sr-users December 2007

sr-users@lists.kamailio.org

130 participants
234 discussions

[Serusers] Problem for compiling with LCR

by Gerson A. Matiolli

Hi all I am compiling the SER-0.96 that downloaded the cvs with following command: make include_modules="mysql lcr" modules The following error occurred: make[1]: Entrando no diretÃ³rio `/home/ser/sip_router/modules/mysql' make[1]: `mysql.so' estÃ¡ atualizado. make[1]: Saindo do diretÃ³rio `/home/ser/sip_router/modules/mysql' make[1]: Entrando no diretÃ³rio `/home/ser/sip_router/modules/lcr' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev13"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c lcr_mod.c -o lcr_mod.o In file included from ../mysql/dbase.h:42, from lcr_mod.c:53: ../mysql/../../db/db_val.h:40: error: redeclaration of enumerator âDB_INTâ ../../db/db_fld.h:47: error: previous definition of âDB_INTâ was here ../mysql/../../db/db_val.h:41: error: redeclaration of enumerator âDB_FLOATâ ../../db/db_fld.h:48: error: previous definition of âDB_FLOATâ was here ../mysql/../../db/db_val.h:42: error: redeclaration of enumerator âDB_DOUBLEâ ../../db/db_fld.h:49: error: previous definition of âDB_DOUBLEâ was here ../mysql/../../db/db_val.h:44: error: redeclaration of enumerator âDB_STRâ ../../db/db_fld.h:51: error: previous definition of âDB_STRâ was here ../mysql/../../db/db_val.h:45: error: redeclaration of enumerator âDB_DATETIMEâ ../../db/db_fld.h:52: error: previous definition of âDB_DATETIMEâ was here ../mysql/../../db/db_val.h:46: error: redeclaration of enumerator âDB_BLOBâ ../../db/db_fld.h:53: error: previous definition of âDB_BLOBâ was here ../mysql/../../db/db_val.h:48: error: redeclaration of enumerator âDB_BITMAPâ ../../db/db_fld.h:55: error: previous definition of âDB_BITMAPâ was here lcr_mod.c:118: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âlcr_dbfâ lcr_mod.c: In function âlcr_db_initâ: lcr_mod.c:242: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:242: error: (Each undeclared identifier is reported only once lcr_mod.c:242: error: for each function it appears in.) lcr_mod.c: In function âlcr_db_bindâ: lcr_mod.c:260: warning: implicit declaration of function âbind_dbmodâ lcr_mod.c:260: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:266: warning: implicit declaration of function âDB_CAPABILITYâ lcr_mod.c:266: error: âDB_CAP_QUERYâ undeclared (first use in this function) lcr_mod.c:272: error: âDB_CAP_RAW_QUERYâ undeclared (first use in this function) lcr_mod.c: In function âlcr_db_closeâ: lcr_mod.c:284: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c: In function âlcr_db_verâ: lcr_mod.c:296: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:305: warning: implicit declaration of function âtable_versionâ lcr_mod.c: In function âreload_gwsâ: lcr_mod.c:503: error: âdb_row_tâ undeclared (first use in this function) lcr_mod.c:503: error: ârowâ undeclared (first use in this function) lcr_mod.c:518: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:534: warning: implicit declaration of function âRES_ROW_Nâ lcr_mod.c:542: warning: implicit declaration of function âRES_ROWSâ lcr_mod.c:543: warning: implicit declaration of function âROW_VALUESâ lcr_mod.c:543: error: invalid type argument of â->â lcr_mod.c:549: error: invalid type argument of â->â lcr_mod.c:550: error: invalid type argument of â->â lcr_mod.c:553: error: invalid type argument of â->â lcr_mod.c:561: error: invalid type argument of â->â lcr_mod.c:564: error: invalid type argument of â->â lcr_mod.c:572: error: invalid type argument of â->â lcr_mod.c:575: error: invalid type argument of â->â lcr_mod.c:584: error: invalid type argument of â->â lcr_mod.c:588: error: invalid type argument of â->â lcr_mod.c: In function âload_gwsâ: lcr_mod.c:636: error: âdb_row_tâ undeclared (first use in this function) lcr_mod.c:636: error: ârowâ undeclared (first use in this function) lcr_mod.c:636: error: ârâ undeclared (first use in this function) lcr_mod.c:636: warning: left-hand operand of comma expression has no effect lcr_mod.c:698: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:705: error: invalid type argument of â->â lcr_mod.c:709: error: invalid type argument of â->â lcr_mod.c:712: error: invalid type argument of â->â lcr_mod.c:714: error: invalid type argument of â->â lcr_mod.c:717: error: invalid type argument of â->â lcr_mod.c:719: error: invalid type argument of â->â lcr_mod.c:722: error: invalid type argument of â->â lcr_mod.c:724: error: invalid type argument of â->â lcr_mod.c:727: error: invalid type argument of â->â lcr_mod.c:729: error: invalid type argument of â->â lcr_mod.c:733: error: invalid type argument of â->â make[1]: ** [lcr_mod.o] Erro 1 make[1]: Saindo do diretÃ³rio `/home/ser/sip_router/modules/lcr' make: ** [modules] Erro 1 Somebody help me Tanks

17 years

[Serusers] Loading attributes question

by Steven C. Blair

I have a locally defined table with attributes which may contain several values. I define each attribute /value pair as a new entry in the table then use avp_load to retrieve these values from the database. The problem is that I don't know how to retrieve the values in a specific order. For example if I have a=1, a=2, a=3 when I load the values for "a" they may be 2,1,3 or 3,1,2, etc. Is there any way to force loading in a specific sequence. Thanks,Steve

17 years

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

by Daniel-Constantin Mierla

On 12/10/07 07:04, Mihail Gluhowchenko wrote: > Hello all work; :) was your version old or there is a bug in sc.dbtext? Can you post the patch of changes? Best regards, Daniel > in script sc.dbtext use old format subscriber. > Little changes and all work > > date_now=$(date +"%s") > GLOB_ID=$(od -N4 -d /dev/random | head -n1 | cut -d' ' -f2,3 | sed -e > 's/ //') > > add) > if [ $# -ne 4 ] ; then > usage > exit 1 > fi > shift > credentials $1 $2 > is_user $1 > if [ $? -eq 0 ] ; then > echo -e "\nuser already exists\n" > exit 1 > fi > #Add to SUBSCRIBER file > #id(int,auto) username(str) domain(str) password(str) > first_name(str,null) last_name(str,null) email_address(str) > datetime_created(int) last_modified(int) ha1(str) ha1b(str) > rpid(str,null) > > line="$GLOB_ID:$SERUSER:$SERDOMAIN:$2:::$3:$date_now:$date_now:$HA1:$HA1B:" > echo $line >> $SUB_FILE > #Add to URI file > #id(int,auto) domain(str) uri_user(str) > last_modified(int) > line="$GLOB_ID:$SERDOMAIN:$SERUSER:$date_now" > echo $line >> $URI_FILE > echo -e "\nNew user ($SERUSER@$SERDOMAIN) added to > DBTEXT files\n" > ;; > -- > Regards, > Mihail Gluhowchenko. > Enterprise ELTEX. >

17 years

[OpenSER-Users] Server Crash - Sending 500

by Kelvin Williams

While simulating a database failure (stopping the service), I noticed that OpenSER would also crash (see log below). While I don't mind the OpenSER crash since in our configuration its highly dependent upon the database, what I would like to do is send a 5xx series message to the client so that it would be forced to look at one of our other proxies to complete its call. As the client continued to send the INVITE to the original server. How can I send a 500 Server Internal Error message back to the client in the event of a database failure? Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: Probable invalid query Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: PGRES_FATAL_ERROR Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: server closed the connection unexpectedly This probably means the server terminated abnormally before or while processing the request. Dec 8 22:36:11 terra /usr/local/sbin/openser[8147]: ERROR: receive_fd: EOF on 23 Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: child process 8134 exited by a signal 11 Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: core was not generated Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: INFO: terminating due to SIGCHLD Dec 8 22:36:11 terra /usr/local/sbin/openser[8147]: INFO: signal 15 received Dec 8 22:36:11 terra /usr/local/sbin/openser[8140]: INFO: signal 15 received Dec 8 22:36:11 terra /usr/local/sbin/openser[8137]: INFO: signal 15 received

17 years

[OpenSER-Users] Timeout between 100 and 180/183 Messages

by Dioris Moreno

I know fr_timer controls the timeout for provisional responses, and fr_inv_timer controls the timeout for final responses. Also, the TM Module documentation explains: "fr_timer - this timer is used when no response was received yet. If there is no response after fr_timer seconds the timer triggers (and failure route will be executed if t_on_failure() was called). If a provisional response was received, the timer is set to fr_inv_timer for INVITE transactions, and RT_T2 for all other transactions. If a final response is received, the transaction has finished." I understand this "If a provisional response was received, the timer is set to fr_inv_timer for INVITE", but how can I set a different timeout for cases when I have received 100 Trying and I didn't get a 180/183 Ringing? This is the example mentioned in documentation: "For example: You want to have fail over if there is no provisional response after 3 seconds, but you want to ring for 60 seconds. Thus, set the fr_timer to 3 and fr_inv_timer to 60." How can I get a fail over if there is no provisional response after 3 seconds, a fail over if there is no 180/183 message after receiving 100 Trying of 10 seconds, and I want the SIP client rings for only 60 seconds? I'll appreciate any help Dioris Moreno

17 years

Re: [Serusers] SIP Express Bundle - let's get started

by Mike Trest - Personal

To All, I offer comments / suggestion from perspective of enterprise user - more precisely from perspective of the IT-folks who will be installing and operating on behalf of their own enterprise. First, no matter what we select, we will not be able to satisfy those whose policy (or learning limitations) restrict them to a specific OS / DISTRO / VERSION / VENDOR. Just as there are myriads of enterprise with unique approaches to their particular industry, so also most have a very specific opinion about what works best for them. Therefore, I suggest we boldly assert, that SER is to be promoted as an appliance where the interests of the harried IT-folks can be best served. Let this bundle be delivered as a as a stand alone CD/DVD that, once it is deployed on a "buy it anywhere" machine, the IT-folks will never need to go beyond the external web / administrator interface into the inner realms of OS dependencies. <DREAM> This may be idealistic! However, broad scale enterprise acceptance can be achieved very quickly when we are able to achieve this goal. Oh, by-the-way, this broad user base who love the basic appliance for it's simplicity will also love those whose talents can deliver upscale versions of the same fundamental appliance. As one with enough grey hair and grandchildren to have loved my 195x vintage black and white television, I absolutely love the HD flat-screens of 2007! </DREAM> IMHO we should focus on an Enterprise level DISTRO. I agree with the statement "... I think the ultimate goal should be something like an ISO that's ready to go on a server..." Furthermore, I think virtual machine distributions are just another in a long line of "Silver Bullets" that may appeal to the developer heart in all of us . . . but which will scare the crap out of the policy focused Enterprise IT-folks. Therefore, I feel there is enough talent among us to create the first bundle in such a way to make it available to multiple Enterprise users so long as they understand it as an appliance! I have stated my preference for the CentOS-RHEL-FCx environment because it has the broadest base of enterprise-quality custom applications which equire mission-critical long-term stability. On the other hand, I will be supportive of the bundled appliance concept even if someone produces a VM - CentOS or another variation on the same theme. ..mike..

17 years

[Serusers] SER and Broadsoft

by Arun Kumar

Hi, I'm trying to configure my ser with broadsoft but to send calls to broadsoft it need to sent via a user no ip based auth. so, is there any why that I can send username/pass from ser to send calls to my broadsoft. thanks arun

17 years

[Serusers] Is SER for me ?

by Charalampos Alexopoulos

Hi Sorry if this is a stupid question, but before i dive to SIP protocol and SER configuration i would like to know if SER doing what i need. What i want to do is to make voip phone calls with my friends without to use some voip provider like skype. Is the SIP the correct think for this ? Is the SER implementation suitable ? Thank you in advance Charalampos Alexopoulos

17 years

[OpenSER-Users] configure openser with cpl module

by Martin Kardi

Hi, I installed openser with cpl module and other dependencies but I don't know how to configure it? Openser is functionally without cpl module. I am beginner in this. Could somebody help me how to? thanks Martin -- View this message in context: http://www.nabble.com/configure-openser-with-cpl-module-tf4942775.html#a141… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

17 years

[OpenSER-Users] loose_route not setting $du ??

by Patrick Baker

I'm having an issue with messages flowing between asterisk and openser and I believe the issue is related to loose_route and having nulls for $du $dd $ds. Does anyone know how to resolve this with my config? I have also referenced my asterisk config along with some debug information. Thanks in advance for anyone that can help! Asterisk ends up dropping the call after 20 seconds as it appears that openSer isnt responding to asterisk OK message. Also BYE's and others arent working as well.. [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical Response) [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet. Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115' Method: INVITE ######################################################################## # This configuration is autogenerated by sip:wizard # (http://www.sipwise.com/wizard) on Fri Dec 07 05:52:34 +0100 2007 # for OpenSER 1.2 # # Copyright (C) 2007 Sipwise (support(a)sipwise.com) ######################################################################## ######################################################################## # By obtaining, using, and/or copying this configuration and/or its # associated documentation, you agree that you have read, understood, # and will comply with the Terms of Usage provided at # http://www.sipwise.com/news/?page_id=6 as well as the following # additions: # # Permission to use, copy, modify, and distribute this configuration and # its associated documentation for any purpose and without fee is hereby # granted, provided that the above copyright notice appears in all # copies, and that both that copyright notice and this permission notice # appear in supporting documentation, and that the name of Sipwise or # the author will not be used in advertising or publicity pertaining to # distribution of the configuration without specific, written prior # permission. ######################################################################## ######################################################################## # Before using this configuration, read the following prerequisites in # order to gain the designated functionallity: # # base: # You have to insert all locally served domains (i.e. # "openserctl domain add your.domain.com"). # # nat-rtpproxy: # You have to install RTPProxy # (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying # RTP traffic. # # offnet-pstn: # You have to add a routing entry for lcr (i.e. "openserctl lcr # addroute '' '' 1 1"). Additionally, you have to add your gateways # (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1"). # ######################################################################## ######################################################################## # Configuration 'sip:wizard - Fri Dec 07 05:52:34 +0100 2007' ######################################################################## listen = udp:127.0.0.1:5060 listen = udp:10.3.1.31:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no tcp_children = 4 mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = no check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes loadmodule "usrloc.so" 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", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") 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_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") 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_db", "use_domain", 0) loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") 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", "strip_column", "strip") 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", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(from_gw()) { xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(3); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } t_on_failure("1"); route(3); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(10); route(1); } ##asterisk sip.conf## [general] matchexterniplocally=yes canreinvite=no externip=xxx.206.xxx.136 localnet=10.3.1.0/255.255.255.0 context=default bindport=5061 bindaddr=0.0.0.0 sipdebug=yes nat=yes [openser] type=friend context=default insecure=very externalnotify=yes allow=all ##ser log## Dec 7 04:04:52 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.31:5061;transport=udp F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:52 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:53 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:54 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:04:56 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:00 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:04 phonesys-slave openser[24602]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:08 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=ACK RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:12 phonesys-slave openser[24589]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:13 phonesys-slave openser[24599]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 Dec 7 04:05:16 phonesys-slave openser[24597]: Request leaving server, D-URI='<null>' - M=BYE RURI=sip:500@167.206.216.136 F=sip:pbaker2@10.3.1.31 T=sip:500@10.3.1.31 IP=10.3.1.115 ID=62B37CB0-053C-17A2-7F0C-1E457AE94833(a)10.3.1.115 ##asterisk sip debug## <--- SIP read from 10.3.1.31:5060 ---> INVITE sip:500@10.3.1.31:5061;transport=udp SIP/2.0 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31> Contact: <sip:pbaker2@10.3.1.115:5060> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE Max-Forwards: 69 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 252 v=0 o=pbaker2 3001829617 3001829777 IN IP4 10.3.1.115 s=X-Lite c=IN IP4 10.3.1.31 t=0 0 m=audio 35128 RTP/AVP 3 97 101 a=rtpmap:3 gsm/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=nortpproxy:yes <-------------> --- (13 headers 12 lines) --- Sending to 10.3.1.31 : 5060 (no NAT) Using INVITE request as basis request - 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 No user 'pbaker2' in SIP users list Found peer 'openser' for 'pbaker2' from 10.3.1.31:5060 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 10.3.1.31:35128 Found audio description format gsm for ID 3 Found audio description format speex for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x202 (gsm|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x202 (gsm|speex) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.3.1.31:35128 Looking for 500 in default (domain 10.3.1.31) list_route: hop: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> <--- Transmitting (no NAT) to 10.3.1.31:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31> Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Length: 0 <------------> Audio is at 167.206.216.136 port 37712 Adding codec 0x2 (gsm) to SDP Adding codec 0x200 (speex) to SDP Adding non-codec 0x1 (telephone-event) to SDP phonesys-slave*CLI> <--- Reliably Transmitting (no NAT) to 10.3.1.31:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Retransmitting #1 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #4 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 7 04:45:50] NOTICE[25642]: rtp.c:998 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.3.1.31 Retransmitting #6 (no NAT) to 10.3.1.31:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.31;branch=z9hG4bK435c.95fa3e24.0;received=10.3.1.31 Via: SIP/2.0/UDP 10.3.1.115:5060;rport=5060;branch=z9hG4bK754BE4A264062E0954ADE98E18502A53 Record-Route: <sip:10.3.1.31;lr;ftag=1516093159;nat=yes> From: Patrick Baker <sip:pbaker2@10.3.1.31>;tag=1516093159 To: <sip:500@10.3.1.31>;tag=as70a2356d Call-ID: 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 CSeq: 8992 INVITE User-Agent: Asterisk PBX SVN-trunk-r91598 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@167.206.216.136> Content-Type: application/sdp Content-Length: 300 v=0 o=root 87430933 87430933 IN IP4 167.206.216.136 s=Asterisk PBX SVN-trunk-r91598 c=IN IP4 167.206.216.136 t=0 0 m=audio 37712 RTP/AVP 3 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2334 retrans_pkt: Maximum retries exceeded on transmission 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 for seqno 8992 (Critical Response) [Dec 7 04:45:55] WARNING[25642]: chan_sip.c:2361 retrans_pkt: Hanging up call 2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115 - no reply to our critical packet. Really destroying SIP dialog '2825F5A8-334C-7B59-8EDA-4C0602180528(a)10.3.1.115' Method: INVITE

17 years

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sr-users December 2007