- sr-users - kamailio.org

kamailio 4.0.4 '408' response
by kamailio＠AaronLux.com 08 Nov '13

08 Nov '13

Below please find a filtered SIP packet capture showing a problem I'm having with callee '408' responses in kamailio 4.0.4. The '408' response occurs after both caller and callee successfully establish 2-way audio. In this setup kamailio is running behind a NAT and both SIP clients connect via WAN and rtpproxy is started with the following command 'rtpproxy -F -l 192.168.1.LAN -A 66.41.221.WAN -s unix:/var/run/rtpproxy.sock -d DBUG:LOG_LOCAL0 -m 10000 -M 20000' Ports 5061 and Ports 10000-20000 are forwarded to the kamailio server and kamailio has TLS enabled. Please let me know if you know if you need more info. Thanks, Aaron # Filtered SIP capture showing only 408 response: http://pastebin.com/pBcZNGA4 # rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Extension 20090810: Support for automatic bridging # kamailio -v version: kamailio 4.0.4 (i386/linux) cabe58 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: cabe58 compiled on 01:07:11 Nov 2 2013 with gcc 4.4.7

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1 via header
by phillman25 08 Nov '13

08 Nov '13

Dear List I am trying to interconnect with a Cisco Call manager express, i am sending the INVITE message with the below format: INVITE sip:22775019@81.21.39.153 SIP/2.0 Record-Route: <sip:81.21.38.33;lr=on;ftag=685329680;did=671.51a2> Via: SIP/2.0/UDP 81.21.38.33;branch=z9hG4bKda1d.291eab1622fb619ec7df3ca647581a19.0 Via: SIP/2.0/UDP 81.21.35.22:5060;branch=z9hG4bKterm-4a32-99665015-6855A22775019-37284 From: 99665015 <sip:99665015@81.21.35.22;user=phone>;tag=685329680 To: 6855A22775019 <sip:6855A22775019@81.21.38.33;user=phone> Call-ID: 6fea2437-2d338fcf-4efc6b90-2d08(a)81.21.35.22 CSeq: 1 INVITE Supported: timer Session-Expires: 1800 Min-SE: 1800 Contact: <sip:99665015@81.21.35.22:5060> Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Max-Forwards: 69 Content-Type: application/sdp Content-Length: 528 P-hint: outbound v=0 o=- 3099173 0 IN IP4 81.21.35.44 s=Cisco SDP 0 c=IN IP4 81.21.35.44 t=0 0 m=audio 52168 RTP/AVP 0 8 18 99 102 103 104 4 105 3 106 107 108 125 101 100 a=rtpmap:99 G.726-16/8000 a=rtpmap:102 G.726-24/8000 a=rtpmap:103 G.726-32/8000 a=rtpmap:104 G.723.1-H/8000 a=rtpmap:105 G.723.1-L/8000 a=rtpmap:106 G.729b/8000 a=rtpmap:107 G.723.1a-H/8000 a=rtpmap:108 G.723.1a-L/8000 a=rtpmap:125 G.nX64/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=nortpproxy:yes The Cisco Manager Express replies by sending the TRYING message in the below format with 1 via header: SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.21.38.33;branch=z9hG4bKda1d.291eab1622fb619ec7df3ca647581a19.0,SIP/2.0/UDP 81.21.35.22:5060 ;branch=z9hG4bKterm-4a32-99665015-6855A22775019-37284 From: 99665015 <sip:99665015@81.21.35.22;user=phone>;tag=685329680 To: 6855A22775019 <sip:6855A22775019@81.21.38.33;user=phone> Date: Tue, 05 Nov 2013 18:24:17 GMT Call-ID: 6fea2437-2d338fcf-4efc6b90-2d08(a)81.21.35.22 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Seems like Kamailio is not accepting this, is the Trying message from the client correct? Is it correct that he is sending 1 via header with 2 IP's as opposed to 2 via headers with 1 IP in each via header? Otherwise is there a solution for this? Thanks in advance Phillip

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Where do I place the logic for modifying auth and call requests?
by erik＠valerius.se 08 Nov '13

08 Nov '13

Hello everyone. Some of my confusion probably comes from me coming from the web development world and not software dev, but I would greatly appreciate any response to the following question: I use the MySQL auth for my subscribers which is working great. However, let's say that I add the column "valid_until" in the subscriber table and want to check if today's date is less than the caller's valid_until value. I can set up the sqlops module without a problem and get the actual value, but I have no idea where I place the logic to compare it to a date and then pass on the OK message to Kamailio to continue the request. Is this done with embedded scripts? Or right there in the config file? Where can I find a list of what parts of the call request are exposed for customization from Kamailio? I'd greatly prefer to do things in Python or JavaScript, if that matters. Many thanks in advance! Erik Eklund

2 2

RTPproxy support in Wireshark
by Peter Lemenkov 08 Nov '13

08 Nov '13

Hello All! First of all I'm sorry for the crossposting. For those who unaware of what's going inside Wireshark project - I'm working on a "classic" RTPproxy protocol dissector. Most of the patches already accepted, and Wireshark's RTPproxy parser is already usable. Few more patches should land in trunk quite soon. As for rtpproxy-ng - I plan to add support for its protocol as well (right after I finish with this one). If you're using Fedora Linux, then Wireshark from the repository already compiled with RTPproxy protocol dissection support. -- With best regards, Peter Lemenkov.

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Mitigation of unavailable rtpproxy
by Alex Balashov 07 Nov '13

07 Nov '13

Hello, (Sorry for cross-posting to -users and -dev; not really sure where this post belongs most.) A few days ago, I ran into an issue with a Kamailio server being somewhat unresponsive, during moderate call volume, on account of a down rtpproxy--the only rtpproxy in the set. This is rtpproxy classic, not ngcp-mediaproxy-ng. Rtpproxy was not actually engaged on any of the initial INVITEs going through the server; the server is configured to invoke it conditionally based on a setting, and the setting was not set for any endpoints. rtpproxy_manage() was never called. However, I call unforce_rtp_proxy() unconditionally in my config when handling CANCELs, reasoning that it can't do any harm if rtpproxy_manage() was not called before[1]. Nevertheless, it seemed to be the case that this situation was clogging up SIP worker threads, because some SIP messages were definitely dropped. Periodic log messages about inability to reach the rtpproxy were echoed as well. This problem cleared up almost immediately when the rtpproxy instance was restored into service. This raised some questions in my mind about the relationship between rtpproxy management and SIP worker thread utilisation. I assume it was my indiscriminate unforce_rtp_proxy() calls that were actually clogging up the worker threads, right? If so, why? I figured that in the unforce_rtp_proxy() case, the rtpproxy module simply sends fire-and-forget UDP messages down the UDP control socket without any sort of blocking for acknowledgement, since in this case the call must be released on the rtpproxy side without doing any rewriting of SDP on the Kamailio side (unlike in the case where rtpproxy is engaged). Thus, there should be no need to wait for ports to substitute into the message. Or is the same response-wait mechanism used regardless, even in the unforce_rtp_proxy() case, for programmatic reasons? More broadly, is there any way that this scenario can be prevented? In other words, is there a way to work around an outage of all rtpproxies in the set without tying up workers, or at least tying them up less severely? Thanks! -- Alex [1] Is this a reasonable assumption? The reason I do this is that I don't see a way to find out if rtpproxy was engaged from the body of a CANCEL message. I do check for a ;proxy_media RR parameter when handling BYEs, but since a CANCEL is not an in-dialog request, I'm not sure what to do except to call unforce_rtp_proxy()/rtpproxy_manage() indiscriminately, without resorting to storing state in htable or other complications I don't want. -- 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/

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Kamctl ping works as my unix user, not in root nor kamailio user
by Benjamin Henrion 07 Nov '13

07 Nov '13

Hi, I am trying to find out why kamctl ping sip:1.2.3.4:5060 works with my unix user (got a 200 OK), while it does not with kamailio nor root user (I got a 408 timeout). Any idea? The permissions of the socket in /tmp are like this: ======================================= srw------- 1 root root 0 Oct 31 01:01 kamailio_ctl prw-rw---- 1 kamailio kamailio 0 Nov 7 13:55 kamailio_fifo srw-rw-rw- 1 kamailio kamailio 0 Oct 31 01:01 kamailio_socket ======================================= Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-4148403 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

2 1

SIP registration
by Mudasser 07 Nov '13

07 Nov '13

Hello I am devloping the chat/call application like viber,whatsapp etc in Android using sip. I want to know how i can register the sip account with ur sevrer. what's code which i need to write in my application.suppose if i want to chat or call between other person how i can get register without telling user that i am going for sip registration means there i cam not going to display sip registration form etc. where i can save user message and how i can maintain user info like dispaly name , display pic etc.is there any serverside databse involvement to save messages,pic etc or sip sever maintain automatically. Kindly describe in details as i am new in Android Thank you Best Regards Mudasser Tufail Senior Software Engineer

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Re: [SR-Users] #cfgutils: sleep(x) does not work in failure_route
by Klaus Feichtinger 07 Nov '13

07 Nov '13

Hello Daniel, I´ve patched the code and found that the return value of the 'sleep' command is - as expected - differing in the scenarios. For scenario 1 (no provisional response), the return value is '0' and for scenario 2, the return value is '2' (see log output): scenario 1: Nov 7 12:40:51 sipsrvnode1 kamailio33[10974]: INFO: -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s Nov 7 12:40:51 sipsrvnode1 kamailio33[10974]: DEBUG: cfgutils [cfgutils.c:650]: sleep 2 seconds Nov 7 12:40:53 sipsrvnode1 kamailio33[10974]: DEBUG: cfgutils [cfgutils.c:652]: sleep-rc 0 Nov 7 12:40:53 sipsrvnode1 kamailio33[10974]: INFO: -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s scenario 2: Nov 7 12:41:38 sipsrvnode1 kamailio33[10975]: INFO: -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s Nov 7 12:41:38 sipsrvnode1 kamailio33[10975]: DEBUG: cfgutils [cfgutils.c:650]: sleep 2 seconds Nov 7 12:41:38 sipsrvnode1 kamailio33[10975]: DEBUG: cfgutils [cfgutils.c:652]: sleep-rc 2 Nov 7 12:41:38 sipsrvnode1 kamailio33[10975]: INFO: -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s This means that the sleep command is interrupted (in scenario 2) by a signal, as the return value is representing the "number of seconds left to sleep, if the call was interrupted by a signal handler.". Which signal is interrupting "sleep" in this scenario (only) :-( ? Klaus > Daniel-Constantin Mierla <miconda(a)gmail.com> hat am 7. November 2013 um 10:36 > geschrieben: > > Hello, > > can you try to see if sleep_us() works? > > It is rather strange because sleep() from cfgutils just uses sleep() from > system library, nothing special internally... sleep is interrupted by signals > as well. Can you patch the function in kamailio to log the return code from > sleep()? > > Cheers, > Daniel > > On 11/7/13 8:40 AM, klaus.lists#inode.at wrote: > > > > Hello, > > > > I have an update: > > > > today I´ve tested with kamailio version 4.0.4, but the behaviour is > > still the same: it is working fine only when no response has been received; > > as soon as a provisional response is received, the failure route is ignoring > > the "sleep" function. > > > > Klaus > > > > > > > Hello list, > > > > > > I have troubles using the function "usleep(x)" or "sleep(x)" (the > > > behaviour is the same) of the "cfgutils" module > > > (<http://kamailio.org/docs/modules/3.2.x/modules_k/cfgutils.html#idp76968> > > > ) in the failure_route. According documentation, this function could be > > > used in any route, including the failure_route, too. However, when I call > > > this function, it does not show any reaction, if a transaction was already > > > answered with a provisional response. It is still working fine, when a > > > transaction is timing out, as no target socket is reachable. This is > > > confusing me! > > > > > > Please find below some xlog messages from my configuration file in > > > two scenarios (function "t_set_fr(20000, 10000);" is identic in both > > > scenarios): > > > (1) scenario 1 is including a primary target, looked up in the > > > registrar DB, which is unreachable; after transaction timeout I have > > > inserted the function "sleep()" in the failure_route for forcing a delay > > > => here it is working fine (see timestamp) > > > > > > (2) scenario 2 is including a primary target which is responding, > > > but not establishing the session; after transaction timeout the procedure > > > is still the same as in scenario (1) > > > => here is no delay visible in the log and practically detectable > > > > > > From my point of view, this is a malfuntion! Does anybody see it > > > different? I´ve tested these scenarios with kamailio-3.2.x and > > > kamailio-3.3.x - no difference. > > > > > > Thanks for any hints! > > > > > > Klaus > > > > > > ################# SCENARIO (1) ORIG TARGET IS UNREACHABLE > > > ########################## > > > Nov 6 15:58:25 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: > > > -<|XLOG|>-: <RELAY> is reached for RU:<sip:117002@10.16.48.226:5678> , > > > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > > > INVITE, Call-ID: 1107651805(a)10.16.48.71 <mailto:1107651805@10.16.48.71> > > > > > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: > > > -<|XLOG|>-: <FAILRELAY> is reached with Code: 408 > > > From:<sip:1101015555@10.16.48.71> To:<sip:115300@10.16.48.44> > > > > > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: > > > -<|XLOG|>-: failure_route <FAILRELAY> is reached because of a > > > BRANCH_TIMEOUT of RU:<sip:117002@10.16.48.226:5678> > > > > > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: > > > -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s > > > > > > Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: > > > -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s > > > > > > Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: > > > -<|XLOG|>-: <RELAYFB> is reached for RU:<sip:117003@10.16.48.226:7001> , > > > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > > > INVITE, Call-ID: 1107651805(a)10.16.48.71 <mailto:1107651805@10.16.48.71> > > > > > > ################# SCENARIO (2) ORIG TARGET IS REACHABLE > > > ########################## > > > > > > Nov 6 16:02:14 sipsrvnode1 /usr/sbin/kamailio[29034]: INFO: > > > -<|XLOG|>-: <RELAY> is reached for RU:<sip:117002@10.16.48.226:5678> , > > > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > > > INVITE, Call-ID: 553330101(a)10.16.48.71 <mailto:553330101@10.16.48.71> > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: <FAILRELAY> is reached with Code: 408 > > > From:<sip:5555@10.16.48.71> To:<sip:4000@10.16.48.44> > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: failure_route <FAILRELAY> is reached because of a > > > BRANCH_TIMEOUT of RU:<sip:117002@10.16.48.44> > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s > > > > > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: > > > -<|XLOG|>-: <RELAYFB> is reached for RU:<sip:117003@10.16.48.44> , > > > From:<sip:5555@10.16.48.71> , To:<sip:4000@10.16.48.44> , Method: INVITE, > > > Call-ID: 553330101(a)10.16.48.71 <mailto:553330101@10.16.48.71> > > > > > > > > > > > > > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > > <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > > > > > > -- > Daniel-Constantin Mierla - <http://www.asipto.com> > <http://twitter.com/#!/miconda> - <http://www.linkedin.com/in/miconda> > Kamailio Advanced Trainings - Berlin, Nov 25-28 > - more details about Kamailio trainings at <http://www.asipto.com> - >

1 0

IM Broadcasting
by sipatse＠in.gr 07 Nov '13

07 Nov '13

1 0

Function to test DB connection from route[]?
by Allen Zhang 07 Nov '13

07 Nov '13

Hi all, Is there a function I can use to test database connection from the routing script in the cfg file? My dispatcher uses OPTIONS probing. When the destination receives the OPTIONS request, I'd like to detect the health status of the database connection before sending a 200 back. Note the OPTIONS request doesn't have a user part in the RURI. Hence, is_subscriber() and alias_db_lookup() don't work. Any suggestion? Regards, Allen Zhang

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Re: [SR-Users] #cfgutils: sleep(x) does not work in failure_route
by klaus.lists#inode.at 07 Nov '13

07 Nov '13

Hello, I have an update: today I´ve tested with kamailio version 4.0.4, but the behaviour is still the same: it is working fine only when no response has been received; as soon as a provisional response is received, the failure route is ignoring the "sleep" function. Klaus > Hello list, > > I have troubles using the function "usleep(x)" or "sleep(x)" (the behaviour > is the same) of the "cfgutils" module > (<http://kamailio.org/docs/modules/3.2.x/modules_k/cfgutils.html#idp76968> ) > in the failure_route. According documentation, this function could be used in > any route, including the failure_route, too. However, when I call this > function, it does not show any reaction, if a transaction was already answered > with a provisional response. It is still working fine, when a transaction is > timing out, as no target socket is reachable. This is confusing me! > > Please find below some xlog messages from my configuration file in two > scenarios (function "t_set_fr(20000, 10000);" is identic in both scenarios): > (1) scenario 1 is including a primary target, looked up in the registrar DB, > which is unreachable; after transaction timeout I have inserted the function > "sleep()" in the failure_route for forcing a delay > => here it is working fine (see timestamp) > > (2) scenario 2 is including a primary target which is responding, but not > establishing the session; after transaction timeout the procedure is still the > same as in scenario (1) > => here is no delay visible in the log and practically detectable > > From my point of view, this is a malfuntion! Does anybody see it different? > I´ve tested these scenarios with kamailio-3.2.x and kamailio-3.3.x - no > difference. > > Thanks for any hints! > > Klaus > > ################# SCENARIO (1) ORIG TARGET IS UNREACHABLE > ########################## > Nov 6 15:58:25 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: -<|XLOG|>-: > <RELAY> is reached for RU:<sip:117002@10.16.48.226:5678> , > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > INVITE, Call-ID: 1107651805(a)10.16.48.71 <mailto:1107651805@10.16.48.71> > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: > <FAILRELAY> is reached with Code: 408 From:<sip:1101015555@10.16.48.71> > To:<sip:115300@10.16.48.44> > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: > failure_route <FAILRELAY> is reached because of a BRANCH_TIMEOUT of > RU:<sip:117002@10.16.48.226:5678> > > Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: > <FAILRELAY> s l e e p 2000ms / 2s > > Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: > <FAILRELAY> s l e p t 2000ms / 2s > > Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: -<|XLOG|>-: > <RELAYFB> is reached for RU:<sip:117003@10.16.48.226:7001> , > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > INVITE, Call-ID: 1107651805(a)10.16.48.71 <mailto:1107651805@10.16.48.71> > > ################# SCENARIO (2) ORIG TARGET IS REACHABLE > ########################## > > Nov 6 16:02:14 sipsrvnode1 /usr/sbin/kamailio[29034]: INFO: -<|XLOG|>-: > <RELAY> is reached for RU:<sip:117002@10.16.48.226:5678> , > From:<sip:1101015555@10.16.48.71> , To:<sip:115300@10.16.48.44> , Method: > INVITE, Call-ID: 553330101(a)10.16.48.71 <mailto:553330101@10.16.48.71> > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > <FAILRELAY> is reached with Code: 408 From:<sip:5555@10.16.48.71> > To:<sip:4000@10.16.48.44> > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > failure_route <FAILRELAY> is reached because of a BRANCH_TIMEOUT of > RU:<sip:117002@10.16.48.44> > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > <FAILRELAY> s l e e p 2000ms / 2s > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > <FAILRELAY> s l e p t 2000ms / 2s > > Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: > <RELAYFB> is reached for RU:<sip:117003@10.16.48.44> , > From:<sip:5555@10.16.48.71> , To:<sip:4000@10.16.48.44> , Method: INVITE, > Call-ID: 553330101(a)10.16.48.71 <mailto:553330101@10.16.48.71> >

2 1

Email alert
by phillman25 07 Nov '13

07 Nov '13

Hi I was wondering which module i could use to configure an email alert upon an event in kamailio.cfg file i.e. if ...... then send to email address xxxx any ideas? thanks! Phillip

3 2

IM Broadcasting
by sipatse＠in.gr 07 Nov '13

07 Nov '13

1 0

408 callee hangup problem
by kamailio＠AaronLux.com 07 Nov '13

07 Nov '13

2 3

Google search
by Carlos Ruiz Díaz 06 Nov '13

06 Nov '13

Hi guys, this happens to me all the time: when I search for documentation about a specific Kamailio module on Google, I always land on outdated entries usually from Kamailio 1.6.x. Although this seems trivial and I could simply update the URL to the current version of Kamailio, this may be not so obvious for new users and they may end up reading expired documentation and/or implementing non-optimal solutions to their problems. is there a way to only allow indexation of the last documentation? Something like removing the version or adding an HTTP 301 to old entries? Regards, -- Carlos http://caruizdiaz.com +595981146623

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#cfgutils: sleep(x) does not work in failure_route
by Klaus Feichtinger 06 Nov '13

06 Nov '13

Hello list, I have troubles using the function "usleep(x)" or "sleep(x)" (the behaviour is the same) of the "cfgutils" module (http://kamailio.org/docs/modules/3.2.x/modules_k/cfgutils.html#idp76968) in the failure_route. According documentation, this function could be used in any route, including the failure_route, too. However, when I call this function, it does not show any reaction, if a transaction was already answered with a provisional response. It is still working fine, when a transaction is timing out, as no target socket is reachable. This is confusing me! Please find below some xlog messages from my configuration file in two scenarios (function "|t_set_fr(20000, 10000);" is identic in both scenarios|): (1) scenario 1 is including a primary target, looked up in the registrar DB, which is unreachable; after transaction timeout I have inserted the function "sleep()" in the failure_route for forcing a delay => here it is working fine (see timestamp) (2) scenario 2 is including a primary target which is responding, but not establishing the session; after transaction timeout the procedure is still the same as in scenario (1) => here is no delay visible in the log and practically detectable From my point of view, this is a malfuntion! Does anybody see it different? I´ve tested these scenarios with kamailio-3.2.x and kamailio-3.3.x - no difference. Thanks for any hints! Klaus ################# SCENARIO (1) ORIG TARGET IS UNREACHABLE ########################## Nov 6 15:58:25 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: -<|XLOG|>-: <RELAY> is reached for RU: sip:117002@10.16.48.226:5678, From: sip:1101015555@10.16.48.71, To: sip:115300@10.16.48.44, Method: INVITE, Call-ID: 1107651805(a)10.16.48.71 Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: <FAILRELAY> is reached with Code: 408 From: sip:1101015555@10.16.48.71 To: sip:115300@10.16.48.44 Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: failure_route <FAILRELAY> is reached because of a BRANCH_TIMEOUT of RU: sip:117002@10.16.48.226:5678 Nov 6 15:58:35 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28880]: INFO: -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s Nov 6 15:58:37 sipsrvnode1 /usr/sbin/kamailio[28879]: INFO: -<|XLOG|>-: <RELAYFB> is reached for RU: sip:117003@10.16.48.226:7001, From: sip:1101015555@10.16.48.71, To: sip:115300@10.16.48.44, Method: INVITE, Call-ID: 1107651805(a)10.16.48.71 ################# SCENARIO (2) ORIG TARGET IS REACHABLE ########################## Nov 6 16:02:14 sipsrvnode1 /usr/sbin/kamailio[29034]: INFO: -<|XLOG|>-: <RELAY> is reached for RU: sip:117002@10.16.48.226:5678, From: sip:1101015555@10.16.48.71, To: sip:115300@10.16.48.44, Method: INVITE, Call-ID: 553330101(a)10.16.48.71 Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: <FAILRELAY> is reached with Code: 408 From: sip:5555@10.16.48.71 To: sip:4000@10.16.48.44 Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: failure_route <FAILRELAY> is reached because of a BRANCH_TIMEOUT of RU: sip:117002@10.16.48.44 Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: <FAILRELAY> s l e e p 2000ms / 2s Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: <FAILRELAY> s l e p t 2000ms / 2s Nov 6 16:02:34 sipsrvnode1 /usr/sbin/kamailio[29036]: INFO: -<|XLOG|>-: <RELAYFB> is reached for RU: sip:117003@10.16.48.44, From: sip:5555@10.16.48.71, To: sip:4000@10.16.48.44, Method: INVITE, Call-ID: 553330101(a)10.16.48.71

1 0

dispatcher: ds_options_callback, Setting the state failed
by Seudin Kasumovic 06 Nov '13

06 Nov '13

Hi, Test configuration uses OPTION for ping method. In log below host 198.51.100.68 probed first but reply come latter then 192.0.2.4. Seams that only last probed uri tried for setting state other hosts. Also, DEBUG log is broken. (kamailio version: kamailio 4.1.0-pre1 (i386/linux)). Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: dispatcher [dispatch.c:2420]: ds_check_timer(): probing set #28, URI sip: 198.51.100.68:5060 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:243]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:172.16.23.130> Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:182]: dlg2hash(): DEBUG: dlg2hash: 33871 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:345]: t_uac_prepare(): executing event_route[tm:local-request] Nov 5 12:48:01 localhost kamailio[23729]: INFO: <script>: 1280:tm:local-request OPTIONS sip:198.51.100.68:5060 --ID= 77db7ea0-23729(a)172.16.23.5 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:410]: t_uac_prepare(): apply new updates to sip msg Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: dispatcher [dispatch.c:2420]: ds_check_timer(): probing set #0, URI sip:192.0.2.4:5090 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:243]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:172.16.23.130> Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:182]: dlg2hash(): DEBUG: dlg2hash: 33872 Nov 5 12:48:01 localhost kamailio[23729]: DEBUG: tm [uac.c:345]: t_uac_prepare(): executing event_route[tm:local-request] Nov 5 12:48:01 localhost kamailio[23729]: INFO: <script>: 1280:tm:local-request OPTIONS sip:192.0.2.4:5090 --ID= 77db7ea1-23729(a)172.16.23.5 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=54 global id=53 T start=(nil) Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 33871 label 0 branch 0 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_lookup.c:1003]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0xb492a8a8)! Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=54 global id=54 T end=0xb492a8a8 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:2205]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=2 is_invite=0) Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:1309]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=200 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:2085]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:2122]: local_reply(): DEBUG: local transaction completed Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0xb492a8a8, callback type 1024, id 0 entered Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: dispatcher [dispatch.c:2359]: ds_options_callback(): OPTIONS-Request was finished with code 200 (to 192.0.2.4:5090> Nov 5 12:48:01 localhost kamailio[23725]: ERROR: dispatcher [dispatch.c:2372]: ds_options_callback(): Setting the state failed ( 192.0.2.4:5090> Nov 5 12:48:01 localhost kamailio[23725]: DEBUG: tm [t_reply.c:1667]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=50 global id=49 T start=(nil) Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 33872 label 0 branch 0 Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_lookup.c:1003]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0xb492c918)! Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=50 global id=50 T end=0xb492c918 Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_reply.c:2205]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=2 is_invite=0) Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_reply.c:1309]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=200 Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_reply.c:2085]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_reply.c:2122]: local_reply(): DEBUG: local transaction completed Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0xb492c918, callback type 1024, id 0 entered Nov 5 12:48:01 localhost kamailio[23726]: DEBUG: dispatcher [dispatch.c:2359]: ds_options_callback(): OPTIONS-Request was finished with code 200 (to sip:192.0.2.4:5090, group 0) ds_list shows: SET_NO:: 2 SET:: 28 URI:: sip:198.51.100.68:5060 flags=TX priority=1 attrs= SET:: 0 URI:: sip:192.0.2.4:5090 flags=AP priority=1 attrs= -- Seudin Kasumovic

2 2

use append_hf() and remove_hf() on the same message
by Camille Oudot 06 Nov '13

06 Nov '13

Hi, I'm using remove_hf() and append_hf() to manipulate the sip headers on my application. Apparently, calling insert_hf("foo") after remove_hf("bar") if bar is the last header (not including Content-Length) is a bad idea, foo doesn't get inserted (I understand that the append_hf() command tries to insert foo after bar, and since bar just disappeared, well, it fails) What is the safest way to both remove and append headers on the same message? Regards -- Camille

2 4

dialog stats reported by app_perl
by mayamatakeshi 06 Nov '13

06 Nov '13

Hello, I am using kamailio 4.0. I am experimenting with module app_perl. Actually, I am not using it, I am just loading it because it solves this problem when loading snmpstats: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader However, before using app_perl i was seeing this: # kamctl fifo get_statistics dialog active_dialogs dialog:active_dialogs = 1 But with app_perl loaded, I see this: # kamctl fifo get_statistics dialog active_dialogs app_perl:active_dialogs = 1 So, is app_perl intercepting stats requests that should be handled by module dialog? Is this expected? And, is there any performance penalty on having this handled by perl? Regards, Takeshi

2 2

TLS - No ServerHello
by Coy Cardwell 05 Nov '13

05 Nov '13

Hi all. We are using Kamailio 4.0.4 in an Edge Proxy to Central Registrar configuration. The system is in a functional state. Oddly, when we enable TLS, the Kamailio server never responds with a Server Hello to the Client Hello for TLS, so a connection is never established. Has anyone seen anything like this? There are no errors anywhere, the server just never responds to the initial TLS Client Hello. We know the packet arrives and is acknowledged at the TCP level. Just a bit stumped.

2 2

Question about call transfer and pickup
by Гергарт Олег (Мобилон) 05 Nov '13

05 Nov '13

Hello! Tell me please, is there in kamailio opportunity to call transfer and pickup? If yes, where can I about this to read? Thanks! Regards, Oleg Gergart

2 1

Segmentaion fault in Kamailio 4.0.3
by Morten Isaksen 05 Nov '13

05 Nov '13

Hi, Our Kamailio has stopped with a segmentation fault 4 times the last week. There has been no changes to the configuration file the last 15 days, so I suspect a SIP phone is sending a SIP packet that Kamailio does not like. We have a core dump file but I cannot read anything usefull from the backtrace. Can you see what is wrong from the backtrace? Regards Morten The output from gdb (bt full): Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -m 256 -M 8 -u kamai'. Program terminated with signal 11, Segmentation fault. #0 0x00007f0e10de17b2 in cancel_branch (t=0x7f0dfbf38e10, branch=0, reason=<value optimized out>, flags=4) at t_cancel.c:284 284 if (cfg_get(tm, tm_cfg, reparse_invite) || Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6.x86_64 hiredis-0.10.1-3.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.10.3-10.el6.x86_64 libcom_err-1.41.12-14.el6.x86_64 libselinux-2.0.94-5.3.el6.x86_64 libxml2-2.7.6-12.el6_4.1.x86_64 mysql-libs-5.1.67-1.el6_3.x86_64 nss-softokn-freebl-3.12.9-11.el6.x86_64 openssl-1.0.0-27.el6_4.2.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt full #0 0x00007f0e10de17b2 in cancel_branch (t=0x7f0dfbf38e10, branch=0, reason=<value optimized out>, flags=4) at t_cancel.c:284 cancel = <value optimized out> len = <value optimized out> crb = 0x7f0dfbf39008 irb = 0x7f0dfbf38f80 ret = 1 tmp_cd = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 0}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 0}}}} pcbuf = <value optimized out> __FUNCTION__ = "cancel_branch" #1 0x00007f0e10e298ab in reply_received (p_msg=0x7f0e124ce760) at t_reply.c:2194 msg_status = <value optimized out> last_uac_status = 408 ack = 0x7f0dfbf38e10 "Ð \267\373\r\177" ack_len = <value optimized out> branch = 0 reply_status = <value optimized out> onreply_route = <value optimized out> cancel_data = {cancel_bitmap = 0, reason = {cause = 408, u = {text = {s = 0x0, len = 307468800}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 307468800}}}} uac = <value optimized out> t = 0x7f0dfbf38e10 lack_dst = {send_sock = 0x0, to = {s = {sa_family = 6704, sa_data = "\\#\377\177\000\000\000\000\000\000\000\000\000"}, sin = {sin_family = 6704, sin_port = 9052, sin_addr = {s_addr = 32767}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 6704, sin6_port = 9052, sin6_flowinfo = 32767, sin6_addr = {__in6_u = { __u6_addr8 = "\000\000\000\000\000\000\000\000p{1\022\016\177\000", __u6_addr16 = {0, 0, 0, 0, 31600, 4657, 32526, 0}, __u6_addr32 = {0, 0, 305232752, 32526}}}, sin6_scope_id = 307029856}}, id = 32526, proto = 28 '\034', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} backup_user_from = <value optimized out> backup_user_to = <value optimized out> backup_domain_from = <value optimized out> backup_domain_to = <value optimized out> backup_uri_from = <value optimized out> backup_uri_to = <value optimized out> backup_xavps = <value optimized out> replies_locked = 0 branch_ret = <value optimized out> prev_branch = <value optimized out> blst_503_timeout = <value optimized out> hf = <value optimized out> onsend_params = {req = 0x8d8a39, rpl = 0x541db4, param = 0x7f0e1253e528, code = 307029856, flags = 32526, branch = 0, t_rbuf = 0x7fff235c1a30, dst = 0x7f0e12317b70, send_buf = { s = 0x375311000000000 <Address 0x375311000000000 out of bounds>, len = 0}} ctx = {rec_lev = 307491008, run_flags = 32526, last_retcode = 5674412, jmp_env = {{__jmpbuf = {140733786626256, 63331951475841423, 139698413054576, 139698413299552, 9276465, 139698039855608, -63254168797292145, 63332490682325391}, __mask_was_saved = 0, __saved_mask = {__val = {139698413734928, 0, 139698411522821, 1, 140733786626608, 6185835, 5972697, 8586176, 9275699, 69026945952, 3, 9276465, 9275673, 139698413738496, 9275961, 139698413760704}}}}} __FUNCTION__ = "reply_received" #2 0x0000000000456444 in do_forward_reply (msg=0x7f0e124ce760, mode=<value optimized out>) at forward.c:799 new_buf = 0x0 dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 0 '\000', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} new_len = <value optimized out> r = <value optimized out> s = <value optimized out> len = <value optimized out> __FUNCTION__ = "do_forward_reply" #3 0x000000000049e15e in receive_msg (buf=<value optimized out>, len=313, rcv_info=0x7fff235c1cd0) at receive.c:270 msg = 0x7f0e124ce760 ctx = {rec_lev = 11, run_flags = 0, last_retcode = 206110737, jmp_env = {{__jmpbuf = {139698036884436, 11, 219309716216, 139698419720192, 140733786627520, 4294967295, 140733786627647, 1}, __mask_was_saved = 8576456, __saved_mask = {__val = {0, 28, 16, 0, 219305533392, 1, 0, 139698411461552, 219309716216, 139698036884436, 139698413732672, 139698419717800, 139698413732680, 140733786627416, 219305559701, 140733786627288}}}}} ret = <value optimized out> inb = { s = 0x8d8900 "SIP/2.0 100 Trying\r\nVia: SIP/2.0/UDP 178.21.249.20;branch=z9hG4bK8149.c6575a95.0\r\nTo: sip:201@78799865.pbx.one-connect.dk;tag=07c44e68\r\nFrom: sip:201@78799865.pbx.one-connect.dk;tag=a6a1c5f60faecf035a"..., len = 313} __FUNCTION__ = "receive_msg" #4 0x0000000000530e46 in udp_rcv_loop () at udp_server.c:557 ---Type <return> to continue, or q <return> to quit--- len = 313 buf = "SIP/2.0 100 Trying\r\nVia: SIP/2.0/UDP 178.21.249.20;branch=z9hG4bK8149.c6575a95.0\r\nTo: sip:201@78799865.pbx.one-connect.dk;tag=07c44e68\r\nFrom: sip:201@78799865.pbx.one-connect.dk;tag=a6a1c5f60faecf035a"... from = 0x7f0e12538340 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {2993962576, 0}, addr32 = {2993962576, 0, 0, 0}, addr16 = {15952, 45684, 0, 0, 0, 0, 0, 0}, addr = "P>t\262", '\000' <repeats 11 times>}}, dst_ip = { af = 2, len = 4, u = {addrl = {351868338, 0}, addr32 = {351868338, 0, 0, 0}, addr16 = {5554, 5369, 0, 0, 0, 0, 0, 0}, addr = "\262\025\371\024", '\000' <repeats 11 times>}}, src_port = 35754, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\213\252P>t\262\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 43659, sin_addr = {s_addr = 2993962576}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 43659, sin6_flowinfo = 2993962576, sin6_addr = {__in6_u = { __u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x7f0e124cfbd0, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #5 0x000000000046716a in main_loop () at main.c:1638 i = <value optimized out> pid = <value optimized out> si = <value optimized out> si_desc = "udp receiver child=2 sock=178.21.249.20:5060 \000\000\000\000\200\303P\022\016\177\000\000\000\000\000\000\000\000\000\000\003\000\000\000\000\000\000\000\001\000\000\000\001\000\000\000@\350\216\000\000\000\000\000\001\000\000\000\000\000\000\000\200\350\216\000\000\000\000\000\000\000\200\020", '\000' <repeats 12 times>, "\005\000\000\000\000\000\000" nrprocs = <value optimized out> __FUNCTION__ = "main_loop" #6 0x000000000046a002 in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:2566 cfg_stream = <value optimized out> c = <value optimized out> r = <value optimized out> tmp = 0x7fff235c377f "" tmp_len = 0 options = 0x5c08c8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1722854551 rfd = <value optimized out> debug_save = <value optimized out> debug_flag = <value optimized out> dont_fork_cnt = <value optimized out> n_lst = <value optimized out> p = <value optimized out> __FUNCTION__ = "main" (gdb) -- Morten Isaksen

2 9

Undefined Symbol Error in TLS.so
by Mark Zeman 05 Nov '13

05 Nov '13

Hi, I'm trying to set up a Kamailio server for a project at university, and have some trouble with it. Since Kamailio isn't starting, I use the 'kamailio -c -f kamailio.cfg' command to figure out what's wrong. This tells me that it cannot load the tls.so file, because of 'undefined symbol: log_color. I've tried to recompile the TLS module, but that hasn't helped. Does anyone know how to fix this? Cheers, Mark

2 4

Callee timeout
by kamailio＠AaronLux.com 05 Nov '13

05 Nov '13

The problem is the callee continues to show the call as 'incoming call' after the call is answered and 2-way audio is established so eventually the callee times out. Network Topography: (ip addresses were changed to make them easier to read.) 66.87.78.CALLER is Caller running csipsimple 66.87.76.CALLEE is Callee running csipsimple 192.168.1.LAN is LAN ip running rtpproxy and kamailio 66.41.221.WAN is WAN ip with ports 5004-5082 forwarded to 192.168.1.LAN And firewall turned off. ###################### kamailio -v version: kamailio 4.0.4 (i386/linux) cabe58 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: cabe58 compiled on 01:07:11 Nov 2 2013 with gcc 4.4.7 KAMAILIO DEBUG LOG: 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=472 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=582 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=577 a=25 n=mf_process_maxfwd_header 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=587 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=582 a=26 n=sanity_check 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=475 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=765 a=55 n=force_rport 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=775 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=766 a=25 n=nat_uac_test 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=771 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=767 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=770 a=24 n=add_contact_alias 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=772 a=39 n=setflag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=775 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=487 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=478 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=487 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=631 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=591 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=491 a=24 n=t_check_trans 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=752 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=739 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=747 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=742 a=27 n=auth_check 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=743 a=26 n=auth_challenge 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=744 a=2 n=exit 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=472 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=582 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=577 a=25 n=mf_process_maxfwd_header 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=587 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=582 a=26 n=sanity_check 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=475 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=765 a=55 n=force_rport 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=775 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=766 a=25 n=nat_uac_test 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=771 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=767 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=770 a=24 n=add_contact_alias 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=772 a=39 n=setflag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=775 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=487 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=478 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=487 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=631 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=591 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=491 a=24 n=t_check_trans 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=494 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=752 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=739 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=747 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=742 a=27 n=auth_check 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=749 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=747 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=748 a=24 n=consume_credentials 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=759 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=759 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=25 n=remove_hf 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=503 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=499 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=500 a=24 n=record_route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=509 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=503 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=505 a=39 n=setflag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=509 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=826 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=514 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=719 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=694 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=695 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=517 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=648 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=635 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=527 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=527 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=860 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=530 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=666 a=63 n=assign 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=683 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=667 a=25 n=lookup 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=688 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=683 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=685 a=39 n=setflag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=688 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=541 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=538 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=540 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=539 a=25 n=t_is_set 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=539 a=25 n=t_on_branch 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=544 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=541 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=543 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=542 a=25 n=t_is_set 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=542 a=25 n=t_on_reply 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=548 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=544 a=25 n=is_method 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=546 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=545 a=25 n=t_is_set 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=545 a=25 n=t_on_failure 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=551 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=548 a=24 n=t_relay 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=907 a=25 n=xdbg 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=908 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=787 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=782 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 52(3514) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=794 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=798 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=795 a=24 n=t_is_branch_route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=796 a=25 n=add_rr_param 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=907 a=25 n=xdbg 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=908 a=5 n=route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=787 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=782 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 52(3514) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=794 a=24 n=has_totag 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=798 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=795 a=24 n=t_is_branch_route 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=796 a=25 n=add_rr_param 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 52(3514) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=551 a=2 n=exit 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 53(3516) NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1383447347;method=INVITE;from_tag=ghp-tpel0iRnq1kdZVv2Ovu0HuNDES6J;to_tag=nZnsaAYjbOaSx2u-knYbcLDvQh2ymtiI;call_id=mh5qsRvw4rQGbPMmJS2oS06Dsof89co6;code=200;reason=OK;src_user=5465;src_domain=66.41.221.WAN;src_ip=66.87.78.CALLER;dst_ouser=5466;dst_user=5466;dst_domain=66.87.76.CALLEE 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 54(3519) ERROR: <core> [tcp_main.c:4247]: handle_tcpconn_ev(): connect 192.168.1.165:43177 failed 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=913 a=25 n=xdbg 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=916 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=915 a=5 n=route 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=781 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=788 a=41 n=isflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=rtpproxy_manage 53(3516) ERROR: rtpproxy [rtpproxy.c:2653]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=793 a=24 n=is_request 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=800 a=24 n=is_reply 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=804 a=16 n=if 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=801 a=25 n=isbflagset 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=802 a=24 n=add_contact_alias 53(3516) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=806 a=2 n=return ################## rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Extension 20090810: Support for automatic bridging COMMAND TO START RTPPROXY: rtpproxy -F -l 66.41.221.WAN/192.168.1.LAN -s unix:/var/run/rtpproxy.sock -d DBUG:LOG_LOCAL0 RTPPROXY DEBUG LOG: Nov 2 21:54:18 localhost rtpproxy[3408]: INFO:handle_command: new session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tag XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1 requested, type strong Nov 2 21:54:18 localhost rtpproxy[3408]: ERR:create_twinlistener: can't bind to the IPv4 port 64918: Cannot assign requested address Nov 2 21:54:18 localhost rtpproxy[3408]: ERR:handle_command: can't create listener Nov 2 21:54:18 localhost rtpproxy[3408]: INFO:handle_command: new session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tag XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1 requested, type strong Nov 2 21:54:18 localhost rtpproxy[3408]: ERR:create_twinlistener: can't bind to the IPv4 port 64918: Cannot assign requested address Nov 2 21:54:18 localhost rtpproxy[3408]: ERR:handle_command: can't create listener Nov 2 21:54:21 localhost rtpproxy[3408]: INFO:handle_command: lookup request failed: session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tags XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1/XgwrtMgNKi0NCb-MAlvB-nvRad6F78rz;1 not found Nov 2 21:54:21 localhost rtpproxy[3408]: INFO:handle_command: lookup request failed: session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tags XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1/XgwrtMgNKi0NCb-MAlvB-nvRad6F78rz;1 not found Nov 2 21:54:22 localhost rtpproxy[3408]: INFO:handle_command: lookup request failed: session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tags XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1/XgwrtMgNKi0NCb-MAlvB-nvRad6F78rz;1 not found Nov 2 21:54:24 localhost rtpproxy[3408]: INFO:handle_command: lookup request failed: session qWqJEynpz5C46nntNShAHgSLlDGm43g6, tags XvTp.jjETla8qXvE1gWIh6HxUCLl3DR2;1/XgwrtMgNKi0NCb-MAlvB-nvRad6F78rz;1 not found ####################Wireshark Packet Capture No. Time Source Destination Protocol Info 1 0.000000 192.168.1.LAN 204.235.61.9 NTP NTP client Frame 1 (90 bytes on wire, 90 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 204.235.61.9 (204.235.61.9) User Datagram Protocol, Src Port: ntp (123), Dst Port: ntp (123) Network Time Protocol No. Time Source Destination Protocol Info 2 0.069184 204.235.61.9 192.168.1.LAN NTP NTP server Frame 2 (90 bytes on wire, 90 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 204.235.61.9 (204.235.61.9), Dst: 192.168.1.LAN (192.168.1.LAN) User Datagram Protocol, Src Port: ntp (123), Dst Port: ntp (123) Network Time Protocol No. Time Source Destination Protocol Info 3 4.812402 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data, Frame 3 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1, Ack: 1, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 4 4.813081 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data Frame 4 (104 bytes on wire, 104 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1189, Ack: 1, Len: 38 [Reassembled TCP Segments (1189 bytes): #3(1151), #4(38)] Secure Socket Layer No. Time Source Destination Protocol Info 5 4.813109 192.168.1.LAN 66.87.78.CALLER TCP sip-tls > 49232 [ACK] Seq=1 Ack=1227 Win=1002 Len=0 TSV=1433086 TSER=34152821 Frame 5 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 1, Ack: 1227, Len: 0 No. Time Source Destination Protocol Info 6 4.814100 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 6 (631 bytes on wire, 631 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 1, Ack: 1227, Len: 565 Secure Socket Layer No. Time Source Destination Protocol Info 7 4.893013 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=1227 Ack=566 Win=2504 Len=0 TSV=34152854 TSER=1433086 Frame 7 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1227, Ack: 566, Len: 0 No. Time Source Destination Protocol Info 8 4.905470 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data, Application Data Frame 8 (556 bytes on wire, 556 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1227, Ack: 566, Len: 490 Secure Socket Layer No. Time Source Destination Protocol Info 9 4.923685 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data, Frame 9 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 1717, Ack: 566, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 10 4.923755 192.168.1.LAN 66.87.78.CALLER TCP sip-tls > 49232 [ACK] Seq=566 Ack=2905 Win=1002 Len=0 TSV=1433196 TSER=34152854 Frame 10 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 566, Ack: 2905, Len: 0 No. Time Source Destination Protocol Info 11 4.996184 66.87.78.CALLER 192.168.1.LAN TLSv1 Application Data Frame 11 (280 bytes on wire, 280 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 2905, Ack: 566, Len: 214 [Reassembled TCP Segments (1365 bytes): #9(1151), #11(214)] Secure Socket Layer No. Time Source Destination Protocol Info 12 4.999143 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 12 (503 bytes on wire, 503 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 566, Ack: 3119, Len: 437 Secure Socket Layer No. Time Source Destination Protocol Info 13 5.002222 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 13 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 14 5.002329 192.168.1.LAN 66.87.76.CALLEE TLSv1 Application Data Frame 14 (1383 bytes on wire, 1383 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1, Ack: 1, Len: 1317 Secure Socket Layer No. Time Source Destination Protocol Info 15 5.103116 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=1003 Win=2539 Len=0 TSV=34152876 TSER=1433272 Frame 15 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 1003, Len: 0 No. Time Source Destination Protocol Info 16 5.700164 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Application Data Frame 16 (620 bytes on wire, 620 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 1, Ack: 1318, Len: 554 Secure Socket Layer No. Time Source Destination Protocol Info 17 5.700204 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=555 Win=998 Len=0 TSV=1433973 TSER=6923455 Frame 17 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 555, Len: 0 No. Time Source Destination Protocol Info 18 5.837983 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Application Data Frame 18 (812 bytes on wire, 812 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 555, Ack: 1318, Len: 746 Secure Socket Layer No. Time Source Destination Protocol Info 19 5.838020 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=1301 Win=998 Len=0 TSV=1434111 TSER=6923469 Frame 19 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 1301, Len: 0 No. Time Source Destination Protocol Info 20 5.838533 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 20 (695 bytes on wire, 695 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 1003, Ack: 3119, Len: 629 Secure Socket Layer No. Time Source Destination Protocol Info 21 6.002208 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 21 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 22 6.002946 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=1632 Win=2574 Len=0 TSV=34152966 TSER=1434111 Frame 22 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 1632, Len: 0 No. Time Source Destination Protocol Info 23 7.001986 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 23 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 24 7.988054 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 24 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 1301, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 25 7.988112 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=2489 Win=998 Len=0 TSV=1436261 TSER=6923682 Frame 25 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 2489, Len: 0 No. Time Source Destination Protocol Info 26 7.996188 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 26 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 2489, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #24(1151), #26(70)] Secure Socket Layer No. Time Source Destination Protocol Info 27 7.996206 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=2559 Win=1002 Len=0 TSV=1436269 TSER=6923682 Frame 27 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 2559, Len: 0 No. Time Source Destination Protocol Info 28 7.997972 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 28 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 1632, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 29 8.003228 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 29 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 30 8.174409 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=2757 Win=2619 Len=0 TSV=34153182 TSER=1436271 Frame 30 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 2757, Len: 0 No. Time Source Destination Protocol Info 31 8.515198 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 31 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 2559, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 32 8.515251 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=3747 Win=998 Len=0 TSV=1436788 TSER=6923734 Frame 32 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 3747, Len: 0 No. Time Source Destination Protocol Info 33 8.516100 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 33 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 3747, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #31(1151), #33(70)] Secure Socket Layer No. Time Source Destination Protocol Info 34 8.516116 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=3817 Win=1002 Len=0 TSV=1436789 TSER=6923734 Frame 34 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 3817, Len: 0 No. Time Source Destination Protocol Info 35 8.517743 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 35 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 2757, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 36 8.583042 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=3882 Win=2665 Len=0 TSV=34153224 TSER=1436790 Frame 36 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 3882, Len: 0 No. Time Source Destination Protocol Info 37 9.003201 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 37 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 38 9.621092 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 38 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 3817, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 39 9.621125 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=5005 Win=998 Len=0 TSV=1437894 TSER=6923837 Frame 39 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 5005, Len: 0 No. Time Source Destination Protocol Info 40 9.621136 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 40 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 5005, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #38(1151), #40(70)] Secure Socket Layer No. Time Source Destination Protocol Info 41 9.621143 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=5075 Win=997 Len=0 TSV=1437894 TSER=6923837 Frame 41 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 5075, Len: 0 No. Time Source Destination Protocol Info 42 9.622121 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 42 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 3882, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 43 9.684112 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=5007 Win=2710 Len=0 TSV=34153334 TSER=1437895 Frame 43 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 5007, Len: 0 No. Time Source Destination Protocol Info 44 10.003000 Dell_70:ce:3a Broadcast ARP Who has 192.168.1.165? Tell 192.168.1.LAN Frame 44 (42 bytes on wire, 42 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Broadcast (ff:ff:ff:ff:ff:ff) Address Resolution Protocol (request) No. Time Source Destination Protocol Info 45 11.545138 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 45 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 5075, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 46 11.545187 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=6263 Win=998 Len=0 TSV=1439818 TSER=6924040 Frame 46 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 6263, Len: 0 No. Time Source Destination Protocol Info 47 11.545212 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 47 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 6263, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #45(1151), #47(70)] Secure Socket Layer No. Time Source Destination Protocol Info 48 11.545222 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=6333 Win=997 Len=0 TSV=1439818 TSER=6924040 Frame 48 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 6333, Len: 0 No. Time Source Destination Protocol Info 49 11.546755 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 49 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 5007, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 50 11.623085 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=6132 Win=2755 Len=0 TSV=34153528 TSER=1439819 Frame 50 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 6132, Len: 0 No. Time Source Destination Protocol Info 51 15.600662 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data, Frame 51 (1254 bytes on wire, 1254 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 6333, Ack: 1318, Len: 1188 Secure Socket Layer No. Time Source Destination Protocol Info 52 15.600719 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=7521 Win=998 Len=0 TSV=1443873 TSER=6924444 Frame 52 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 7521, Len: 0 No. Time Source Destination Protocol Info 53 15.601394 66.87.76.CALLEE 192.168.1.LAN TLSv1 Application Data Frame 53 (136 bytes on wire, 136 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.76.CALLEE (66.87.76.CALLEE), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 58818 (58818), Dst Port: sip-tls (5061), Seq: 7521, Ack: 1318, Len: 70 [Reassembled TCP Segments (1221 bytes): #51(1151), #53(70)] Secure Socket Layer No. Time Source Destination Protocol Info 54 15.601411 192.168.1.LAN 66.87.76.CALLEE TCP sip-tls > 58818 [ACK] Seq=1318 Ack=7591 Win=1002 Len=0 TSV=1443874 TSER=6924444 Frame 54 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.76.CALLEE (66.87.76.CALLEE) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 58818 (58818), Seq: 1318, Ack: 7591, Len: 0 No. Time Source Destination Protocol Info 55 15.601851 192.168.1.LAN 66.87.78.CALLER TLSv1 Application Data Frame 55 (1191 bytes on wire, 1191 bytes captured) Ethernet II, Src: Dell_70:ce:3a (00:19:b9:70:ce:3a), Dst: Intel_da:5d:dd (00:02:b3:da:5d:dd) Internet Protocol, Src: 192.168.1.LAN (192.168.1.LAN), Dst: 66.87.78.CALLER (66.87.78.CALLER) Transmission Control Protocol, Src Port: sip-tls (5061), Dst Port: 49232 (49232), Seq: 6132, Ack: 3119, Len: 1125 Secure Socket Layer No. Time Source Destination Protocol Info 56 15.666175 66.87.78.CALLER 192.168.1.LAN TCP 49232 > sip-tls [ACK] Seq=3119 Ack=7257 Win=2800 Len=0 TSV=34153932 TSER=1443874 Frame 56 (66 bytes on wire, 66 bytes captured) Ethernet II, Src: Intel_da:5d:dd (00:02:b3:da:5d:dd), Dst: Dell_70:ce:3a (00:19:b9:70:ce:3a) Internet Protocol, Src: 66.87.78.CALLER (66.87.78.CALLER), Dst: 192.168.1.LAN (192.168.1.LAN) Transmission Control Protocol, Src Port: 49232 (49232), Dst Port: sip-tls (5061), Seq: 3119, Ack: 7257, Len: 0

2 1

Msilo: remove messages from db immediately after successful delivery
by Marc M. 05 Nov '13

05 Nov '13

Hi! I have a running msilo config. However, messages remain in the db after after successful delivery, and get removed at the next clean period. Unfortunatelly, this makes support cases dificult, as we do not see from the db which messages are delivered and which are not. My goal is is either: Remove the message from the db immediately after successful delivery (preferred) and/or be able to see the individual message delivery status. Can you help me, how should I proceed? Many thanks in advance! Marc

2 1

How to configure MySQL?
by Gus Monti 05 Nov '13

05 Nov '13

Hi, I can see that we are missing a lot of Sip messages in or db, so, checking Homer FAQ there is a way to compile MySql with Insert Delayed for partitioning table. The problem is that I don't understand how to recompile MySql and make the necessary changes. The URL is https://code.google.com/p/homer/wiki/FAQ#Q:_I_have_a_lot_of_SIP_traffic_an d_some_SIP_messages_are_missing The question is, can you send me a step by step how to do this? I really appreciate your help. I have CentOS 6.4: [root@Homerdb02 ~]# kamailio -v version: kamailio 4.1.0-dev9 (x86_64/linux) f8f3d3 [root@Homerdb02 ~]# cat /etc/redhat-release Red Hat Enterprise Linux Server release 6.4 (Santiago) [root@Homerdb02 ~]# uname -a Linux Homerdb02 2.6.32-358.18.1.el6.x86_64 #1 SMP Fri Aug 2 17:04:38 EDT 2013 x86_64 x86_64 x86_64 GNU/Linux [root@Homerdb02 ~]# mysql -V mysql Ver 14.14 Distrib 5.5.34, for Linux (x86_64) using readline 5.1 Thanks, Gus

2 1

Event route sl:local-response is not executed in kamailio 4.1
by Sergey Okhapkin 04 Nov '13

04 Nov '13

I just upgraded one server to latest git (version: kamailio 4.1.0-pre0 (x86_64/linux) 26fa46-dirty), sl:local-response route is no longer executed. I have in kamailio.cfg event_route[sl:local-response] { xlog("L_INFO","Local response\n$mb\n"); } No local responses are logged in kamailio log file. It was working fine in kamailio 4.1.0-dev7 (x86_64/linux) b2a536-dirty

2 2

Status : 401 Unauthorized (0 bindings) Reason
by Owais ul Haq 04 Nov '13

04 Nov '13

Hello, I am using kamailio server. Problem is when my soft phone tries to connect to Kamailio server, it connects in second attempt. First it gets a sip reply saying "Status : 401 Unauthorized (0 bindings)". I checked this using Wire Shark. Can some one please help me in finding the reason for this error message found all the time. Thanks Best Regards

2 1

Reason for 401 Unauthorized message
by Owais ul Haq 04 Nov '13

04 Nov '13

Hello, I am using kamailio server. Problem is when my soft phone tries to connect to Kamailio server, it connects in second attempt. First it gets a sip reply saying "Status : 401 Unauthorized (0 bindings)". I checked this using Wire Shark. Can some one please help me in finding the reason for this error message found all the time. Thanks Best Regards

2 1

Limit the nr. of transaction retries
by Marc M. 04 Nov '13

04 Nov '13

Hi! I Have a questions concerning t_realy() and failure route. If the t_realy() is not successful, it makes 12 re-tries before failure route is called. Can you help me, how can I lower the number of retries before move to failure route? My goal is to make 3 re-tries and then call failure route. (I gues it is a tm param?) Thansk you for your help in advance! regards Marc

2 1

Kamailio callee timeout problem
by kamailio＠AaronLux.com 04 Nov '13

04 Nov '13

I'm pretty sure I'm just missing something very simple here like a port forward. Let me know if you have any ideas! This setup completely works when both csipsiimple clients are inside my LAN and kamailio is configured with alias=192.168.1.LAN. The problem begins when I connect both clients from the WAN and configure kamailio with alias=66.41.221.WAN. The problem is the callee continues to show the call as 'incoming call' after the call is answered and 2-way audio is established so eventually the callee times out. Summary:: The callee and caller both establish 2-way audio and zrtp keys. TLS is working. Both SIP clients are using the OSTN wizard in csipsimple. Please review my Debug logs: http://pastebin.com/9Rw7zSQ0

2 1

Regex question
by Oliver Roth 04 Nov '13

04 Nov '13

Hi all Maybe a very simple question - but I cannot see the solution - I am more or less newbie ;) I need to do some string operations in kamailio.cfg. How can I get the cli from the following string: sip:+41523940347@195.216.67.103;user=phone I only would need +41523940347 in a variable Something like $avp(s:myCli) = ^sip:\+(\d{11})@.{1,40}$ Any idea? Thanks in advance Oli

4 4

Re: [SR-Users] Capturing-200 Okey(with SDP) from cfg (Daniel Grotti)
by Surendra 04 Nov '13

04 Nov '13

Hi Daniel, Issue got solved. I am just sharing it. My requirement is I have to parse the SDP from cfg which is coming in 200 OKEY(Initiated by UAC once the called party answers the call). On reply route we can capture the responses for those requests which we set t_on_reply();(externally we have to call for each request). But reply route we can capture all replies. So that we can parse our required headers or msg body. Regards Surendra ******************************

1 0

TLS: support TLS v1.2 ?
by Marc M. 03 Nov '13

03 Nov '13

Hi! I have question concerning TLS module. I would like to enable TLS 1.2 protocol. As I understand the TLS module is compiled with openssl, which as far as i know has TLS 1.2 support since openssl 1.0.1. I am wondering if there is way to enable TLS 1.2 on the TLS module? Many thanks Marc

2 1

Incoming call problem
by Jason Sia 02 Nov '13

02 Nov '13

Hi, I have kamailio server and a freeswitch server during a call the kamailio passes the call to freeswitch to play a media and back to kamailio for routing to the destination, this works fine however for samsung devices (any ROM using linphone). On other device brands, it does not work it always returns USER_BUSY or NUMBER_NOT_FOUND. After investigation, I found out that if the transport is UDP then it works on all phones, the problem happens if I use TCP as the transport. Anyone had experienced on this? Thanks, Jason

3 4

Re: [SR-Users] xHttp_PI Module
by Ovidiu Sas 02 Nov '13

02 Nov '13

Please post your questions to the mailing list instead of sending private e-mails. The subscriber provisioning via xhttp is supported only for plaintext passwords. Set modparam("auth_db", "calculate_ha1", 1) and it will work fine. Regards, Ovidiu Sas On Sat, Nov 2, 2013 at 12:22 PM, Abdul Hakeem <alhakeem(a)gmail.com> wrote: > Hello, > > I just want to enquire if the subscriber full provisioning is ready. > Best regards, > Abdul Hakeem > > -- VoIP Embedded, Inc. http://www.voipembedded.com

1 0

dispatcher: how to set the transport protocol
by Karimi-Cherkandi, Bizhan 01 Nov '13

01 Nov '13

kamailio dispatcher module in my environment uses only UDP I have not been able to find how I can configure kamailio dispatcher module to use TCP Would appreciate any help Bijan

3 6

Kamailio - RTPProxy - Asymmetric RTP
by Lucas Girard 01 Nov '13

01 Nov '13

Hi all, I have an issue with a Kamailio and rtpProxy, when Asymmetric RTP is used. I have the system running, lines registered in Kamailio are able to call to several destinations, but I have a problem with one provider that use Asymmetric RTP (others providers use Symmetric RTP and I dont have any issues). A -----> Kamailio & RTPProxy ----------> Trunk -------> Provider (Asymmetric RTP) -------> B A is calling B. The problem is that line in kamailio A can listen to B, but B can not listen what A says. I have captured the sip dialog and rtp traffic and I can see that: Kamailio tells to provider that is using the port 52388 (media port in SDP). Providers tells Kamailio that is using the port 5394 (media port in SDP). RTP traffic from Providers to kamailio goes from port 5392 to 52388. RTP traffic from Kamailio to Provider goes from port 52388 to 5392. I have not found an RFC regarding Asymmetric RTP. The question are, why RTP is sent from Kamailio (rtpproxy) to Provider to destination port 5392? It should use 5394 as SDP indicate? or, port 5392 is learned when RTP traffic was received from 5392 to 52388? # kamailio -V version: kamailio 3.3.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 22:17:07 Jun 18 2012 with gcc 4.4.5 # rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications I hope to be clear with this explanation. Regards, Lucas Girard

3 7

msilo - use case: registered but not reachable
by Marc M. 01 Nov '13

01 Nov '13

Hi! I have a simple msilo setup which is working fine: -If the user is registered, the messages are delivered right away -If the user is not registered, the messages are stored in msilo and got deliverd at the next register Here is the relevant part of the cfg: --------------- if (is_method("MESSAGE")) { log(1,"-------------------------------------------------------->WE HAVE A NEW MESSAGE\n"); route(MSILO_MESSAGE); } . . . route[MSILO_MESSAGE]{ if (lookup("location")){ t_relay(); }else{ m_store("$ru"); sl_send_reply("200", "OK"); } exit; } . . . route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); # MSILO - dumping user's offline messages m_dump(); exit; } } ------------------- Thats working fine. Now, we have a use case where a user is registered, but got terminated in between abd not reachable. So the user is in the userloc table, but not reachable. In this case ,since the user is still in userloc, the t_relay() will be executed, and it drops a 408 (timeout) code which is fine. However, the process end here. The message is then dropped. What i would like to achieve is to make sure that t_relay() was executed with 200, otherwise store the message and deliver it at the next register. I was trying this: ------------ ##storing offline messages (new) route[MSILO_MESSAGE]{ if (lookup("location")){ if (!t_relay()) { m_store("$ru"); sl_send_reply("200", "OK"); } exit; } else{ m_store("$ru"); sl_send_reply("200", "OK"); } exit; } -------- but i have still the same behaviour. Can you help me, what do I wrong? Many thnaks Marc. M

2 1

Re: [SR-Users] Capturing-200 Okey(with SDP) from cfg
by Daniel Grotti 01 Nov '13

01 Nov '13

Hi, what do you mean with "capture from cfg file" ? Do you want to log 200OK message in kamailio log file ? Daniel On Friday, November 1, 2013 08:36 CET, "Surendra" <surendra.pullaiah(a)plintron.com> wrote: > Hi all, > > > > I have a requirement , i.e. I need to capture 200 ok(with > SDP) from cfg file. This 200 okey will be getting once the called party > answers the call. > > Can anyone help me to get out of this. > > > > Regards > > Surendra > > > > >

1 0

Capturing-200 Okey(with SDP) from cfg
by Surendra 01 Nov '13

01 Nov '13

Hi all, I have a requirement , i.e. I need to capture 200 ok(with SDP) from cfg file. This 200 okey will be getting once the called party answers the call. Can anyone help me to get out of this. Regards Surendra

1 0

Help : Kamailio RTP Proxy Issue...
by Mahmoud Ramadan Ali 31 Oct '13

31 Oct '13

Hello , This is the first time to put a question on sr-users mailing list and i hope to help me to solve my issue...i'm following this post http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html to setup RTP Proxy for kamailio 4 but it's NOT working as predicted and i'm receiving this error message on asterisk... [2013-10-16 05:20:04] ERROR[3216][C-0000002c]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("192.168.1.80192.168.1.80", "(null)", ...): Name or service not known [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10873 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 192.168.1.80192.168.1.80' [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10464 process_sdp: Insufficient information in SDP (c=)... The problem is NOT at the asterisk side but at the proxy side...when kamailio rewrites the SDP message it adds the public IP twice in the "C" filed of the SDP message instead of only one time like this 'IN IP4 192.168.1.80192.168.1.80' this is because the scripte is calling rtpproxy_manage() in the routes TOASTERISK /FROMASTERISK and in MANAGE_BRANCH / NATMANAGE. It should be either or,not both....but i don't know how to do this since i don't have any scripting skills so if anyone can modify the config file in the attachment i will be thankful... Best Regards' Mahmoud Ramadan Ali

2 2

Hardware requirements/recommandations
by Igor Potjevlesch 30 Oct '13

30 Oct '13

Hello, I'm looking for the best practises for setting Kamailio and the associated hardware requirements. Kamailio must deliver registrar and routing functions for 500k aliases, 5k registered accounts and ~10k routes set in carrierroute module or similar. My worries mainly concern DB integration. Is it better to setup Kamailio DB on the same server (no network issue, etc.) or on another server? Your opinion/feedback experience would be appreciated. Thanks! Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 7

Upgrade v4.0.x to 4.1.0 guidelines
by Daniel-Constantin Mierla 30 Oct '13

30 Oct '13

Hello, this is a call for developers and community users to update the page collecting upgrade guidelines from current stable (4.0.x) to upcoming major release (4.1.0): - https://www.kamailio.org/wiki/install/upgrade/4.0.x-to-4.1.0 If you are aware of a change in the behaviour for some parameters, functions or statements, list them there. I added the sql statements for MySQL that will allow to get to the new database structure. Would be good to have the same for other popular sql servers (maybe postgres, sqlite ...). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Inser Delayed in MySQL
by Gus Monti 29 Oct '13

29 Oct '13

Hi, I can see that we are missing a lot of Sip messages in or db, so, checking Homer FAQ there is a way to compile MySql with Insert Delayed for partitioning table. The problem is that I don't understand how to recompile MySql and make the necessary changes. The URL is https://code.google.com/p/homer/wiki/FAQ#Q:_I_have_a_lot_of_SIP_traffic_an d_some_SIP_messages_are_missing The question is, can you send me a step by step how to do this? I really appreciate your help. I have CentOS 6.4: [root@Homerdb02 ~]# kamailio -v version: kamailio 4.1.0-dev9 (x86_64/linux) f8f3d3 [root@Homerdb02 ~]# cat /etc/redhat-release Red Hat Enterprise Linux Server release 6.4 (Santiago) [root@Homerdb02 ~]# uname -a Linux Homerdb02 2.6.32-358.18.1.el6.x86_64 #1 SMP Fri Aug 2 17:04:38 EDT 2013 x86_64 x86_64 x86_64 GNU/Linux [root@Homerdb02 ~]# mysql -V mysql Ver 14.14 Distrib 5.5.34, for Linux (x86_64) using readline 5.1 Thanks, Gus

2 2

Kamailio Tech Admin Group
by Daniel-Constantin Mierla 29 Oct '13

29 Oct '13

Hello, being discussed during last Devel IRC Meeting, we are planing to build a Kamailio Project Technical Administration Group: https://www.kamailio.org/wiki/devel/irc-meetings/2013blog#technical_adminis… Its goal is to get a bunch of people that volunteer to do administration tasks for the project, such as: - helping with releases (e.g., patch backports, packaging, uploading files for download, etc) - doing sysadmin tasks for our servers (e.g., performing upgrades to wiki, web site, etc) - preparing technical decisions and doing them (e.g., what applications to use to make operations easier, cloning git repository to github, ...) From the devel meeting, so far we have Victor Seva, Fred Posner, Peter Dunkley and Olle Johansson. Existing people doing admin tasks will probably stay in (if they don't opt out): me, Elena-Ramona Modroiu, Henning Westerholt (owner of devel.kamailio.org hardware), Jan Janak (owner of sip-router.org hardware), Jesus Rodriguez and Oriol Capsada (owners of kamailio.org hardware). Requirements for candidates and other details: - volunteer to do the work, it is not a paid job - an existing record of activity within the project is a plus (e.g., developer, active mailing list member) - reply to the lists detailing where and how you can help - possibility to spend 1-4 hours a week for project administration (more is welcome, sometime is not necessary at all) Rewards: - you will be listed as part of project administration on the website - get to interact more with the project and the nice guys around it ;-) - more spam - admin list address will be public and the list open so everyone can send in case of critical situations (content/archive will be kept private) Note that we will try to build a group of an adequate size, thus not everyone willing to participate may get in (at least on the first phase). One criteria is to have skills that complement existing team knowledge. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

5 5

ACK not sending to the invite to header but to the contact.
by anfecora 29 Oct '13

29 Oct '13

Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside. when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok propperly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok. any help is apreciated . thanks. my setup request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") { #Account BYE transactions }; if (method=="CANCEL") { if (t_check_trans()) t_relay(); exit; }; if (loose_route()) { t_relay(); exit; } if (is_method("INVITE")) { record_route(); } f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit }; here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks. asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2 U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060 INVITE sip:23276341079@2.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060 INVITE sip:76890723276341079@3.1.1.1:5060 SIP/2.0. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: gProxy (1.8.3 (i386/Linux)). Content-Length: 0. . U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP] .....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967',' 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1 ','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060',' sip:23276341079@2.0.0.1','OUT') U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. . U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 16. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:176822213@1.1.1.1>;tag=as4bc322e9. From: 0 <sip:079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. .

2 2

Dialog callback after restart
by Kristian F. Høgh 29 Oct '13

29 Oct '13

Hi, After restarting kamailio, pua_dialoginfo doesn't get updated when active dialogs change status. I see the dialogs are read from the database, but when they receive eg. BYE the callback are not called. I use db_mode 0 in pua, but tried mode 2 with no luck. Should pua_dialoginfo register DLGCB_LOADED? Regards, Kristian Høgh.

2 2

Re: [SR-Users] Kamailio network edge for registration and rtp pass-through
by JR Richardson 29 Oct '13

29 Oct '13

Date: Fri, 25 Oct 2013 17:48:13 -0400 From: Fred Posner <fred(a)palner.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: SR-Users <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio network edge for registration and rtp pass-through Message-ID: <6D231DE3-5BED-4601-A097-5A79CEF452F0(a)palner.com> Content-Type: text/plain; charset=us-ascii Do you need the registration be local to the asterisk? I would have all the asterisks send calls to the Kamailio. You can have a lookup on endpoint outbound to decide which asterisk should handle the outbound call for that did. Also a lookup for incoming DIDs, etc. ---Fred > On Oct 25, 2013, at 5:43 PM, Jr Richardson <jmr.richardson(a)gmail.com> wrote: > > Hi All, > > Starting a new project, roll your own SBC, not a full SBC, just need some minor functionality. I'm interested in deploying Kamailio as a edge device on a VSP for single entry point for hosted PBX's, Asterisk based. I had some wonderful and informative conversations at Astricon 2013, several folks assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow up to delve more into details. > > I've been researching configs, topology, modules needed, ect... Most of the examples I'm reading about for this scenario are spreading the registrations across many PBX's without distinction. One concept I'm struggling with is having a specific phone register to a specific PBX. > > phone-customer-A-x101><internet><kamailio><PBX-customer-A-x101 > phone-customer-B-x101><internet><kamailio><PBX-customer-B-x101 > phone-customer-C-x101><internet><kamailio><PBX-customer-B-x101 > > I could add a unique identifier to some part of the registration of each phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect. What I'm not clear on is when the request comes to kamailio, where would I identify what PBX the phone should register to and how to re-write the 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX and ensure rtp flows through kamailio. > > Could this function be derived using dbaliases or possibly using dispatcher with group number for each customer PBX? > > So assuming I can get the registrations to work properly, would standard invite for calling just work or would I also have to have specific config in place to ensure an invite from customer A phone also reaches the correct customer A PBX? > > A point in the right direction? > > Thanks. > > JR I would like kamailio to be transparent in the process, i.e. phones register locally to their respective PBX. Thanks. JR

1 0

Fwd: Presence Proxy
by Jan Gaida 29 Oct '13

29 Oct '13

Hello, I'd like to explain my scenario and the module I'd like to develop. I am hoping for comments whether there are already modules like that or which modules' API I could use to make the development easier. Scenario: I have two domains (A and B) with a Kamailio server in each of them configured as presence servers. The agents from both domains want to subscribe to few resources from domain A. Idea: In order to reduce traffic between the domains, the presence server from domain B shall handle the subscriptions internally and use an internal virtual presence-user-agent to subscribe to the resources in domain A. SIP Messages: Subscription: 1) Subscribe resource.domain.a, from each user agent ua_n.domain.b 2) 202 OK from kamailio.domain.b 3) If not already subscribed, Subscribe resource.domain.a from kamailio.domain.b with virtual user-agent 4) 202 OK from kamailio.domain.a (Repeat for each subscriber) (Subscription has to be refreshed by an internal timer) Events: 1) Publish resource.domain.a to kamailio.domain.a 2) 200 OK from kamailio.domain.a 3) Notify resource.domain.a to kamailio.domain.b 4) 200 OK from kamailio.domain.b 5) Notify resource.domain.a to each user agent ua_n.domain.b 6) 200 OK from each ua_n.domain.b (Internally, Kamailio of domain B has to forward the incoming Notify to all locally subscribed user-agents) I tried using the PUA module but it does not fit all my needs. Now I think I will have to develop my own module in order to achieve everything. Any ideas? Kind regards Jan Gaida PS: Sorry for mailing to both mail-lists. But I was not sure which list was the correct for this topic. -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida(a)grupoamper.com | www.grupoamper.com -- This message and any attachments are intended only for the use of the individual to whom they are addressed and it may contain information that is privileged or confidential. If you have received this communication by mistake, please notify us immediately by e-mail or telephone.The storage, recording, use or disclosure of this e-mail and its attachments by anyone other than the intended recipient is strictly prohibited. This message has been verified using antivirus software; however, the sender is not responsible for any damage to hardware or software resulting from the presence of any virus. Este mensaje y cualquier anexo son exclusivamente para la persona a quien van dirigidos y pueden contener información privilegiada o confidencial. Si usted ha recibido esta comunicación por error, le agradecemos notificarlo de inmediato por esta misma vía o por teléfono. Está prohibida su retención, grabación, utilización o divulgación con cualquier propósito. Este mensaje ha sido verificado con software antivirus; sin embargo, el remitente no se hace responsable en caso de que en éste o en los archivos adjuntos haya presencia de algún virus que pueda generar daños en los equipos o programas del destinatario.

2 2

Re: [SR-Users] Pickup group implementation in Kamailio
by Daniel-Constantin Mierla 29 Oct '13

29 Oct '13

Hello, for pickup group look at pua_/presence_dialoginfo or sca modules. Cheers, Daniel On 10/23/13 11:52 AM, John Murray wrote: > > Hi, > > I am also looking for a 'group pickup' function with Kamailio. > > Once I have the information on the call I want to get how would I > 'grab' the call. > > My calls are being processed with the tm module. > > I imagine I send an INVITE with the 'replaces' header but what > Kamailio module would process that and what would the config look like? > > Many thanks > > John > > >/ Hello,/ > > >// > > >/this is a custom implementation, only for phones that have programmable/ > > >/buttons. It is not really recommended way to do it./ > > >// > > >/Better rely on standard specs out there and use presence extensions for/ > > >/such feature -- in kamailio you would need to use modules such as > dialog,/ > > >/presence, pua, presence_dialoginfo and pua_dialoginfo./ > > >// > > >/Cheers,/ > > >/Daniel/ > > >// > > >// > > >/On 5/14/12 1:15 AM, M.C. wrote:/ > > >// > > >/Hi all,/ > > >/I want to implement a "group pickup" (asterisk like) with Kamailio and/ > > >/Snom phones./ > > >/I found in the changelog this comment:/ > > >/"- an useful application - call pickup:/ > > >/ - INVITE comes in, callee is in a pickup group/ > > >/ - store callid, cseq, etc in database or notify the other phones in/ > > >// > > >/ the group (easy to do for snom phones and extra programmable/ > > >/ buttons) via uac_req_send()/ > > >/ - send a remote control command telling which call to pickup/ > > >/ (callid, cseq) and where to redirect. In config, call/ > > >/ t_cancel_callid(). in failure route for INVITE catch the > cancelled/ > > >/ transaction (flag is set), get the new destination (e.g., > stored in/ > > >/ htable) and forward the invite there""/ > > >// > > >/regarding new functions in tmx module, but I have found no examples > in kamailio documentation./ > > >// > > >/Is this the way to implement a group pickup in kamailio?/ > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

ALIAS_DB module resolving to multiple user@domain does not work when use_domain is set to 1
by Vassilis Radis 29 Oct '13

29 Oct '13

The documentation of the append_brances(int) function in ALIAS _DB module says: "If the alias resolves to many SIP IDs, the first is replacing the R-URI, the rest are added as branches." However this only works if not using the domain for resolving, i.e. use_domain is set to 0. This happens because there is a unique constraint on the dbaliases table for the columns (alias_username, alias_domain). Would it hurt to drop the constraint so that I can have multiple branches when resolving username AND domain? Any other suggestions/comments? Thanks.

2 1

How to add debugger module in kamailio.cfg
by Owais ul Haq 29 Oct '13

29 Oct '13

Hello, I want to add "dubugger module" in kamailio. As I want to know which part of kamailio.cfg is responsible for reply of some of incoming sip messages. Can some one please help me in this regard. Regards, Kamailio kid.

2 1

Help with 200k responses to the contact header not the record-route
by anfecora 29 Oct '13

29 Oct '13

Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside. when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok properly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok. I am sorry i sent a too long message before i will try skim it a bit. any help is appreciated . thanks. my setup request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") { #Account BYE transactions }; if (method=="CANCEL") { if (t_check_trans()) t_relay(); exit; }; if (loose_route()) { t_relay(); exit; } if (is_method("INVITE")) { record_route(); } f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit }; here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks. asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2 U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0.

2 1

Regarding Exporting a function
by premchandiran.marimuthu 29 Oct '13

29 Oct '13

Hi All, I was able to export a function to cfg file with int return type but when i try to export a function with char/char-ptr as return type , in cfg file i was not able to print/use the characters which i returned in the exported function. can i get a clue/tip to overcome above said issue?. Regards, Prem Chandiran M -- Open WebMail Project (http://openwebmail.org)

2 1

Presence issues
by Wingsravi R 29 Oct '13

29 Oct '13

Hi kamailio'ns I am working on Kamailio V 4.0.3, installed from Ubuntu 12.04(precise) repository. My SIP clients(jitsi) are registering to kamailio sever successfully but their presence is not sharing between the clients. I am frequently logging following presence related errors in syslog while trying to register SIP clients to kamailio server: Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15274]: INFO: pua_usrloc [ul_publish.c:221]: ul_publish(): not marked for publish Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15273]: WARNING: presence [publish.c:481]: handle_publish(): Missing or unsupported event header field value Oct 25 20:01:05 kamailio /usr/sbin/kamailio[15273]: ERROR: presence [publish.c:484]: handle_publish(): event=[reg] what could be the wrong ? How can i reslove this issue ? please find in the attachment is my kamailio config file. (suggest me if any changes to be made to this configuration) Any help will greatly appreciate. Regards, Ravi

2 3

kamailio 3.3.2 db_mysql.so problem after mysql upgrade to 5.6
by Uri Shacked 28 Oct '13

28 Oct '13

Hi, I upgraded mysql from 5.5 to 5.6. After the upgrade kamailio fails to start and throws the following error: "..... /usr/lib64/libmysqlclient.so.18: version `libmysqlclient_16` not found....." Any ideas what to do? Thanks, Uri

2 2

media between 2 clients behind separate NAT
by Vassilis Radis 28 Oct '13

28 Oct '13

Hello, When I have 2 clients using a kamailio proxy, and both of the clients are behind their own NAT, then my only options for relaying media between them is using some kind of intermediate rtp proxy or STUN etc? Proxy is on a public IP and essentially I am asking if I can avoid setting up stuff to proxy the media or mess with STUN, by using some kind of request response mangling. i.e. have both clients's sdp address/port changed to their public facing ip/port and then each client can send an initial packet to the other end. Those initial packets will be blocked by the other receiver router, but they will open a nat hole so that next package from the peer will pass through. Is that feasible? Thanks.

2 1

How can I use a module from the devel branch in a stable varsion installation
by Vassilis Radis 27 Oct '13

27 Oct '13

Hello, I would like to use rtpproxy-ng module from the devel branch in an 4.0.4 installation. I have succesfully compiled the mediaproxy-ng kernel modules. I would like to know if there is a way to compile the rtpproxy-ng module in the devel branch in such a way that I dont get the following error "kamailio: ERROR: <core> [sr_module.c:422]: version_control(): ERROR: module version mismatch for /usr/local/lib/kamailio/modules/rtpproxy-ng.so; core: kamailio 4.0.4 (i386/linux); module: kamailio 4.1.0-pre0 (i386/linux)" Thanks.

2 2

Issue with failure_route on kamailio 4.0
by mayamatakeshi 27 Oct '13

27 Oct '13

Hello, I am trying to upgrade from kamailio 1.4 to 4.0. I am having an issue with call forwarding: I send the call to a subscriber (location) and if it doesn't answer, there is a failure_route to do the forward. It is something like this: failure_route[1] { $ru = "sip:0313241342@192.168.2.2" append_branch(); t_relay(); } However, the INVITE is sent to both the subscriber location and the new branch. So, is it now necessary to somehow to "drop" the original branch? If so, how do I do that? Regards, Takeshi

2 2

Usrloc errors
by Wingsravi R 26 Oct '13

26 Oct '13

Hi kamailio'ns I am working on MSRP module of Kamailio serverV 4.0.3, which is installed from Ubuntu 12.04(precise) repository. When i am trying to register SIP clients(Blink) to kamailio server, frequently logging following user location related errors in syslog, SIP clients are registering to kamailio sever successfully, but still logging these errors and client presence is not visible to other client. Oct 25 20:09:05 kamailio /usr/sbin/kamailio[15277]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Duplicate entry 'uloc-526a7929-3bab-91' for key 'ruid_idx' Oct 25 20:09:05 kamailio /usr/sbin/kamailio[15277]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Oct 25 20:09:05 kamailio /usr/sbin/kamailio[15277]: ERROR: usrloc [ucontact.c:625]: db_insert_ucontact(): inserting contact in db failed Oct 25 20:09:05 kamailio /usr/sbin/kamailio[15277]: ERROR: usrloc [urecord.c:368]: wb_timer(): updating contact in db failed (aor: 1112(a)kamailio.test) Below attachment is my Kamailio config file (please suggest me if any changes to be made). Anybody please help me in resolving the issues PS: With the same script when i used to register the clients (jitsi) to kamailio i am logging presence related errors (which i posted too in the mailing list with the subject as : Presence issues) Regards, Ravi

2 2

Kamailio network edge for registration and rtp pass-through
by Jr Richardson 25 Oct '13

25 Oct '13

Hi All, Starting a new project, roll your own SBC, not a full SBC, just need some minor functionality. I'm interested in deploying Kamailio as a edge device on a VSP for single entry point for hosted PBX's, Asterisk based. I had some wonderful and informative conversations at Astricon 2013, several folks assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow up to delve more into details. I've been researching configs, topology, modules needed, ect... Most of the examples I'm reading about for this scenario are spreading the registrations across many PBX's without distinction. One concept I'm struggling with is having a specific phone register to a specific PBX. phone-customer-A-x101><internet><kamailio><PBX-customer-A-x101 phone-customer-B-x101><internet><kamailio><PBX-customer-B-x101 phone-customer-C-x101><internet><kamailio><PBX-customer-B-x101 I could add a unique identifier to some part of the registration of each phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect. What I'm not clear on is when the request comes to kamailio, where would I identify what PBX the phone should register to and how to re-write the 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX and ensure rtp flows through kamailio. Could this function be derived using dbaliases or possibly using dispatcher with group number for each customer PBX? So assuming I can get the registrations to work properly, would standard invite for calling just work or would I also have to have specific config in place to ensure an invite from customer A phone also reaches the correct customer A PBX? A point in the right direction? Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Help With kamailio not responding ack propperly.
by anfecora 25 Oct '13

25 Oct '13

Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside. when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok properly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok. any help is appreciated . thanks. my setup request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") { #Account BYE transactions }; if (method=="CANCEL") { if (t_check_trans()) t_relay(); exit; }; if (loose_route()) { t_relay(); exit; } if (is_method("INVITE")) { record_route(); } f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit }; here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks. asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2 U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060 INVITE sip:23276341079@2.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060 INVITE sip:76890723276341079@3.1.1.1:5060 SIP/2.0. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: gProxy (1.8.3 (i386/Linux)). Content-Length: 0. . U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP] .....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967',' 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1 ','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060',' sip:23276341079@2.0.0.1','OUT') U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. . U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 16. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:176822213@1.1.1.1>;tag=as4bc322e9. From: 0 <sip:079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. .

2 3

Invite Fragmented
by Gus Monti 25 Oct '13

25 Oct '13

Hi, There are often times where key packets are missing from homer traces. This seems to be caused by packet fragmentation. It is most readily evidenced by the trace showing an Invite entering the SBC from the outside and then no invite on the inside being seen but the 100 trying is seen on both sides. Any Idea why? Thanks, Gus

3 2

RTP Proxy issue on Kamailio 4
by Mahmoud Ramadan Ali 25 Oct '13

25 Oct '13

*Hello , This is the first time to put a question on *sr-users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> mailing list * and i hope to help me to solve my issue...i'm following this post http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html to setup RTP Proxy for kamailio 4 but it's NOT working as predicted and i'm receiving this error message on asterisk...* [2013-10-16 05:20:04] ERROR[3216][C-0000002c]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("192.168.1.80192.168.1.80", "(null)", ...): Name or service not known [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10873 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 192.168.1.80192.168.1.80' [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10464 process_sdp: Insufficient information in SDP (c=)... The problem is NOT at the asterisk side but at the proxy side...when kamailio rewrites the SDP message it adds the public IP twice in the "C" filed of the SDP message instead of only one time like this 'IN IP4 192.168.1.80192.168.1.80' but i don't know why it's doing this...? if u have any idea or recommendations plz tell me cuz i'm new to kamailio...waiting ur reply and thanks in advance... also here's my config file if u want to take a look at it... #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://root:passw0rd@192.168.1.10/mya2billing" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 mhomed=1 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.1.10" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.1.40" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "ipops.so" loadmodule "sqlops.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "secret") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif # ------- Load-balancer params ------ modparam("dispatcher", "db_url","mysql://kamailio:kamailiorw@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags_col", "flags") modparam("dispatcher", "priority_col", "priority") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:Kamailio-Proxy@192.168.1.40") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_threshhold", 3) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") # ----- SQLops params ----- modparam("sqlops","sqlcon","ca=>mysql://kamailio:kamailiorw@localhost /kamailio") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))){ return; } route(NATMANAGE); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "cc_sip_buddies", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { if (is_method("INVITE")){ sql_query("ca", "select destination from dispatcher where destination like '%$dd%'","ra"); if($dbr(ra=>rows)>0){ $avp(duip)=$(du{s.select,-2,:}); if (is_ip_rfc1918("$avp(duip)")) { xlog("L_INFO", "Call is going to private IPv4 Media Server Engage RTPProxy Now\n"); #rtpproxy_manage("crwie","192.168.1.40"); rtpproxy_manage("rwie"); } } else if(ds_is_from_list()){ if (is_ip_rfc1918("$si")) { xlog("L_INFO", " Call is coming from a private IPv4 Media Server Engage RTPProxy Now\n"); #rtpproxy_manage("crwei","192.168.1.80"); rtpproxy_manage("rwei"); } }else if(!ds_is_from_list()){ rtpproxy_manage("rwie"); } } } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if(ds_is_from_list()){ rtpproxy_manage("cawie"); xlog("L_INFO","[$fU@$si:$sp]{$rm} Call from Media-Server Cluster\n"); return 1; } return -1; } # Send to Asterisk route[TOASTERISK] { ds_mark_dst("IP"); if(!ds_select_dst("1", "9")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n"); exit; } xlog("L_INFO","[$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); rtpproxy_manage("cawei"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif

2 2

Debugging Logs Location
by Owais ul Haq 25 Oct '13

25 Oct '13

Hi all, I have loaded debugger module in my kamailio server and set "cfgparameter" to 1. Can some one please tell me where can I now find logs of this cfg trace. Any help will be appreciated. Thanks. Regards, M. Owais ul Haq

2 1

Audio Codec option in Siremis 4.0
by Adarsh Pillai 25 Oct '13

25 Oct '13

Hi Where do i find the option for enabling and disabling audio codecs in Siremis 4.0. Regards Adarsh Pillai

2 1

MSRP module - Ubuntu repo
by Wingsravi R 24 Oct '13

24 Oct '13

Dear Kamailio'ns, I have Installed kamailio V 4.0.3 from Ubuntu 12.04 (precise) repository and running successfully. now i have to work around MSRP relay module of kamailio sever to achieve file transfer between two SIP clients. so, i just tried to install MSRP module, but i get logged as below: /$ sudo apt-get install kamailio-msrp-modules Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package kamailio-msrp-modules Is MSRP module is included in Ubuntu repository? how can i install it ? Thank you and regards,

2 6

proxy is not forwarding 180 ringing reply
by Juha Heinanen 24 Oct '13

24 Oct '13

i have been wondering why my proxy is not forwarding 180 ringing reply to caller that it receives from callee. invite was forwarded to callee by t_relay(). proxy does not print any error messages to syslog. the only thing i get is the message i have in onreply_route: onreply_route [REPLY] { xlog("L_INFO", "Received reply\n"); return; } level 3 debug of 180 reply processing is below. does it tell anything to tm gurus that would explain why kamailio does nothing to the reply? 180 reply itself looks like this: Session Initiation Protocol Status-Line: SIP/2.0 180 Ringing Message Header Record-Route: <sip:192.98.102.30;transport=tcp;r2=on;lr> Record-Route: <sip:192.98.102.30;r2=on;lr> Via: SIP/2.0/TCP 192.98.102.30;branch=z9hG4bK84cf.498ea35cdde08fdfad989b 9212a2a9a5.0 Via: SIP/2.0/UDP 192.98.102.31:55419;branch=z9hG4bKd6e2f0db78e2df94;rpor t=55419 To: <sip:test@test.tutpro.com>;tag=8e143c1b780cfb1b From: <sip:jh@test.tutpro.com>;tag=630450233fa9bc92 Call-ID: d0b328ba01bcc483 CSeq: 22643 INVITE Server: baresip v0.4.6 (i586/linux) Contact: <sip:0x94fa6e8@192.98.102.30:49240;transport=tcp> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,NOTIFY,SUBSCRIBE,INFO Content-Length: 0 -- juha Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [tcp_read.c:1510]: handle_io(): received n=4 con=0xb4252638, fd=15 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [tcp_read.c:1317]: tcp_read_req(): tcp_read_req: content-length= 0 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status): Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <180> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <Ringing> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK84cf.498ea35cdde08fdfad989b9212a2a9a5.0>; state=16 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [receive.c:151]: receive_msg(): After parse_msg... Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffff Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKd6e2f0db78e2df94>; state=6 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 235, <rport> = <55419>; state=16 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=62 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=8e143c1b780cfb1b Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/parse_addr_spec.c:885]: parse_addr_spec(): end of header reached, state=29 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:test@test.tutpro.com] Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:test@test.tutpro.com>] Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <22643> <INVITE> Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 64584 label 0 branch 0 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1003]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0xb4256d88)! Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0xb4256d88, callback type 2, id 0 entered Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: acc [acc_logic.c:538]: tmcb_func(): acc callback called for t(0xb4256d88) event type 2, reply code 180 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_lookup.c:1140]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=5 T end=0xb4256d88 Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: tm [t_reply.c:2258]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=1) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: INFO: Received reply Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [receive.c:295]: receive_msg(): receive_msg: cleaning up Oct 23 17:40:51 rautu /usr/sbin/sip-proxy[8730]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x82ca540, 15, 2, 0xb4252638), fd_no=1

4 12

clearing pua table
by SIP Guru 24 Oct '13

24 Oct '13

Hi Kristian, thank you. I'll check this. I think in some cases we are using even a higher lifetime. Could you please send me a patch with your changes for the pua_dialoginfo module in case we run into the same issue. Toni On Tuesday 22 October 2013 13:19:01 Kristian Høgh wrote: >Hi, > >I think I've got the same issue. >At call creation we set dialog lifetime to 6 hours. >pua_dialoginfo uses that value and send PUBLISH for each hour, even after the dialog terminates. >After 6 hours the entry is deleted. > >To prevent the updates, I've made a small change to pua_dialoginfo module, which set pua-dialog timeout to 300 sec. when the dialog terminates. >pua_dialoginfo then sends PUBLISH, which updates presence. > >Regards, >Kristian Høgh > > >On Monday 21 October 2013 15:21:16 SIP Guru wrote: >>/ Hi, />>/ />>/ expired entries in pua table won't be deleted. /<>/ And kamailio keep sending PUBLISH's for expired entries. />>/ />>/ Am I missing something ? />>/ />>/ Toni />>/ />>/ />>/ />>/ _______________________________________________ />>/ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list />>/ sr-users at lists.sip-router.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> />>/ http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users />

3 3

Pickup group implementation in Kamailio
by John Murray 23 Oct '13

23 Oct '13

Hi, I am also looking for a 'group pickup' function with Kamailio. Once I have the information on the call I want to get how would I 'grab' the call. My calls are being processed with the tm module. I imagine I send an INVITE with the 'replaces' header but what Kamailio module would process that and what would the config look like? Many thanks John > Hello, > > this is a custom implementation, only for phones that have programmable > buttons. It is not really recommended way to do it. > > Better rely on standard specs out there and use presence extensions for > such feature -- in kamailio you would need to use modules such as dialog, > presence, pua, presence_dialoginfo and pua_dialoginfo. > > Cheers, > Daniel > > > On 5/14/12 1:15 AM, M.C. wrote: > > Hi all, > I want to implement a "group pickup" (asterisk like) with Kamailio and > Snom phones. > I found in the changelog this comment: > "- an useful application - call pickup: > - INVITE comes in, callee is in a pickup group > - store callid, cseq, etc in database or notify the other phones in > > the group (easy to do for snom phones and extra programmable > buttons) via uac_req_send() > - send a remote control command telling which call to pickup > (callid, cseq) and where to redirect. In config, call > t_cancel_callid(). in failure route for INVITE catch the cancelled > transaction (flag is set), get the new destination (e.g., stored in > htable) and forward the invite there"" > > regarding new functions in tmx module, but I have found no examples in kamailio documentation. > > Is this the way to implement a group pickup in kamailio?

1 0

Kamailio MSRP relay problem
by mohammed alyaseen 23 Oct '13

23 Oct '13

Hi all... I am trying to do file transfer between IMS clients (web-based WebRTC and android RCS native client). The fisrt test scenario is browser-to-browser (WebRTC) using JSSIP stack. The second scenario is RCS-to-Broswer or Browser-to-RCS. My testbed consists of OpenIMS and OverSIP for WebRTC clients and Kamailio MSRP Relay for file transfer For file transfer I am using the msrp-crocodile library... The scenario browser-to-browser works so far without problems. But in the second scenario I have the following problems: * When I try to transfer a file from the RCS client to WebRTC client , kamailio sends the message "error ""481", "Session-does-not-exist", however the file is sent, when I click send for the second time... * When I try to send a file from WebRTC client to RCS client, WebRTC client sends all file chunks to kamailio, kamailio receives the chunks but it doesn't forward them to the RCS client, in the console I see some errors pointed to the configuration file, however I don't know what exactly the problem is. In the attachment please find the logs and the configuration file of the kamailio msrp relay... Can any one help me to solve this problem or give me useful hints? Thanks a lot in advance.... Medo

2 3

why kamcmd tls.reload is not safe
by Ding Ma 23 Oct '13

23 Oct '13

In the current Kamailio TLS module document, there is a statement about tls.reload being unsafe. But the only way to periodically update CRL without restarting Kamailio is to use tls.reload. In our test with tls.reload for CRL, it seems Kamailio would crash after about 100 times of tls.reload in 5/6 hours. The core dump indicates memory access violation, signal 11. We compiled Kamailio with openssl 1.0.0-fips. Would appreciate some insights on tls.reload and ideas to fix the crash issue. Thanks,

3 4

Dialog profiles using $fU and $rU
by Daniel W. Graham 22 Oct '13

22 Oct '13

Having an odd issue with dialog profiles, if I try to set/check profile based on $fU or $rU call is always rejected due to call limit, but $ru or $fu work without an issue. Example using $fU- $avp(s:checkuser) = $fU; set_dlg_profile("quota","$avp(s:checkuser)"); dlg_manage(); get_profile_size("quota","$avp(s:checkuser)","$avp(s:calls)"); avp_db_load("$avp(s:checkuser)/username", "$avp(s:callquota)"); if ($avp(s:calls) <= $avp(s:callquota)) ....... If I change checkuser to $avp(s:checkuser) = $fu; it returns the correct result. Any explanation for why this is? [cid:image001.gif@01CECF2B.581704D0]

2 1

Monitoring events (presence)
by Paul Belanger 22 Oct '13

22 Oct '13

I'm looking for some information about how best to monitor presence events within kamailio? Ideally I'd be writing a python application that will monitor specific endpoints within kamailio and once there state changes I do my thing. I know tapping into the database is an option, but figure I see if there is another method. EG: Opening a websocket to stream events? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

3 5

Help with NAT
by P. S. 22 Oct '13

22 Oct '13

Hello there, I am trying to setup a Kamailio (3.3) + Linphone + Jitsi based private calling system (users can call each other but not call out of the network). Agents can be behind NAT (think Verizon cellphone or home-users behind their router). I started with the default setup and have tried different settings for NAT/rtpproxy. Currently, all my users register onto the Kamailio server. Users who are not behind a NAT (Jitsi on the internet) i.e. with Public IPs are able to successfully make audio/video calls. I have RTP proxy setup / running as well, and calls to rtpproxy_manage() as required. The process is running and Kamailio is configured to know that the RTPproxy is listening. I had to switch to using TCP on some of the my UAs because of provider blocking UDP traffic. So they are able to register, yet calls are not completed. The call looks like it connects but no media (audio or video goes through). I have read a lot of the forums on NAT - and am trying to solve for a couple of big issues: 1. Figuring out when a UA is behind a NAT, and handling the RTPProxying appropriately. 2. while ngrep-ing UDP packets occassionally I see "Warning: 399 sipalg "Unauthorized" or Bad request. based on what I saw on the internet, it has to do with the service provider (Verizon) messing with the packet. Is there a way arount this? Any help is greatly appreciated! Thanks Pranv

2 1

clearing pua table
by SIP Guru 22 Oct '13

22 Oct '13

Hi, expired entries in pua table won't be deleted. And kamailio keep sending PUBLISH's for expired entries. Am I missing something ? Toni

2 1

[kamailio-carrierroute]
by Dzexolokpli AMOUZOU 22 Oct '13

22 Oct '13

Hello, I installed kamailio 4.0 with packages (apt-get install kamailio). So everything works well. Now I want to install carrierroute modules to route some calls. Please how can I do it? Help!!! thanks. Dzexolokpli AMOUZOU

1 0

XMPP- component mode
by Wingsravi R 22 Oct '13

22 Oct '13

Hi Kamailio community, I am working around with Kamailio (V 4.0.3), its XMPP module- component mode and jabbed2 server, intended to get file transfer feature between two SIP clients. In a way while surfing through the blogs i got some info like this: 'In component mode, a sub domain is diverted to respective component,so you don't need users in XMPP sever'. (http://lists.sip-router.org/pipermail/sr-users/2010-August/065209.html). With this my question is: what does it mean ? Is it mean like i dont need to register xmpp clients to jabbedd2(xmpp) server ? If this is the case, How can i use my SIP users (registered to kamailio server) in jabberd2 server context ? Even XMPP module's man page doesn't give clarity about these questions. Any help will greatly appreciate. Regards, Ravi

3 4

AWS LOOP detected
by julian arsanches 22 Oct '13

22 Oct '13

Daniel, Klaus thanks a lot for your responses i do appreciate them, just got me a bit confuse because on the documents as what Daniel suggest to advertise directive for listen parameter, I do if you see my config below that i advertise it and have an alias, kamailio still sends an invite to itself without recognizing that it came from the internal ip. Is there anything else wrong that i am not putting on my advertise line ? thanks i really apreciate your input. Julian. *listen=udp:1.1.1.1:5060 advertise 2.2.2.2:5060*>>* *>>**>>* alias=1.1.1.1:5060 <http://1.1.1.1:5060>*>>* alias=1.1.1.1*>>* alias=2.2.2.2:5060 <http://2.2.2.2:5060>*>>* alias=2.2.2.2* >Hello, > >for reference, in case kamailio is running behind nat, like on AWS, one >should just use advertise directive for listen parameter, like:> > >listen=proto:privateip:port advertise publicip:port> > > >See: > >http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen > >It simplifies the config by not needing to use alias parameter, >advertised_address/set_addvertised_address() and record_route_preset().> > >Cheers, >Daniel > >On 10/18/13 8:23 AM, Klaus Darilion wrote: >**>**>* On 17.10.2013 17:31, julian arsanches wrote:*>>* Hi all, before hand thanks for all the support received on this channel.*>>**>>* I have an issue with an installed server on a aws instance which is*>>* giving me routing loops, my setup is simple, i have alias set for both*>>* ips internal and external and the loop still present when t_relay don't*>>* find any routes to relay it just sent it to itself with the public ip*>>* then the loop starts.*>>**>>* can any one please advise how can i prevent that or just tell t_relay*>>* that if there is no more routes just respond with a cancel or stop*>>* searching.*>**>* If you call t_relay, then the message will be forwarded to $ru (or to *>* $du if $du is set). t_relay itself does not have any logic inside.*>**>* Thus, if you do not want a message to be forwarded, then you must not *>* call t_relay.*>**>* If you do not want to send a message to a certain destination, then *>* set $ru (the domain part) to the respective destination.*>**>* One issue may be the aliases. If Kamailio thinks that it is not *>* authoritative for a domain, then it forwards the message to the *>* domain. Kamailio by default feels authoritative for the IP addresses *>* on which it listens (the private IP addresses). Thus if a request is *>* received by Kamailio, but the RURI (request URI, $ru) indicated the *>* public IP address, KAmailio by default forwards the message to the *>* respective RURI (which my be forward again by AWS NAT to Kamailio).*>**>* Thus, alias=public.ip.addr.ess must be added to the config.*>**>* If there are still problems you have set debug=4 and track in the log *>* file how the message is processed, and where the routing decision *>* takes the wrong direction.*>**>* regards*>* Klaus*>**>>**>>* same thing happens when a gw code is not found on the dispatcher list *>>* file.*>>**>>* as i had read by just specifying the alias=domain it should take care of*>>* it but it is not happening perhaps because it is a AWS instance, or most*>>* likely i am missing something after the T_relay to stop the call from*>>* looping.*>>**>>* I do have several headers fix because i have an asterisk sending calls*>>* with lots of signalling calls added to the call that i need to remove*>>* before the call is passed out to a carrier or another UA, Kamailio is*>>* just proxing the calls between asterisk and the outside world no Media*>>* handling, No registrations.*>>**>>* Again thanks for any tips to find my answer or if anyone had a similar*>>* issue thanks again.*>>**>>* thanks for any help .*>>**>>* Julian.*>>**>>* on this setup i have*>>**>>* public ip as 2.2.2.2*>>**>>* private IP as 1.1.1.1*>>**>>**>>* #!KAMAILIO*>>* ####### Global Parameters #########*>>**>>* #!ifdef WITH_DEBUG*>>* debug=4*>>* log_stderror=yes*>>* #!else*>>* #debug=2*>>* #log_stderror=yes*>>* #!endif*>>**>>* memdbg=5*>>* memlog=5*>>**>>* log_facility=LOG_LOCAL7*>>**>>* fork=yes*>>* children=6*>>**>>* auto_aliases=no*>>**>>**>>* listen=udp:1.1.1.1:5060 <http://1.1.1.1:5060> advertise 2.2.2.2:5060*>>* <http://2.2.2.2:5060>*>>**>>* alias=1.1.1.1:5060 <http://1.1.1.1:5060>*>>* alias=1.1.1.1*>>* alias=2.2.2.2:5060 <http://2.2.2.2:5060>*>>* alias=2.2.2.2*>>**>>**>>* port=5060*>>**>>* enable_tls=no*>>**>>* #tcp_connection_lifetime=3605*>>**>>* #!ifdef WITH_SRCPATH*>>* mpath="modules_k:modules"*>>* #!else*>>* mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/"*>>**>>* #!endif*>>* loadmodule "db_mysql.so"*>>* ##loadmodule "mi_fifo.so"*>>* loadmodule "tm.so"*>>* loadmodule "sl.so"*>>* loadmodule "acc.so"*>>* loadmodule "rr.so"*>>* loadmodule "maxfwd.so"*>>* loadmodule "uac.so"*>>* loadmodule "textops.so"*>>* loadmodule "pv.so"*>>* loadmodule "dispatcher.so"*>>* loadmodule "usrloc.so"*>>* loadmodule "registrar.so"*>>* loadmodule "siputils.so"*>>* loadmodule "sanity.so"*>>* loadmodule "ctl.so"*>>* loadmodule "mi_rpc.so"*>>* loadmodule "avpops.so"*>>* #!ifdef WITH_NAT*>>* loadmodule "nathelper.so"*>>* loadmodule "rtpproxy.so"*>>* #!endif*>>* loadmodule "xlog.so"*>>* ##mysqlcdr*>>* modparam("acc", "db_flag", 2)*>>* modparam("acc", "db_missed_flag", 3)*>>* modparam("acc", "report_cancels", 1)*>>* modparam("acc", "db_url", "mysql://openser:password@1.1.1.2/openser*>>* <http://openser:password@1.1.1.2/openser>")*>>* modparam("acc", "db_extra", "from_uri=$fu; to_uri=$tu; intid=$fU;*>>* type_call=$si; dst_ip=$ru; carriercode=$tu;callmode=$var(out)" )*>>**>>* #loadmodule "carrierroute"*>>* modparam("tm", "cancel_b_method", 1)*>>* modparam("tm", "failure_reply_mode", 3)*>>* modparam("tm", "fr_timer", 30000)*>>* modparam("tm", "fr_inv_timer", 120000)*>>**>>* modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.lst")*>>* modparam("rr", "enable_full_lr", 1)*>>**>>* #!ifdef WITH_NAT*>>* # ----- rtpproxy params -----*>>* modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722*>>* <http://127.0.0.1:7722>")*>>**>>* # ----- nathelper params -----*>>* modparam("nathelper", "natping_interval", 30)*>>* modparam("nathelper", "ping_nated_only", 1)*>>* modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)*>>* modparam("nathelper", "sipping_from", "sip:pinger at kamailio.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>>* <mailto:sip%3Apinger at kamailio.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>")*>>* # ----- mi_fifo params -----*>>**>>* ##modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")*>>**>>* # params needed for NAT traversal in other modules*>>* modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")*>>* modparam("usrloc", "nat_bflag", FLB_NATB)*>>* #!endif*>>* modparam("xlog", "buf_size", 8192)*>>* modparam("xlog", "force_color", 1)*>>* #carrierroute*>>* #modparam("carrierroute", "config_file", *>>* "/etc/kamailio/carrierroute.conf")*>>* #end carrierroute*>>**>>* modparam("pv", "shvset", "ruta=s:$ru")*>>* modparam("pv", "varset","in=s:IN")*>>* modparam("pv", "varset","out=s:OUT")*>>* #################BEGINING MAINROUTE##################*>>* request_route {*>>**>>* if (!mf_process_maxfwd_header("4")) {*>>* sl_send_reply("483","Too Many Hops");*>>* drop;*>>* exit;*>>* }*>>* ##*>>* # initial sanity checks -- too long messages*>>* if (msg:len >= 2048 ) {*>>* sl_send_reply("513", "Message too big");*>>* xlog("out header CHECK 513 baderror from_uri=$fu;($ct++)<$tU---=$var(n)>*>>* to_uri=$tu; intid=$fU; type_call=$si; dst_ip=$ru;*>>* carriercode=$var(z);callmode=$var(out)");*>>* drop;*>>* exit;*>>* };*>>* #*>>* if(is_method("OPTIONS")) {*>>* # send reply for each options request*>>* sl_send_reply("200", "ok");*>>* exit();*>>* }*>>**>>**>>**>>* #*>>* if ( is_method("ACK") ) {*>>* if ( t_check_trans() ) {*>>* # no loose-route, but stateful ACK;*>>* # must be an ACK after a 487*>>* # or e.g. 404 from upstream server*>>* $var(n)=$(tU{s.substr,3,0});*>>* $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(r)=$rd; ##to domain*>>* $fU=$(fU{s.substr,14,0});*>>* $fn=$var(t);*>>* $tU=$var(n);*>>* $td=$var(r);*>>**>>* subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig');*>>**>>* # Remove prefixes numbers from Contact\*>>* t_relay();*>>* exit;*>>* } else {*>>* # ACK without matching transaction ... ignore and discard*>>* $var(n)=$(tU{s.substr,3,0});*>>* $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(r)=$rd; ##to domain*>>* $fU=$(fU{s.substr,14,0});*>>* $fn=$var(t);*>>* $tU=$var(n);*>>* $td=$var(r);*>>**>>* subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig');*>>* t_relay();*>>* exit;*>>* }*>>* }*>>* ##responding to bye requests and modifying headers to match invite*>>* if(method=="BYE") {*>>* #Account BYE transactions*>>* #$var(d) = $tu;*>>* #$var(e)=$(var(d){s.substr,4,3});*>>* setflag(2);*>>* setflag(10);*>>* $var(n)=$(tU{s.substr,3,0});*>>* $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(r)=$rd; ##to domain*>>* $fU=$(fU{s.substr,14,0});*>>* $fn=$var(t);*>>* $tU=$var(n);*>>* $td=$var(r);*>>**>>* subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig');*>>* t_relay();*>>* exit;*>>* };*>>**>>* #CANCEL processing*>>* if (method=="CANCEL") {*>>* setflag(2);*>>* setflag(3);*>>* $var(n)=$(tU{s.substr,3,0});*>>* $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(r)=$rd; ##to domain*>>* $fU=$(fU{s.substr,14,0});*>>* $fn=$var(t);*>>* $tU=$var(n);*>>* $td=$var(r);*>>**>>* subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig');*>>* t_relay();*>>* exit;*>>* ##if (t_check_trans()) t_relay();*>>**>>* exit;*>>* };*>>**>>**>>* #*>>**>>**>>* #*>>* if (loose_route()) {*>>**>>**>>* ##*>>* if(is_method("OPTIONS")) {*>>* # send reply for each options request*>>* sl_send_reply("200", "ok");*>>* exit();*>>* }*>>* ##*>>* if(method=="BYE") {*>>* #Account BYE transactions*>>* setflag(2);*>>* };*>>**>>**>>* ##CANCEL processing*>>* if (method=="CANCEL") {*>>* setflag(3);*>>* $var(n)=$(tU{s.substr,3,0});*>>* $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/});*>>**>>* $var(r)=$rd; ##to domain*>>* $fU=$(fU{s.substr,14,0});*>>* $fn=$var(t);*>>* $tU=$var(n);*>>* $td=$var(r);*>>* subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig');*>>* if (t_check_trans()) t_relay();*>>* exit;*>>* };*>>**>>* if (!t_relay()) {*>>* sl_reply_error();*>>* }*>>* exit;*>>* }*>>**>>**>>* if (is_method("INVITE")) { #Procesing invites*>>**>>**>>* record_route();*>>* }*>>* ##LOGGING CALL BEFORE IS PROCESSED*>>* xlog("before strip,< time [$Tf] > Aip= $src_ip, r-uri($ru\n");*>>* xlog("before from_uri=$fu; to_uri=$tu; pai=<<$ct>>;intid=$fU;*>>* type_call=$si; dst_ip=$ru; carriercode=$tu;callmode=$var(out) *>>* --$$--$ou");*>>* #creando rutas*>>* setflag(2);*>>* # Account Missed calls*>>* setflag(3);*>>* ###VARIABLES FOR MODIFICATIONS*>>* $var(x) = $(ru);*>>**>>* $var(y)=$(var(x){s.substr,4,3});*>>**>>* $var(z)=$(var(y){s.int <http://s.int>});*>>**>>* $var(t)=$(var(x){s.substr,7,0});*>>**>>* $var(h)=$(var(x){s.substr,0,7});*>>**>>* $var(o)=$(fU{s.substr,14,0});*>>**>>* if($(var(x){s.len}) < 3)*>>* {*>>* $var(z)='0'+ $var(z);*>>* }*>>**>>* xlog("dsroute $var(z)fullruri($ru)\n");*>>**>>**>>* strip(3);*>>**>>* xlog("after strip time [$Tf] method ($rm) via el code*>>* ($var(z)--varhex($var(h))-- ($var(y))\n");*>>* xlog("after from_uri=$fu; to_uri=$tu; intid=$fU; type_call=$si;*>>* dst_ip=$ru; carriercode=$var(z);callmode=$var(out)");*>>**>>* ds_select_domain ("$var(z)", "4");#carrer dynamic*>>**>>**>>* ###########KILLING BUGGIES#######when t_realy fails to find a route a*>>* loop is generate then strip takes off a digit until 0 at 2.2.2.2 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>>* <mailto:0 at 2.2.2.2 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> is left for to: user this kills that call.#####*>>* if ($(tU{s.len}) < 4) {*>>* xlog("CHECK TU 1bad kILLING BUGGIES>>> $tu--$td-$tU-$tt--$dd-$dd-$du-*>>* -<<$rd>$ru --- $oU> pai+++ <<$ct>>++ from_uri=$fu;<$tU---=$var(n)>*>>* to_uri=$tu; }pai<$ai> id=$fU; type_call=$si; dst_ip=$ru;*>>* carriercode=$var(z);callmode=$var(out)");*>>* ##sl_reply_error();*>>* ##sl_send_reply("513", "Message too big");*>>* ##sl_send_reply("488","Not Acceptable Here");*>>* sl_send_reply("488","Not Acceptable Here");*>>* #t_reset_retr();*>>* #t_release();*>>* ##if(!t_relay()){ sl_reply_error(); drop; exit;}*>>**>>* #if (!t_relay()) { sl_reply_error(); break; };*>>* # t_relay();*>>* #setflag(2);*>>* drop;*>>* exit;*>>**>>* }; ## Endif less than 4*>>* ##############END KILLING###########*>>* ################ROUTE SELECTION###############*>>* if($(fn{s.len}) > 10) { # IF WE CHECK FOR FROM: ROUTEID MUST BE LONGER*>>* THAN 10 DIGITS TO BE A VALID ROUTE*>>**>>* xlog("out header CHECK more than 10 digits BEFORE*>>* from_uri=$fu;($ct++)<$tU---=$var(n)> to_uri=$tu; intid=$fU;*>>* type_call=$si; dst_ip=$ru; carriercode=$var(z);callmode=$var(out)");*>>* $var(n)=$(tU{s.substr,3,0});*>>* $var(i)=$(ct{re.subst,/^.*00700(.*)/"\1/});*>>* $var(t)=$(fn{re.subst,/^.*00700(.*)/"\1/});*>>* $var(r)=$rd; ##to domain*>>* $fU=$(fU{s.substr,14,0});*>>* $fn=$var(t);*>>* $tU=$var(n);*>>* $td=$var(r);*>>**>>* subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig'); # Remove number*>>* from Contact*>>**>>* ##if(!t_relay()){ sl_reply_error(); drop; exit;}*>>**>>* if (!t_relay()) { sl_reply_error(); break; };*>>**>>* # t_relay();*>>**>>* drop;*>>* exit;*>>**>>* };*>>**>>* ################ENDrouteselect##############################*>>**>>**>>* ###############AFTER LAST IF##############reply 488 ANY CALL THAT DID*>>* NOT FOUND A ROUTE.*>>* sl_send_reply("488","Not Acceptable Here");*>>* drop;*>>* exit;*>>* }*>>**>>* ###################END MAINROUTE#################*>>**>>**>>**>>**>>**>>* On My dispatcher.lst i have*>>**>>**>>* 1 sip:1.1.1.1:5060 <http://1.1.1.1:5060>*>>* 2 sip:2.2.2.2:5060 <http://2.2.2.2:5060>*>>* 3 sip:3.3.3.3:5060 <http://3.3.3.3:5060>*>>* 3 sip:3.3.3.5:5060 <http://3.3.3.5:5060>*>>* ...*>>**>>**>>**>>**>>* _______________________________________________*>>* SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list*>>* sr-users at lists.sip-router.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>>* http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users*>>**>**>* _______________________________________________*>* SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list*>* sr-users at lists.sip-router.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users* -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28; Miami, Nov 18-20, 2013 - more details about Kamailio trainings at http://www.asipto.com - ------------------------------ - Previous message: [SR-Users] Loop detected on aws setup<http://lists.sip-router.org/pipermail/sr-users/2013-October/080128.html> - Next message: [SR-Users] Proxy-Registrar call issues<http://lists.sip-router.org/pipermail/sr-users/2013-October/080126.html> - *Messages sorted by:* [ date ]<http://lists.sip-router.org/pipermail/sr-users/2013-October/date.html#80129> [ thread ]<http://lists.sip-router.org/pipermail/sr-users/2013-October/thread.html#801…> [ subject ]<http://lists.sip-router.org/pipermail/sr-users/2013-October/subject.html#80…> [ author ]<http://lists.sip-router.org/pipermail/sr-users/2013-October/author.html#801…> ------------------------------ More information about the sr-users mailing list<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>

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Re: [SR-Users] SNMP
by Keith 22 Oct '13

22 Oct '13

Hi, I am looking to use SNMP with Zabbix to monitor Kamailio. However the module is not in the directory so won't load. I built Kamailio from YUM, is there amyway I can include the SNMP module without having to reinstall? Thanks, Keith

3 2

Proxy-Registrar call issues
by Coy Cardwell 21 Oct '13

21 Oct '13

Hello again. Compiling from source, using the proxy-to-central registrar configurations outlined here http://www.kamailio.org/docs/modules/4.0.x/modules/outbound.html and adding in the additional "append_hf("Supported: path, outbound\r\n");" on the proxy, registration works. However, when a call is made, the debug traces show problems when entering this area of the configuration outlined in the Outbound Module: if (!t_load_contacts()) { send_reply("500", "Server Internal Error"); exit; } if (!t_next_contacts()) { send_reply("500", "Server Internal Error"); exit; } as shown here: 1(26960) ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=700 a=16 n=if 1(26960) ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=695 a=24 n=t_load_contacts 1(26960) DEBUG: tm [t_serial.c:166]: t_load_contacts(): nr_branches is 0 1(26960) DEBUG: tm [t_serial.c:169]: t_load_contacts(): nothing to do - only one contact! 1(26960) ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=705 a=16 n=if 1(26960) ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=700 a=24 n=t_next_contacts 1(26960) DEBUG: tm [t_serial.c:376]: t_next_contacts(): no contacts in contacts_avp - we are done! 1(26960) ERROR: *** cfgtrace: c=[//etc/kamailio/kamailio.cfg] l=701 a=26 n=send_reply The result from t_load_contacts seems to imply one contact exists, which would be correct as there should be one, but then t_next_contacts sees no cotacts and the sever returns an error, breaking the call. Any ideas?

2 3

Cannot write to file (/var/www/siremis-4.0.0/siremis/Config.xml)
by Cory Sanders 21 Oct '13

21 Oct '13

I am at the Database configuration screen and I get the Subject line message when I try to submit. I used directions from the http://liuchangjun.com/2011/04/08/howto-kamailio-openser-installation-guide/ and edited /usr/local/etc/kamailio/kamctlrc per "2. Configuration Config the enviroment variables" I was able to successfully # /usr/local/sbin/kamdbctl create And I get all the tables in db openser when I do a mysql>show full tables from openser; But I get this error when I try to do a #/usr/local/sbin/kamailio loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib64/kamailio/modules/ ERROR: bad config file (3 errors) root@proxy:/usr/local/lib64/kamailio/modules# Little stuck. Thanks.

2 1

syntax error and invalid arguments when trying to start Kamailio
by Cory Sanders 21 Oct '13

21 Oct '13

Sorry if this is a duplicate email. First one bounced. "I am having trouble starting Kamailio. Please see below for the error output and the content of /usr/local/etc/kamailio/kamailio.cfg at the lines noted in the error. Following instrucions at http://www.kamailio.org/wiki/install/4.0.x/git root@proxy:/etc/default# /etc/init.d/kamailio start Not starting Kamailio: invalid configuration file! -e 0(15561) : <core> [cfg.y:3567]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 356, column 30-34: syntax error 0(15561) : <core> [cfg.y:3567]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 356, column 30-34: Invalid arguments 0(15561) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 356, column 35: ERROR: bad config file (3 errors) # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif Thanks."

2 1

Record-route in dialog requests
by Spencer Thomason 21 Oct '13

21 Oct '13

Hello, In our network topology we use Kamailio at the edge of our network proxying requests from a core set of FreeSWITCH servers. I have a new termination provider where a session timer ReINVITE fails because the ACK is sent to the contact address of the FreeSWITCH server, not the proxy. Can I Record-Route these in dialog requests? If so, where should I do it in the config? A quick google search returned this thread: https://lists.cs.columbia.edu/pipermail/sip-implementors/2003-June/005089.h… which confused me a little. Thanks, Spencer My in dialog config block looks like this: # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails setflag(FLT_SIPTRACE); } if (is_method("ACK")) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if ( is_method("ACK") ) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } }

3 3

Error loading carrierroute in Kamailio 4.0.x
by Aaron Seelye 20 Oct '13

20 Oct '13

On debian 7/wheezy, updated this morning, running kamailio 4.0.x (pulled this morning, git commit cabe585ea64c03d29bcdbe5c46b0190c41bf18a5), I'm running into an error loading carrierroute.so. After setting the loadmodule "carrierroute.so" in the configuration, and starting kamailio, I'm given the error: Not starting Kamailio: invalid configuration file! -e 0(28740) ERROR: <core> [sr_module.c:572]: load_module(): ERROR: load_module: could not open module </usr/lib64/kamailio/modules/carrierroute.so>: /usr/lib64/kamailio/modules/carrierroute.so: undefined symbol: cfg_set_error_function 0(28740) : <core> [cfg.y:3567]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 244, column 12-28: failed to load module 0(28740) ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <carrierroute> found 0(28740) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 459, column 47: Can't set module parameter 0(28740) ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <carrierroute> found 0(28740) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 460, column 90: Can't set module parameter ERROR: bad config file (3 errors) It starts fine without the carrierroute module (though I won't accomplish what I'd like to without it). I just subscribed to the list, so I can't reply to previous emails, but it looks like Oliver Roth is seeing the same thing at http://lists.sip-router.org/pipermail/sr-users/2013-October/080096.html. I saw a message from august that was vague regarding possibly something similar on CentOS needing libconfuse, but having installed that and remade/installed the debs, that doesn't appear to be it. At this point I'm fairly stuck, and would love a tip in the right direction. TIA, -Aaron

2 4

Luis, let's connect on LinkedIn
by Glenn Dalgliesh via LinkedIn 20 Oct '13

20 Oct '13

LinkedIn ------------ Glenn Dalgliesh requested to add you as a connection on LinkedIn: ------------------------------------------ Luis, I'd like to add you to my professional network on LinkedIn. - Glenn Accept invitation from Glenn Dalgliesh http://www.linkedin.com/e/-y8k3ym-hmzk91hv-q/XkppUi8mK5TI9VJsZIpnfBovln4SuK… View profile of Glenn Dalgliesh http://www.linkedin.com/e/-y8k3ym-hmzk91hv-q/rso/462300/eOye/name/23995329_… ------------------------------------------ You are receiving Invitation emails. This email was intended for Luis Guaman. Learn why this is included: http://www.linkedin.com/e/-y8k3ym-hmzk91hv-q/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

Kamailio failed to start
by Wingsravi R 19 Oct '13

19 Oct '13

Dear kamailio'ns I am running kamailio 4.0.3 on ubuntu 12.04 and installed kamailio from its repository. I had changed the default kamailio config file, and inserted xmpp and presence related script. But after all when i tried to start my kamailio server I am getting the following errors: $ sudo /etc/init.d/kamailio start Not starting Kamailio: invalid configuration file! -e 0(5502) INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... 0(5502) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 816, column 6: syntax error 0(5502) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 816, column 6: missing '(' or ')' ? 0(5502) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 816, column 6: bad command: missing ';'? 0(5502) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 816, column 15: '('')' expected (function call) 0(5502) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 816, column 15: bad command: missing ';'? 0(5502) : <core> [cfg.y:3570]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 816, column 17: bad command ERROR: bad config file (6 errors) All of the above errors are pointing at route[Presence] block in the script. What could be the reason for the errors and is there any fault in my config file (please find the attachment for the same). Thanks in advance. Regards. Ravi

2 5

Re: [SR-Users] Kamailio failed to start
by Fred Posner 19 Oct '13

19 Oct '13

It appears as though you might have multiple syntax errors in the script -- such as not opening or not closing certain tags. The best way to troubleshoot this (or at least the way I like best) is to go back to your last known good config and make the changes one at a time. --fred Wingsravi R <wingsravi777(a)gmail.com> wrote: >_______________________________________________ >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

1 0

Parameters list transformations comma and semi column conflict
by Seudin Kasumovic 18 Oct '13

18 Oct '13

Hi, Transformation {param.value, param_name} returns incomplete or empty values when parameter value contains comma (,). See next example: *$avp*(my_var)="a=val_a1,val_a2,val_a3;b=val_b"; in next transformations: $(*avp*(my_var){param.value, a}) returns 'val_a1' $(*avp*(my_var){param.value, b}) returns empty string Seams that comma in parameter value conflicts with semi column separator. Is this bug or wrong documented? Related link: http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#paramvalue_name -- Seudin Kasumovic Tuzla, Bosnia

3 6

Loop detected on aws setup
by julian arsanches 18 Oct '13

18 Oct '13

Hi all, before hand thanks for all the support received on this channel. I have an issue with an installed server on a aws instance which is giving me routing loops, my setup is simple, i have alias set for both ips internal and external and the loop still present when t_relay don't find any routes to relay it just sent it to itself with the public ip then the loop starts. can any one please advise how can i prevent that or just tell t_relay that if there is no more routes just respond with a cancel or stop searching. same thing happens when a gw code is not found on the dispatcher list file. as i had read by just specifying the alias=domain it should take care of it but it is not happening perhaps because it is a AWS instance, or most likely i am missing something after the T_relay to stop the call from looping. I do have several headers fix because i have an asterisk sending calls with lots of signalling calls added to the call that i need to remove before the call is passed out to a carrier or another UA, Kamailio is just proxing the calls between asterisk and the outside world no Media handling, No registrations. Again thanks for any tips to find my answer or if anyone had a similar issue thanks again. thanks for any help . Julian. on this setup i have public ip as 2.2.2.2 private IP as 1.1.1.1 #!KAMAILIO ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else #debug=2 #log_stderror=yes #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 fork=yes children=6 auto_aliases=no listen=udp:1.1.1.1:5060 advertise 2.2.2.2:5060 alias=1.1.1.1:5060 alias=1.1.1.1 alias=2.2.2.2:5060 alias=2.2.2.2 port=5060 enable_tls=no #tcp_connection_lifetime=3605 #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/" #!endif loadmodule "db_mysql.so" ##loadmodule "mi_fifo.so" loadmodule "tm.so" loadmodule "sl.so" loadmodule "acc.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "uac.so" loadmodule "textops.so" loadmodule "pv.so" loadmodule "dispatcher.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "siputils.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "avpops.so" #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif loadmodule "xlog.so" ##mysqlcdr modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 3) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://openser:password@1.1.1.2/openser") modparam("acc", "db_extra", "from_uri=$fu; to_uri=$tu; intid=$fU; type_call=$si; dst_ip=$ru; carriercode=$tu;callmode=$var(out)" ) #loadmodule "carrierroute" modparam("tm", "cancel_b_method", 1) modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.lst") modparam("rr", "enable_full_lr", 1) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # ----- mi_fifo params ----- ##modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif modparam("xlog", "buf_size", 8192) modparam("xlog", "force_color", 1) #carrierroute #modparam("carrierroute", "config_file", "/etc/kamailio/carrierroute.conf") #end carrierroute modparam("pv", "shvset", "ruta=s:$ru") modparam("pv", "varset","in=s:IN") modparam("pv", "varset","out=s:OUT") #################BEGINING MAINROUTE################## request_route { if (!mf_process_maxfwd_header("4")) { sl_send_reply("483","Too Many Hops"); drop; exit; } ## # initial sanity checks -- too long messages if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("out header CHECK 513 baderror from_uri=$fu;($ct++)<$tU---=$var(n)> to_uri=$tu; intid=$fU; type_call=$si; dst_ip=$ru; carriercode=$var(z);callmode=$var(out)"); drop; exit; }; # if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } # if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server $var(n)=$(tU{s.substr,3,0}); $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/}); $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/}); $var(r)=$rd; ##to domain $fU=$(fU{s.substr,14,0}); $fn=$var(t); $tU=$var(n); $td=$var(r); subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig'); # Remove prefixes numbers from Contact\ t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard $var(n)=$(tU{s.substr,3,0}); $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/}); $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/}); $var(r)=$rd; ##to domain $fU=$(fU{s.substr,14,0}); $fn=$var(t); $tU=$var(n); $td=$var(r); subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig'); t_relay(); exit; } } ##responding to bye requests and modifying headers to match invite if(method=="BYE") { #Account BYE transactions #$var(d) = $tu; #$var(e)=$(var(d){s.substr,4,3}); setflag(2); setflag(10); $var(n)=$(tU{s.substr,3,0}); $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/}); $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/}); $var(r)=$rd; ##to domain $fU=$(fU{s.substr,14,0}); $fn=$var(t); $tU=$var(n); $td=$var(r); subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig'); t_relay(); exit; }; #CANCEL processing if (method=="CANCEL") { setflag(2); setflag(3); $var(n)=$(tU{s.substr,3,0}); $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/}); $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/}); $var(r)=$rd; ##to domain $fU=$(fU{s.substr,14,0}); $fn=$var(t); $tU=$var(n); $td=$var(r); subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig'); t_relay(); exit; ##if (t_check_trans()) t_relay(); exit; }; # # if (loose_route()) { ## if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } ## if(method=="BYE") { #Account BYE transactions setflag(2); }; ##CANCEL processing if (method=="CANCEL") { setflag(3); $var(n)=$(tU{s.substr,3,0}); $var(i)=$(ct{re.subst,/^.*11811(.*)/"\1/}); $var(t)=$(fn{re.subst,/^.*11811(.*)/"\1/}); $var(r)=$rd; ##to domain $fU=$(fU{s.substr,14,0}); $fn=$var(t); $tU=$var(n); $td=$var(r); subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig'); if (t_check_trans()) t_relay(); exit; }; if (!t_relay()) { sl_reply_error(); } exit; } if (is_method("INVITE")) { #Procesing invites record_route(); } ##LOGGING CALL BEFORE IS PROCESSED xlog("before strip,< time [$Tf] > Aip= $src_ip, r-uri($ru\n"); xlog("before from_uri=$fu; to_uri=$tu; pai=<<$ct>>;intid=$fU; type_call=$si; dst_ip=$ru; carriercode=$tu;callmode=$var(out) --$$--$ou"); #creando rutas setflag(2); # Account Missed calls setflag(3); ###VARIABLES FOR MODIFICATIONS $var(x) = $(ru); $var(y)=$(var(x){s.substr,4,3}); $var(z)=$(var(y){s.int}); $var(t)=$(var(x){s.substr,7,0}); $var(h)=$(var(x){s.substr,0,7}); $var(o)=$(fU{s.substr,14,0}); if($(var(x){s.len}) < 3) { $var(z)='0'+ $var(z); } xlog("dsroute $var(z)fullruri($ru)\n"); strip(3); xlog("after strip time [$Tf] method ($rm) via el code ($var(z)--varhex($var(h))-- ($var(y))\n"); xlog("after from_uri=$fu; to_uri=$tu; intid=$fU; type_call=$si; dst_ip=$ru; carriercode=$var(z);callmode=$var(out)"); ds_select_domain ("$var(z)", "4");#carrer dynamic ###########KILLING BUGGIES#######when t_realy fails to find a route a loop is generate then strip takes off a digit until 0(a)2.2.2.2 is left for to: user this kills that call.##### if ($(tU{s.len}) < 4) { xlog("CHECK TU 1bad kILLING BUGGIES>>> $tu--$td-$tU-$tt--$dd-$dd-$du- -<<$rd>$ru --- $oU> pai+++ <<$ct>>++ from_uri=$fu;<$tU---=$var(n)> to_uri=$tu; }pai<$ai> id=$fU; type_call=$si; dst_ip=$ru; carriercode=$var(z);callmode=$var(out)"); ##sl_reply_error(); ##sl_send_reply("513", "Message too big"); ##sl_send_reply("488","Not Acceptable Here"); sl_send_reply("488","Not Acceptable Here"); #t_reset_retr(); #t_release(); ##if(!t_relay()){ sl_reply_error(); drop; exit;} #if (!t_relay()) { sl_reply_error(); break; }; # t_relay(); #setflag(2); drop; exit; }; ## Endif less than 4 ##############END KILLING########### ################ROUTE SELECTION############### if($(fn{s.len}) > 10) { # IF WE CHECK FOR FROM: ROUTEID MUST BE LONGER THAN 10 DIGITS TO BE A VALID ROUTE xlog("out header CHECK more than 10 digits BEFORE from_uri=$fu;($ct++)<$tU---=$var(n)> to_uri=$tu; intid=$fU; type_call=$si; dst_ip=$ru; carriercode=$var(z);callmode=$var(out)"); $var(n)=$(tU{s.substr,3,0}); $var(i)=$(ct{re.subst,/^.*00700(.*)/"\1/}); $var(t)=$(fn{re.subst,/^.*00700(.*)/"\1/}); $var(r)=$rd; ##to domain $fU=$(fU{s.substr,14,0}); $fn=$var(t); $tU=$var(n); $td=$var(r); subst('/^Contact:.*11811(.*)$/Contact: <sip:\1\2/ig'); # Remove number from Contact ##if(!t_relay()){ sl_reply_error(); drop; exit;} if (!t_relay()) { sl_reply_error(); break; }; # t_relay(); drop; exit; }; ################ENDrouteselect############################## ###############AFTER LAST IF##############reply 488 ANY CALL THAT DID NOT FOUND A ROUTE. sl_send_reply("488","Not Acceptable Here"); drop; exit; } ###################END MAINROUTE################# On My dispatcher.lst i have 1 sip:1.1.1.1:5060 2 sip:2.2.2.2:5060 3 sip:3.3.3.3:5060 3 sip:3.3.3.5:5060 ...

3 2

how to replace sdp c line?
by Juha Heinanen 17 Oct '13

17 Oct '13

i have tried various textops functions to replace sdp c lines with c=IN IP4 0.0.0.0 but so far all have appended the above line to the end of sdp rather than replaced existing c lines. any hints on a solution? -- juha

3 8

Need help on RTP port opening!!!
by Vinay Krishna 17 Oct '13

17 Oct '13

Hi Sir/Madam, I want to make a call across firewall device. I am using nf_conntrack_sip and nf_nat_sip(2.6.32) kernel modules as sip proxy. Connection got established from private network to public network but *no voice exchange is happening*. We can see RTP ports related UDP entry in conntrack -L. How can I fix this problem? Test Scenario: --------------- Device LAN IP -- 10.0.0.1 Device WAN IP -- 192.168.1.1 LAN side of device ------------------ Analog phone with ATA(10.0.0.3) Trixbox(Asterisk) - 10.0.0.2 WAN side of device ------------------ PC with SJphone - 192.168.1.2 I am trying to establish the connection from Analog phone to SJphone. In the device iptable rule added in FORWARD chain to allow SIP packet(5060 port). In NAT table under POSTROUTING chain MASQUERADE rule to allow packets to traverse from LAN side to WAN side. Thanks in advance. Regards, Vinay Krishna K

1 0

Memory leak in 3.3.4
by David Cunningham 17 Oct '13

17 Oct '13

Hello, We're running a Kamailio 3.3.4 system, and Kamailio is slowly using more and more memory. Over a couple of weeks it will run out of system memory. We tried to enable memory debugging doing the following, but it resulted in Kamailio not responding to any SIP packets. Would anyone have advice please on how to debug the situation? 1. In Makefile.defs set MEMDBG to 1 and recompile Kamailio. 2. In kamailio.cfg add the line: #!define DEBUG_MEMORY 1 While this was running and Kamailio didn't respond to packets, it logged lots of lines like this: Jul 22 21:32:22 hostname kamailio: : <core> [mem/q_malloc.c:369]: qm_malloc(0x4000e008, 128) called from <core>: cfg.lex: addstr(1438) Jul 22 21:32:22 hostname kamailio: : <core> [mem/q_malloc.c:413]: qm_malloc(0x4000e008, 128) returns address 0x40048918 frag. 0x40048900 (size=128) on 1 -th hit Jul 22 21:32:22 hostname kamailio: : <core> [mem/q_malloc.c:369]: qm_malloc(0x4000e008, 128) called from <core>: cfg.lex: addstr(1438) Jul 22 21:32:22 hostname kamailio: : <core> [mem/q_malloc.c:413]: qm_malloc(0x4000e008, 128) returns address 0x400489c8 frag. 0x400489b0 (size=128) on 1 -th hit Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

3 38

Kamilio and AWS Route 53 latency regions
by Coy Cardwell 16 Oct '13

16 Oct '13

Hi All. Fist time poster here, long time geek. I am trying to get Kamilio to work in the following configuration and after days of googling and configuration attempts, I have come here for help. Two servers, both sharing the same database. Server1, Server2. Both servers are behind a NAT (AWS). If I set the DNS to return both IP addresses for the domain on the A record, everything works, all is well. If I set the DNS to return values based on latency and the Calling Client gets Server1's IP Address but the Receiving Client is registered to Server2, the call fails from a timeout. I can see the attempt to go through at the packet level and a 'non-local' socket message as well. If the DNS returns Server2's IP Address and the Receiving Client is registered to Server2, the call completes. I have tried numerous variations of the various 'path' and 'registrar' options with any joy. I tried making one of the servers the central registrar as well. I am not entirely sure I ever set everything correctly, but I have tried numerous combinations. Any thoughts would be appreciated, I only ask this as I think I have exhausted my own list of things to try. Thanks. - Coy Cardwell

3 7

Openser and freeswitch for media scenario
by Miha 16 Oct '13

16 Oct '13

Hi, I need a little help with best practice scenario. I am using freeswitch for pickups, ivrs and etc. What is best pratice, if I would like to use freeswitch for pbx functionality and I am not using media proxy, than I would need relay every call throught freeswitch. Problem in this case is accouting as freeswitch change callid for everycall and I can not do proper accouting. thank you for help! miha

2 2

Re: [SR-Users] Regarding rtp proxy
by Alex Balashov 16 Oct '13

16 Oct '13

The one that you provided as an argument to -s. Premchandiran <premchandiran.marimuthu(a)plintron.com> wrote: >Hi All, > >In modparam("rtpproxy", "rtpproxy_sock", "udp:ip:port") which ip I have >to >give if I use this command to run "rtpproxy -F -l ip1 -s udp:ip:port >-d >DBUG" > > > >Regards, > > > >Prem Chandiran M > > > > > >------------------------------------------------------------------------ > >_______________________________________________ >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 -- Sent from my mobile, and thus lacking in the refinement one might expect from a fully fledged keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com

1 0

Regarding rtp proxy
by Premchandiran 16 Oct '13

16 Oct '13

Hi All, In modparam("rtpproxy", "rtpproxy_sock", "udp:ip:port") which ip I have to give if I use this command to run "rtpproxy -F -l ip1 -s udp:ip:port -d DBUG" Regards, Prem Chandiran M

1 0

kamailio with rtp proxy error
by Premchandiran 15 Oct '13

15 Oct '13

Hi All, I am getting " incorrect port 0 in reply from rtp proxy" when I run kamailio with rtp proxy. May I know how to overcome this issue. Thanks, Prem Chandiran M

1 0

(no subject)
by Sébastien Cramatte 15 Oct '13

15 Oct '13

Hello, I've setup Kamailio 4 + Asterisk 11 following asipto guide ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb) I've got my Kamailio listening on Public IP+ loopback and Asterisk only on loopback. Configuring RPTProxy in bridge mode I've been able to get it everything working. My question is hwo can I replace rtpproxy by ngcp-mediaproxy-ng in bridge mode ? Does it possible ? I use "rwie" and "rwei" flags but in ngcp-mediaproxy-ng e and i seems to be used for IPv4 / IPv6 .. Thank you for your help Best regards

3 2

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