sr-users March 2006

sr-users@lists.kamailio.org

223 participants
505 discussions

[Serusers] Carrier-grade framework for SER
by Andreas Granig 27 Sep '06

27 Sep '06

Hi guys, I'd like to propose another possibility for a highly-available and scalable system design based on SER: The problems I've encountered for scalable systems are: - Distribution of the user location and alias location among the nodes (user location is based on registrations, alias location comes from web interfaces and is used for call forwarding). - Reloading up to date location tables after breakdown and recovery of a node So I'm just thinking loud about the following provisioning system: - Write a client which fulfills the this demands: - Receive one or more locations from SER via a SER module or from a web application and distribute them to other known clients. Take care of retransmissions if a client isn't reachable or reports a temporary failure. - Receive one or more locations from other clients and write them into the SER FIFO. If writing into the FIFO fails, try to write directly into the database (location-table, alias-table etc.). Report a temporary failure if this also fails. Maybe a centralized server should be used which receives the locations from the clients and distributes them to other clients, so that the nodes just know about the server and nothing about other nodes. This would make integration of new nodes easy. On the other hand, it's another single point of failure, so a decentralized solution should be considered. But that would mean that you've to inform every node about the existence of a new node. The protocol used between the nodes should be simple and fast. So I think SOAP drops out here. Maybe XMLRPC or ICE (http://www.zeroc.com/ice.html) could be used. One might think now why not just use replication on SIP layer, but t_replicate only supports one peer and you've no possibility to get locations on a node while it's down. Replication of other location tables like the alias-table is also not possible. I'd be willing to release these parts as GPL for creating an open framework for carrier-grade SER integration, so any feedback, improvements or flames are highly welcome. Cheers, Andy

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[Serusers] OT: DIDs from Europe.
by Juan 15 Sep '06

15 Sep '06

Hi Gang. The company I work for is interested in buying DIDs from Europe. Of course the connection should be SIP. If any of you is selling them, please contact me off the list. Regards Juan Ferrari VoIP & Telecom Specialist United World Telecom Phone: +1561-276 7156 ext. 302

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[Serusers] Routing users to other SIP server
by KaveH Aasaraai 03 Aug '06

03 Aug '06

Hi, I was wondering how I can route my SER users to other SIP servers, without need of authentication to other server. I mean this: My User -> Auth -> My SER My SER -> Auth -> Other SIP Server My User --------make call--------> My SER --------route call-------> Other SIP Server Thank you. Kaveh __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

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[Users] Parallel ringing
by Andreas M. 19 Jul '06

19 Jul '06

Hello, how could it be done through AVP´s and database lookup, to do a parallel ringing. Following setup: If uri eg example(a)test.dom is called, this call should be forwarded in parallel to a pstn-gateway. That means, receive the AVP with the telephonnumber associated to uri example(a)test.dom, rewrite uri, rewrite host/port and forward in parallel to gateway. I tried "append_branch", but was not able to put in the AVP (phonenumber), rewrite host/port points directly to the gateway. I have no troubles to load the AVP, only to make is possible, that the SIP Client and PSTN Phone are ringing in parallel. Some suggestions ? regards, Andreas

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RE: [Serusers] RTP proxy between two subnetworks with private @s
by Jose Soler 14 Jul '06

14 Jul '06

Hi Joao, No I was not able to solve the issue. It seems (this is my guess tough) that the Portaone RTP proxy assumes that it has one public IP adress, so the valid configuration to use it is Public Nt-Private Nt. I was not able to make it work in other configurations (neiher I got feedback from Portaone to do so). Nevertheless the code is available, so it could be modified...as long as you have the time and will to do so. I did not ;). Best regards, josé -----Original Message----- From: Joao Pereira [mailto:joao.pereira@fccn.pt] Sent: 19. oktober 2005 20:17 To: Jose Soler; serusers(a)lists.iptel.org Subject: Re: [Serusers] RTP proxy between two subnetworks with private @s Hello, did you made it to put the clients of networks A and B to call each other? I want to do the same, and tried a lot of SER/RTPproxy configurations, including the one in: /ser-0.9.0/modules/nathelper/examples/alg.cfg and also tried to run rtpproxy with the "-l 10.0.0.135/193.136.2.2" option. But I just was able to ring the phones (wen calling between networks), but the RTP doesnt pass... If you found the solution, please tell me. Thanks Joao Pereia www.fccn.pt Jose Soler wrote: > Hi, > > I am trying to figure out how to solve the follwoing problem. I have > two subnetworks, A and B, with different private ip adressing schemes > (IP@A <mailto:IP@A>) and (IP@B <mailto:IP@B>). > > SER is installed in a computer with network interfaces towards both > subnetworks. > SER's SIP signalling proxying operation works properly within the > subnetworks and when trying to set up a communication between users in > A and B. But in that last case, obviously there is no media at all > circulating among the subnetworks. > > Portaone's RTP proxy has been installed and configured in the computer > with interfaces towards both subnetworks where SER is installed. > > I am trying to configure SER so that, based on the nathelper module, > when communication between both subnetworks occurs, the RTP proxy is > involved and the communication (also media and not only signalling) is > possible. BUT I am making something wrong, becouse it does not work ... > > Can anyone give me a hand /hint? > Thanks a lot in advance / in any case. > > My SER config file is the following: > > > # > > # ----------- global configuration parameters ------------------------ > > /* 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) > > fifo="/tmp/ser_fifo" > > fifo_mode=0662 > > alias=wirelessip.x.x.x > > alias=sip..x.x.x > > alias=x.x.x > > log_stderror=no > > debug=3 > > children=3 > > mhomed=1 > > # ------------------ module loading ---------------------------------- > > # Uncomment this if you want to use SQL database > > loadmodule "/lib/ser/modules/mysql.so" > > loadmodule "/lib/ser/modules/sl.so" > > loadmodule "/lib/ser/modules/tm.so" > > loadmodule "/lib/ser/modules/rr.so" > > loadmodule "/lib/ser/modules/maxfwd.so" > > loadmodule "/lib/ser/modules/usrloc.so" > > loadmodule "/lib/ser/modules/textops.so" > > loadmodule "/lib/ser/modules/registrar.so" > > # Uncomment this if you want digest authentication > > # mysql.so must be loaded ! > > loadmodule "/lib/ser/modules/auth.so" > > loadmodule "/lib/ser/modules/auth_db.so" > > # For NAT support / media proxying > > loadmodule "/lib/ser/modules/nathelper.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) > > # For NAT > > # We will use 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; > > }; > > lookup("aliases"); > > # 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("com.dtu.dk", "subscriber")) { > > www_challenge("com.dtu.dk", "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(); # I force everything through the 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(); > > }; > > } > > > > > > > > >----------------------------------------------------------------------- >- > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] SER + B2BUA
by roger leszczynski 26 Jun '06

26 Jun '06

Anyone here use ser + b2bua for billing reasons? if so can you provide the config files? ..... i want to be able to capture cdrs even if i restart ser.

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[Serusers] Modifying "From" header field of an INVITE message
by scm-j＠nuntius.com 04 May '06

04 May '06

I am relatively new to SER. My SIP service provider requires me to present the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows: From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx> They want it in E.164 format as follows: From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx> I see function only to modify the SIP URI but not the other header or SDP fields. Could someone help me with a code snippet that I can use in ser.cfg to do this conversion? Regards, SCM

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RE: [Serusers] loose_route's bug?
by Vitaly Nikolaev 30 Apr '06

30 Apr '06

Cant help without config (did u do record_route() before relying INVITE?) ________________________________ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ramin Nikaeen Sent: Thursday, March 09, 2006 2:17 PM To: Serusers Subject: [Serusers] loose_route's bug? Hi Everyone, I am really stuck here. I really need help! I receive the following ACK request in SER from a Linksys/PAP user agent that SER should forward to the destination User Agent: 2010000002. ACK sip:2010000002@10.1.10.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.51:5060;branch=z9hG4bK-5fb5b16;rport From: 2224440133 <sip:2224440133@host.domain.com>;tag=692c5bafde7b5655o0 To: <sip:2224449999@host.domain.com>;tag=3b968d2d Call-ID: c813c237-8673e16d(a)172.16.15.51 CSeq: 101 ACK Max-Forwards: 70 Route: <sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr>, <sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr> Contact: 2224440133 <sip:2224440133@172.16.15.51:5060> User-Agent: Linksys/PAP2-3.1.5(LS) Content-Length: 0 But SER's loose_route incorrectly modifies this ACK as follows: ACK sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr SIP/2.0 Record-Route: <sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr=on> Via: SIP/2.0/UDP 10.1.10.65;branch=0 Via: SIP/2.0/UDP 172.16.15.51:5060;branch=z9hG4bK-5fb5b16;rport=5060 From: 2224440133 <sip:2224440133@host.domain.com>;tag=692c5bafde7b5655o0 To: <sip:2224449999@host.domain.com>;tag=3b968d2d Call-ID: c813c237-8673e16d(a)172.16.15.51 CSeq: 101 ACK Max-Forwards: 16 Route: Contact: 2224440133 <sip:2224440133@172.16.15.51:5060> User-Agent: Linksys/PAP2-3.1.5(LS) Content-Length: 0 Which causes the ACK to keep looping through SER: ACK sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr SIP/2.0 Record-Route: <sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr=on> Record-Route: <sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr=on> Via: SIP/2.0/UDP 10.1.10.65;branch=0 Via: SIP/2.0/UDP 10.1.10.65;branch=0 Via: SIP/2.0/UDP 172.16.15.51:5060;branch=z9hG4bK-5fb5b16;rport=5060 From: 2224440133 <sip:2224440133@host.domain.com>;tag=692c5bafde7b5655o0 To: <sip:2224449999@host.domain.com>;tag=3b968d2d Call-ID: c813c237-8673e16d(a)172.16.15.51 CSeq: 101 ACK Max-Forwards: 15 Route: Contact: 2224440133 <sip:2224440133@172.16.15.51:5060> User-Agent: Linksys/PAP2-3.1.5(LS) Content-Length: 0 ACK sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr SIP/2.0 Record-Route: <sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr=on> Record-Route: <sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr=on> Record-Route: <sip:10.1.10.65;ftag=692c5bafde7b5655o0;lr=on> Via: SIP/2.0/UDP 10.1.10.65;branch=0 Via: SIP/2.0/UDP 10.1.10.65;branch=0 Via: SIP/2.0/UDP 10.1.10.65;branch=0 Via: SIP/2.0/UDP 172.16.15.51:5060;branch=z9hG4bK-5fb5b16;rport=5060 From: 2224440133 <sip:2224440133@host.domain.com>;tag=692c5bafde7b5655o0 To: <sip:2224449999@host.domain.com>;tag=3b968d2d Call-ID: c813c237-8673e16d(a)172.16.15.51 CSeq: 101 ACK Max-Forwards: 14 Route: Contact: 2224440133 <sip:2224440133@172.16.15.51:5060> User-Agent: Linksys/PAP2-3.1.5(LS) Content-Length: 0 I am really stuck. Can anyone tell me what is happening?! Thanks ramin

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[Users] cancel processing
by Jose Antonio Garvayo 26 Apr '06

26 Apr '06

*Hi, I am experiencing a problem regarding CANCEL processing, I have the normal config file plus the following (I got that in a previous mail) * if (is_method("CANCEL")) { t_relay(); break; } The problem is that openserver is relaying the CANCEL, but putting a completely new branch tag in the Via header, instead of the same branch than in the INVITE that is to be cancel. I have debug dump, that shows that the CANCEL message is matched with the INVITE, but it still put a new branch in the relayed cancel, and also in the 200 OK response. If somebody could help me it would be great 0(5309) DEBUG: callid: '2ef6f3df-5309(a)10.161.14.110' 0(5309) DEBUG:init_mod_child: CHILD , rank=1, module=usrloc 0(5309) DEBUG:init_mod_child: CHILD , rank=1, module=xlog 0(5309) XLOG: init_child [1] pid [5309] 0(5309) SIP Request: 0(5309) method: <INVITE> 0(5309) uri: <sip:user4@home2.jagarvayo.mot.dev> 0(5309) version: <SIP/2.0> 0(5309) parse_headers: flags=2 0(5309) Found param type 232, <branch> = <z9hG4bK4f9932cde5bad6c36f103563aa96f449>; state=6 0(5309) Found param type 238, <xxxST_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK6f59072ad3fe644fcf851c16d3168bee>; state=6 0(5309) Found param type 238, <xxxIT_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK80752856b61f3aaf3f1d17e57067d067>; state=6 0(5309) Found param type 238, <xxxSO_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK8b46a43071f3848c711cb4bcf4b42335>; state=6 0(5309) Found param type 238, <xxxPO_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) end of header reached, state=5 0(5309) parse_headers: Via found, flags=2 0(5309) parse_headers: this is the first via 0(5309) After parse_msg... 0(5309) preparing to run routing scripts... 0(5309) parse_headers: flags=100 0(5309) DEBUG:parse_to:end of header reached, state=9 0(5309) DEBUG: get_hdr_field: <To> [37]; uri=[ sip:user4@home2.jagarvayo.mot.dev] 0(5309) DEBUG: to body [<sip:user4@home2.jagarvayo.mot.dev> ] 0(5309) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(5309) DEBUG: get_hdr_body : content_length=95 0(5309) DEBUG:maxfwd:is_maxfwd_present: value = 66 0(5309) DEBUG: add_param: tag=6661 0(5309) DEBUG:parse_to:end of header reached, state=29 0(5309) xl_printf: final buffer length 112 0(5309) INVITE: 10.161.14.59:5060 ci=1-26284(a)10.161.14.105 ft=6661 tt=<null> cseq=1 sip:user4@home2.jagarvayo.mot.dev: 0(5309) parse_headers: flags=200 0(5309) is_preloaded: Yes 0(5309) grep_sock_info - checking if host==us: 13==13 && [10.161.14.110] == [10.161.14.110] 0(5309) grep_sock_info - checking if port 7000 matches port 7000 0(5309) after_loose: Topmost route URI: 'sip:10.161.14.110:7000' is me 0(5309) after_loose: URI to be processed: ' sip:csm002_c638dcee898a6df9b8a460a45e020ce4@10.161.14.59;lr' 0(5309) after_loose: Next URI is a loose router 0(5309) xl_printf: final buffer length 78 0(5309) Relying to Route: sip:csm002_c638dcee898a6df9b8a460a45e020ce4@10.161.14.59;lr 0(5309) parse_headers: flags=ffffffffffffffff 0(5309) found end of header 0(5309) DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffff 0(5309) parse_headers: flags=ffffffffffffffff 0(5309) parse_headers: flags=78 0(5309) t_lookup_request: start searching: hash=61655, isACK=0 0(5309) DEBUG: RFC3261 transaction matching failed 0(5309) DEBUG: t_lookup_request: no transaction found 0(5309) parse_headers: flags=78 0(5309) DEBUG: MD5 calculated: cf43897e3c46f1bb8c4ab6702426d007 0(5309) DEBUG:tm:t_relay: new INVITE 0(5309) parse_headers: flags=ffffffffffffffff 0(5309) check_via_address(10.161.14.59, 10.161.14.59, 0) 0(5309) WARNING:vqm_resize: resize(0) called 0(5309) DEBUG:tm:_reply_light: reply sent out. buf=0x8137e38: SIP/2.0 1..., shmem=0x40357658: SIP/2.0 1 0(5309) DEBUG:tm:_reply_light: finished 0(5309) DEBUG: mk_proxy: doing DNS lookup... 0(5309) check_via_address(10.161.14.59, 10.161.14.59, 0) 0(5309) DEBUG: add_to_tail_of_timer[4]: 0x40355bf4 0(5309) DEBUG: add_to_tail_of_timer[0]: 0x40355c04 0(5309) SER: new transaction fwd'ed 0(5309) DEBUG:destroy_avp_list: destroying list (nil) 0(5309) receive_msg: cleaning up 0(5309) SIP Reply (status): 0(5309) version: <SIP/2.0> 0(5309) status: <100> 0(5309) reason: <Trying> 0(5309) parse_headers: flags=2 0(5309) DEBUG:parse_to:end of header reached, state=9 0(5309) DEBUG: get_hdr_field: <To> [37]; uri=[ sip:user4@home2.jagarvayo.mot.dev] 0(5309) DEBUG: to body [<sip:user4@home2.jagarvayo.mot.dev> ] 0(5309) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(5309) Found param type 232, <branch> = < z9hG4bK7d0f.cf43897e3c46f1bb8c4ab6702426d007.0>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK4f9932cde5bad6c36f103563aa96f449>; state=6 0(5309) Found param type 238, <xxxST_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK6f59072ad3fe644fcf851c16d3168bee>; state=6 0(5309) Found param type 238, <xxxIT_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK80752856b61f3aaf3f1d17e57067d067>; state=6 0(5309) Found param type 238, <xxxSO_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK8b46a43071f3848c711cb4bcf4b42335>; state=6 0(5309) Found param type 238, <xxxPO_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) end of header reached, state=5 0(5309) parse_headers: Via found, flags=2 0(5309) parse_headers: this is the first via 0(5309) After parse_msg... 0(5309) forward_reply: found module tm, passing reply to it 0(5309) DEBUG: t_check: msg id=2 global id=1 T start=0xffffffff 0(5309) parse_headers: flags=22 0(5309) parse_headers: flags=8 0(5309) DEBUG: t_reply_matching: hash 61655 label 0 branch 0 0(5309) DEBUG: t_reply_matching: reply matched (T=0x40355ae0)! 0(5309) DEBUG: t_check: msg id=2 global id=2 T end=0x40355ae0 0(5309) DEBUG:tm:reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) 0(5309) DEBUG:tm:t_should_relay_response: T_code=100, new_code=100 0(5309) DEBUG:tm:relay_reply: branch=0, save=0, relay=-1 0(5309) DEBUG: add_to_tail_of_timer[1]: 0x40355c04 0(5309) DEBUG:destroy_avp_list: destroying list (nil) 0(5309) receive_msg: cleaning up 0(5309) SIP Reply (status): 0(5309) version: <SIP/2.0> 0(5309) status: <183> 0(5309) reason: <Session Progress> 0(5309) parse_headers: flags=2 0(5309) DEBUG: add_param: tag=9991 0(5309) DEBUG:parse_to:end of header reached, state=29 0(5309) DEBUG: get_hdr_field: <To> [46]; uri=[ sip:user4@home2.jagarvayo.mot.dev] 0(5309) DEBUG: to body [<sip:user4@home2.jagarvayo.mot.dev>] 0(5309) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(5309) Found param type 232, <branch> = < z9hG4bK7d0f.cf43897e3c46f1bb8c4ab6702426d007.0>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK4f9932cde5bad6c36f103563aa96f449>; state=6 0(5309) Found param type 238, <xxxST_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK6f59072ad3fe644fcf851c16d3168bee>; state=6 0(5309) Found param type 238, <xxxIT_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK80752856b61f3aaf3f1d17e57067d067>; state=6 0(5309) Found param type 238, <xxxSO_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK8b46a43071f3848c711cb4bcf4b42335>; state=6 0(5309) Found param type 238, <xxxPO_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) end of header reached, state=5 0(5309) parse_headers: Via found, flags=2 0(5309) parse_headers: this is the first via 0(5309) After parse_msg... 0(5309) forward_reply: found module tm, passing reply to it 0(5309) DEBUG: t_check: msg id=3 global id=2 T start=0xffffffff 0(5309) parse_headers: flags=22 0(5309) parse_headers: flags=8 0(5309) DEBUG: t_reply_matching: hash 61655 label 0 branch 0 0(5309) DEBUG: t_reply_matching: reply matched (T=0x40355ae0)! 0(5309) DEBUG: t_check: msg id=3 global id=3 T end=0x40355ae0 0(5309) DEBUG:tm:reply_received: org. status uas=100, uac[0]=100 local=0 is_invite=1) 0(5309) DEBUG:tm:t_should_relay_response: T_code=100, new_code=183 0(5309) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(5309) old size: 1120, new size: 1035 0(5309) build_res_from_sip_res: copied size: orig:549, new: 464, rest: 571 msg= SIP/2.0 183 Session Progress From: <sip:user3@home2.jagarvayo.mot.dev>;tag=6661 To: <sip:user4@home2.jagarvayo.mot.dev>;tag=9991 Call-ID: 1-26284(a)10.161.14.105 CSeq: 1 INVITE Contact: <sip:10.161.14.105:6020> Content-Type: application/sdp Record-Route: <sip:xxxPT_002@10.161.14.59;lr>,<sip:xxxST_002@10.161.14.59 ;lr>,<sip:10.161.14.110:7000;ftag=6661;lr>,<sip:xxxST_002@10.161.14.59;lr>,< sip:xxxSO_002@10.161.14.59;lr>,<sip:xxxPO_002@10.161.14.59;lr> Via: SIP/2.0/UDP 10.161.14.59;branch=z9hG4bK4f9932cde5bad6c36f103563aa96f449;xxxST_002,SIP/2.0/UDP 10.161.14.59;branch=z9hG4bK6f59072ad3fe644fcf851c16d3168bee;xxxIT_002,SIP/2.0/UDP 10.161.14.59;branch=z9hG4bK80752856b61f3aaf3f1d17e57067d067;xxxSO_002,SIP/2.0/UDP 10.161.14.59;branch=z9hG4bK8b46a43071f3848c711cb4bcf4b42335;xxxPO_002,SIP/2.0/UDP 10.161.14.105:6020 P-Charging-Vector: icid-value= 43d9d426:f1135:7e37002e@pcscf.home2.jagarvayo.mot.dev;term-ioi=ims.XXX.com v=0 o=srini - 0 IN IP4 - s=SIPtest - - c=IN IP4 10.161.14.59 t=0 0 m=audio 2410 RTP/AVP 0 0(5309) DEBUG:tm:relay_reply: sent buf=0x8138178: SIP/2.0 1..., shmem=0x40357f90: SIP/2.0 1 0(5309) DEBUG: add_to_tail_of_timer[1]: 0x40355c04 0(5309) DEBUG:destroy_avp_list: destroying list (nil) 0(5309) receive_msg: cleaning up 1(5311) DEBUG: timer routine:4,tl=0x40355bf4 next=(nil) 0(5309) SIP Reply (status): 0(5309) version: <SIP/2.0> 0(5309) status: <180> 0(5309) reason: <Ringing> 0(5309) parse_headers: flags=2 0(5309) DEBUG: add_param: tag=9991 0(5309) DEBUG:parse_to:end of header reached, state=29 0(5309) DEBUG: get_hdr_field: <To> [46]; uri=[ sip:user4@home2.jagarvayo.mot.dev] 0(5309) DEBUG: to body [<sip:user4@home2.jagarvayo.mot.dev>] 0(5309) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(5309) DEBUG: get_hdr_body : content_length=0 0(5309) Found param type 232, <branch> = < z9hG4bK7d0f.cf43897e3c46f1bb8c4ab6702426d007.0>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK4f9932cde5bad6c36f103563aa96f449>; state=6 0(5309) Found param type 238, <xxxST_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK6f59072ad3fe644fcf851c16d3168bee>; state=6 0(5309) Found param type 238, <xxxIT_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK80752856b61f3aaf3f1d17e57067d067>; state=6 0(5309) Found param type 238, <xxxSO_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) Found param type 232, <branch> = <z9hG4bK8b46a43071f3848c711cb4bcf4b42335>; state=6 0(5309) Found param type 238, <xxxPO_002> = <n/a>; state=9 0(5309) parse_via: next_via 0(5309) end of header reached, state=5 0(5309) parse_headers: Via found, flags=2 0(5309) parse_headers: this is the first via 0(5309) After parse_msg... 0(5309) forward_reply: found module tm, passing reply to it 0(5309) DEBUG: t_check: msg id=4 global id=3 T start=0xffffffff 0(5309) parse_headers: flags=22 0(5309) parse_headers: flags=8 0(5309) DEBUG: t_reply_matching: hash 61655 label 0 branch 0 0(5309) DEBUG: t_reply_matching: reply matched (T=0x40355ae0)! 0(5309) DEBUG: t_check: msg id=4 global id=4 T end=0x40355ae0 0(5309) DEBUG:tm:reply_received: org. status uas=183, uac[0]=183 local=0 is_invite=1) 0(5309) DEBUG:tm:t_should_relay_response: T_code=183, new_code=180 0(5309) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(5309) old size: 1001, new size: 916 0(5309) build_res_from_sip_res: copied size: orig:528, new: 443, rest: 473 msg= SIP/2.0 180 Ringing From: <sip:user3@home2.jagarvayo.mot.dev>;tag=6661 To: <sip:user4@home2.jagarvayo.mot.dev>;tag=9991 Call-ID: 1-26284(a)10.161.14.105 CSeq: 1 INVITE Contact: <sip:10.161.14.105:6020> Content-Length: 0 Record-Route: <sip:xxxPT_002@10.161.14.59;lr>,<sip:xxxST_002@10.161.14.59 ;lr>,<sip:10.161.14.110:7000;ftag=6661;lr>,<sip:xxxST_002@10.161.14.59;lr>,< sip:xxxSO_002@10.161.14.59;lr>,<sip:xxxPO_002@10.161.14.59;lr> Via: SIP/2.0/UDP 10.161.14.59;branch=z9hG4bK4f9932cde5bad6c36f103563aa96f449;xxxST_002,SIP/2.0/UDP 10.161.14.59;branch=z9hG4bK6f59072ad3fe644fcf851c16d3168bee;xxxIT_002,SIP/2.0/UDP 10.161.14.59;branch=z9hG4bK80752856b61f3aaf3f1d17e57067d067;xxxSO_002,SIP/2.0/UDP 10.161.14.59;branch=z9hG4bK8b46a43071f3848c711cb4bcf4b42335;xxxPO_002,SIP/2.0/UDP 10.161.14.105:6020 P-Charging-Vector: icid-value= 43d9d426:f1135:7e37002e@pcscf.home2.jagarvayo.mot.dev;term-ioi=ims.XXX.com 0(5309) DEBUG:tm:relay_reply: sent buf=0x8138178: SIP/2.0 1..., shmem=0x40357f90: SIP/2.0 1 0(5309) DEBUG: add_to_tail_of_timer[1]: 0x40355c04 0(5309) DEBUG:destroy_avp_list: destroying list (nil) 0(5309) receive_msg: cleaning up 0(5309) SIP Request: 0(5309) method: <CANCEL> 0(5309) uri: <sip:10.161.14.104:6030> 0(5309) version: <SIP/2.0> 0(5309) parse_headers: flags=2 0(5309) Found param type 232, <branch> = <z9hG4bK4f9932cde5bad6c36f103563aa96f449>; state=6 0(5309) Found param type 238, <xxxST_002> = <n/a>; state=17 0(5309) end of header reached, state=5 0(5309) parse_headers: Via found, flags=2 0(5309) parse_headers: this is the first via 0(5309) After parse_msg... 0(5309) preparing to run routing scripts... 0(5309) parse_headers: flags=100 0(5309) DEBUG:parse_to:end of header reached, state=9 0(5309) DEBUG: get_hdr_field: <To> [37]; uri=[ sip:user4@home2.jagarvayo.mot.dev] 0(5309) DEBUG: to body [<sip:user4@home2.jagarvayo.mot.dev> ] 0(5309) get_hdr_field: cseq <CSeq>: <1> <CANCEL> 0(5309) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(5309) DEBUG: add_param: tag=6661 0(5309) DEBUG:parse_to:end of header reached, state=29 0(5309) xl_printf: final buffer length 101 0(5309) CANCEL: 10.161.14.59:5060 ci=1-26284(a)10.161.14.105 ft=6661 tt=<null> cseq=1 sip:10.161.14.104:6030: 0(5309) parse_headers: flags=200 0(5309) DEBUG: get_hdr_body : content_length=0 0(5309) found end of header 0(5309) find_first_route: No Route headers found 0(5309) loose_route: There is no Route HF 0(5309) DEBUG: t_newtran: msg id=5 , global msg id=4 , T on entrance=0xffffffff 0(5309) parse_headers: flags=ffffffffffffffff 0(5309) parse_headers: flags=78 0(5309) t_lookup_request: start searching: hash=61655, isACK=0 0(5309) DEBUG: RFC3261 transaction matching failed 0(5309) DEBUG: t_lookup_request: no transaction found 0(5309) parse_headers: flags=78 0(5309) DEBUG: MD5 calculated: 1b7635cf6840ddd15282505b57290b27 0(5309) DEBUG: t_lookupOriginalT: searching on hash entry 61655 0(5309) DEBUG: RFC3261 transaction matched, tid=4f9932cde5bad6c36f103563aa96f449 0(5309) DEBUG: t_lookupOriginalT: canceled transaction found (0x40355ae0)! 0(5309) DEBUG: t_lookupOriginalT completed 0(5309) check_via_address(10.161.14.59, 10.161.14.59, 0) 0(5309) DEBUG: add_to_tail_of_timer[4]: 0x4035855c 0(5309) DEBUG: add_to_tail_of_timer[0]: 0x4035856c 0(5309) DEBUG: e2e_cancel: e2e cancel proceeding 0(5309) parse_headers: flags=ffffffffffffffff 0(5309) check_via_address(10.161.14.59, 10.161.14.59, 0) 0(5309) WARNING:vqm_resize: resize(0) called 0(5309) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(5309) DEBUG: add_to_tail_of_timer[2]: 0x40358490 0(5309) DEBUG:tm:_reply_light: reply sent out. buf=0x8138320: SIP/2.0 2..., shmem=0x40359558: SIP/2.0 2 0(5309) DEBUG:tm:_reply_light: finished 0(5309) SER: new transaction fwd'ed 0(5309) DEBUG:destroy_avp_list: destroying list (nil) 0(5309) receive_msg: cleaning up 0(5309) SIP Reply (status): 0(5309) version: <SIP/2.0> 0(5309) status: <481> 0(5309) reason: <Call Leg/Transaction Does Not Exist> 0(5309) parse_headers: flags=2 0(5309) DEBUG:parse_to:end of header reached, state=9 0(5309) DEBUG: get_hdr_field: <To> [37]; uri=[ sip:user4@home2.jagarvayo.mot.dev] 0(5309) DEBUG: to body [<sip:user4@home2.jagarvayo.mot.dev> ]

4 9

[Serusers] ERROR message : MySQL server has gone away
by "한승전" 13 Apr '06

13 Apr '06

please help..... If i restart MySQL(version 5.0.19), SER shows next error message. Mar 31 22:37:18 localhost /usr/local/sbin/ser[32701]: submit_query(): MySQL server has gone away Mar 31 22:37:18 localhost /usr/local/sbin/ser[32701]: submit_query(): Error while submitting query Mar 31 22:37:18 localhost /usr/local/sbin/ser[32701]: get_ha1(): Error while querying database Mar 31 22:37:22 localhost /usr/local/sbin/ser[32703]: Binding '1234(a)192.9.2.1','sip:1234@192.9.2.11:5060' has expired As appear in upside message, SER does not auto-reconnect to mysql. If MySQL restarted, SER must be able to do Auto-Reconnect. Is no there way that SER can do Auto-Reconnect to MySQL?

4 4

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sr-users March 2006