- sr-users - kamailio.org

Asterisk re-INVITE race condition, error 500.
by David K 04 Feb '14

04 Feb '14

Hello all, So I have three machines, we don't care about audio for this problem, so everything I mention here is SIP related. Freeswitch <--> Kamailio 3.3.2 <--> Asterisk 1. Asterisk sends an INVITE to Freeswitch through the Kamailio proxy. 2. Kamailio replies 100 Trying and forwards to Freeswitch 3. Freeswitch replies 100 Trying 4. Freeswitch replies 180 Ringing to Kamailio 5. Kamailio routes the answer to Asterisk 6. Freeswitch replies 200 OK to Kamailio 7. Kamailio replies 200 OK to Asterisk 8. Asterisk replies ACK to Kamailio 9. Asterisk sends a re-INVITE to Freeswitch through Kamailio 10. Kamailio routes the re-INVITE to freeswitch 11. Kamailio routes the ACK to freeswitch. 12. Freeswitch replies 500 Server error because it got a re-INVITE before the ACK. So, my problem is that Kamailio seems to process my re-INVITE more quickly than the ACK. So Freeswitch replies an error because it got the re-INVITE before the ACK. So my "solution" is to add a usleep(20); for re-INVITEs on Kamailio, but I think this is a lousy solution. Has anyone here had to deal with problems where Kamailio routes a re-INVITE faster than an ACK causing endpoints to return error messages? Has anyone had to deal with a similar issue? Thanks, David

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Kamailio World 2014 - New speakers and presentations
by Daniel-Constantin Mierla 04 Feb '14

04 Feb '14

Slightly less than two months to the start of Kamailio World 2014 <http://www.kamailioworld.com> [1] the organization of the event is getting accelerated. We received plenty of interesting proposals so far, we expect to end the Call for Speakers very soon, earlier than previously announced. If you have plans to submit, you should really hurry up. The profiles of about 20 speakers have been published in Speakers page <http://conference.kamailio.com/k02/speakers/> [2], few others that were accepted already will be added soon. Details for about 15 sessions during the conference days (April 3-4, 2014) are available in the Schedule page <http://conference.kamailio.com/k02/schedule/> [3]. The technical workshop on the 2nd of April is getting to the final shape as well. Similar to last year, expect several exhibition spots where various companies will present their services or products during the conference days. In the evening of 3rd of April it will be the cocktail party, a relaxing event for social networking around good food and drinks. We are grateful to our sponsors [4], they are making possible this event: FhG Fokus <http://www.fokus.fraunhofer.de/en/fokus/index.html>, Asipto <http://www.asipto.com>, Sipwise <http://www.sipwise.com>, Sipgate <http://www.sipgate.de>, IT Center <http://www.itcenter.com.pt/> and NG Voice <http://www.ng-voice.com/>. Should you consider sponsoring, contact us via email at <conference [at] kamailio.com>**and we can present you more details. The event take place again at the very city center of Berlin, next to Island of Museums and Alexanderplatz. The number of seats is limited and they can fill up very quickly, if you plan to attend, do not delay the registration – see more details in Registration page <http://conference.kamailio.com/k02/registration/> [5]. /Looking forward to meeting many of you in Berlin! / Daniel/ / [1] - http://conference.kamailio.com [2] - http://conference.kamailio.com/k02/speakers/ [3] - http://conference.kamailio.com/k02/schedule/ [4] - http://conference.kamailio.com/k02/sponsors/ [5] - http://conference.kamailio.com/k02/registration/ -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Proxied authentication
by Pete Ashdown 04 Feb '14

04 Feb '14

I'm having trouble divining the proper way to authenticate calls to remote destinations that require it. I'm hoping someone can clear it up. Freeswitch client server A is authenticated to Kamailio. Freeswitch test server B is authenticated to Kamailio. Calls from A -> Kamailio -> B fail with rejection of the authentication information that A is using to Kamailio. As far as I understand, I need to rewrite the authentication information with UAC? However, UAC's uacreg table seems to tie remote users to remote destinations, when all I want to do is authenticate all calls routed to the destination regardless of what the user or the source is. Am I using the wrong module?

2 1

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Shankar 04 Feb '14

04 Feb '14

Hi, Can someone clarify here please? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Monday, February 03, 2014 10:53 AM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: Regd. MAX_LDG_LOCKS Hello, Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if the hash size is increased? Regards, Shankar

1 0

Media-proxy Query
by Nandini madhu 04 Feb '14

04 Feb '14

Dear All, Greetings, I have configured Kamailio (4.0.3) with mediaproxy server, with this set-up audio/video calls are going good and hence NAT issue has solved. I have used two soft clients: 1) IMSDroid 2) Bria. But i have a query about media-proxy statistics i.e in the syslog, the info about media-proxy activity is shown as below: *Case 1*: when Call between two IMSDroid clients. caller_codec : 'Unknown(73)' callee_codec : 'Unknown(73)' callee_ua : 'unknown agent'. *Case 2* : When call between two Bria clients. 'callee_ua': 'unknown agent'. (Calls using Bria clients media-proxy's syslog shows Audio codec names, but using IMSdroid clients it is showing 'Unknown' for Audio codec names ). Find the attachment below for the full syslog about the mentioned unknown behaviour. And also find my kamailio.cfg file. And also logging in the syslog as shown like RTP: Unknown , RTCP: Unknown. In this context, there is a jitter in audio calls and pixelled video sometimes, So is there any wrong with the above mentioned "unknown' factor in this problem of jitter/pixelled audio/video calls ? (AFAIK these payload type 73 (i.e 72-76) is reserved for RTCP conflict avoidance. right? ) What could be the problem for this 'Unknown' issue? How can i solve this issue ? Anything can be done in kamailio script? And how these RTCP payload types (72-76) plays a role in audio/video calls? Any suggestions will greatly help. please help me in clarifying these issues. Thanks in advance. Regards, Nandini

2 3

Buffer Size
by Wingsravi R 04 Feb '14

04 Feb '14

Dear Kamailio'ns I am working on Kamailio (V 4.0)+ Mediaproxy server (2.5.2) running on Ubuntu 12.04. I am experiencing Latency, jitter, pixel led, choppy video errors in my set-up. So i wanted increase buffer sizes in my kamailio server , Mediaproxy server and Ubuntu also. In that regards i have increased Shared memory value to 512 MB and Package memory value 16 MB. BUFFER SIZE value to 8 MB in *config.h* file, But where can i change the value for 'MAX_RECV_BUFFER_SIZE' ? And are there any settings to increase Buffer Size of Mediaproxy server ? Is this changes will affect on my issues ? PS: I even changed ubuntu settings also : net.core.rmem_max= 8388608 net.core.wmem_max= 8388608 (but these values reset to default values again when i restarted system.) Please help me in resolving this issues. Any help will appreciate. Regards, Ravi

1 0

Re: [SR-Users] dialog list query
by John Murray 03 Feb '14

03 Feb '14

From: Charles Chance <charles.chance(a)sipcentric.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] dialog list query Message-ID: Charles hi, >What db_mode are you using for dialog module? Mode 5. I Don't need to persist the dialog state so means table maintained in memory and not flushed to disk until shutdown but I can access it through normal avp_db_query right? >Are the entries in the db if you check manually? If I do sercmd dlg.dlg_list I see the call fine including callid etc. >Are you able to fetch other data using avp_db_query? Yes lots of queries to non dialog tables working fine. >And do you see any errors in the log? No logs show query happening fine and returning Null. Thanks John

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Re: [SR-Users] Kamailio not mdifying Contact in multi-homed setup
by *sm1Ly 03 Feb '14

03 Feb '14

hello, I got the same issue. my sip device getting ok from kamailio, but asterisk reply 401. but. this is my config: http://pastebin.com/jGCak01E and I changed regfwd route to $du = "sip:50.0.0.10:5060;transport=udp"; forward(); and now I see this in logs: Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <200> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <OK> Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <0>; state=16 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global id=24 T start=(nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=006cfccc318be31188fc19977b3a5651 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/parse_addr_spec.c:885]: parse_addr_spec(): end of header reached, state=29 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [63]; uri=[sip:77.37.241.151:5060] Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip: 77.37.241.151:5060>] Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <OPTIONS> *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1045]: t_reply_matching(): DEBUG: t_reply_matching: failure to match a transaction* Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=25 global id=25 T end=(nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header *Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [forward.c:784]: do_forward_reply(): reply cannot be forwarded - no 2nd via* Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Feb 3 15:13:38 kamaz /usr/sbin/kamailio[14502]: DEBUG: <core> [receive.c:296]: receive_msg(): receive_msg: cleaning up any ideas? and show me ur conf please. -- Hi All, I'm running into an issue, I'm not sure whether any of you seen this yourselves and resolved it. Please share some pointers. My network is: clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 (Freeswitch) The 200 OK response from Freeswitch (on the way back from called party to caller) to Kamailio is shown below. Notice the Contact header URI host part contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change that to Public IP before forwarding the 200 OK (copied below) to Caller in public domain. But, it's not. As a result, ACK from Caller is not reaching back to Kamailio. How did you or anybody out there using Kamailio resolve this problem? If needed, I can copy/paste my kamailio.cfg. -- yours respectfully, Alexander Vasin. 8 926 1437200 icq: 9906064

2 1

Change of source ip address
by Diego Alejandro Ozuna Escalada 03 Feb '14

03 Feb '14

Dear list, I'm having a problem. I have an SBC server with two network interfaces, one external IP (X.X.248.194) and other internal IP (X.X.248.21). Messages are sent to the proxy sip server through internal ip with this configuration: route [sip-serverr] { rewritehostport ("ip_sip_proxy: 5060"); force_send_socket (X.X.248.21: 5060); t_relay (); exit; } When a UA sends an INVITE to the SBC, the responses (back) to UA are being sent with the correct external source IP (X.X.248.194) but if there are retransmissions of these responses, after 6 seconds, the source IP changes to the internal ip (X.X.248.21) Does anyone from the list know how I can solve this? Attached picture of wireshark capture for better understanding. -- Diego Alejandro Ozuna

1 0

Security hygiene for Kamailio
by davy 03 Feb '14

03 Feb '14

Hi all, we all enjoy our FAIL2BAN and snippets of our Kamailio config when we see it successfully fight off the "friendly-scanner", and multiple futile attempts to fool our systems. But it got me thinking… What is a sufficient level of security on our Kamailio machinery… ? Are we all just doing whatever, or is the nature of the beast, that every setup is different? Eventually while having a beer, we will end up in the discussion Kamailio is as good (and even much better) as most of the commercially available SBCs. But, imho, that all depends on the configuration. There are a few good reads available, and on the security front I personally love Pike, Topoh, Dnssec, Htable and recently I think I'm doing rather clever stuff with CNXCC… And I do feel comfortable on my setups, them won't be hacked… But do we have a-sort -of stake in the ground example configuration which we can consider as being more than sufficiently secure? Some config where we can tick off all the known security risks for SIP (as chapter 26 of rfc3261 gives a state of the art back in 2002) Or would that be a nice idea for a micro project? Grtz, Davy

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Carrierroute - scan_prefix
by Oliver Roth 03 Feb '14

03 Feb '14

Hi all I need a solution to get the used scan_prefix in carrierroute module. We have different carriers for our destinations, sometimes split up 50%/50% or other splittings - all routes with failure routes. Now if one destination is not reachable, then I would like to get this information and reroute to another carrier. As we use the carrierroute module, we use the "description" to get the carriername (for billing purposes). Is there a solution to get the used scan_prefix from the carrierroute module? Regards, Oli

2 2

kamailio answers instead asterisk
by *sm1Ly 03 Feb '14

03 Feb '14

hello all. this is my kamailio conf: http://pastebin.com/jGCak01E when I try to register any sip device. it looks like. device -> ext IP kamailio -> int IP kamailio -> asterisk. so I trying to nat something to lan. but on asterisk I see that register, it answer 401, than register again and it again answer 401. but on my sip device I see register and than kamailio answer 200 !! why ??? 1. #!KAMAILIO 2. # 3. # Kamailio (OpenSER) SIP Server v4.1 - default configuration script 4. # - web: http://www.kamailio.org 5. # - git: http://sip-router.org 6. # 7. # Direct your questions about this file to: < sr-users(a)lists.sip-router.org> 8. # 9. # Refer to the Core CookBook at http://www.kamailio.org/wiki/ 10. # for an explanation of possible statements, functions and parameters. 11. # 12. # Several features can be enabled using '#!define WITH_FEATURE' directives: 13. # 14. # *** To run in debug mode: 15. # - define WITH_DEBUG 16. # 17. # *** To enable mysql: 18. # - define WITH_MYSQL 19. # 20. # *** To enable authentication execute: 21. # - enable mysql 22. # - define WITH_AUTH 23. # - add users using 'kamctl' 24. # 25. # *** To enable IP authentication execute: 26. # - enable mysql 27. # - enable authentication 28. # - define WITH_IPAUTH 29. # - add IP addresses with group id '1' to 'address' table 30. # 31. # *** To enable persistent user location execute: 32. # - enable mysql 33. # - define WITH_USRLOCDB 34. # 35. # *** To enable presence server execute: 36. # - enable mysql 37. # - define WITH_PRESENCE 38. # 39. # *** To enable nat traversal execute: 40. # - define WITH_NAT 41. # - install RTPProxy: http://www.rtpproxy.org 42. # - start RTPProxy: 43. # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 44. # 45. # *** To enable PSTN gateway routing execute: 46. # - define WITH_PSTN 47. # - set the value of pstn.gw_ip 48. # - check route[PSTN] for regexp routing condition 49. # 50. # *** To enable database aliases lookup execute: 51. # - enable mysql 52. # - define WITH_ALIASDB 53. # 54. # *** To enable speed dial lookup execute: 55. # - enable mysql 56. # - define WITH_SPEEDDIAL 57. # 58. # *** To enable multi-domain support execute: 59. # - enable mysql 60. # - define WITH_MULTIDOMAIN 61. # 62. # *** To enable TLS support execute: 63. # - adjust CFGDIR/tls.cfg as needed 64. # - define WITH_TLS 65. # 66. # *** To enable XMLRPC support execute: 67. # - define WITH_XMLRPC 68. # - adjust route[XMLRPC] for access policy 69. # 70. # *** To enable anti-flood detection execute: 71. # - adjust pike and htable=>ipban settings as needed (default is 72. # block if more than 16 requests in 2 seconds and ban for 300 seconds) 73. # - define WITH_ANTIFLOOD 74. # 75. # *** To block 3XX redirect replies execute: 76. # - define WITH_BLOCK3XX 77. # 78. # *** To enable VoiceMail routing execute: 79. # - define WITH_VOICEMAIL 80. # - set the value of voicemail.srv_ip 81. # - adjust the value of voicemail.srv_port 82. # 83. # *** To enhance accounting execute: 84. # - enable mysql 85. # - define WITH_ACCDB 86. # - add following columns to database 87. #!ifdef ACCDB_COMMENT 88. ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; 89. ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; 90. ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; 91. ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; 92. ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; 93. ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; 94. ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; 95. ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; 96. ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; 97. ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; 98. ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; 99. ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; 100. #!endif 101. 102. # #!define WITH_MYSQL 103. # #!define WITH_AUTH 104. # #!define WITH_USRLOCDB 105. #!define WITH_NAT 106. #!define WITH_ASTERISK 107. # #!define WITH_DEBUG 108. 109. ####### Include Local Config If Exists ######### 110. import_file "kamailio-local.cfg" 111. 112. ####### Defined Values ######### 113. 114. # *** Value defines - IDs used later in config 115. #!ifdef WITH_MYSQL 116. # - database URL - used to connect to database server by modules such 117. # as: auth_db, acc, usrloc, a.s.o. 118. #!ifndef DBURL 119. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" 120. #!endif 121. #!endif 122. #!ifdef WITH_MULTIDOMAIN 123. # - the value for 'use_domain' parameters 124. #!define MULTIDOMAIN 1 125. #!else 126. #!define MULTIDOMAIN 0 127. #!endif 128. 129. # - flags 130. # FLT_ - per transaction (message) flags 131. # FLB_ - per branch flags 132. #!define FLT_ACC 1 133. #!define FLT_ACCMISSED 2 134. #!define FLT_ACCFAILED 3 135. #!define FLT_NATS 5 136. 137. #!define FLB_NATB 6 138. #!define FLB_NATSIPPING 7 139. 140. ####### Global Parameters ######### 141. 142. ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR 143. #!ifdef WITH_DEBUG 144. debug=4 145. log_stderror=yes 146. #!else 147. debug=3 148. log_stderror=no 149. #!endif 150. 151. memdbg=5 152. memlog=5 153. 154. log_facility=LOG_LOCAL0 155. 156. fork=yes 157. children=4 158. 159. /* uncomment the next line to disable TCP (default on) */ 160. #disable_tcp=yes 161. 162. /* uncomment the next line to disable the auto discovery of local aliases 163. based on reverse DNS on IPs (default on) */ 164. #auto_aliases=no 165. 166. /* add local domain aliases */ 167. #alias="sip.mydomain.com" 168. 169. /* uncomment and configure the following line if you want Kamailio to 170. bind on a specific interface/port/proto (default bind on all available) */ 171. 172. #listen=udp:50.0.0.1:5060 advertise 194.190.8.171:5060 173. listen=udp:194.190.8.171:5060 advertise 50.0.0.1:5060 174. 175. /* port to listen to 176. * - can be specified more than once if needed to listen on many ports */ 177. port=5060 178. 179. mhomed=1 180. 181. #!ifdef WITH_TLS 182. enable_tls=yes 183. #!endif 184. 185. # life time of TCP connection when there is no traffic 186. # - a bit higher than registration expires to cope with UA behind NAT 187. tcp_connection_lifetime=3605 188. 189. ####### Custom Parameters ######### 190. 191. # These parameters can be modified runtime via RPC interface 192. # - see the documentation of 'cfg_rpc' module. 193. # 194. # Format: group.id = value 'desc' description 195. # Access: $sel(cfg_get.group.id) or @cfg_get.group.id 196. # 197. 198. #!ifdef WITH_PSTN 199. # PSTN GW Routing 200. # 201. # - pstn.gw_ip: valid IP or hostname as string value, example: 202. # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" 203. # 204. # - by default is empty to avoid misrouting 205. #pstn.gw_ip = "" desc "PSTN GW Address" 206. #pstn.gw_port = "" desc "PSTN GW Port" 207. #!endif 208. 209. #!ifdef WITH_VOICEMAIL 210. # VoiceMail Routing on offline, busy or no answer 211. # 212. # - by default Voicemail server IP is empty to avoid misrouting 213. #voicemail.srv_ip = "" desc "VoiceMail IP Address" 214. #voicemail.srv_port = "5060" desc "VoiceMail Port" 215. #!endif 216. 217. #!ifdef WITH_ASTERISK 218. asterisk.bindip = "50.0.0.10" desc "Asterisk IP Address" 219. asterisk.bindport = "5060" desc "Asterisk Port" 220. kamailio.bindip = "50.0.0.1" desc "Kamailio IP Address" 221. kamailio.bindport = "5060" desc "Kamailio Port" 222. #!endif 223. 224. ####### Modules Section ######## 225. 226. # set paths to location of modules (to sources or installation folders) 227. #!ifdef WITH_SRCPATH 228. mpath="modules/" 229. #!else 230. mpath="/usr/lib64/kamailio/modules/" 231. #!endif 232. 233. #!ifdef WITH_MYSQL 234. loadmodule "db_mysql.so" 235. #!endif 236. 237. loadmodule "mi_fifo.so" 238. loadmodule "kex.so" 239. loadmodule "tm.so" 240. loadmodule "tmx.so" 241. loadmodule "sl.so" 242. loadmodule "rr.so" 243. loadmodule "pv.so" 244. loadmodule "maxfwd.so" 245. loadmodule "usrloc.so" 246. loadmodule "registrar.so" 247. loadmodule "textops.so" 248. loadmodule "siputils.so" 249. loadmodule "xlog.so" 250. loadmodule "sanity.so" 251. loadmodule "ctl.so" 252. loadmodule "cfg_rpc.so" 253. loadmodule "mi_rpc.so" 254. loadmodule "acc.so" 255. 256. #!ifdef WITH_AUTH 257. loadmodule "auth.so" 258. loadmodule "auth_db.so" 259. #!ifdef WITH_IPAUTH 260. loadmodule "permissions.so" 261. #!endif 262. #!endif 263. 264. #!ifdef WITH_ALIASDB 265. loadmodule "alias_db.so" 266. #!endif 267. 268. #!ifdef WITH_SPEEDDIAL 269. loadmodule "speeddial.so" 270. #!endif 271. 272. #!ifdef WITH_MULTIDOMAIN 273. loadmodule "domain.so" 274. #!endif 275. 276. #!ifdef WITH_PRESENCE 277. loadmodule "presence.so" 278. loadmodule "presence_xml.so" 279. #!endif 280. 281. #!ifdef WITH_NAT 282. loadmodule "nathelper.so" 283. loadmodule "rtpproxy.so" 284. #!endif 285. 286. #!ifdef WITH_TLS 287. loadmodule "tls.so" 288. #!endif 289. 290. #!ifdef WITH_ANTIFLOOD 291. loadmodule "htable.so" 292. loadmodule "pike.so" 293. #!endif 294. 295. #!ifdef WITH_XMLRPC 296. loadmodule "xmlrpc.so" 297. #!endif 298. 299. #!ifdef WITH_DEBUG 300. loadmodule "debugger.so" 301. #!endif 302. 303. #!ifdef WITH_ASTERISK 304. loadmodule "uac.so" 305. #!endif 306. 307. 308. #loadmodule "topoh.so" 309. #modparam("topoh", "mask_key", "balalayka") 310. #modparam("topoh", "mask_ip", "50.0.0.1") 311. 312. # ----------------- setting module-specific parameters --------------- 313. 314. 315. # ----- mi_fifo params ----- 316. modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") 317. 318. 319. # ----- tm params ----- 320. # auto-discard branches from previous serial forking leg 321. modparam("tm", "failure_reply_mode", 3) 322. # default retransmission timeout: 30sec 323. modparam("tm", "fr_timer", 30000) 324. # default invite retransmission timeout after 1xx: 120sec 325. modparam("tm", "fr_inv_timer", 120000) 326. 327. 328. # ----- rr params ----- 329. # add value to ;lr param to cope with most of the UAs 330. modparam("rr", "enable_full_lr", 1) 331. # do not append from tag to the RR (no need for this script) 332. #!ifdef WITH_ASTERISK 333. modparam("rr", "append_fromtag", 1) 334. #!else 335. modparam("rr", "append_fromtag", 0) 336. #!endif 337. 338. 339. # ----- registrar params ----- 340. modparam("registrar", "method_filtering", 1) 341. /* uncomment the next line to disable parallel forking via location */ 342. # modparam("registrar", "append_branches", 0) 343. /* uncomment the next line not to allow more than 10 contacts per AOR */ 344. #modparam("registrar", "max_contacts", 10) 345. # max value for expires of registrations 346. modparam("registrar", "max_expires", 3600) 347. # set it to 1 to enable GRUU 348. modparam("registrar", "gruu_enabled", 0) 349. 350. 351. # ----- acc params ----- 352. /* what special events should be accounted ? */ 353. modparam("acc", "early_media", 0) 354. modparam("acc", "report_ack", 0) 355. modparam("acc", "report_cancels", 0) 356. /* by default ww do not adjust the direct of the sequential requests. 357. if you enable this parameter, be sure the enable "append_fromtag" 358. in "rr" module */ 359. modparam("acc", "detect_direction", 0) 360. /* account triggers (flags) */ 361. modparam("acc", "log_flag", FLT_ACC) 362. modparam("acc", "log_missed_flag", FLT_ACCMISSED) 363. modparam("acc", "log_extra", 364. "src_user=$fU;src_domain=$fd;src_ip=$si;" 365. "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") 366. modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) 367. /* enhanced DB accounting */ 368. #!ifdef WITH_ACCDB 369. modparam("acc", "db_flag", FLT_ACC) 370. modparam("acc", "db_missed_flag", FLT_ACCMISSED) 371. modparam("acc", "db_url", DBURL) 372. modparam("acc", "db_extra", 373. "src_user=$fU;src_domain=$fd;src_ip=$si;" 374. "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") 375. #!endif 376. 377. 378. # ----- usrloc params ----- 379. /* enable DB persistency for location entries */ 380. #!ifdef WITH_USRLOCDB 381. modparam("usrloc", "db_url", DBURL) 382. modparam("usrloc", "db_mode", 2) 383. modparam("usrloc", "use_domain", MULTIDOMAIN) 384. #!endif 385. 386. 387. # ----- auth_db params ----- 388. #!ifdef WITH_AUTH 389. modparam("auth_db", "db_url", DBURL) 390. modparam("auth_db", "calculate_ha1", yes) 391. modparam("auth_db", "password_column", "password") 392. modparam("auth_db", "load_credentials", "") 393. modparam("auth_db", "use_domain", MULTIDOMAIN) 394. 395. # ----- permissions params ----- 396. #!ifdef WITH_IPAUTH 397. modparam("permissions", "db_url", DBURL) 398. modparam("permissions", "db_mode", 1) 399. #!endif 400. 401. #!endif 402. 403. 404. # ----- alias_db params ----- 405. #!ifdef WITH_ALIASDB 406. modparam("alias_db", "db_url", DBURL) 407. modparam("alias_db", "use_domain", MULTIDOMAIN) 408. #!endif 409. 410. 411. # ----- speeddial params ----- 412. #!ifdef WITH_SPEEDDIAL 413. modparam("speeddial", "db_url", DBURL) 414. modparam("speeddial", "use_domain", MULTIDOMAIN) 415. #!endif 416. 417. 418. # ----- domain params ----- 419. #!ifdef WITH_MULTIDOMAIN 420. modparam("domain", "db_url", DBURL) 421. # register callback to match myself condition with domains list 422. modparam("domain", "register_myself", 1) 423. #!endif 424. 425. 426. #!ifdef WITH_PRESENCE 427. # ----- presence params ----- 428. modparam("presence", "db_url", DBURL) 429. 430. # ----- presence_xml params ----- 431. modparam("presence_xml", "db_url", DBURL) 432. modparam("presence_xml", "force_active", 1) 433. #!endif 434. 435. 436. #!ifdef WITH_NAT 437. # ----- rtpproxy params ----- 438. modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") 439. 440. # ----- nathelper params ----- 441. modparam("nathelper", "natping_interval", 30) 442. modparam("nathelper", "ping_nated_only", 1) 443. modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) 444. modparam("nathelper", "sipping_from", "sip:pinger@kamaz") 445. 446. # params needed for NAT traversal in other modules 447. modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") 448. modparam("usrloc", "nat_bflag", FLB_NATB) 449. #!endif 450. 451. 452. #!ifdef WITH_TLS 453. # ----- tls params ----- 454. modparam("tls", "config", "//etc/kamailio/tls.cfg") 455. #!endif 456. 457. #!ifdef WITH_ANTIFLOOD 458. # ----- pike params ----- 459. modparam("pike", "sampling_time_unit", 2) 460. modparam("pike", "reqs_density_per_unit", 16) 461. modparam("pike", "remove_latency", 4) 462. 463. # ----- htable params ----- 464. # ip ban htable with autoexpire after 5 minutes 465. modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") 466. #!endif 467. 468. #!ifdef WITH_XMLRPC 469. # ----- xmlrpc params ----- 470. modparam("xmlrpc", "route", "XMLRPC"); 471. modparam("xmlrpc", "url_match", "^/RPC") 472. #!endif 473. 474. #!ifdef WITH_DEBUG 475. # ----- debugger params ----- 476. modparam("debugger", "cfgtrace", 1) 477. #!endif 478. 479. ####### Routing Logic ######## 480. 481. 482. # Main SIP request routing logic 483. # - processing of any incoming SIP request starts with this route 484. # - note: this is the same as route { ... } 485. request_route { 486. 487. # per request initial checks 488. route(REQINIT); 489. 490. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) Main Route before ---NAT---\n"); 491. route(DEBUG); 492. # NAT detection 493. route(NATDETECT); 494. 495. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in Route[NATDETECT] fix_nat-register\n"); 496. route(DEBUG); 497. 498. # CANCEL processing 499. if (is_method("CANCEL")) 500. { 501. if (t_check_trans()) { 502. route(RELAY); 503. } 504. exit; 505. } 506. 507. route(DEBUG); 508. 509. # handle requests within SIP dialogs 510. route(WITHINDLG); 511. 512. route(DEBUG); 513. 514. ### only initial requests (no To tag) 515. 516. t_check_trans(); 517. 518. route(DEBUG); 519. 520. # authentication 521. route(AUTH); 522. 523. route(DEBUG); 524. 525. # record routing for dialog forming requests (in case they are routed) 526. # - remove preloaded route headers 527. remove_hf("Route"); 528. if (is_method("INVITE|SUBSCRIBE")) 529. record_route(); 530. 531. # account only INVITEs 532. if (is_method("INVITE")) 533. { 534. setflag(FLT_ACC); # do accounting 535. } 536. 537. # dispatch requests to foreign domains 538. route(SIPOUT); 539. 540. ### requests for my local domains 541. 542. # handle presence related requests 543. route(PRESENCE); 544. 545. route(DEBUG); 546. 547. # handle registrations 548. route(REGISTRAR); 549. 550. route(DEBUG); 551. 552. if ($rU==$null) 553. { 554. # request with no Username in RURI 555. sl_send_reply("484","Address Incomplete"); 556. exit; 557. } 558. 559. # dispatch destinations to PSTN 560. # route(PSTN); 561. 562. route(DEBUG); 563. 564. # user location service 565. route(LOCATION); 566. 567. route(DEBUG); 568. 569. route(RELAY); 570. 571. } 572. 573. 574. route[RELAY] { 575. 576. # enable additional event routes for forwarded requests 577. # - serial forking, RTP relaying handling, a.s.o. 578. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { 579. if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); 580. } 581. if (is_method("INVITE|SUBSCRIBE|UPDATE")) { 582. if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); 583. } 584. if (is_method("INVITE")) { 585. if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); 586. } 587. 588. if (!t_relay()) { 589. sl_reply_error(); 590. } 591. exit; 592. } 593. 594. # Per SIP request initial checks 595. route[REQINIT] { 596. #!ifdef WITH_ANTIFLOOD 597. # flood dection from same IP and traffic ban for a while 598. # be sure you exclude checking trusted peers, such as pstn gateways 599. # - local host excluded (e.g., loop to self) 600. if(src_ip!=myself) 601. { 602. if($sht(ipban=>$si)!=$null) 603. { 604. # ip is already blocked 605. xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); 606. exit; 607. } 608. if (!pike_check_req()) 609. { 610. xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); 611. $sht(ipban=>$si) = 1; 612. exit; 613. } 614. } 615. #!endif 616. 617. if (!mf_process_maxfwd_header("10")) { 618. sl_send_reply("483","Too Many Hops"); 619. exit; 620. } 621. 622. if(!sanity_check("1511", "7")) 623. { 624. xlog("Malformed SIP message from $si:$sp\n"); 625. exit; 626. } 627. } 628. 629. # Handle requests within SIP dialogs 630. route[WITHINDLG] { 631. if (has_totag()) { 632. # sequential request withing a dialog should 633. # take the path determined by record-routing 634. if (loose_route()) { 635. route(DLGURI); 636. if (is_method("BYE")) { 637. setflag(FLT_ACC); # do accounting ... 638. setflag(FLT_ACCFAILED); # ... even if the transaction fails 639. } 640. else if ( is_method("ACK") ) { 641. # ACK is forwarded statelessy 642. route(NATMANAGE); 643. } 644. else if ( is_method("NOTIFY") ) { 645. # Add Record-Route for in-dialog NOTIFY as per RFC 6665. 646. record_route(); 647. } 648. route(RELAY); 649. } else { 650. if (is_method("SUBSCRIBE") && uri == myself) { 651. # in-dialog subscribe requests 652. route(PRESENCE); 653. exit; 654. } 655. if ( is_method("ACK") ) { 656. if ( t_check_trans() ) { 657. # no loose-route, but stateful ACK; 658. # must be an ACK after a 487 659. # or e.g. 404 from upstream server 660. route(RELAY); 661. exit; 662. } else { 663. # ACK without matching transaction ... ignore and discard 664. exit; 665. } 666. } 667. sl_send_reply("404","Not here"); 668. } 669. exit; 670. } 671. } 672. 673. # Handle SIP registrations 674. route[REGISTRAR] { 675. if (is_method("REGISTER")) 676. { 677. if(isflagset(FLT_NATS)) 678. { 679. setbflag(FLB_NATB); 680. # uncomment next line to do SIP NAT pinging 681. setbflag(FLB_NATSIPPING); 682. } 683. if (!save("location")) 684. sl_reply_error(); 685. 686. 687. #!ifdef WITH_ASTERISK 688. route(REGFWD); 689. #!endif 690. 691. exit; 692. } 693. } 694. 695. 696. #!ifdef WITH_ASTERISK 697. # Test if coming from Asterisk 698. route[FROMASTERISK] { 699. if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; 700. return -1; 701. } 702. 703. 704. # Send to Asterisk 705. route[TOASTERISK] { 706. $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); 707. xlog("L_INFO","[$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); 708. route(RELAY); 709. exit; 710. } 711. 712. # Forward REGISTER to Asterisk 713. route[REGFWD] { 714. if(!is_method("REGISTER")) 715. { 716. return; 717. } 718. 719. xlog("[$mi] PRINT AU: $au PRINT fU: $fU Received SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 720. 721. t_relay_to_udp("50.0.0.10", "5060"); 722. 723. xlog("[$mi] PRINT AU: $au PRINT fU: $fU Sended SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 724. 725. } 726. #!endif 727. 728. # USER location service 729. route[LOCATION] { 730. 731. #!ifdef WITH_SPEEDDIAL 732. # search for short dialing - 2-digit extension 733. if($rU=~"^[0-9][0-9]$") 734. if(sd_lookup("speed_dial")) 735. route(SIPOUT); 736. #!endif 737. 738. #!ifdef WITH_ALIASDB 739. # search in DB-based aliases 740. if(alias_db_lookup("dbaliases")) 741. route(SIPOUT); 742. #!endif 743. 744. #!ifdef WITH_ASTERISK 745. if(is_method("INVITE") && (!route(FROMASTERISK))) { 746. # if new call from out there - send to Asterisk 747. # - non-INVITE request are routed directly by Kamailio 748. # - traffic from Asterisk is routed also directy by Kamailio 749. route(TOASTERISK); 750. exit; 751. } 752. #!endif 753. 754. $avp(oexten) = $rU; 755. if (!lookup("location")) { 756. $var(rc) = $rc; 757. route(TOVOICEMAIL); 758. t_newtran(); 759. switch ($var(rc)) { 760. case -1: 761. case -3: 762. send_reply("404", "Not Found"); 763. exit; 764. case -2: 765. send_reply("405", "Method Not Allowed"); 766. exit; 767. } 768. } 769. 770. # when routing via usrloc, log the missed calls also 771. if (is_method("INVITE")) 772. { 773. setflag(FLT_ACCMISSED); 774. } 775. 776. route(RELAY); 777. exit; 778. } 779. 780. # Presence server route 781. route[PRESENCE] { 782. if(!is_method("PUBLISH|SUBSCRIBE")) 783. return; 784. 785. if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { 786. route(TOVOICEMAIL); 787. # returns here if no voicemail server is configured 788. sl_send_reply("404", "No voicemail service"); 789. exit; 790. } 791. 792. #!ifdef WITH_PRESENCE 793. if (!t_newtran()) 794. { 795. sl_reply_error(); 796. exit; 797. } 798. 799. if(is_method("PUBLISH")) 800. { 801. handle_publish(); 802. t_release(); 803. } else if(is_method("SUBSCRIBE")) { 804. handle_subscribe(); 805. t_release(); 806. } 807. exit; 808. #!endif 809. 810. # if presence enabled, this part will not be executed 811. if (is_method("PUBLISH") || $rU==$null) 812. { 813. sl_send_reply("404", "Not here"); 814. exit; 815. } 816. return; 817. } 818. 819. # Authentication route 820. route[AUTH] { 821. 822. #!ifdef WITH_ASTERISK 823. # do not auth traffic from Asterisk - trusted! 824. if(route(FROMASTERISK)) 825. return; 826. #!endif 827. 828. 829. #!ifdef WITH_AUTH 830. 831. #!ifdef WITH_IPAUTH 832. if((!is_method("REGISTER")) && allow_source_address()) 833. { 834. # source IP allowed 835. return; 836. } 837. #!endif 838. 839. if (is_method("REGISTER") || from_uri==myself) 840. { 841. # authenticate requests 842. if (!auth_check("$fd", "subscriber", "1")) { 843. auth_challenge("$fd", "0"); 844. exit; 845. } 846. # user authenticated - remove auth header 847. if(!is_method("REGISTER|PUBLISH")) 848. consume_credentials(); 849. } 850. # if caller is not local subscriber, then check if it calls 851. # a local destination, otherwise deny, not an open relay here 852. if (from_uri!=myself && uri!=myself) 853. { 854. sl_send_reply("403","Not relaying"); 855. exit; 856. } 857. 858. #!endif 859. return; 860. } 861. 862. # Caller NAT detection route 863. route[NATDETECT] { 864. #!ifdef WITH_NAT 865. force_rport(); 866. if (nat_uac_test("19")) { 867. if (is_method("REGISTER")) { 868. fix_nated_register(); 869. } else { 870. if(is_first_hop()) 871. set_contact_alias(); 872. } 873. setflag(FLT_NATS); 874. 875. } 876. #!endif 877. return; 878. } 879. 880. # RTPProxy control 881. route[NATMANAGE] { 882. #!ifdef WITH_NAT 883. if (is_request()) { 884. if(has_totag()) { 885. if(check_route_param("nat=yes")) { 886. setbflag(FLB_NATB); 887. } 888. } 889. } 890. if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) 891. return; 892. 893. rtpproxy_manage(); 894. 895. if (is_request()) { 896. if (!has_totag()) { 897. if(t_is_branch_route()) { 898. add_rr_param(";nat=yes"); 899. } 900. } 901. } 902. if (is_reply()) { 903. if(isbflagset(FLB_NATB)) { 904. if(is_first_hop()) 905. set_contact_alias(); 906. } 907. 908. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[NATMANAGE] RTPproxy with EI Flags\n"); 909. 910. } 911. #!endif 912. return; 913. } 914. 915. # URI update for dialog requests 916. route[DLGURI] { 917. #!ifdef WITH_NAT 918. if(!isdsturiset()) { 919. handle_ruri_alias(); 920. 921. xlog("L_NOTICE","$rm from $fu (IP:$si:$sp) in route[DLGURI] RTPproxy with EI Flags\n"); 922. 923. } 924. #!endif 925. return; 926. } 927. 928. # Routing to foreign domains 929. route[SIPOUT] { 930. if (!uri==myself) 931. { 932. append_hf("P-hint: outbound\r\n"); 933. route(RELAY); 934. } 935. } 936. 937. # PSTN GW routing 938. route[PSTN] { 939. #!ifdef WITH_PSTN 940. # check if PSTN GW IP is defined 941. if (strempty($sel(cfg_get.pstn.gw_ip))) { 942. xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); 943. return; 944. } 945. 946. # route to PSTN dialed numbers starting with '+' or '00' 947. # (international format) 948. # - update the condition to match your dialing rules for PSTN routing 949. if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) 950. return; 951. 952. # only local users allowed to call 953. if(from_uri!=myself) { 954. sl_send_reply("403", "Not Allowed"); 955. exit; 956. } 957. 958. if (strempty($sel(cfg_get.pstn.gw_port))) { 959. $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); 960. } else { 961. $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" 962. + $sel(cfg_get.pstn.gw_port); 963. } 964. 965. route(RELAY); 966. exit; 967. #!endif 968. 969. return; 970. } 971. 972. # XMLRPC routing 973. #!ifdef WITH_XMLRPC 974. route[XMLRPC] { 975. # allow XMLRPC from localhost 976. if ((method=="POST" || method=="GET") 977. && (src_ip==127.0.0.1)) { 978. # close connection only for xmlrpclib user agents (there is a bug in 979. # xmlrpclib: it waits for EOF before interpreting the response). 980. if ($hdr(User-Agent) =~ "xmlrpclib") 981. set_reply_close(); 982. set_reply_no_connect(); 983. dispatch_rpc(); 984. exit; 985. } 986. send_reply("403", "Forbidden"); 987. exit; 988. } 989. #!endif 990. 991. # route to voicemail server 992. route[TOVOICEMAIL] { 993. #!ifdef WITH_VOICEMAIL 994. if(!is_method("INVITE|SUBSCRIBE")) 995. return; 996. 997. # check if VoiceMail server IP is defined 998. if (strempty($sel(cfg_get.voicemail.srv_ip))) { 999. xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); 1000. return; 1001. } 1002. if(is_method("INVITE")) { 1003. if($avp(oexten)==$null) 1004. return; 1005. $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) 1006. + ":" + $sel(cfg_get.voicemail.srv_port); 1007. } else { 1008. if($rU==$null) 1009. return; 1010. $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) 1011. + ":" + $sel(cfg_get.voicemail.srv_port); 1012. } 1013. route(RELAY); 1014. exit; 1015. #!endif 1016. 1017. return; 1018. } 1019. 1020. # manage outgoing branches 1021. branch_route[MANAGE_BRANCH] { 1022. xdbg("new branch [$T_branch_idx] to $ru\n"); 1023. route(NATMANAGE); 1024. } 1025. 1026. # manage incoming replies 1027. onreply_route[MANAGE_REPLY] { 1028. xdbg("incoming reply\n"); 1029. if(status=~"[12][0-9][0-9]") 1030. route(NATMANAGE); 1031. } 1032. 1033. # manage failure routing cases 1034. failure_route[MANAGE_FAILURE] { 1035. route(NATMANAGE); 1036. 1037. if (t_is_canceled()) { 1038. exit; 1039. } 1040. 1041. #!ifdef WITH_BLOCK3XX 1042. # block call redirect based on 3xx replies. 1043. if (t_check_status("3[0-9][0-9]")) { 1044. t_reply("404","Not found"); 1045. exit; 1046. } 1047. #!endif 1048. 1049. #!ifdef WITH_VOICEMAIL 1050. # serial forking 1051. # - route to voicemail on busy or no answer (timeout) 1052. if (t_check_status("486|408")) { 1053. $du = $null; 1054. route(TOVOICEMAIL); 1055. exit; 1056. } 1057. #!endif 1058. } 1059. 1060. 1061. route[DEBUG] { 1062. if (method==("PUBLISH|SUBSCRIBE|REGISTER|OPTIONS")) { 1063. return; 1064. } 1065. xlog("[$mi] Received SIP Message (method: $rm) ($ml bytes) to $Ri:$Rp from $si:$sp:\n$mb\n"); 1066. } -- С уважением, Александр Васин. 8 926 1437200 icq: 9906064

1 0

ngcp-mediaproxy-ng media address
by Kelvin Chua 03 Feb '14

03 Feb '14

somehow, mediaproxy is ignoring the media address set via rtpproxy_manage("co","10.17.0.102"); anybody tried this before? Jan 31 09:03:29 kam2 mediaproxy-ng[20455]: Got valid command from 127.0.0.1:46077: answer - { "sdp": "v=0#015#012o=root 345956998 345956998 IN IP4 10.17.0.105#015#012s=Asterisk PBX 11.7.0#015#012c=IN IP4 10.17.0.105#015#012t=0 0#015#012m=audio 15712 RTP/AVP 97 3 0 8 101#015#012a=rtpmap:97 speex/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=sendrecv#015#012", "replace": [ "session-connection", "origin" ], "call-id": "yqtupxcwsnbmcbo@X340saucy", "received-from": [ "IP4", "10.17.0.105" ], "media address": "22.9.18.15", "from-tag": "kjoel", "to-tag": "as478f57f3", "command": "answer" } Jan 31 09:03:29 kam2 mediaproxy-ng[20455]: Returning to SIP proxy: d3:sdp541:v=0#015#012o=root 345956998 345956998 IN IP4 10.17.0.102#015#012s=Asterisk PBX 11.7.0#015#012c=IN IP4 10.17.0.102#015#012t=0 0#015#012a=ice-lite#015#012m=audio 10002 RTP/AVP 97 3 0 8 101#015#012a=rtpmap:97 speex/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:10003#015#012a=ice-ufrag:QocOqhlR#015#012a=ice-pwd:HZ7WUgsHAKlLGp1n2FAzEwWsS9gj#015#012a=candidate:pMoXfVtH0dDfxpoa 1 UDP 2130706432 10.17.0.102 10002 typ host#015#012a=candidate:pMoXfVtH0dDfxpoa 2 UDP 2130706431 10.17.0.102 10003 typ host#015#0126:result2:oke Kelvin Chua

2 4

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Shankar 03 Feb '14

03 Feb '14

Hi Alex, We set the hash size to 32 MB and found that the hash size is not the same as configured one. At the max it was taking 64KB. Regards, Shankar Date: Mon, 03 Feb 2014 00:27:29 -0500 From: Alex Balashov <abalashov(a)evaristesys.com> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Regd. MAX_LDG_LOCKS Message-ID: <52EF28C1.5010204(a)evaristesys.com> Content-Type: text/plain; charset=windows-1252; format=flowed On 02/03/2014 12:22 AM, Shankar wrote: > Also we found that in ?dialog.c? the variable dlg_hash_size should be > ?unsigned int?. Hope this will be taken care in next release of Kamailio. How did you arrive at that conclusion? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ From: Shankar [mailto:shankar.rk@plintron.com] Sent: Monday, February 03, 2014 10:53 AM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: Regd. MAX_LDG_LOCKS Hello, Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if the hash size is increased? Also we found that in 'dialog.c' the variable dlg_hash_size should be 'unsigned int'. Hope this will be taken care in next release of Kamailio. Regards, Shankar

1 0

Re: [SR-Users] Regd. MAX_LDG_LOCKS
by Alex Balashov 03 Feb '14

03 Feb '14

On 02/03/2014 12:22 AM, Shankar wrote: > Also we found that in ‘dialog.c’ the variable dlg_hash_size should be > ‘unsigned int’. Hope this will be taken care in next release of Kamailio. How did you arrive at that conclusion? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

Regd. MAX_LDG_LOCKS
by Shankar 03 Feb '14

03 Feb '14

Hello, Can anyone brief about MAX_LDG_LOCKS in dlg_hash.c? Should I increase the value of MAX_LDG_LOCKS (hardcoded to 2048) if the hash size is increased? Also we found that in 'dialog.c' the variable dlg_hash_size should be 'unsigned int'. Hope this will be taken care in next release of Kamailio. Regards, Shankar

1 0

Kamailio behind DSL router
by Muhammad Shahzad 03 Feb '14

03 Feb '14

Hi, I am setting up Raspberry Pi to run Kamailio with mediaproxy-ng. This machine is running on local LAN behind a DSL router. Using dynamic DNS and DMZ services of router, i can access this box from the Internet. However, i do not know how to define advertised_address parameter to public IP of router. The public IP is dynamic and changes roughly every 12 hours or so. Is there any way kamailio can automatically learn this dynamic public IP and use that as advertised_address? Something similar to Asterisk's res_stun_monitor module, https://reviewboard.asterisk.org/r/854/ Any help will be greatly appreciated. Thank you.

2 6

Packet loss / Jitter / Latency issues
by Wingsravi R 02 Feb '14

02 Feb '14

Dear Daniel and Kamailio'ns Greetings, I am working on kamailio server (4.0.x),whicch is integreted with media-proxy server (2.5.2) and it is all working fine in the point of SIP functionality. And i have a set-up like this below, with this when the calls (audio/video) sessions started i am experiencing latency,jitter, echo's in audio calls and a choppy ,pixelled video : KAMAILIO | Android SIP client (IMSDroid) ---- Wi-Fi router ---- Switch (level 2) ---- Wi-Fi router ---- Android SIP client (IMSDroid) (Total set-up runs on Intranet infrastructure). When i call between two SIP clients, Calls are getting established successfully and RTP flow is also relaying using Media-proxy server. But the audio/video performnce is not good enough. I am experiencing all this Latency , jitter , Choppy pixelled video. How can i resolve all this issues? What could be the problem ? Ofcourse SIP proxy is not responsible all this problems (as i know ), But is there anything can be done on Media-proxy server ? Or anything Buffer settings can be done on Kamailio server side (or anything else) ? Even i changed Shared memory and Package memories to 512MB and 16MB respectively. Anybody can guess what may be the problems that causes these issues ? PS: When i tried 'ping' Clients IP addresses in Kamailio server PC, there is 20-30% packet loss also. Please help me in resolving these issues. Any help will greatly appreciate. Regards, Ravi.

4 3

Re: [SR-Users] dialog list query
by Charles Chance 01 Feb '14

01 Feb '14

John, What db_mode are you using for dialog module? Are the entries in the db if you check manually? Are you able to fetch other data using avp_db_query? And do you see any errors in the log? Regards, Charles On 1 Feb 2014 16:24, "John Murray" <john.murray(a)skyracktelecom.com> wrote: > Hello, > > > > I am managing calls with the dialog module in kamailio 4.0.4. > > > > However if is try to get the callid and from_tag of existing calls using: > > > > avp_db_query("select callid, from_tag from dialog", "$avp(s:s_callid), > $avp(s:s_from_tag)"); > > > > I get NULL, yet if I use: > > > > sercmd proxy dlg.list > > > > I get the calls listed correctly. > > > > What am I doing wrong? > > > > Thanks > > > > John > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- www.sipcentric.com Follow us on twitter @sipcentric <http://twitter.com/sipcentric> Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ.

1 0

dialog list query
by John Murray 01 Feb '14

01 Feb '14

Hello, I am managing calls with the dialog module in kamailio 4.0.4. However if is try to get the callid and from_tag of existing calls using: avp_db_query("select callid, from_tag from dialog", "$avp(s:s_callid), $avp(s:s_from_tag)"); I get NULL, yet if I use: sercmd proxy dlg.list I get the calls listed correctly. What am I doing wrong? Thanks John

1 0

Kamailio without tls
by Neville D'Souza 01 Feb '14

01 Feb '14

Hi Support, I have recently kamailio 4.0.x version with tls on port 5061. Now our development team needs to connect to it using unencrypted on port 5060 to test an app for features and then work towards connecting through tls connection. So my question is how do I temporarily disable tls to test all the features on the kamailio server and then once feature testing is done revert back to tls connection? Best Regards, Neville D'Souza

2 1

Re: [SR-Users] [Kamailio-Business] Fosdem this year
by Ivo V 01 Feb '14

01 Feb '14

Hi, My 2 colleague and me (3 total) like to join the Kamailio dinner. We went to Beermania 2 years ago with the Jitsi guys and DanB and I can confirm the perfect fries! Ivo

4 6

kamailio 4.1.1 failure routes issue
by José Seabra 01 Feb '14

01 Feb '14

Hello there, I think that i found a bug within failure routes on version 4.1.1, i have a call flow that needs to use the failure routes 4 times if nobody answer the call, but when the call goes to the 3º attempt to same failure route kamailio doesn't handle the transaction, With the same kamailio script but using the version 4.0.1 kamailio works correctly. Please check the attachment with kamailio debug level 3 -- Cumprimentos José Seabra

1 0

Kamailio not mdifying Contact in multi-homed setup
by Biswas, Dipak 31 Jan '14

31 Jan '14

Hi All, I'm running into an issue, I'm not sure whether any of you seen this yourselves and resolved it. Please share some pointers. My network is: clients <--> Public IP(Kamailio/RTPProxy)10.1.128.11 <--> 10.1.128.34 (Freeswitch) The 200 OK response from Freeswitch (on the way back from called party to caller) to Kamailio is shown below. Notice the Contact header URI host part contains Freeswitch Private IP (10.1.128.34). Kamailio suppose to change that to Public IP before forwarding the 200 OK (copied below) to Caller in public domain. But, it's not. As a result, ACK from Caller is not reaching back to Kamailio. How did you or anybody out there using Kamailio resolve this problem? If needed, I can copy/paste my kamailio.cfg. SIP 200 OK -> Via: SIP/2.0/UDP 10.1.128.11;branch=z9hG4bKa7ea.4013d9881c1b7fe7b4c6c0f0e8f9d6b6.0 Via: SIP/2.0/UDP <Caller Public IP>:5060;rport=5060;branch=z9hG4bK-383736-9b4621118533d3ccea17992738433249 Record-Route: <sip:10.1.128.11;r2=on;lr=on;ftag=791b5ae3;nat=yes> Record-Route: <sip:<Kamailio Pub IP>;r2=on;lr=on;ftag=791b5ae3;nat=yes> From: \"Dipak Biswas\" <sip:dipak.biswas@<Kamailio Pub IP>>;tag=791b5ae3 To: <sip:dipak.biswas.test2@<Kamailio Pub IP>>;tag=atXF5443gQj9p Call-ID: 20d5d6d366fcb06c259a0895b3e44b52@0:0:0:0:0:0:0:0 CSeq: 2 INVITE Contact: <sip:dipak.biswas.test2@10.1.128.34:5060;transport=udp> User-Agent: FreeSWITCH mod_sofia/1.4.2+git~20140108T200418Z~d8fc8469b4~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 269 -- Thanks, Dipak

1 0

sip reroute question
by Øystein Rudi 31 Jan '14

31 Jan '14

Hi, I have a scenario where multiple companies with multiple sip and smtp domains want to also use one common domain. Ie. Users from domain1.com have their own Exchange and Lync topology, this is also true for users from domain domain2.com Is there a way to use SER on commondomain.com to forward Lync traffic, presence, chat, rtp to these servers/users B.r, Øystein Rudi Teknisk leder (CTO) / Seljestadveien 78 A, 9407 Harstad sentralbord: +47 77 00 27 50 / fax: +47 77 00 27 51 / web: www.serit.no epost: oystein.rudi(a)itpartner.no / direkte: +47 77 00 27 81 / mobil: +47 93 00 89 99 [Serit_ITPartner_Harstad]

2 1

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