sr-users September 2005

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[Serusers] SER and OpenSER

by Waldo Rubinstein

Can anyone tell me what the difference between SER and OpenSER is? Thanks, Waldo

19 years, 2 months

[Serusers] Issues when using LCR module

by Anthony Dean

I am getting a weird issue when I use the LCR modules and I am not sure if its a bug in the module or my config. It only happens when the first leg has failed the invite and it jumps to the failure route and tries the next gateway on the list. The bug is in the "c=IN IPV4" line in the invite to the second gateway where there should only be one ip address. So it sends the following: c=IN IPV4 192.168.8.35192.168.8.35. when it should be: c=IN IPV4 192.168.8.35. This makes the second gateway give back a 488 not acceptable back regardless if it can process the call or not as the header is malformed. If the first gateway of choice doesn't give back an error the call will connect just fine as the first invite is correctly formatted (which makes me think it is something to do with how I process the failure route that is at fault or how the lcr does it - just can't figure out which). Below is the sip message flow that I get (I made the first gateway give back a 404 not found to test the lcr failover). Call flow is supposed to be: PH-A = 2958770000 (Caller, 192.168.10.30 [Nat Inside], 192.168.7.250 [Nat Outside]) PH-B = 2959770000 (Called, PSTN) GW-A = 192.168.8.6 GW-B = 192.168.35.18 SER = 192.168.8.35 PH-A -> SER (INVITE - Good) PH-A <- SER (100 Trying - Good) SER -> GW-A (INVITE - Good) SER <- GW-A (100 Trying - Good) SER <- GW-A (404 Not Found - Good) SER -> GW-A (ACK - Good) SER -> GW-B (INVITE - Malformed) SER <- GW-B (488 Not Acceptable - Due To Malformed Header) ... Then the call fails due to no more gateways as expected. Any ideas what is causing this (my config is mainly based on the onsip pstn example with lcr extras)? U 192.168.7.250:61903 -> 192.168.8.35:5060 INVITE sip:2959770000@192.168.8.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK-701a9765. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Max-Forwards: 70. Contact: 2958770000 <sip:2958770000@192.168.10.30:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.13(LSb). Content-Length: 286. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 712750 712750 IN IP4 192.168.10.30. s=-. c=IN IP4 192.168.10.30. t=0 0. m=audio 16430 RTP/AVP 18 0 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.8.35:5060 -> 192.168.7.250:61903 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK-701a9765;rport=61903;received=192.168.7.25 0. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Server: Sip Express Router (0.9.4 - i386/linux). Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.8.6:5060 INVITE sip:12959770000@192.168.8.6 SIP/2.0. Record-Route: <sip:2959770000@192.168.8.35:5060;nat=yes;ftag=d0ff6100e33025ado0;lr=on>. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.0. Via: SIP/2.0/UDP 192.168.10.30:5060;rport=61903;received=192.168.7.250;branch=z9hG4bK-701a976 5. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Max-Forwards: 16. Contact: 2958770000 <sip:2958770000@192.168.7.250:61903>. Expires: 240. User-Agent: Linksys/PAP2-2.0.13(LSb). Content-Length: 284. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 712750 712750 IN IP4 192.168.10.30. s=-. c=IN IP4 192.168.8.35. t=0 0. m=audio 35040 RTP/AVP 18 0 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.8.6:50183 -> 192.168.8.35:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.0,SIP/2.0/UDP 192.168.10.30:5060;rport=61903;received=192.168.7.250;branch=z9hG4bK-701a976 5. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=22C61F2C-2339. Date: Thu, 08 Sep 2005 21:06:38 GMT. Call-ID: c19d05bc-82926821(a)192.168.10.30. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Length: 0. . U 192.168.8.6:50183 -> 192.168.8.35:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.0,SIP/2.0/UDP 192.168.10.30:5060;rport=61903;received=192.168.7.250;branch=z9hG4bK-701a976 5. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=22C61F2C-2339. Date: Thu, 08 Sep 2005 21:06:38 GMT. Call-ID: c19d05bc-82926821(a)192.168.10.30. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Reason: Q.850;cause=3. Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.8.6:5060 ACK sip:12959770000@192.168.8.6 SIP/2.0. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.0. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. Call-ID: c19d05bc-82926821(a)192.168.10.30. To: <sip:2959770000@192.168.8.35>;tag=22C61F2C-2339. CSeq: 101 ACK. User-Agent: Sip Express Router (0.9.4 - i386/linux). Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.35.18:5060 INVITE sip:12959770000@192.168.35.18 SIP/2.0. Record-Route: <sip:2959770000@192.168.8.35:5060;nat=yes;ftag=d0ff6100e33025ado0;lr=on>. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.1. Via: SIP/2.0/UDP 192.168.10.30:5060;rport=61903;received=192.168.7.250;branch=z9hG4bK-701a976 5. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Max-Forwards: 16. Contact: 2958770000 <sip:2958770000@192.168.7.250:61903>. Expires: 240. User-Agent: Linksys/PAP2-2.0.13(LSb). Content-Length: 300. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 712750 712750 IN IP4 192.168.10.30. s=-. c=IN IP4 192.168.8.35192.168.8.35. t=0 0. m=audio 3504035040 RTP/AVP 18 0 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.35.18:5060 -> 192.168.8.35:5060 SIP/2.0 488 Not acceptable here. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.1. Via: SIP/2.0/UDP 192.168.10.30:5060;received=192.168.7.250;branch=z9hG4bK-701a9765. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=as113c57f8. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. User-Agent: Video PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: <sip:12959770000@192.168.35.18>. Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.35.18:5060 ACK sip:12959770000@192.168.35.18 SIP/2.0. Via: SIP/2.0/UDP 192.168.8.35;branch=z9hG4bK1128.6389a122.1. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. Call-ID: c19d05bc-82926821(a)192.168.10.30. To: <sip:2959770000@192.168.8.35>;tag=as113c57f8. CSeq: 101 ACK. User-Agent: Sip Express Router (0.9.4 - i386/linux). Content-Length: 0. . U 192.168.8.35:5060 -> 192.168.7.250:61903 SIP/2.0 503 Service Temporarilly Unavailable, No More Available Gateways. Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK-701a9765;rport=61903;received=192.168.7.25 0. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=d1bd3b96fe18e5f9baf4949f8e4468f7-0b42. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 INVITE. Server: Sip Express Router (0.9.4 - i386/linux). Content-Length: 0. . U 192.168.7.250:61903 -> 192.168.8.35:5060 ACK sip:2959770000@192.168.8.35 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.30:5060;branch=z9hG4bK-701a9765. From: 2958770000 <sip:2958770000@192.168.8.35>;tag=d0ff6100e33025ado0. To: <sip:2959770000@192.168.8.35>;tag=d1bd3b96fe18e5f9baf4949f8e4468f7-0b42. Call-ID: c19d05bc-82926821(a)192.168.10.30. CSeq: 101 ACK. Max-Forwards: 70. Contact: 2958770000 <sip:2958770000@192.168.10.30:5060>. User-Agent: Linksys/PAP2-2.0.13(LSb). Content-Length: 0. . # ----------------------------------------------------------------- # Standard Settings # ----------------------------------------------------------------- debug = 3 children = 1 fork = no log_stderror = yes check_via = no dns = no rev_dns = no sip_warning = no # ----------------------------------------------------------------- # Server Settings # ----------------------------------------------------------------- port = 5060 listen = 192.168.8.35 alias = 192.168.8.35 # ----------------------------------------------------------------- # Fifo Settings # ----------------------------------------------------------------- fifo = "/tmp/ser_fifo" fifo_mode = 0666 fifo_db_url = "mysql://root:password@192.168.8.35/ser" # ----------------------------------------------------------------- # Module Settings # ----------------------------------------------------------------- loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/auth_radius.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/uri_radius.so" loadmodule "/usr/lib/ser/modules/uac.so" loadmodule "/usr/lib/ser/modules/lcr.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/group_radius.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/avp_radius.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/options.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/speeddial.so" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/msilo.so" loadmodule "/usr/lib/ser/modules/xlog.so" # ----------------------------------------------------------------- # Module Parameters # ----------------------------------------------------------------- modparam("auth", "secret", "password") modparam("auth", "nonce_expire", 60) modparam("auth", "rpid_prefix", "") modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "realm_prefix", "sip.") modparam("auth", "rpid_avp", "i:13") modparam("auth_db", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_radius", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("uri_db", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_radius", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("uri_radius", "service_type", 15) modparam("domain", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("domain", "db_mode", 0) modparam("domain", "domain_table", "domain") modparam("group", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("group", "table", "grp") modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") modparam("group", "use_domain", 0) modparam("group_radius", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("group_radius", "use_domain", 0) modparam("registrar", "nat_flag", 6) modparam("registrar", "append_branches", 0) modparam("registrar", "case_sensitive", 0) modparam("registrar", "default_expires", 300) modparam("registrar", "default_q", 100) modparam("registrar", "max_contacts", 0) modparam("registrar", "max_expires", 300) modparam("registrar", "min_expires", 60) modparam("registrar", "use_domain", 0) modparam("registrar", "received_avp", 42) modparam("registrar", "received_param", "received") modparam("registrar", "retry_after", 0) modparam("permissions", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_allow_file", "/etc/ser/permissions/allow") modparam("permissions", "default_deny_file", "/etc/ser/permissions/deny") modparam("permissions", "check_all_branches", 1) modparam("permissions", "allow_suffix", ".allow") modparam("permissions", "deny_suffix", ".deny") modparam("usrloc", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "timer_interval", 30) modparam("usrloc", "use_domain", 0) modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "method_column", "method") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("avpops", "avp_url", "mysql://root:password@192.168.8.35/ser") modparam("avpops", "avp_aliases", "voicemail=i:500") modparam("avpops", "avp_table", "preferences") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") modparam("lcr", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 30) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "rpid") modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) modparam("tm", "fr_timer", 10) modparam("tm", "fr_inv_timer", 30) modparam("tm", "wt_timer", 10) modparam("tm", "delete_timer", 2) modparam("tm", "retr_timer1p1", 1) modparam("tm", "retr_timer1p2", 2) modparam("tm", "retr_timer1p3", 4) modparam("tm", "retr_timer2", 4) modparam("tm", "noisy_ctimer", 0) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "fr_timer_avp", "callee_fr_timer") modparam("tm", "fr_inv_timer_avp", "callee_fr_inv_timer") modparam("tm", "tw_append", "") modparam("acc", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("acc", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "service_type", 15) modparam("acc", "log_flag", 21) modparam("acc", "log_missed_flag", 22) modparam("acc", "db_flag", 23) modparam("acc", "db_missed_flag", 24) modparam("acc", "radius_flag", 25) modparam("acc", "radius_missed_flag", 26) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "early_media", 0) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "received_avp", 42) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "force_socket", "") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/ser/mediaproxy/sip") modparam("mediaproxy", "rtp_asymmetrics", "/etc/ser/mediaproxy/rtp") modparam("mediaproxy", "natping_interval", 30) modparam("exec", "setvars", 1) modparam("speeddial", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("speeddial", "use_domain", 0) modparam("speeddial", "user_column", "username") modparam("speeddial", "domain_column", "domain") modparam("speeddial", "sd_user_column", "sd_username") modparam("speeddial", "sd_domain_column", "sd_domain") modparam("speeddial", "new_uri_column", "new_uri") modparam("speeddial", "domain_prefix", "sip.") modparam("maxfwd", "max_limit", 16) modparam("options", "accept", "*/*") modparam("options", "accept_encoding", "") modparam("options", "accept_language", "en") modparam("xlog", "buf_size", 8192) modparam("uac", "from_store_param", "vsf") modparam("uac", "credential", "username:domain:password") modparam("msilo", "db_url", "mysql://root:password@192.168.8.35/ser") modparam("msilo", "db_table", "silo") modparam("msilo", "registrar", "sip:registrar@tristarcorp.net") modparam("msilo", "expire_time", 259200) modparam("msilo", "check_time", 30) modparam("msilo", "use_contact", 1) modparam("avp_radius", "radius_config", "/etc/ser/radius/radiusclient.conf") modparam("avp_radius", "caller_service_type", 15) modparam("avp_radius", "callee_service_type", 15) # ----------------------------------------------------------------- # Main Route # ----------------------------------------------------------------- route { # ----------------------------------------------------------------- # Sanity Check # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { if (method != "ACK") { sl_send_reply("483", "Too Many Hops"); }; break; }; if (msg:len > max_len) { if (method != "ACK") { sl_send_reply("513", "Message Overflow"); }; break; }; # ----------------------------------------------------------------- # Record Route # ----------------------------------------------------------------- if (method == "INVITE" && client_nat_test("3")) { record_route_preset("192.168.8.35:5060;nat=yes"); } else if (method != "REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down # ----------------------------------------------------------------- if (method == "BYE" || method == "CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Routing # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method == "INVITE" || method == "ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Processing # ----------------------------------------------------------------- if (uri != myself) { route(1); break; }; if (uri == myself) { if (method == "REGISTER") { route(2); break; } else if (method == "REFER" || method == "INVITE" || method == "CANCEL") { route(3); break; } else if (method == "OPTIONS") { options_reply(); break; }; lookup("aliases"); if (uri != myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); break; } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { # ----------------------------------------------------------------- # Reply Routing # ----------------------------------------------------------------- t_on_reply("1"); # ----------------------------------------------------------------- # Call Log & Accounting (log=21, sql=23, radius=25) # ----------------------------------------------------------------- setflag(23); # ----------------------------------------------------------------- # Relay Call # ----------------------------------------------------------------- if (!t_relay()) { if (method == "INVITE" || method == "ACK") { end_media_session(); }; if (method != "REFER" && method != "INVITE" && method != "CANCEL" && method != "ACK") { sl_reply_error(); }; }; } # ----------------------------------------------------------------- # Register Message Handler # ----------------------------------------------------------------- route[2] { # ----------------------------------------------------------------- # Nat Test # ----------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; # ----------------------------------------------------------------- # Authenticate User # ----------------------------------------------------------------- if (!allow_trusted() && !from_gw()) { if (!www_authorize("", "subscriber")) { www_challenge("", "1"); break; }; consume_credentials(); }; # ----------------------------------------------------------------- # Save Registration # ----------------------------------------------------------------- if (!save("location")) { sl_reply_error(); }; } # ----------------------------------------------------------------- # Invite/Cancel Message Handler # ----------------------------------------------------------------- route[3] { # ----------------------------------------------------------------- # Nat Fixup # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; # ----------------------------------------------------------------- # Authorize Users / Setup Voicemail For Gateways # ----------------------------------------------------------------- if (!allow_trusted() || !from_gw()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "1"); break; }; consume_credentials(); }; # ----------------------------------------------------------------- # Check For Alias # ----------------------------------------------------------------- lookup("aliases"); if (uri != myself) { route(4); break; }; # ----------------------------------------------------------------- # Cancel Call # ----------------------------------------------------------------- if (method == "CANCEL") { route(1); break; }; # ----------------------------------------------------------------- # Check For Local User # ----------------------------------------------------------------- lookup("location"); if (uri != myself) { route(4); break; }; # ----------------------------------------------------------------- # Route Call # ----------------------------------------------------------------- if (uri =~ "^sip:1[0-9]{10}@" || uri =~ "^sip:011") { route(5); break; } else { sl_send_reply("404", "User Not Found"); break; }; } # ----------------------------------------------------------------- # Media Proxy Enabler # ----------------------------------------------------------------- route[4] { # ----------------------------------------------------------------- # Enable Media Proxy # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); break; } # ----------------------------------------------------------------- # Gateway LCR Route Handler # ----------------------------------------------------------------- route[5] { # ----------------------------------------------------------------- # Load Pstn LCR Gateways # ----------------------------------------------------------------- if (!load_gws()) { sl_send_reply("500", "Internal Server Error, No PSTN Gateways"); break; }; # ----------------------------------------------------------------- # Find Next Available Gateway # ----------------------------------------------------------------- if (!next_gw()) { sl_send_reply("503", "Service Temporarilly Unavailable"); break; } # ----------------------------------------------------------------- # Route Call # ----------------------------------------------------------------- t_on_failure("1"); route(4); break; } # ----------------------------------------------------------------- # LCR Route Failure Handler # ----------------------------------------------------------------- failure_route[1] { # ----------------------------------------------------------------- # Failed Call Log & Accounting (log=22, sql=24, radius=26) # ----------------------------------------------------------------- setflag(24); # ----------------------------------------------------------------- # Find Next Available Gateway # ----------------------------------------------------------------- if (method == "INVITE" && t_check_status("404|408|500|503")) { if (!next_gw()) { t_reply("503", "Service Temporarilly Unavailable, No More Available Gateways"); break; }; t_on_failure("1"); route(4); break; }; } # ----------------------------------------------------------------- # Reply Route # ----------------------------------------------------------------- onreply_route[1] { # ----------------------------------------------------------------- # Enable Media Proxy # ----------------------------------------------------------------- if ((isflagset(6) || isflagset(7)) && (status =~ "(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; # ----------------------------------------------------------------- # Nat Fixup # ----------------------------------------------------------------- if (client_nat_test("1")) { fix_nated_contact(); }; } Thank You Anthony Dean This correspondence is considered confidential and any reproduction for the purpose of public disclosure is forbidden without written permission by the author signed above. If you are not the intended recipient, please immediately notify the sender and delete any copies. If you are responding to this e-mail, please append or attach all previous correspondence as a point of reference.

19 years, 2 months

[Serusers] 503 service unavailable

by zhaomin

I use asterisk as a pstn and voicemail,but I get this error from asterisk 's CLI blow: Verbosity is at least 3 -- Executing AbsoluteTimeout("SIP/61.183.23.216-08a3b558", "15") in new stack -- Set Absolute Timeout to 15 -- Executing Congestion("SIP/61.183.23.216-08a3b558", "") in new stack == Spawn extension (from-sip-external, 2000, 2) exited non-zero on 'SIP/61.183.23.216-08a3b558' -- Executing AbsoluteTimeout("SIP/61.183.23.216-08a3b558", "15") in new stack -- Set Absolute Timeout to 15 -- Executing Congestion("SIP/61.183.23.216-08a3b558", "") in new stack == Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/61.183.23.216-08a3b558' What's wrong in it? ser or asterisk? I am not sure .anyones would help me? Thanks Zhaommin

19 years, 2 months

[Serusers] one way voice on a public IP address

by Iqbal

Hi Has anyone had a problem like this, everything is on a public IP, even the UA, I get voice from UA --> PSTN, not the other way, the SDP all looks fine, c= has the IP address of the UA in it, I was thinking it might be a port problem, but the logs are not showing anything there. Iqbal

19 years, 2 months

[Serusers] Problem with ACK's (branch=0)

by Rodrigo P. Telles

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've been experiencing some troubles with ACK's with branch=0. I found a thread about it but I didn't find a 'solution' folowing the thread. http://lists.iptel.org/pipermail/serdev/2005-April/004296.html Some one said that ACK/branch=0 is ok for non-negative (2xx) replies (RFC3261), but I couldn't find that text in the RFC3261. Can some one point me to the correct answer for that question? Thanks in advance. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> TI Manager Devel-IT - http://www.devel.it ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDIeGgiLK8unYgEMQRAsxDAJwOKxDAIHyr4Ll8hEEOIpav9FzqgACfT9Ej fSsQEexrILyfA3u7JxU6lkg= =nkWg -----END PGP SIGNATURE-----

19 years, 2 months

[Serusers] SIP TEST

by fedora＠voice.dyndns.tv

Example Mailbox, There is a new voicemail in mailbox 1234: From: "Japan SIP" <8000> Length: 0:10 seconds Date: Saturday, September 10, 2005 at 12:39:17 AM Dial *98 to access your voicemail by phone. Visit http://www.voice.dyndns.tv/cgi-bin/vmail.cgi?action=login&mailbox=1234 to check your voicemail with a web browser.

19 years, 2 months

RE: [Serusers] Problem with command "serctl add"

by Alberto

Sorry, I use a start-up script and the /var/log/messages : Sep 9 16:53:25 sip ser: WARNING: fix_socket_list: could not rev. resolve 192.168.100.199 Sep 9 16:53:25 sip ser: WARNING: fix_socket_list: could not rev. resolve 192.168.100.199 Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: Maxfwd module- initializing Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: new_connection(): Access denied for user 'ser'@'localhost' (using password: YES) Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: db_init(): Could not create a connection Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: ERROR: uridb_db_ver: unable to open database connection Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: ERROR: uri_db:mod_init(): Error while querying table version Sep 9 16:53:25 sip /usr/local/sbin/ser[434]: init_mod(): Error while initializing module uri_db The script: #!/bin/sh # # Start/stop/restart SIP EXpress Router (SER) # # Version: 1.0 - Paul Belanger <pabelanger(a)gmail.com> # # Directions: # copy ser script to /usr/local/etc/rc.d/ # edit /etc/rc.conf and add the following: # ser_enable="YES" # # # 05.05.2005 - Initial Version . /etc/rc.subr name="ser" rcvar="`set_rcvar`" command="/usr/local/sbin/${name}" pidfile="/var/run/${name}.pid" load_rc_config $name ser_flags="$cron_flags -P $pidfile" run_rc_command "$1"

19 years, 2 months

[Users] Call failed due to 403 Forbidden

by Subashini C V - CTD, Chennai

hi, i have registered with Asterisk, After that when i try to call my other extension, i am getting an "403 Forbidden" response. Below is the dump of SIP messages. Is there any SIP message is in conflict with the Asterisk rules..? I need your help.... >> SENT: SOURCE [0.0.0.0:1035] <-> DESTINATION [10.100.12.201:5060] INVITE sip:6442161001@10.100.12.201:5060 SIP/2.0 Via: SIP/2.0/UDP 10.100.12.209:5060;rport;branch=z9hG4bKce95dd81f969e71b577c8037f9bbdd63 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24891 INVITE Contact: "Phone1"<sip:6442091001@10.100.12.209> Content-Length: 287 Content-Type: application/sdp Expires: 180 Max-Forwards: 70 Organization: PMC-SIERRA Proxy-Authorization: DIGEST username="6442091001",realm="asterisk",nonce="51afa269",uri="sip:6442161001@ 10.100.12.201",response="4e3255ca53d7cd894c98fac21132a991",algorithm=MD5 User-Agent: Stein Authorization: DIGEST username="6442091001",realm="asterisk",nonce="51afa269",uri="sip:6442161001@ 10.100.12.201",response="4e3255ca53d7cd894c98fac21132a991",algorithm=MD5 v=0 o=UserName 584103545 0 IN IP4 10.100.12.209 s=Voice Session c=IN IP4 10.100.12.209 t=0 0 m=audio 8002 RTP/AVP 0 4 8 18 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=yes a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=sendrecv a=rtcp: 8003 << RECEIVED: SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060] SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201;tag=as7f9a42c9 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24892 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:6442161001@10.100.12.201> Content-Length: 0 << RECEIVED: SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060] SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201;tag=as7f9a42c9 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24892 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:6442161001@10.100.12.201> Content-Length: 0 << RECEIVED: SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060] SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.100.12.209:5060;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201;tag=as7f9a42c9 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24892 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:6442161001@10.100.12.201> Content-Length: 0 >> SENT: SOURCE [0.0.0.0:1036] <-> DESTINATION [10.100.12.201:5060] ACK sip:6442161001@10.100.12.201:5060 SIP/2.0 Via: SIP/2.0/UDP 10.100.12.209:5060;rport;branch=z9hG4bK39757cbf1864977ffda3e87bf0072942 From: "Phone1"<sip:6442091001@10.100.12.209>;tag=1ed0181d0aae037030318fc1e955ccb4 To: sip:6442161001@10.100.12.201;tag=as7f9a42c9 Call-ID: 2d8bda61e762aedd29f8431fbaa3d1c2(a)10.100.12.209 CSeq: 24892 ACK Content-Length: 0 Organization: PMC-SIERRA User-Agent: Stein Thanks, Subashini DISCLAIMER This message and any attachment(s) contained here are information that is confidential, proprietary to HCL Technologies and its customers. Contents may be privileged or otherwise protected by law. The information is solely intended for the individual or the entity it is addressed to. If you are not the intended recipient of this message, you are not authorized to read, forward, print, retain, copy or disseminate this message or any part of it. If you have received this e-mail in error, please notify the sender immediately by return e-mail and delete it from your computer

19 years, 2 months

[Users] Scalability in presence notification

by Vincent Luba

Hi, I need some advices about a design issue. I'm trying to figure out how to manage my presence information notifications, using RFC 3856. I have a situation in which end-to-end presence management wouldn't be suitable. Still I'm wondering if server based presence management is a good idea. My thought is that it present a scalability issue. The point is that having the presence server would have to handle every single SUBSCRIBE & PUBLISH request then sends NOTIFY to every watchers. Is it an efficient way to handle presence in a network with hundreds online users, everyone having tens of contact on their list. Thank you for your comments Luba Vincent

19 years, 2 months

[Serusers] RFC3621

by harry gaillac

Hello, I've some troubles for incoming calls from others domains When I dial numbers in my domain all is ok (see internal_sip_call ethereal file) no problem onreply route However when somebody call numbers in my domain something is wrong with status 200 OK / SIP/SDP message Public ip address is not set in sdp message see (incoming_call ethereal file). Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

19 years, 2 months

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sr-users September 2005