November 2008 - sr-users

[Serusers] Fw: SER COLP & COLR implementation
by Erman ÇAKMAK 13 Nov '08

13 Nov '08

Hello all, I could not find any information about SER whether it supports the COLP & COLR call features or not. Does anyone have an idea about it? Regards, Erman Cakmak

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[Kamailio-Users] 488 Not acceptable here
by Ricky Gutierrez 13 Nov '08

13 Nov '08

Hi list, I am giving him returned script from openser trying to find the problem to this message " 488 acceptable Not here" , I have my openser integrated recently with asterisk and the voicemail recently adds rtpproxy to solve problems of nat and was almost a success, some details to improve, but after adding the rtpproxy when I call to UAC and it does not answer the call on the telephone shows a 488 message to me No Aceptable, it does not jump to the voicemai.. I have 2 networks cards in my server, and looking at the sip log, I see twice in the sdp the interfaz it public any idea as solving this problem? regards rickygm # U +1.675624 192.168.10.1:5060 -> 192.168.10.1:5070 INVITE sip:u120@192.168.10.1:5070 SIP/2.0 Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663> Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1> Contact: <sip:119@192.168.10.28:5060;nat=yes;nat=yes> Supported: replaces, timer, path Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Grandstream GXV3000 1.1.3.14 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 604 P-hint: inbound->inbound P-hint: Route[20]: Rtpproxy P-hint: Route[20]: Rtpproxy v=0 o=119 8000 8001 IN IP4 192.168.10.28 s=SIP Call c=IN IP4 192.168.1.64192.168.1.64 t=0 0 m=audio 3504835048 RTP/AVP 18 4 3 2 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 3505035050 RTP/AVP 99 34 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4CXID= a=rtpmap:34 H263/90000 a=fmtp:34 CIF=2 MaxBR=1280 a=framerate:20 a=nortpproxy:yes a=nortpproxy:yes # U +0.000052 192.168.10.1:5060 -> 192.168.10.27:5060 CANCEL sip:120@192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1> CSeq: 12216 CANCEL Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000411 192.168.10.1:5070 -> 192.168.10.1:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1;received=192.168.10.1 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U +0.000120 192.168.10.1:5060 -> 192.168.10.1:5070 ACK sip:u120@192.168.10.1:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1>;tag=as4f1165d5 CSeq: 12216 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000144 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.000787 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 CANCEL User-Agent: Grandstream GXV3000 1.1.3.14 Supported: replaces, timer, 100rel, path Content-Length: 0 # U +0.000544 192.168.10.27:5060 -> 192.168.10.1:5060 SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663> From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Grandstream GXV3000 1.1.3.14 Content-Length: 0 # U +0.000067 192.168.10.1:5060 -> 192.168.10.27:5060 ACK sip:120@192.168.10.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 Call-ID: 7c55e4667d473f76(a)192.168.10.28 To: <sip:120@192.168.10.1>;tag=fb538483d0333de1 CSeq: 12216 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.001390 192.168.10.28:5060 -> 192.168.10.1:5060 ACK sip:120@192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Contact: <sip:119@192.168.10.28:5060> Proxy-Authorization: Digest username="119", realm="192.168.10.1", algorithm=MD5, uri="sip:120@192.168.10.1", nonce="4919edaf0a873106bfd0b18f347709360f00f7eb", response="a5a26c257b491718def2b43488a39853" Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 ACK User-Agent: Grandstream GXV3000 1.1.3.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.497023 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact # U +0.999813 192.168.10.1:5060 -> 192.168.10.28:5060 SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3 From: <sip:119@192.168.10.1>;tag=69372fe200a41663 To: <sip:120@192.168.10.1>;tag=as4f1165d5 Call-ID: 7c55e4667d473f76(a)192.168.10.28 CSeq: 12216 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 P-hint: Onreply-route - fixcontact exit 67 received, 0 dropped

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[Kamailio-Users] issues with rtpproxy and kamailio
by Fabian Borot 13 Nov '08

13 Nov '08

I am trying to set up rtpproxy with kamailio and are having these 2 issues: 1- the BYEs are not relayed to the other party, I have to hang up the call at both ends 2- kamailio receives the INVITE from the calling party [behind a nat] and relays it to the called party but the connection information in the SDP has to IPs [the source IP and the RTPproxy IP together c=IN IP4 204.9.239.6765.111.185.187 user 100 is behind a NAT and user 101 is not but I want to force the RTPs to go thru the rtpproxy server [ in the same IP as kamailio] the good and weird thing is that there is audio both ways. this is my config file: if (loose_route()) { # mark routing logic in request xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("1")){ xlog("L_INFO","mylog: REGISTER-Device behind NAT.\n"); fix_nated_register(); } force_rport(); save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } else if (method=="INVITE") { xlog("L_INFO","mylog: Processing INVITE.\n"); if (!proxy_authorize("","subscriber")) { xlog("L_INFO","mylog: INVITE came without auth, sending challenge.\n"); proxy_challenge("","0"); exit; }; consume_credentials(); if (nat_uac_test("1")){ xlog("L_INFO","mylog: INVITE-Device behind NAT.\n"); #fix_nated_contact(); fix_nated_sdp("10"); } force_rtp_proxy(); if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP t_on_reply("1"); if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } onreply_route[1] { force_rtp_proxy("","65.111.185.187"); } thank you in advance Fabian _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Kamailio-Users] problem with fix_nated_register() function
by Fabian Borot 13 Nov '08

13 Nov '08

Hello, I am having probs with the fix_nated_register() function, I still see the private IP, even thought the nat_uac_test(31) shows the user behind a NAT This is the REGISTER section in kamailio.cfg: if (method=="REGISTER") { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; if (nat_uac_test("31")){ xlog("L_INFO","mylog: Device behind NAT.\n"); fix_nated_register(); force_rport(); } save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; this is the debug log [level 6], it shows the via with a public IP Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: mylog: starting to process REGISTER.Info: [100]. Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_nonce: comparing [491a03a9000000041ba5231fd93870c44415f01d1a0d1963] and [491a03a9000000041ba5231fd93870c44415f01d1a0d1963] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_new_result: allocate 28 bytes for result set at 0x81ab8a0 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: 1 columns returned from the query Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_allocate_columns: allocate 4 bytes for result names at 0x81a7288 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_allocate_columns: allocate 4 bytes for result types at 0x81a7258 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[0] at 0x81a7268 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x81a7268)[0]=[password] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_convert_rows: allocate 8 bytes for rows at 0x81a7278 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_convert_row: allocate 20 bytes for row values at 0x81ab880 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:db_mysql:db_mysql_str2val: converting STRING [100] Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth_db:get_ha1: HA1 string calculated: e8f9a8a478e49a182e9cceeb3e43277f Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_response: our result = '84a320db7e04e3ddd3ef84a5173187cd' Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:check_response: authorization is OK Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:auth:post_auth: nonce index= 4 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing 1 columns Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x81a7268 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing result names at 0x81a7288 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_columns: freeing result types at 0x81a7258 Nov 11 17:09:04 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_rows: freeing 1 rows Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_row: freeing row values at 0x81ab880 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_rows: freeing rows at 0x81a7278 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:db_free_result: freeing result set at 0x81ab8a0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:check_via_address: params 204.9.239.67, 192.168.1.108, 0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: mylog: Device behind NAT. Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=8000000 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:registrar:build_contact: created Contact HF: Contact: <sip:100@192.168.1.108 7566;rinstance=89990b40ab560825>;expires=3341, <sip:100@192.168.1.108:44568;rinstance=f0942ae1a596ea83>;expires=3597 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:check_via_address: params 204.9.239.67, 192.168.1.108, 0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: mylog: save-location successful. Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:destroy_avp_list: destroying list 0xb60e11c0 Nov 11 17:09:05 sp1094a ../../sbin/kamailio[14219]: DBG:core:receive_msg: cleaning up and this is how the AOR is stored in the location db [ private IP] [root@sp1094a kamailio]# ../../sbin/kamctl ul show database engine 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd ul_dump Domain:: location table=512 records=1 max_slot=1 AOR:: 100 Contact:: sip:100@192.168.1.108:44568;rinstance=f0942ae1a596ea83 Q= Expires:: 3365 Callid:: 7a32d234b04d4b69M2IzODRhODkwM2M5NTM0MzUwMGM4ODY3NTgzYzFjMmE. Cseq:: 2 User-agent:: eyeBeam release 1003l stamp 30937 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:65.111.185.187:5060 Methods:: 5951 FIFO command was: :ul_dump:openser_receiver_14270 [root@sp1094a kamailio]# Am I missing something? Thanks in advance Fabian _________________________________________________________________ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_11…

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[Kamailio-Users] Tomcat won't start serMyAdmin - SEVERE: Error listenerStart
by rjreb 13 Nov '08

13 Nov '08

I'm trying to install serMyAdmin but it refuses to start. In the logs I see the following: Nov 12, 2008 2:25:31 AM org.apache.catalina.startup.HostConfig deployWAR INFO: Deploying web application archive serMyAdmin.war Nov 12, 2008 2:25:34 AM org.apache.catalina.core.StandardContext start SEVERE: Error listenerStart Nov 12, 2008 2:25:34 AM org.apache.catalina.core.StandardContext start SEVERE: Context [/serMyAdmin] startup failed due to previous errors I've been searching for this for a day now and I can't find where anyone else is having this problem. Any suggestions? Debian Etch Sun Java 1.5.0-16-3 Tomcat 6.0.16 -- View this message in context: http://www.nabble.com/Tomcat-won%27t-start-serMyAdmin---SEVERE%3A-Error-lis… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] Is msilo module necessary to make IM work in SER? SER doesn't forward MESSAGE
by lin9k＠yahoo.com 12 Nov '08

12 Nov '08

Hi, I used two IM (SIMPLE based) clients to communicate through SER. However, I could only see clients kept sending MESSAGE to SER but SER never forwarded them to the peer. I was not using msilo. I am not sure if that's the cause that SER didn't forward to the other end. Or, do I need any special configuration in ser.cfg, or any different version of SER? I was using ser-0.9.6. Thanks in advance! Ling

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[Serusers] compiling ser from source in ubuntu
by amar mahmoud 12 Nov '08

12 Nov '08

Hi, when i compile SER from the debian packages embedded with ubuntu it works fine, but i want to compile from the source because i want to add one module i have invented. my problem is : when i compile from source mysql module does not get compiled, i have tried to change the Makefile and delete mysql module from those excluded modules, also i have tried to compile it alone with the command: make modules=modules/mysql modules but still there is problem. does any one met that before. thanks Amar _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

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[Kamailio-Users] Dialog profile persistence - DB vs. memory.
by Alex Balashov 12 Nov '08

12 Nov '08

Greetings, Is some of the dialog profile information kept in in-memory structures inside the Kamailio instance even when one is using a database? I am using the dialog module in an HA configuration and fail over the database and proxy between two machines. The database and the proxy are both started on the secondary node during failover, and the database uses on-disk data and indices on a shared filesystem. What I am finding is that if I initiate some dialogs on the primary and then trigger the failover, the proxy on the secondary node is not aware of them. I see the entries for them in the database table from the secondary node - they're definitely there, and I am using db_mode 1 (real-time). However, the profile sizes are all showing as 0. I have also tried it with different match_modes with the same results whether I am using the RR cookie or SIP attributes purely. It's rather important that I get this working because I require dialog tracking for secure loose-routing of subsequent (non-initial) in-dialog requests and for concurrent call limit control. I was imagining that storing all the dialog info in the database would create the necessary persistence layer for another proxy instance to have the exact same profile information as the first. Is this a bug, or is it supposed to work this way? Cheers, -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] question about register
by Fabian Borot 12 Nov '08

12 Nov '08

Hello I am having an issue handling UN-REGISTERs. This is my config file: route{ xlog("L_INFO", "mylog: starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","mylog: Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_INFO","mylog: Message too big.\n"); exit; }; if (!method=="REGISTER") { xlog("L_INFO","mylog: Recording Route. Method: [$rm].\n"); record_route(); } if (loose_route()) { # mark routing logic in request xlog("L_INFO","mylog: Loose Route section. Method: [$rm].\n"); append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER" && (!search("expires=0"))) { xlog("L_INFO","mylog: starting to process REGISTER.Info: [$au].\n"); if (!www_authorize("", "subscriber")) { xlog("L_INFO","mylog: REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; save("location"); xlog("L_INFO","mylog: save-location successful.\n"); exit; } else if (method=="INVITE") { xlog("L_INFO","mylog: Processing INVITE.\n"); if (!proxy_authorize("","subscriber")) { xlog("L_INFO","mylog: INVITE came without auth, sending challenge.\n"); proxy_challenge("","0"); exit; }; consume_credentials(); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (t_relay()) { xlog("L_INFO","mylog: Route 1 section. Method [$rm]."); } else { sl_reply_error(); xlog("L_INFO","mylog: Route 1 section. T_Relay failed. Method [$rm]."); }; exit; } I added the "&& (!search("expires=0"))" to make sure that only the REGISTER requests will be processed by the next lines and the AOR will be saved in those cases. The X-lite sends "expires=0" in the Contact HF when it is unregistering. This is how the processing of a REGISTER looks like as per my code: Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting to process REGISTER.Info: [<null>]. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: REGISTER came without auth, sending challenge. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: starting to process REGISTER.Info: [101]. Nov 10 12:33:08 sp1094a ../../sbin/kamailio[26069]: mylog: save-location successful. and when the phone UN-REGISTERs this is what I see: Nov 10 12:33:17 sp1094a ../../sbin/kamailio[26069]: mylog: starting_main_logic. Nov 10 12:33:17 sp1094a ../../sbin/kamailio[26069]: mylog: lookup-location failed, sending 404 Not Found. Method: [REGISTER]. My question is, do I need to use the lookup section for REGISTERs and UN-REGISTERs? If I comment out that part this is how the log looks like: #if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # xlog("L_INFO","mylog: lookup-location failed, sending 404 Not Found. Method: [$rm].\n"); # exit; #}; log: the 1st part [REGISTER] looks the same [root@sp1094a kamailio]# tail -f /var/log/kamailio_logs | grep mylog Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting to process REGISTER.Info: [<null>]. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: REGISTER came without auth, sending challenge. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: starting to process REGISTER.Info: [101]. Nov 10 12:36:35 sp1094a ../../sbin/kamailio[26203]: mylog: save-location successful. and this is what the log shows when the phone is UN-REGISTERING Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:42 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Route 1 section. Method [REGISTER]. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: starting_main_logic. Nov 10 12:36:43 sp1094a ../../sbin/kamailio[26203]: mylog: Message too big. [root@sp1094a kamailio]# Regards Fabian _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Kamailio-Users] sip router meeting in Karlsruhe
by Daniel-Constantin Mierla 12 Nov '08

12 Nov '08

Hello, because of taking several days off, a complete report might show up a bit later, apologizes for that. Now a (short) summary... First, many thanks thanks to 1&1 for hosting the event and everyone that participated in such short notice. There were representatives from 10 companies: - 1&1 - FhG Fokus - Telio - Asipto - iptelorg/Tekelec - Voztelecom - Iptego - Itsyscom - Longphone - Basis AudioNet Some pictures should be published in the near future. After going in short introduction and presentation of the goals from the point of view of each project, we focused on: - identification of potential points of conflicts and how to get to a resolution in such case - code integration for common layer of the first phase - future development and proposals of new features - management of the larger eco-system that includes related projects and business entities I will send several emails detailing what was discussed and proposed there in few days. meanwhile, the outlines: - it is hard to avoid conflicts just by some clear and strict rules, so the common sense should lead the collaboration and discussions - GIT repository should be up in several days so the work can start, with a time line of 2-3 months from now to get core and tm in a very good shape of integration - another meeting shall be set in about 3 months time, to allow enough time for people to be able to attend, adjust the development and look more deep at the future. While the a lot of focus in the next months will be on integration, development of new features won't stop -- for examples, steps to a partial asynchronous processing are undertaking, couple of new modules are planned for release, several other modules to bring new functionalities - we should encourage and promote the development of related applications, like web interfaces, management tools, applications servers -- the add value for community and business Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] irc devel meeting summary
by Daniel-Constantin Mierla 11 Nov '08

11 Nov '08

Hello, sorry for delay... but better later than never, so I am trying to summarize the outcome of the IRC devel meeting from last Thursday, Nov 06, 2008. The log is available at: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-minutes-0… Regarding the 1.3.x series, we set the 2oth of November as the date for 1.3.4 release. There are several fixes and this branch is still used in many production environments, so it worths investing time in it. There are some updates in 1.4.x to be ported and some issues to be investigated. For a new 1.4.x release (the 1.4.3) we look for the beginning of December as the time frame for setting the date of the release. The 1.5.x rises in the first phase the question of a shorter time release, as we have lot of new features already implemented, see: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x http://www.kamailio.org/dokuwiki/doku.php/roadmap:1.5.x Unfortunately some of the new features are not completed yet (e.g., htable, carrierroute updates, PV migration) so we will stick to 6-8 months release cycle. The next major one that will use the common layer (core and tm) from sip-router could be done in a shorter time than usual. So, beginning to mid of January should be the time to freeze the code for a release date somewhere in end of January or February. An important aspect approached during the discussions was the release management and testing management. Perhaps we need to appoint a person to manage releasing process in order to have a better synchronization of the developers. As everyone agrees, testing is very important and we try to improve it. There is a testing suite included in the source tree right now, used for static testing. The idea of a dynamic testing and testing community was introduced and hopefully we can attract people and build a group of beta testers. There is a proposal of how to implement and what to take care of, probably this one has to continue on devel and users mailing lists for a proper organizational structure. GIT became as a subject of migrating from SVN, because of its features for a distributed development environment. However, sourceforge does not have such offerings, so it is not possible now anyhow. However the sip-router.org will use it. Regarding the sip-router.org project, we focused on short term goals, acknowledging that we have several modules offering same functionality but relying on different underlaying data structures, this have to co-exists for a longer time. For the first phase, core + tm the overlapping is rather small, the two projects insisting in the past to improve different aspects of them. Unfortunately I forgot to bring up some punctual topics proposed, mainly related to tm failure route calling in case of sending error and reply handling/dropping in onreply_routes. We can continue those subjects on the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Question regarding config file operation
by Siddhardha Garige 11 Nov '08

11 Nov '08

Hello all, I have a basic question regarding config file operation. Here is my understanding. At build time bison and flex work on config file and generates cfg.tab.c. This file is equivalent to c version of config file. Make uses cfg.tab.c along with other files and loads a series of actions into action structures. These actions will be performed on messages in a predefined order. If i make a change to config file, do I have to rebuild openser? But I do see just by stopping and starting these changes get reflected in operation. How does these changes get into action structure with out bison and flex generating cfg.tab.c file and make compiling with other files? Any help in this regard is appreciated. Thanks Sid

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[Serusers] Information request about IPv6
by Mary Allen (mrallen) 11 Nov '08

11 Nov '08

Hi, If somebody is using SERi as a Proxy with IPv6, would you please let me know the modifications/additions needed to have it working with the IPv6 and IPv4. We have three type of trunks to the Proxy (IPv4, IPv6 and Dual). IPv4 and IPv6 are configured for the machine. For the Config file do we just have to add another line for listening on one for the IPv4 and another for the IPv6? and for the routes based on the phone number I will have to entered one for an IPv6. Will that be all? Thanks, Mary

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[Kamailio-Users] OpenSER HA example with vrrp/ucarp?
by sergejf 11 Nov '08

11 Nov '08

We would like to set up two OpenSER 1.4.x servers in a failover configuration. The active (primary) server has two main roles: 1) load balancing SIP INVITEs between SBCs from various carriers using the CarrierRoute module 2) collecting CDRs from all these PSTN calls using the Acc module and storing them in a MySQL database for later billing reconciliation. Right now we're using a F5 Big-IP load balancer to handle the pool of two SIP proxies which receives the SIP INVITEs from our Media Servers and dispatches them to the SBCs. However, should the primary proxy fail, the secondary proxy would never receive (for example) the SIP BYE from an SBC. Solving that issue with the Big-IP seems complicated and a simpler vrrp/ucarp solution looks attractive. Any (hands-on) advice on how to set up vrrp/ucarp to advertise a single virtual IP address, then detect that the active SIP proxy has gone down and switch to the secondary IP, would be most welcome. We're using Centos 5.x (64 bit) on our servers. Thanks a lot. Serge -- View this message in context: http://www.nabble.com/OpenSER-HA-example-with-vrrp-ucarp--tp20443679p204436… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] SIP/2.0 500 Server Internal Error using Mysql
by Nel Tulaylay 11 Nov '08

11 Nov '08

Hi Tech support, I just completed configuring my SER SIP Server, I'm experiencing an intermittent error of "SIP/2.0 500 Server Internal Error" during REGISTER and whenever im using a mysql server for authentication. Most of the time, i can register successfully to the SIP Server but there are times i encounter error "SIP/2.0 500 Server Internal Error". Do you have an idea what's causing this error. I have here a trace log of SIP Messages. I will greatly appreciate any support from you. Thanks Nel U CLIENT-IP:1384 -> SERVER-IP:5060 REGISTER sip:SERVER-IP SIP/2.0..Via: SIP/2.0/UDP 192.168.1.136:5060;br anch=z9hG4bK63235644e9b491928f5..From: 638871680007 <sip:638871680007@116.6 8.141.251>;tag=63283ff..To: 638871680007 <sip:638871680007@SERVER-IP>. .Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Contact: "638871680007" <sip:638871680007@192.168.1.136:5060>..Authorization: Digest username="638871680007", realm="SERVER-IP", nonce="49199956fed351b86a 5c11c83d849a62144c5dcd", uri="sip:SERVER-IP", response="6a29e5839028f9 bc690c77fda9031054", algorithm=MD5..User-Agent: VaxSIPUserAgent/3.0..Expire s: 3600..Max-Forwards: 70..Content-Length: 0.... U SERVER-IP:5060 -> CLIENT-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK63235 644e9b491928f5;received=CLIENT-IP..From: 638871680007 <sip:638871680007 @SERVER-IP>;tag=63283ff..To: 638871680007 <sip:638871680007@116.68.141 .251>..Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Ser ver: Sip EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 SERVER-IP:5060 "Noisy feedback tells: pid=16275 req_src_ip=203.1 89.8.149 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:116.68.141 .251 via_cnt==1".... U SERVER-IP:5060 -> CLIENT-IP:1384 SIP/2.0 500 Server Internal Error..Via: SIP/2.0/UDP 192.168.1.136:5060 ;bran ch=z9hG4bK63235644e9b491928f5;rport=1384;received=CLIENT-IP..From: 6388 71680007 <sip:638871680007@SERVER-IP>;tag=63283ff..To: 638871680007 <s ip:638871680007@SERVER-IP>;tag=04b7610a616ee809532470b3b2dd8840.0e13.. Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Server: Si

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[Serusers] SIP/2.0 500 Server Internal Error using Mysql
by Nel Tulaylay 11 Nov '08

11 Nov '08

Hi Tech support, I just completed configuring my SER SIP Server, I'm experiencing an intermittent error of "SIP/2.0 500 Server Internal Error" during REGISTER and whenever im using a mysql server for authentication. Most of the time, i can register successfully to the SIP Server but there are times i encounter error "SIP/2.0 500 Server Internal Error". Do you have an idea what's causing this error. I have here a trace log of SIP Messages. I will greatly appreciate any support from you. Thanks Nel U CLIENT-IP:1384 -> SERVER-IP:5060 REGISTER sip:SERVER-IP SIP/2.0..Via: SIP/2.0/UDP 192.168.1.136:5060;br anch=z9hG4bK63235644e9b491928f5..From: 638871680007 <sip:638871680007@116.6 8.141.251>;tag=63283ff..To: 638871680007 <sip:638871680007@SERVER-IP>. .Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Contact: "638871680007" <sip:638871680007@192.168.1.136:5060>..Authorization: Digest username="638871680007", realm="SERVER-IP", nonce="49199956fed351b86a 5c11c83d849a62144c5dcd", uri="sip:SERVER-IP", response="6a29e5839028f9 bc690c77fda9031054", algorithm=MD5..User-Agent: VaxSIPUserAgent/3.0..Expire s: 3600..Max-Forwards: 70..Content-Length: 0.... U SERVER-IP:5060 -> CLIENT-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK63235 644e9b491928f5;received=CLIENT-IP..From: 638871680007 <sip:638871680007 @SERVER-IP>;tag=63283ff..To: 638871680007 <sip:638871680007@116.68.141 .251>..Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Ser ver: Sip EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 SERVER-IP:5060 "Noisy feedback tells: pid=16275 req_src_ip=203.1 89.8.149 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:116.68.141 .251 via_cnt==1".... U SERVER-IP:5060 -> CLIENT-IP:1384 SIP/2.0 500 Server Internal Error..Via: SIP/2.0/UDP 192.168.1.136:5060 ;bran ch=z9hG4bK63235644e9b491928f5;rport=1384;received=CLIENT-IP..From: 6388 71680007 <sip:638871680007@SERVER-IP>;tag=63283ff..To: 638871680007 <s ip:638871680007@SERVER-IP>;tag=04b7610a616ee809532470b3b2dd8840.0e13.. Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Server: Si p EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 11 6.68.141.251:5060 "Noisy feedback tells: pid=16275 req_src_ip= 203.189.8.14 9 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:SERVER-IP vi a_cnt==1"....

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Re: [Kamailio-Users] CDRTool problem
by Ricky Gutierrez 10 Nov '08

10 Nov '08

hello, I change the values inside the section db_class, but it follows me showing the error class DB_radius extends DB_Sql { var $Host = "localhost"; var $Database = "radius"; var $User = "radius"; var $Password = "mypassword"; var $Halt_On_Error ="no"; } class DB_radius2 extends DB_Sql { var $Host = "localhost"; var $Database = "radius"; var $User = "radius"; var $Password = "mypassword"; var $Halt_On_Error ="no"; } "ser_radius"=>array( "name" => "OpenSER", "class" => "CDRS_ser_radius", "table" => "radacct", # "table" => "radacct".date("Ym"), # "db_class" => array("DB_radius2","DB_radius"), "db_class" => "DB_radius", "rating" => "1", "normalizedField" => "Normalized", "UserQuotaClass" => "OpenSERQuota", "AccountsDBClass" => "DB_openser", "UserQuotaNotify" => "1", "purgeCDRsAfter" => 120, // how many days to keep the CDRs "SIPProxies" => array("10.0.0.1" => "node01", // friendly names in sip trace table "10.0.0.2" => "node02", ), "db_class_siponline" => "DB_openser", "mediaDispatcher" => "10.0.0.1" // where to get the active media sessions from MediaProxy 2.0 I am using the version of freeradius 1.1.3 and cdrtool 6.6.10 greetingss ________________________________ Sunday, November 9, 2008 4:45:43 PM Tiago Rés Wrote: Hello, I had the same error. The problem is in the section db_class in global.inc. The default config file has in that section an array with radius1 and radius2. You have to configure it properly. That problem didn't occur more, but now When I try to read the tables it always checks cdrtool.radacct instead of radius.radacct. I don't know why it happens but I think someone else had that problem. Hope I helped.

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[Serusers] Instant Messaging? Too Many Hops
by lin9k＠yahoo.com 10 Nov '08

10 Nov '08

Hi, How do I set up the instant messaging in my SER? When I use a SIP client to send a simple message the SER responds back "Too Many Hops". How do I make it work in my SER? Ling

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[Serusers] RTP Proxy used without NAT
by Marc Bechtel 10 Nov '08

10 Nov '08

Hi, we are testing the iptel.org service using the sip server provided by iptel (213.192.59.75). Sjphonev. 1.65 is used as client. We noticed a strange behavior concerning the RTP Proxy. All SIP messages pass the iptel sip proxy as record route is used. Also all RTP packets pass the sip/rtp proxy. The question bothering us is why the RTP packets are using a proxy? (sniffs show that all RTP packets are send to the proxy) To make our case clear I will explain some details: 1. We are definitely using public IP addresses. (no RFC 1918 addresses) 2. There is no NAT. 3. Sniffs show that in the first invite packet the SDP media attribute is set to nortpproxy:yes 4. In the Sip Header via section the "rport" attribute is used. (not sure why) As I understand the SER documentation RTP proxy should only be used, if the nat_uac_test function in the nathelper module returns true. This function is checking for RFC 1918 addresses and differences in IP and port values. Thanks a lot for the help. Marc

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[Kamailio-Users] branch route
by Alex R.S.M 10 Nov '08

10 Nov '08

Hi, I am trying to modify the header of both main and the branch routes, more specifically trying to modify FROM field on both routes to different values. The change on main route is fine but couldn't make the change on the branch side. if (is_method("INVITE")) { .... seturi("sip:1111111111@x.x.x.x:5060;transport=UDP"); // main route append_branch("sip:2222222222@y.y.y.y:5060"); // branch route if ( uri=~"sip:.+@x.x.x.x.*" ) { uac_replace_from("..."); // make changes on main route, works } if ( uri=~"sip:.+@y.y.y.y.*" ) { uac_replace_from("..."); // make changes on the branch, not working } .... When I move the logic for the branch into "branch_route" it would created two FROM fields. Is this a bug? Thanks, Alex _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Serusers] STUN servers?
by SIP 10 Nov '08

10 Nov '08

Until recently, we've been using the Mystun server because it was a much more stable and reliable version of a stun server than the Vovida implementation (which hasn't been developed in a LONG time). The Mystun server, however, doesn't support XOR_MAPPED_ADDRESS or some of the other new things from the draft-ietf-behave-rfc3489bis-15. I'm wondering if anyone knows of any good stun servers out there that do. N.

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[Kamailio-Users] Routing Call
by paulo leonardo 10 Nov '08

10 Nov '08

Good Morning list, i have kamailio installed in my LAN (192.168.170.101 - machine 1), and i have eyebeam (192.168.46.101 - machine 2) registred in my kamailio, when eyebeam make a call i would like to route this call to a gateway (XXX.XXX.XXX.XXX - machine 3), how can i do that? i would like to read a practical manual. thanks

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[Kamailio-Users] Core dump on writing dialog profile size into script var.
by Alex Balashov 09 Nov '08

09 Nov '08

I also have another interesting problem with the aforementioned configuration (http://pastebin.com/f28051a5). When I write the dialog profile size into an AVP, it works fine. When I write it into a script var, i.e. replace $avp(S:dlg_sz) with $var(dlg_sz), it crashes: Relaying INVITE from 210.23.22.23 to sip:2122222322@210.23.22.23:5060 [D] Added new dialog for 7709600101 [ONREPLY-ROUTE 1] Provisional reply 100 received. [ONREPLY-ROUTE 1] 200 OK received for 7709600101 [ONREPLY-ROUTE 1] INVITE/200 is part of dialog for 7709600101 Segmentation fault (core dumped) GDB reveals: Program received signal SIGSEGV, Segmentation fault. 0xb7e7f2f8 in strncpy () from /lib/i686/nosegneg/libc.so.6 (gdb) where #0 0xb7e7f2f8 in strncpy () from /lib/i686/nosegneg/libc.so.6 #1 0x080a4a07 in set_var_value (var=0x8188d98, value=0xbfae5f28, flags=<value o at script_var.c:122 #2 0xb7d6c4fe in w_get_profile_size (msg=0x81902b8, profile=0xb5ede180 "\034??\ value=0x818b6b8 "@?\030\b", result=0x818b6f8 "N") at dialog.c:668 #3 0x08054f15 in do_action (a=0x818b850, msg=0x81902b8) at action.c:850 #4 0x08053ed2 in run_action_list (a=0x818b5c8, msg=0x81902b8) at action.c:138 #5 0x08056365 in do_action (a=0x818bab8, msg=0x81902b8) at action.c:722 #6 0x08053ed2 in run_action_list (a=0x818b178, msg=0x81902b8) at action.c:138 #7 0x080572c2 in run_top_route (a=0x818b178, msg=0x81902b8) at action.c:118 #8 0xb7de3a64 in reply_received (p_msg=0x81902b8) at t_reply.c:1361 #9 0x08064793 in forward_reply (msg=0x81902b8) at forward.c:507 #10 0x08090d5b in receive_msg ( buf=0x81600e0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 215.22.22.22;branch=z9hG4 08.52.173.18\r\nVia: SIP/2.0/UDP 198.225.86.10:5060;branch=z9hG4bKragjo0207gm0dc p:208.52."..., len=814, rcv_info=0xbfae65d4) at receive.c:203 #11 0x080cccfb in udp_rcv_loop () at udp_server.c:449 #12 0x0806b78d in main (argc=1, argv=0xbfae6764) at main.c:693 It would seem to me that there is some sort of buffer overflow issue that results in the garbage seen above. Not sure that it makes a difference, but the glibc being linked against is a Xen-safe one that disables TLS functionality. This is all running inside a Xen DomU. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

4 9

[Serusers] Assistance with SER setup
by Scott Seltzer 09 Nov '08

09 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com/> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

2 1

[Kamailio-Users] CDRTool problem
by Ricky Gutierrez 09 Nov '08

09 Nov '08

Hi forum, I am trying to configure cdrtool with openser and radius, but in the administration console it shows me this message: "Error instantiating db class: ". I have created the radius database and I see that the openser with the I modulate ACC, he writes data in the chart radacct any suggestion?, something that me this forgetting the file global.inc? best regards rickygm

1 0

[Kamailio-Users] About 'avp_check' and strings formed with pseudo-variables
by Iñaki Baz Castillo 08 Nov '08

08 Nov '08

Hi, I've stored in "usr_preferences" some entries with: - attribute: 'bla' - value: 'alice(a)domain.org' <--- With no "sip:" in front. I use 'avp_db_load' to get that value and compare using 'avp_check' with the To uri without protocol (without "sip:"). I get this behaviour by doing: -------------------------- if avp_db_load("$fu/uri", "$avp(s:bla)") { $var(To) = $tU + '@' + $td; if avp_check("$avp(s:bla)","eq/$var(To)/gi") { [...] -------------------------- But I would like to avoid using a variable $var(To), something as: -------------------------- if avp_db_load("$fu/uri", "$avp(s:bla)") { if avp_check("$avp(s:bla)","eq/$tU + '@' + $td/gi") { # or: if avp_check("$avp(s:bla)","eq/$tU@$td/gi") { [...] -------------------------- Unfortunatelly it doesn't work, it's not valid. Do I really need to use a variable (or AVP) for "avp_check"? Thanks. -- Iñaki Baz Castillo

1 0

[Kamailio-Users] Openser configured as outbound proxy - retransmissions issue.
by Siddhardha Garige 08 Nov '08

08 Nov '08

Hello all, I am working on configuring openser as a outbound proxy. User agents register to service provider through openser outbound proxy. I am experiencing issues with retransmissions. Case 1. UA sends an INVITE to Proxy and proxy is down. Openser sends 100 Trying to UA and retransmits INVITE messages to proxy. After 8 retry attempts sends 408 Request time out to handset. All as expected. Case 2: Openser received INVITE for UA from proxy. UA is down. Openser sent 100 Trying to proxy and routes reqest to UA. No response from UA and ** Openser did not send any retransmissions of INVITE to UA**. After a while CANCLE arrived at openser and cleared transaction. why openser does retransmissions in Case 1 and why not in case 2? Can openser proxy be operated in stateful outbound proxy to take care of retransmissions? I am using default openser config file. Is there any thing I am doing wrong? Any help is greatly appreciated. Thanks Sid

2 2

[Kamailio-Users] [Fwd: Re: [Sip-implementors] REGISTER R-URI with port parameter]
by Alex Balashov 07 Nov '08

07 Nov '08

Does Kamailio match 'uri == myself' if the domain is the same as the canonical hostnames/aliases but the port is different than the one Kamailio knows itself to be listening on? -------- Original Message -------- Subject: Re: [Sip-implementors] REGISTER R-URI with port parameter Date: Tue, 28 Oct 2008 16:46:04 -0500 From: Dale.Worley(a)comcast.net To: sip-implementors(a)lists.cs.columbia.edu References: <00D42150952F70458C66072322F7FE2502EDA940(a)saturn2.aculab.com> <200810241652.m9OGqx6s027493(a)dragon.ariadne.com> <200810241902.43581.ibc(a)aliax.net> From: =?iso-8859-1?q?I=F1aki_Baz_Castillo?= <ibc(a)aliax.net> El Viernes, 24 de Octubre de 2008, Dale.Worley(a)comcast.net escribió: > sip:xxx@host:5060 > sip:xxx@host:15060 > sip:xxx@host:25060 > sip:xxx@host:35060 > > In that case, the request-URI of the REGISTER needs to contain the > proper port number. Well, not totally needed. A UA could construct a REGISTER like this: REGISTER sip:xxx@host SIP/2.0 and sent it to host:15060. You can't depend on that working -- It's never been settled what must happen if a SIP message arrives at a UAS which isn't the UAS that the RFC 3263 rules would send it to. It's possible that the UAS will forward it based on the request-URI. It's safer to include the port, if the destination port is not 5060. For example Twinkle does it if you set the registrar in a port different than 5060. I wouldn't depend on that working. Dale _______________________________________________ Sip-implementors mailing list Sip-implementors(a)lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

3 3

[Kamailio-Users] Passing REGISTERs up stream
by Daniel Corbe 07 Nov '08

07 Nov '08

Guys, When attempting to pass REGISTER requests up stream I get the following error messages on the console (below). This also brings up the question of "what is the best way to simply pass REGISTER requests on to an upstream registrar?" Thanks for the help. -Daniel REGISTER: NATed client, enabling NAT Oct 29 18:03:58 [48175] ERROR:tm:t_forward_nonack: no branch for forwarding Oct 29 18:03:58 [48175] ERROR:tm:w_t_relay: t_forward_nonack failed Oct 29 18:03:58 [48175] ERROR:tm:t_forward_nonack: no branch for forwarding Oct 29 18:03:58 [48175] ERROR:tm:w_t_relay: t_forward_nonack failed Oct 29 18:03:59 [48175] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 401 Here's the relevant parts of my ser config; in route[1] route[1] { ... if (method == "REGISTER") { route(2); } route(1); } route[2] { # Check to see if the UAC is trying to UNREGISTER. If not test # for NAT. If NAT is present, we mark it as such before we save() # the location, that way flag 6 is always set for NATed UAs. if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { xlog("L_NOTICE", "REGISTER: NATed client, enabling NAT\n"); setflag(6); fix_nated_register(); force_rport(); }; # We must handle replies for registrations, for caching and location # tracking purposes. t_on_reply("1"); t_on_failure("1"); # Check for digest #if (radius_www_authorize("")) #{ # xlog("L_NOTICE", "REGISTER: No Digest, sending challenge\n"); # www_challenge("", "0"); # exit; #} # Digest was good if we get here #save("location"); # Relay register to porta t_relay("216.151.143.69"); } onreply_route[2] { if (status =~ "2[0-9][0-9]") { # save("location"); }; }

3 3

[Serusers] Problem with SIP Identity
by jerome.herve＠orange-ftgroup.com 07 Nov '08

07 Nov '08

> Hello everyone, > I'm currently working on the implementation of RFC4474 in SER. > I didn't manage to set up the auth_identity module in SER. > Is there someone able to tell me what do I have to write in ser.cfg to make this possible? > An example of an entire ser.cfg will be great. > I hope you will help me. > Thanks & regards, > > Jérôme HERVE > tél. 02 96 05 27 41 > mob. 06 76 15 18 49 > >

3 2

[Serusers] Problem with SIP Identity
by jerome.herve＠orange-ftgroup.com 07 Nov '08

07 Nov '08

Hello everyone, I'm currently working on the implementation of RFC4474 in SER. I didn't manage to set up the auth_identity module in SER. Is there someone able to tell me what do I have to write in ser.cfg to make this possible? An example of an entire ser.cfg will be great. I hope you will help me. Thanks & regards, Jérôme HERVE tél. 02 96 05 27 41 mob. 06 76 15 18 49

2 1

[Kamailio-Users] $du has no effect in failure route
by jh＠tutpro.com 07 Nov '08

07 Nov '08

i have in failure route: t_on_branch("0"); $du = "sip:127.0.0.1:5070"; xlog("L_INFO", "Pseudo variable du has value <$du>\n"); t_relay("0x01"); and i get to syslog: Nov 6 16:33:15 test /usr/sbin/kamailio[22539]: Pseudo variable du has value <sip:127.0.0.1:5070> but wireshark shows that the invite is sent according to uri host of the branch added in failure route and not to 127.0.0.1:5070. is this a bug or is it so that $du is not supposed to have any effect in failure route? -- juha

3 3

[Kamailio-Users] Kamailio + eyebeam + TLS
by paulo leonardo 07 Nov '08

07 Nov '08

Hi list, I have a litle problem, i would like to setup eyebeam with kamailio using TLS, but i always get *Registration error: 503 - certificate validation failure* =( !! I read the README of TLS. How certificate (.crt) i need to install in eyebeam? # kamailio -V ------------------------------------------------------------------------------------------- version: kamailio 1.4.1-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP,* USE_TLS*, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 4623 2008-08-07 07:31:30Z miconda $ main.c compiled on 16:31:23 Oct 24 2008 with gcc 4.2.3 ------------------------------------------------------------------------------------------ disable_tls = 0 listen = tls:192.168.170.101:5061 tls_port_no = 5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = TLSv1 tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" ------------------------------------------------------------------------------------------------------------ thanks list!

2 1

Re: [Kamailio-Users] Remote-Party-ID header
by ingdavidcespedes＠cable.net.co 07 Nov '08

07 Nov '08

2 1

Re: [Kamailio-Users] CANCEL and 487 Request Terminated
by ingdavidcespedes＠cable.net.co 06 Nov '08

06 Nov '08

I think that's the way it should be. First it receive 487 from SBC and immediately it responds it to SBC with the ACK. Then it passes the 487 to the UAC and receive the ACK, that's all. It doesn't have to forward that last ACK to the SBC. ----- Mensaje original ----- De: Krom <kyle.romulas(a)gmail.com> Fecha: Jueves, Noviembre 6, 2008 4:06 pm Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > My apologies for the drawing. Openser is receiving the 487 from the > SBC, ACKing back to the SBC and then passing the 487 to the UAC and > the the UAC ACK's back to Openser. At which time the error occurs > because the ACK will not be forwarded since it already ACK'd for the > 487 to the SBC. > > -Kyle > > On Thu, Nov 6, 2008 at 2:42 PM, <ingdavidcespedes(a)cable.net.co> > wrote:> The final 487 should go from openser to UAC, not from SBC > to UAC. That's why the final ACK is getting lost, because it was > allready ACK'd by openser. > > > > ----- Mensaje original ----- > > De: Krom <kyle.romulas(a)gmail.com> > > Fecha: Jueves, Noviembre 6, 2008 1:18 pm > > Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > > > >> Klaus, > >> > >> Some additional information. After viewing a trace I can see > that the > >> error is being generated from the last ACK received from the UAC. > >> OpenSER/Kamailio is ACK'ing to the 487 from the PSTN provider > and then > >> relaying the 487 to the UAC which in turn ACK's back. It is the > last>> ACK from the UAC that is getting the error on t_relay. I > can assume > >> because the ACK to the 487 was already sent by OpenSER/Kamailio > to the > >> PSTN upon receiving the initial 487. Something in the script I can > >> change to prevent this or just ignore the error and move on. :-) > >> > >> UAC OSER SBC > >> > >> CANCEL ---> | > >> > >> CANCEL ---> | > >> > >> <--- 200 canceling > >> > >> | <--- 200 OK > >> > >> | <--- 487 Request Terminated > >> > >> ACK ---> > >> > >> <--- 487 Reqest Terminated > >> > >> ACK ---> | > >> > >> > >> Regards, > >> -Kyle > >> > >> > >> On Thu, Nov 6, 2008 at 9:05 AM, Krom <kyle.romulas(a)gmail.com> > wrote:>> > Klaus, > >> > > >> > Thank you for the reply and the link. The link help in > checking my > >> > work. My script is very simple and even checking against the > >> compiled> scripted from sipwise.com the error persists. > >> > > >> > Here is my CANCEL and ACK handling. > >> > if(is_method("CANCEL") || is_method("ACK")) { > >> > if(t_check_trans()) { > >> > if(!t_relay()) sl_reply_error(); > >> > } > >> > exit; > >> > } > >> > > >> > Here is my route block for PSTN calls. > >> > route[4] > >> > { > >> > if (!load_gws()) { > >> > sl_send_reply("500", "Server Internal Error"); > >> > exit; > >> > } > >> > if (!next_gw()) { > >> > sl_send_reply("503", "Service Unavailable"); > >> > exit; > >> > } > >> > t_on_failure("2"); > >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder > >> Controller> > >> > exit; > >> > > >> > } > >> > > >> > Here is my failure route. > >> > failure_route[2] > >> > { > >> > > >> > if(t_check_status("408|500|503")) { > >> > if (!next_gw()) { > >> > t_reply("503", "Service Unavailable"); > >> > exit; > >> > } > >> > t_on_failure("2"); > >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder > >> Controller> } > >> > > >> > exit; > >> > > >> > } > >> > > >> > Any thoughts or errors you see? > >> > > >> > Regards. > >> > > >> > > >> > On Thu, Nov 6, 2008 at 3:56 AM, Klaus Darilion > >> > <klaus.mailinglists(a)pernau.at> wrote: > >> >> looks like the ACK is miss-routed. > >> >> > >> >> Usually calling t_relay() for ACK is sufficient (take a look at > >> the config > >> >> file generated with sipwise.com wizard for correct CANCEL/ACK > >> handling.>> > >> >> regards > >> >> klaus > >> >> > >> >> Krom schrieb: > >> >>> > >> >>> Et al, > >> >>> > >> >>> Version: openser-1.1.1-notls > >> >>> > >> >>> I am receiving the following after calling t_relay within the > >> default>>> route block. > >> >>> ERROR: t_newtran: transaction already in process 0xb0139638 > >> >>> WARNING:sl:sl_send_reply: I won't send a reply for ACK!! > >> >>> WARNING: script writer didn't release transaction > >> >>> > >> >>> This is happening when a downstream UAC dials outbound to the > >> PSTN and > >> >>> then terminates the call prior to the far ending answering. > >> >>> > >> >>> UAC send a CANCEL to the proxy, the proxy responds to the UAC > >> with a > >> >>> '200 canceling' and forwards the CANCEL to the PSTN provider. > >> >>> The PSTN provider sends 487 Request Terminated to the proxy and > >> the>>> proxy sends it on to the UAC. UAC responds with an ACK to > >> the proxy. > >> >>> > >> >>> Is this an issue? How do I handle this dialog withou > getting the > >> >>> t_relay error message? > >> >>> > >> >>> > >> >> > >> > > >> > > >> > > >> > -- > >> > Thanks, > >> > -- Kyle > >> > > >> > >> > >> > >> -- > >> Thanks, > >> -- Kyle > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.kamailio.org > >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >> > > > > > > > > -- > Thanks, > -- Kyle >

2 1

Re: [Kamailio-Users] CANCEL and 487 Request Terminated
by Krom 06 Nov '08

06 Nov '08

My apologies for the drawing. Openser is receiving the 487 from the SBC, ACKing back to the SBC and then passing the 487 to the UAC and the the UAC ACK's back to Openser. At which time the error occurs because the ACK will not be forwarded since it already ACK'd for the 487 to the SBC. -Kyle On Thu, Nov 6, 2008 at 2:42 PM, <ingdavidcespedes(a)cable.net.co> wrote: > The final 487 should go from openser to UAC, not from SBC to UAC. That's why the final ACK is getting lost, because it was allready ACK'd by openser. > > ----- Mensaje original ----- > De: Krom <kyle.romulas(a)gmail.com> > Fecha: Jueves, Noviembre 6, 2008 1:18 pm > Asunto: Re: [Kamailio-Users] CANCEL and 487 Request Terminated > >> Klaus, >> >> Some additional information. After viewing a trace I can see that the >> error is being generated from the last ACK received from the UAC. >> OpenSER/Kamailio is ACK'ing to the 487 from the PSTN provider and then >> relaying the 487 to the UAC which in turn ACK's back. It is the last >> ACK from the UAC that is getting the error on t_relay. I can assume >> because the ACK to the 487 was already sent by OpenSER/Kamailio to the >> PSTN upon receiving the initial 487. Something in the script I can >> change to prevent this or just ignore the error and move on. :-) >> >> UAC OSER SBC >> >> CANCEL ---> | >> >> CANCEL ---> | >> >> <--- 200 canceling >> >> | <--- 200 OK >> >> | <--- 487 Request Terminated >> >> ACK ---> >> >> <--- 487 Reqest Terminated >> >> ACK ---> | >> >> >> Regards, >> -Kyle >> >> >> On Thu, Nov 6, 2008 at 9:05 AM, Krom <kyle.romulas(a)gmail.com> wrote: >> > Klaus, >> > >> > Thank you for the reply and the link. The link help in checking my >> > work. My script is very simple and even checking against the >> compiled> scripted from sipwise.com the error persists. >> > >> > Here is my CANCEL and ACK handling. >> > if(is_method("CANCEL") || is_method("ACK")) { >> > if(t_check_trans()) { >> > if(!t_relay()) sl_reply_error(); >> > } >> > exit; >> > } >> > >> > Here is my route block for PSTN calls. >> > route[4] >> > { >> > if (!load_gws()) { >> > sl_send_reply("500", "Server Internal Error"); >> > exit; >> > } >> > if (!next_gw()) { >> > sl_send_reply("503", "Service Unavailable"); >> > exit; >> > } >> > t_on_failure("2"); >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder >> Controller> >> > exit; >> > >> > } >> > >> > Here is my failure route. >> > failure_route[2] >> > { >> > >> > if(t_check_status("408|500|503")) { >> > if (!next_gw()) { >> > t_reply("503", "Service Unavailable"); >> > exit; >> > } >> > t_on_failure("2"); >> > t_relay("udp:<IPADDR>:5060"); # Relay to the Session Boarder >> Controller> } >> > >> > exit; >> > >> > } >> > >> > Any thoughts or errors you see? >> > >> > Regards. >> > >> > >> > On Thu, Nov 6, 2008 at 3:56 AM, Klaus Darilion >> > <klaus.mailinglists(a)pernau.at> wrote: >> >> looks like the ACK is miss-routed. >> >> >> >> Usually calling t_relay() for ACK is sufficient (take a look at >> the config >> >> file generated with sipwise.com wizard for correct CANCEL/ACK >> handling.>> >> >> regards >> >> klaus >> >> >> >> Krom schrieb: >> >>> >> >>> Et al, >> >>> >> >>> Version: openser-1.1.1-notls >> >>> >> >>> I am receiving the following after calling t_relay within the >> default>>> route block. >> >>> ERROR: t_newtran: transaction already in process 0xb0139638 >> >>> WARNING:sl:sl_send_reply: I won't send a reply for ACK!! >> >>> WARNING: script writer didn't release transaction >> >>> >> >>> This is happening when a downstream UAC dials outbound to the >> PSTN and >> >>> then terminates the call prior to the far ending answering. >> >>> >> >>> UAC send a CANCEL to the proxy, the proxy responds to the UAC >> with a >> >>> '200 canceling' and forwards the CANCEL to the PSTN provider. >> >>> The PSTN provider sends 487 Request Terminated to the proxy and >> the>>> proxy sends it on to the UAC. UAC responds with an ACK to >> the proxy. >> >>> >> >>> Is this an issue? How do I handle this dialog withou getting the >> >>> t_relay error message? >> >>> >> >>> >> >> >> > >> > >> > >> > -- >> > Thanks, >> > -- Kyle >> > >> >> >> >> -- >> Thanks, >> -- Kyle >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> > > -- Thanks, -- Kyle

1 0

[Kamailio-Users] CANCEL and 487 Request Terminated
by Krom 06 Nov '08

06 Nov '08

Et al, Version: openser-1.1.1-notls I am receiving the following after calling t_relay within the default route block. ERROR: t_newtran: transaction already in process 0xb0139638 WARNING:sl:sl_send_reply: I won't send a reply for ACK!! WARNING: script writer didn't release transaction This is happening when a downstream UAC dials outbound to the PSTN and then terminates the call prior to the far ending answering. UAC send a CANCEL to the proxy, the proxy responds to the UAC with a '200 canceling' and forwards the CANCEL to the PSTN provider. The PSTN provider sends 487 Request Terminated to the proxy and the proxy sends it on to the UAC. UAC responds with an ACK to the proxy. Is this an issue? How do I handle this dialog withou getting the t_relay error message? -- Thanks, -- Kyle

2 3

[Serusers] Assistance with SER setup
by Scott Seltzer 06 Nov '08

06 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com/> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

1 0

[Serusers] FW: Support for SER setup
by Scott Seltzer 06 Nov '08

06 Nov '08

Hello, my name is Scott Seltzer, from ConnectMe (www.connectmevoice.com <http://www.connectmevoice.com> ) Connectme is currently in the process of setting up SER for use as a Proxy and Registrar. We will ultimately need to be able to use an API to setup new SIP URIs for each new customer. We are not familiar at all with SER, and would like to find out what support options there may be out there. We would also be interested in using some consulting services to help get us started, and support us along the way. Please let me know if there is anybody that can help us out with this. Thanks! Scott Seltzer

1 0

[Kamailio-Users] two endpoints
by Alex R.S.M 06 Nov '08

06 Nov '08

Hi, If ENUM lookup returns 2 destinations, does kamailio fork the message and send two INVITE? What function can be used to force kamailio to fork a message? Regards, Alex _________________________________________________________________ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster…

2 1

[Kamailio-Users] rewriteuser with AVP
by Samuel Muller 06 Nov '08

06 Nov '08

Hello all, I've a little question : I got a username in an AVP by avp_load_radius, and I would rewrite the uri with it. The objectives is to replace the r-uri by an ipbx uri, then forward the call to it by a new branch (to not loose the phone number requested). in this case : 0123452000 is behind a spa9000. The user 0123452000 has an AVP called "ipbx", that i'm using to do some groups and permissions (plus the rewrite). actually, it does not work, i tried many ways : # load radius attributes of the callee if (!avp_load_radius("callee")) { xlog("L_INFO","-> user unknown in radius usr db : $ru"); route(15); # route PSTN } # verify the AVPs we got (caller and callee) xlog("L_INFO", " -AVP------------------------------------- "); avp_print(); xlog("L_INFO", " ----------------------------------------- "); # callee is behind an ipbx (avp ipbx) ? if (is_avp_set("$avp(s:callee_ipbx)")) { xlog("L_INFO", "-> callee is behind an ipbx : $avp(s:callee_ipbx)"); setflag(14); # flag IPBX #subst_user('/$rU$/$avp(s:callee_ipbx)/'); #rewriteuri("sip:$avp(s:callee_ipbx)@$rd"**); rewriteuser($avp(s:callee_ipbx)); } # callee in usrloc ? if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); xlog("L_INFO","-> registered user called : $tu"); route(7); # route FORWARD } else { xlog("L_INFO","-> 480: local user not found in usrloc : $tu"); sl_send_reply("480","Temporarily Unavailable"); drop; } and in the debugs log : Nov 5 13:04:51 ser0 kamailio[2930]: -ROUTE--INBOUND-------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: rc_auth Success Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP 'callee_asserted_id'/0='0123452000'/0 has been added Nov 5 13:04:51 ser0 kamailio[2930]: DBG:avp_radius:load_avp_user: AVP 'callee_ipbx'/0='spa9000'/0 has been added Nov 5 13:04:51 ser0 kamailio[2930]: -AVP------------------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5cfe8, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<callee_ipbx> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<spa9000 / 7> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5cf88, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<callee_asserted_id> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<0123452000 / 10> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5ced8, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Iname=<caller_asserted_id> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: ^I^I^Ival_str=<0123451011 / 10> Nov 5 13:04:51 ser0 kamailio[2930]: INFO:avpops:ops_print_avp: p=0x7fb3d7e5ce38, flags=0x0003 Nov 5 13:04:51 ser0 kamailio[2930]: ----------------------------------------- Nov 5 13:04:51 ser0 kamailio[2930]: -> callee is behind an ipbx : spa9000 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:registrar:lookup: '$avp( s(a)sip.720.fr' Not found in usrloc Nov 5 13:04:51 ser0 kamailio[2930]: -> 480: local user not found in usrloc : sip:0123452000@sip.720.fr <sip%3A0123452000(a)sip.720.fr> Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:parse_headers: flags=ffffffffffffffff Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:check_via_address: params 77.246.81.162, 192.168.0.134, 0 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:destroy_avp_list: destroying list 0x7fb3d7e5d040 Nov 5 13:04:51 ser0 kamailio[2930]: DBG:core:receive_msg: cleaning up -> But, if I did it by : if (uri =~ "^sip:012345200{1}") { rewriteuri("sip:spa9000@sip.720.fr<sip%3Aspa9000(a)sip.720.fr> "); } if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); xlog("L_INFO","-> registered user called : $tu"); route(7); # route FORWARD } it works great, and the call is ok (the ipbx forward correctly the request the phone behind him). Anyone has an idea ? thanks in advance ! -- Samuel MULLER

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[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 06 Nov '08

06 Nov '08

Hi David, thank you for you answer, 2008/11/6, ingdavidcespedes(a)cable.net.co <ingdavidcespedes(a)cable.net.co>: > > Hi! > > I had the same problem (My PC was responding everything it receives through > the default gateway "eth0" ). Everybody say that it was an Asterisk bug, > but it was a problem in the routing table. Yes, I'm new to Openser but I know quite well Asterisk, and I don't think it is a bug or a misconfiguration . Type route in a terminal and check if they are the way you are thinking. If > not, add the correct route. Tell me if it solve your problem. The route table is correct, but a strange thing happens since the SIP packet is sent from the correct nic, but un the UDP header of that packet I see as source the IP of the other nic (the one that received the orinal packet) I have to investigate a little more on this aspect anyway. C. David Cespedes > > ----- Mensaje original ----- > De: Cosimo Fadda <cfadda.lists(a)gmail.com> > Fecha: Miércoles, Noviembre 5, 2008 9:36 am > Asunto: [Kamailio-Users] Openser as Proxy and Asterisk as Registrar > > > Hi everybody > > I'd like to implement this scenario: > > > > ----------- ------------------- ---------- > > ----- > > | GW/FW > > |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| -- > > >PSTN----------- ------------------- | ----- > > ---------- > > |_______________________________________________| > > > > > > > > > > > > Where: > > - Openser is reachable from the outside with a public IP > > forwarded to > > eth0 private address; > > > > - Asterisk is connected to Openser using another separated LAN > > - Openser acts as proxy (for requests coming from the outside) and > > Asterisk act as registrar and gateway. > > > > > > This is the description, now the questions: > > -since opneser has two different nics, how can configure it to > > properly > > send register request to asterisk? > > This is what currently happens: > > 192.168.40.68:5060: is eth0 Openser Address > > 192.168.12.165 is eth1 Openser Address > > 192.168.12.106 is Asterisk address > > 82.187.X.X is Openser public address > > Network 192.168.40.X is not reachable from net 192.168.12.X > > > > <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106 > > SIP/2.0 // 12.106 is Asterisk address > > Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: > > SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> > >;tag=3189722840 > > To: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68>> > > Contact: "Cfadda" <sip:225@192.168.40.254:2660> > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > Expires: 1800 > > Max-Forwards: 69 > > User-Agent: X-PRO release 1105x > > Content-Length: 0 > > > > > > Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) -- > > - > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER > > request > > as basis request > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : > > 5060 > > (non-NAT) > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to > > 192.168.40.68:5060: > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 82.187.X.X > > ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 > > Via: SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> > >;tag=3189722840 > > To: Cfadda <sip:225@82.187.X.X > > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > Contact: <sip:225@192.168.12.106 <sip%3A225(a)192.168.12.106>> > > Content-Length: 0 > > > > > > --- > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to > > 192.168.40.68:5060: > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP 82.187.X.X > > ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 > > Via: SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 > > To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > > nonce="05f14a3e"Content-Length: 0 > > > > So asterisk sends response to the wrong interface. > > > > How can I solve this? > > > > Thanks in advance, > > > > C. > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > >

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[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 06 Nov '08

06 Nov '08

Hi Daniel, I'll try to use the force_send_sock function, maybe this solve the issue; btw I don't think it is a strictly Asterisk related problem, thank you for the hint, C. 2008/11/5, Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, > > On 11/05/08 16:36, Cosimo Fadda wrote: > >> Hi everybody >> I'd like to implement this scenario: >> >> ----------- ------------------- --------------- >> | GW/FW |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| >> -->PSTN >> ----------- ------------------- | --------------- >> |_______________________________________________| >> >> >> >> Where: >> - Openser is reachable from the outside with a public IP forwarded to >> eth0 private address; >> >> - Asterisk is connected to Openser using another separated LAN >> - Openser acts as proxy (for requests coming from the outside) and >> Asterisk act as registrar and gateway. >> >> >> This is the description, now the questions: >> -since opneser has two different nics, how can configure it to properly >> send register request to asterisk? >> >> > if you want to force an interface for outgoing sip requests, see > force_send_sock() function: > > http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.4.x#force_send_so… > > This is what currently happens: >> 192.168.40.68:5060: is eth0 Openser Address >> 192.168.12.165 is eth1 Openser Address >> 192.168.12.106 is Asterisk address >> 82.187.X.X is Openser public address >> Network 192.168.40.X is not reachable from net 192.168.12.X >> >> <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106SIP/2.0 // 12.106 is Asterisk address >> Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: SIP/2.0/UDP >> 192.168.40.254:2660;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> >> >;tag=3189722840 >> To: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68>> >> Contact: "Cfadda" <sip:225@192.168.40.254:2660> >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> Expires: 1800 >> Max-Forwards: 69 >> User-Agent: X-PRO release 1105x >> Content-Length: 0 >> >> >> Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) --- >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER request as >> basis request >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : 5060 >> (non-NAT) >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to >> 192.168.40.68:5060: >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received= >> 192.168.40.68 >> Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> >> >;tag=3189722840 >> To: Cfadda <sip:225@82.187.X.X > >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Contact: <sip:225@192.168.12.106 <sip%3A225(a)192.168.12.106>> >> Content-Length: 0 >> >> >> --- >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to >> 192.168.40.68:5060: >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received= >> 192.168.40.68 >> Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 >> To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05f14a3e" >> Content-Length: 0 >> >> So asterisk sends response to the wrong interface. >> >> How can I solve this? >> >> > > Probably the right answer you will get on the asterisk mailing lists. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

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[Kamailio-Users] reminder: irc devel meeting in 1h20
by Daniel-Constantin Mierla 06 Nov '08

06 Nov '08

Hello, just a quick reminder about the irc devel meeting: - channel: #kamailio - server: irc.freenode.net - time: 15:00UTC Deatils and agenda: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-dr… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] SER/Asterisk interworking mailing list
by luzango mfupe 06 Nov '08

06 Nov '08

Hi Alex,This is good news coming from you!!! Yes We Can....!!! Rgds, Lu. On Wed, Nov 5, 2008 at 7:48 PM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > > 5. SER/Asterisk interworking mailing list. (Alex Balashov) > > > Message: 5 > Date: Wed, 05 Nov 2008 12:04:32 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: [Kamailio-Users] SER/Asterisk interworking mailing list. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users(a)lists.digium.com>, Commercial and > Business-Oriented > Asterisk Discussion <asterisk-biz(a)lists.digium.com>, users > <users(a)lists.kamailio.org>, openser users < > users(a)lists.opensips.org> > Message-ID: <4911D220.1040506(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Greetings, > > As a developer and consultant who spends considerable time on projects > involving the fusion of Asterisk and products derived from the SER > ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have > found that there is a great volume of interest in this topic on the > mailing lists associated with all communities involved, but a > comparative lack of focus that results in duplicated effort and lack of > specialised response. > > This is mainly due, I think, to the fact that detailed Asterisk > experience - while common - is not a prerequisite for working with the > SER products, while for Asterisk people SER can often be a next step in > scalability and VoIP service delivery platform enhancement that they are > just getting into. And so on. There's pollution in the respective > discursive spaces; a lot of Asterisk people posting to the SER lists > ask a lot of Asterisk-specific questions in addition to any they may > have about SER which can be construed as potentially off-topic by some > members, and the opposite is true on the Asterisk lists when detailed, > involved discussion about SER occurs. > > We need to capture that discussion that exists at the overlap and is > specifically concerned with making these two systems work together, > requiring somewhat detailed and esoteric understanding of both and a > community of user support and knowledge that focuses on both of these > conceptual and product universes. > > Toward that end, I am hosting a new mailing list with this succinct > purpose, if slightly unwieldy name, and encourage all interested to > join. It is called 'SER-Asterisk-Interwork' and can be accessed for > subscription here: > > http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork > > The archives are available here: > > http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ > > You can post to the list at: > > ser-asterisk-interwork(a)lists.evaristesys.com > > It's the same GNU Mailman stuff you are already used to. > > While it could be argued that this cross-product discussion is valuable > to retain in both communities, I think there is considerable benefit to > creating a specialised mailing list that focuses specifically on this > integration path and the unique interoperation and configuration issues > it creates. I think it would be good to get some of this discussion off > of the SER and Asterisk-specific mailing lists where it has somewhat > marginal relevance at times and refocus it. If you agree and are > interested in this topic, you are invited to join the list. > > One last note: The SER/OpenSER community has been in a state of flux > recently, with OpenSER undergoing a name change to Kamailio and > subsequently seeing a fork. The incumbent Kamailio project is now > in the process of merging with the original SER project. The choice of > nomenclature for list is not meant to imply an endorsement of or > affinity for the IPTel SER project per se. It is just that right now it > serves the aim of terseness to use a common denominator, to refer to > this family of projects as the "SER ecosystem." Whether you are a SER, > OpenSER, Kamailio, or OpenSIPS user, you are part of that "SER > ecosystem." That is why the list is named what it is. > > Thank you, > > -- Alex > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 42, Issue 15 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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Re: [Kamailio-Users] Is it possible to break a "if" stament?
by ingdavidcespedes＠cable.net.co 06 Nov '08

06 Nov '08

----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Miércoles, Noviembre 5, 2008 6:50 pm Asunto: [Kamailio-Users] Is it possible to break a "if" stament? > Hi, is it possible to break the body of an "if"? something like: > > -------------- > if $fU == "alice" { > xlog "hello"; > break; > xlog "bye"; > } > xlog "this is after the 'if'" > -------------- > > So the result would be: > > ------- > hello > this is after the 'if' > ------- > > Is it possible? > Not, you can't do brake inside an If, only for Switch. Why don't you do if $fU == "alice" { xlog "hello"; } else{ xlog "bye"; } xlog "this is after the 'if'" > -- > Iñaki Baz Castillo > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] high processing delay
by Fabian Borot 06 Nov '08

06 Nov '08

hello All my Register request take longer than normal to process, what I mean is that an UAC sends a Register and kamailio responds almost 0.5 secs later, causing the UAC to re-transmit its request several times 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 0.469256 192.168.1.20 -> 192.168.10.10 SIP Status: 401 Unauthorized (0 bindings) 0.518263 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 0.628294 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 1.130722 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 1.269301 192.168.1.20 -> 192.168.10.10 SIP Status: 401 Unauthorized (0 bindings) 2.224942 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 2.325358 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) 3.457509 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) 4.497532 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) the cpu shows almost 100 % idle at that time and the pc were kamailio is running has nothing else running but kamailio. this is my config file: ####### Routing Logic ######## # main request routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests xlog("L_DBG", "mylog, time [$Tf] : starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_DGB","mylog, time [$Tf] : Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_DGB","mylog, time [$Tf] : Message too big.\n"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_DGB","mylog, time [$Tf] : starting to process REGISTER.Info: [$au,$ad,$ci,$ct,$cs,$rd,$si,$sp].\n"); if (!www_authorize("", "subscriber")) { xlog("L_DGB","mylog, time [$Tf] : REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; save("location"); xlog("L_DBG","mylog, time [$Tf] : save-location successful.\n"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_DGB","mylog, time [$Tf] : lookup-location failed, sending 404 Not Found.\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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[Kamailio-Users] Is it possible to break a "if" stament?
by Iñaki Baz Castillo 05 Nov '08

05 Nov '08

Hi, is it possible to break the body of an "if"? something like: -------------- if $fU == "alice" { xlog "hello"; break; xlog "bye"; } xlog "this is after the 'if'" -------------- So the result would be: ------- hello this is after the 'if' ------- Is it possible? -- Iñaki Baz Castillo

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Re: [Kamailio-Users] high processing delay
by Daniel-Constantin Mierla 05 Nov '08

05 Nov '08

On 11/05/08 20:15, Fabian Borot wrote: > debug was set to 9, I commented that line out and now it is fine: ok > > > 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > sip:192.168.1.20 > 0.001650 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > Unauthorized (0 bindings) > 0.191670 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > sip:192.168.1.20 > 0.192518 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > bindings) > > but how can I check about this "syslog set to asynchronous mode" ? check your syslog.conf or so, depends on you distribution. Probably the safest is to check the manual, the usual way for old versions of syslog was to put a - (dash) in front of the file for logging. Cheers, Daniel > > regards > fabian > > > > > > Date: Wed, 5 Nov 2008 19:37:35 +0200 > > From: miconda(a)gmail.com > > To: fborot(a)hotmail.com > > CC: users(a)lists.kamailio.org > > Subject: Re: [Kamailio-Users] high processing delay > > > > hello, > > > > what is the debug level? is you syslog set to asynchronous mode? > > > > Cheers, > > Daniel > > > > > > On 11/05/08 17:43, Fabian Borot wrote: > > > hello > > > > > > All my Register request take longer than normal to process, what I > > > mean is that an UAC sends a Register and kamailio responds almost 0.5 > > > secs later, causing the UAC to re-transmit its request several times > > > > > > 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 0.469256 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > > > Unauthorized (0 bindings) > > > 0.518263 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 0.628294 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 1.130722 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 1.269301 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > > > Unauthorized (0 bindings) > > > 2.224942 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 2.325358 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > 3.457509 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > 4.497532 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > > > > > > > the cpu shows almost 100 % idle at that time and the pc were kamailio > > > is running has nothing else running but kamailio. this is my > config file: > > > > > > > > > > > > ####### Routing Logic ######## > > > > > > > > > # main request routing logic > > > > > > > > > route{ > > > # initial sanity checks -- messages with > > > # max_forwards==0, or excessively long requests > > > xlog("L_DBG", "mylog, time [$Tf] : starting_main_logic.\n"); > > > if (!mf_process_maxfwd_header("10")) { > > > sl_send_reply("483","Too Many Hops"); > > > xlog("L_DGB","mylog, time [$Tf] : Too Many Hops.\n"); > > > exit; > > > }; > > > > > > if (msg:len >= 2048 ) { > > > sl_send_reply("513", "Message too big"); > > > xlog("L_DGB","mylog, time [$Tf] : Message too big.\n"); > > > exit; > > > }; > > > > > > > > > if (!method=="REGISTER") > > > record_route(); > > > > > > if (loose_route()) { > > > # mark routing logic in request > > > append_hf("P-hint: rr-enforced\r\n"); > > > route(1); > > > }; > > > > > > > > > if (uri==myself) { > > > if (method=="REGISTER") { > > > xlog("L_DGB","mylog, time [$Tf] : starting to > > > process REGISTER.Info: [$au,$ad,$ci,$ct,$cs,$rd,$si,$sp].\n"); > > > if (!www_authorize("", "subscriber")) { > > > xlog("L_DGB","mylog, time [$Tf] : REGISTER > > > came without auth, sending challenge.\n"); > > > www_challenge("", "0"); > > > exit; > > > }; > > > save("location"); > > > xlog("L_DBG","mylog, time [$Tf] : save-location successful.\n"); > > > exit; > > > }; > > > > > > > > > if (!lookup("location")) { > > > sl_send_reply("404", "Not Found"); > > > xlog("L_DGB","mylog, time [$Tf] : lookup-location > > > failed, sending 404 Not Found.\n"); > > > exit; > > > }; > > > append_hf("P-hint: usrloc applied\r\n"); > > > }; > > > route(1); > > > } > > > > > > route[1] { > > > if (!t_relay()) { > > > sl_reply_error(); > > > }; > > > exit; > > > } > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > Stay up to date on your PC, the Web, and your mobile phone with > > > Windows Live Click here > > > <http://clk.atdmt.com/MRT/go/119462413/direct/01/> > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)lists.kamailio.org > > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Daniel-Constantin Mierla > > http://www.asipto.com > > > > ------------------------------------------------------------------------ > Color coding for safety: Windows Live Hotmail alerts you to suspicious > email. Sign up today. > <http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety…> -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Openser as Proxy and Asterisk as Registrar
by Cosimo Fadda 05 Nov '08

05 Nov '08

Hi everybody I'd like to implement this scenario: ----------- ------------------- --------------- | GW/FW |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| -->PSTN ----------- ------------------- | --------------- |_______________________________________________| Where: - Openser is reachable from the outside with a public IP forwarded to eth0 private address; - Asterisk is connected to Openser using another separated LAN - Openser acts as proxy (for requests coming from the outside) and Asterisk act as registrar and gateway. This is the description, now the questions: -since opneser has two different nics, how can configure it to properly send register request to asterisk? This is what currently happens: 192.168.40.68:5060: is eth0 Openser Address 192.168.12.165 is eth1 Openser Address 192.168.12.106 is Asterisk address 82.187.X.X is Openser public address Network 192.168.40.X is not reachable from net 192.168.12.X <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106 SIP/2.0 // 12.106 is Asterisk address Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.90.68>;tag=3189722840 To: Cfadda <sip:225@82.187.90.68> Contact: "Cfadda" <sip:225@192.168.40.254:2660> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER Expires: 1800 Max-Forwards: 69 User-Agent: X-PRO release 1105x Content-Length: 0 Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) --- Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER request as basis request Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : 5060 (non-NAT) Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to 192.168.40.68:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.90.68>;tag=3189722840 To: Cfadda <sip:225@82.187.X.X > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:225@192.168.12.106> Content-Length: 0 --- Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to 192.168.40.68:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05f14a3e" Content-Length: 0 So asterisk sends response to the wrong interface. How can I solve this? Thanks in advance, C.

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sr-users November 2008