which then connects to extension 1009. The remote phone makes a call to the user agent application via the remote SIP server and the SER instance. It dials 2311 which gets automagically converted to extension 1009 The call is established but the user application times out and detects a disconnect despite the calling phone not hanging up. In my wireshark trace it seems that SER is ignoring some form of keep-alive from the remote SIP server and this is causing the timeout. The lines of interest include 39, 42, 45, 50, 53, 56, 59 Lines 28 and others similar are on the loopback interface and are packets between port 5060 and 5070 Can someone please advise if the SIP protocol has problems and/or if SER configuration has problems? 22 0.218826 10.50.32.10 -> 127.0.0.1 SIP Request: REGISTER sip:localhost.localdomain 23 0.219570 127.0.0.1 -> 10.50.32.10 SIP Status: 200 OK (1 bindings) 24 19.472555 10.1.39.10 -> 10.50.32.10 SIP/SDP Request: INVITE sip:1001@butler.fesa.sto;transport=udp, with session description 25 19.473394 10.50.32.10 -> 10.1.39.10 SIP Status: 100 trying -- your call is important to us 26 19.473596 10.50.32.10 -> 10.50.32.10 SIP/SDP Request: INVITE sip:1009@10.50.32.10:5070;LINEID=1d5a67e38809, with session description 27 19.484244 10.50.32.10 -> 10.50.32.10 SIP Status: 100 Trying 28 19.725476 10.50.32.10 -> 10.50.32.10 SIP Status: 180 Ringing 29 19.725828 10.50.32.10 -> 10.1.39.10 SIP Status: 180 Ringing 32 21.770798 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 33 21.782437 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 38 21.789396 10.50.32.10 -> 10.1.39.202 RTCP Source port: 8001 Destination port: 16391 39 21.800069 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 40 22.278096 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 41 22.278713 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 42 22.298101 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 43 23.285536 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 44 23.288988 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 45 23.307839 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 48 25.298214 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 49 25.299105 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 50 25.317892 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 51 29.306496 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 52 29.306758 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 53 29.328420 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 54 33.314736 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 55 33.318736 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 56 33.338302 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 57 37.323175 10.50.32.10 -> 10.50.32.10 SIP/SDP Status: 200 OK, with session description 58 37.324791 10.50.32.10 -> 10.1.39.10 SIP/SDP Status: 200 OK, with session description 59 37.347936 10.1.39.10 -> 10.50.32.10 SIP Request: ACK sip:2311@10.50.32.10:5070;LINEID=1d5a67e38809 --> client detects disconnect here Thanks jeremy

16 years, 5 months

[Serusers] Presence in SIP

by Carlos Angeles

Hello I have installed Ser-2.0 with the PA module.... I am testing the server with x-lite... When a user is NOTIFYING some event change the server is answering with a Status: 420 Bad Extension.... Is thst SER doesn´t support this kind of presence events? Thanks Carlos _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

16 years, 5 months

[OpenSER-Users] Handling re-INVITES --hellp required

by srinivas Antarvedi

Hello all, i have users one is on global ip and another behind NAT am using asterisk as media server leg 1: caller : Global ip UAC. callee: asterisk leg2: caller :asterisk callee: NATed UAC. sdp of NATed client is handled at openser reply route at first stage when asterisk re-invites the NATed UAC to bridge the two call-Leg's the sdp from NATed UAC is not changed ,, even if i call t_on_reply in the loose route section of the script.. it is still showing privat ip so finally after 2 or 3 sec's there was an end to the dialog can anybody have any idea to handle re-invite's 200 ok SDP mangling? please help me out.. Thanks in advance regards srinivas -- Srinivas Antarvedi

16 years, 5 months

[OpenSER-Users] auth_db module - get_ha1(): Error while querying database

by Mihail Gluhowchenko

Hello All; I`m trying to tune openser-1.2.2 with dbtext data base But have some problems. I make db use - textdb.sh Then add users use - sc.dbtext and add to openser.cfg: #set module path mpath="/usr/local/lib/openser/modules/" # use dbtext database loadmodule "dbtext.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- 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 -- # use dbtext database for persistent storage modparam("usrloc", "db_mode", 2) modparam("usrloc|auth_db", "db_url", "dbtext:///usr/local/etc/openser/dbtext") # 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", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "domain_column", "domain") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 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"); log(1, "Too Many Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); log(1, "Message too big\n"); exit; }; # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # 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") { log(1, "this is a request REGISTER users\n"); if (!www_authorize("my.domain", "subscriber")) { log(1, "this is a www_challenge\n"); www_challenge("my.domain", "0"); exit; }; save("location"); exit; }; but when trying REGISTER user use(sjphone) in log i see: Nov 28 17:30:03 devel s-eltex[12797]: this is a request for my.domain users Nov 28 17:30:03 devel s-eltex[12797]: this is a request REGISTER users Nov 28 17:30:03 devel s-eltex[12797]: this is a www_challenge Nov 28 17:30:03 devel s-eltex[12794]: this is a request for my.domain users Nov 28 17:30:03 devel s-eltex[12794]: this is a request REGISTER users Nov 28 17:30:03 devel s-eltex[12794]: get_ha1(): Error while querying database In sjphone LOG i see: Service unavailable server error; What do I adjust not correctly?

16 years, 5 months

[OpenSER-Users] t_relay can not handle PRACK retransmission

by Klaus Darilion

Hi! I have encountered this problem at an openser 1.1.1 proxy: Client Proxy Client ---PRACK------> ------PRACK-----> retansmission -----PRACK-------> here at this point (when openser receives the PRACK retransmission) t_relay fails: ERROR: t_newtran: transaction already in process 0xb66e7880 Is this a known problem? Were there some fixes to 1.1 or 1.2 branch? regards klaus

16 years, 5 months

[OpenSER-Users] Branch limitations

by Daniel

Hi all, I've read this from iptel website. It is true for SER 0.11.0. What about Openser 1.2 ? " * Script processing of multiple branches on forking Warning ser's request processing language allows to make request decisions based on current URI. When a request if forked to multiple destinations, only the first branch's URI is used as input for script processing. This might lead to unexpected results. Whenever a URI resolves to multiple different next-hop URIs, only the first is processed which may result in handling not appropriate for the other branch. For example, a URI might resolve to an IP phone SIP address and PSTN gateway SIP address. If the IP phone address is the first, then script execution ignores the second branch. If a script includes checking gateway address in request URI, the checks never match. That might result in ignoring of gateway admission control rules or applying them unnecessarily to non-gateway destinations. List of known problems is publicly available at the ser webpage at [70]http://www.iptel.org/ser/ . See the "ISSUES" link. -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

16 years, 5 months

[OpenSER-Users] OpenSER: Support of advanced RegExpr?

by Helmut Kuper

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is openser supporting advanced regular expressions as described here: http://www.regular-expressions.info/refadv.html regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHTVg64tZeNddg3dwRAl+JAKCst6W4QdNjJ6ZT6fWRaE9otPcX+wCdEKbw iQy1J1Md3CKE7pUgx4lld3k= =sRw7 -----END PGP SIGNATURE-----

16 years, 5 months

[OpenSER-Users] OpenXCAP: 'presence-rules' PUT sent: '409 Conflict' received

by ddonate

Hi, I am using OpenSER and openXCAP to handle 'resource list' and 'presence rules' of some buddies. The 'resource-list' works fine: I can GET my file (404 the 1st time, and I PUT one). But I have a problem with the 'presence-rules' PUT, when I receive a 'winfo' NOTIFY (when the user 'john' subscribes to my presence): I send an HTTP request but I receive a '409 Conflict': - My request: PUT /pres-rules/users/diego(a)openser/presence_rules.xml HTTP/1.1 User-Agent: IMS_DialcomClient Accept: text/* Host: 192.168.1.208:8000 Content-Length: 284 ... <?xml version='1.0' encoding='UTF-8'?><ruleset xmlns='urn:ietf:params:xml:ns:presence-rules'><rule id='sip:john@openser_rule'><conditions><identity><one id='sip:john@openser'/></identity></conditions><actions><sub-handling>allow</sub-handling></actions></rule></ruleset> - The traces in the openXCAP are: -------------------- Failed to validate document against XML schema: <string>:1:ERROR:SCHEMASV:SCHEMAV_CVC_ELT_1: Element '{urn:ietf:params:xml:ns:pres-rules}ruleset': No matching global declaration available for the validation root. -------------------- I know it is related to the schema, but I have not managed to fix it... Thanks in advance Diego -- View this message in context: http://www.nabble.com/OpenXCAP%3A-%27presence-rules%27-PUT-sent%3A-%27409-C… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

16 years, 5 months

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sr-users November 2007