- sr-users - kamailio.org

signaling for NON SIP environment
by fb111＠web.de 03 Nov '14

03 Nov '14

2 1

question
by Javier Ricke 03 Nov '14

03 Nov '14

i work with asterisk, but i have problem at the moment to send audio rtp, because ip is private, kamailio will could solution this problem? regards

2 1

mysql db error
by shmick＠riseup.net 03 Nov '14

03 Nov '14

2 1

Address already in use error when starting Kamailio service
by Mahmoud Ramadan Ali 03 Nov '14

03 Nov '14

Hiiiiii everyone, I'm receiving this error when starting Kamailio saying that it can not resolve my ip addresses used by Kamailio and another "127.0.0.1:5060 : Address already in use " error message...so how can i fix this please ? Nov 2 15:39:26 Kamailio /usr/local/sbin/kamailio[7692]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 35 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.100.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.50.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.100.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.50.1 Nov 2 15:39:44 Kamailio /usr/local/sbin/kamailio[7726]: ERROR: <core> [tcp_main.c:2876]: tcp_init(): bind(b, 0x7faebd135ca4, 16) on 127.0.0.1:5060 : Address already in use

3 3

Kamailio behind NAT - Wrong VIA Header
by Marko Seidenglanz 03 Nov '14

03 Nov '14

Hello, We are using Kamailio 4.2 and RTPEngine 3.3 hosted on a VM behind NAT in Google Cloud Engine. Asterisk 11 + Kamailio + RTPEngine are on the same machine and Kamailio is used as WebRTC Gateway. INVITES, that are rewritten by Kamailio look like the following: INVITE sip:eTden7KoNSdMVCUNYIwmz@wh2.24dial.com SIP/2.0 Record-Route: <sip:0.0.0.0:5064;lr=on;nat=yes> Via: SIP/2.0/UDP 0.0.0.0:5064 ;branch=z9hG4bK86f5.88fb7f8810c91758b5f3962420d3a4ef.0 Via: SIP/2.0/UDP 10.240.1.95:5060 ;received=127.0.0.1;branch=z9hG4bK05952ee6;rport=5060 Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as7900b63f To: <sip:eTden7KoNSdMVCUNYIwmz@wh2.24dial.com> Contact: <sip:anonymous@10.240.1.95:5060;alias=127.0.0.1~5060~1> Call-ID: 0662d336694c60ed60ed5310571a1aee@10.240.1.95:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Mon, 03 Nov 2014 09:01:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676 P-hint: outbound As you can see, Kamailio uses 0.0.0.0 for Record-Route as well as for VIA Header. It uses the address configured in kamilio.cfg (listen=udp:0.0.0.0:5064) I know, that we could use the advertise parameter to tell kamailio the external IP. The problem is, that IPs in GCE are dynamically assigned to the VMs. Thus static configuration of external IP is not sufficient for our setup. Is there a way, to let Kamailio gather it's external IP dynamically. (E.g. using a STUN server or something like that?) Another possibility would be to tell Kamailio it's external IP via a start parameter. Is that possible? Any help would be grateful. Kind regards, Marko Seidenglanz

2 1

using time stamp
by bob sacamano 03 Nov '14

03 Nov '14

hi, I am trying to see how long kamailio takes to process messages I saw the commands: set_time_stamp diff_time_stamp in older versions of kamailio but they dont seem to be available. I have been searching for info but have been unable to find anything. Is the command no longer available or is there some other way to use? thanks

2 3

FIFO Error When Running Kamctl dispatcher reload
by Mahmoud Ramadan Ali 02 Nov '14

02 Nov '14

Hiiiii Everyone I have encountered that error whenever i try to use the #Kamctl dispatcher reload even thought that i have the "modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")" and loadmodule "mi_fifo.so" configured...any ideas how to solve this issue ?! root@Kamailio:/usr/local/etc/kamailio# kamctl dispatcher reload ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module.

2 1

Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 01 Nov '14

01 Nov '14

Hi all.. I have solve the problem . Cheers, vivi From: vivi [mailto:vivi.hilton@gmail.com] Sent: Saturday, June 06, 2009 11:29 PM To: 'users(a)lists.kamailio.org' Subject: /usr/local/sbin/kamdbctl create error Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

2 1

Trouble with LCR removing $ru after upgrading to 4.1.6
by Will Ferrer 01 Nov '14

01 Nov '14

Hi We are having an odd issue with our LCR. This started after we upgraded from kamailio 4.1.0 to 4.1.6. Here is the situation: We populate our gw_uri_avp (stored in: $avp(i:709)) as follows: 16|1||||[gateway ip address]||5060||1|1. The $ru before running next gateway shows: sip:[destination number]@[our domain that the call came into]. This is how it should look before lcr occurs. Then we run next_gw(). We get these message in the logs during the running of next_gw DEBUG: lcr [lcr_mod.c:2156]: generate_uris(): r_uri <*1*[destination number]@[gateway ip address]:5060*1*>, dst_uri <> DEBUG: lcr [lcr_mod.c:2437]: next_gw(): added ruri_user_avp <[destination number]> DEBUG: lcr [lcr_mod.c:2473]: next_gw(): added flags_avp <1> DEBUG: lcr [lcr_mod.c:2480]: next_gw(): added tag_avp <> DEBUG: lcr [lcr_mod.c:2488]: next_gw(): added defunct_gw_avp <1253613379> The main thing to notice there is that there is now a 1 at the start and end of the r_uri shown in generate_uris. I am fairly certain those are new, and maybe related to what happens next. Immediately following the next_gw function running the $ru becomes: <null>, so of course the call doesn't go through. I have looked at the code for the lcr module and it looks like it wasn't changed in version 4.1.6. I found alot of other notes in the change log that had to do with the way uris are handled so it is possible one of those changes caused the problem. I am stumped currently. Any help any one can offer me would be very appreciated. All the best. Will Ferrer Swithsoft Inc

2 4

Installing rtpengine on CentOS 6.5
by Juan Perez 31 Oct '14

31 Oct '14

Using Kamailio 4.2, CentOS 6.5, as per the documentation here: https://github.com/sipwise/rtpengine "Running make will compile the binary, which will be called rtpengine." When trying to install rtpengine I got the following error when building "daemon" [root@ip-10-0-2-68 daemon]# make Usage: awk [POSIX or GNU style options] -f progfile [--] file ... Usage: awk [POSIX or GNU style options] [--] 'program' file ... POSIX options: GNU long options: -f progfile --file=progfile -F fs --field-separator=fs -v var=val --assign=var=val -m[fr] val -O --optimize -W compat --compat -W copyleft --copyleft -W copyright --copyright -W dump-variables[=file] --dump-variables[=file] -W exec=file --exec=file -W gen-po --gen-po -W help --help -W lint[=fatal] --lint[=fatal] -W lint-old --lint-old -W non-decimal-data --non-decimal-data -W profile[=file] --profile[=file] -W posix --posix -W re-interval --re-interval -W source=program-text --source=program-text -W traditional --traditional -W usage --usage -W use-lc-numeric --use-lc-numeric -W version --version To report bugs, see node `Bugs' in `gawk.info', which is section `Reporting Problems and Bugs' in the printed version. gawk is a pattern scanning and processing language. By default it reads standard input and writes standard output. Examples: gawk '{ sum += $1 }; END { print sum }' file gawk -F: '{ print $1 }' /etc/passwd Makefile:88: .depend: No such file or directory make: *** No rule to make target `aux.c', needed by `.depend'. Stop. [root@ip-10-0-2-68 daemon]# I was able to build the iptables-extension, the kernel failed because I dont have the dependencies but the daemon is required. I think that I have all the dependencies: [root@ip-10-0-2-68 daemon]# rpm -qa | egrep "pcre|pkgconfig|xml|glib|zlib|openssl|curl" openssl-1.0.1e-30.el6_6.2.x86_64 glibc-common-2.12-1.132.el6_5.4.x86_64 zlib-1.2.3-29.el6.x86_64 curl-7.19.7-37.el6_5.3.x86_64 glibc-devel-2.12-1.132.el6_5.4.x86_64 libcurl-devel-7.19.7-37.el6_5.3.x86_64 openssl-devel-1.0.1e-30.el6_6.2.x86_64 glibc-2.12-1.132.el6_5.4.x86_64 pcre-7.8-6.el6.x86_64 libxml2-2.7.6-14.el6_5.2.x86_64 libcurl-7.19.7-37.el6_5.3.x86_64 python-pycurl-7.19.0-8.el6.x86_64 pkgconfig-0.23-9.1.el6.x86_64 glib2-2.26.1-7.el6_5.x86_64 dbus-glib-0.86-6.el6.x86_64 glibc-headers-2.12-1.132.el6_5.4.x86_64 zlib-devel-1.2.3-29.el6.x86_64 libxml2-devel-2.7.6-14.el6_5.2.x86_64 pcre-devel-7.8-6.el6.x86_64 [root@ip-10-0-2-68 daemon]# Am I missing something here?, thank you jp

2 4

pipelimit rate question
by Alex Balashov 31 Oct '14

31 Oct '14

Hi, There does not seem to be a way to update the limits of pipes dynamically. If I don't use DB backing, there is no MI/RPC command to flush existing pipes and cause them to be recreated with reconstituted limits. If I use DB backing and add pipes to the pl_pipes table, an SQL update to the limit column does not influence existing pipes that have been loaded from the DB on startup. Am I missing something, or does the existing feature set of the module simply not allow this? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 4

async workers causing unexplained high load average
by Alex Balashov 31 Oct '14

31 Oct '14

Another thing I have found is that having a certain amount of async workers running, even if they are not doing anything, appears to cause unexplained CPU load, even if the Kamailio instance is completely idle and not processing any calls. Here is the baseline load with no async workers: [root@centosity6 ~]# uptime 14:06:28 up 2 min, 2 users, load average: 0.01, 0.00, 0.00 If I start Kamailio with 1 async worker... modparam("async", "workers", 1) async_workers=1 14:07:59 up 3 min, 2 users, load average: 0.00, 0.00, 0.00 Still nothing. Now I try 2: 14:08:54 up 4 min, 2 users, load average: 0.45, 0.11, 0.04 14:08:59 up 4 min, 2 users, load average: 0.49, 0.12, 0.04 14:09:04 up 4 min, 2 users, load average: 0.53, 0.14, 0.04 14:09:09 up 4 min, 2 users, load average: 0.57, 0.15, 0.05 14:09:14 up 5 min, 2 users, load average: 0.60, 0.17, 0.05 14:09:19 up 5 min, 2 users, load average: 0.63, 0.18, 0.06 14:09:31 up 5 min, 2 users, load average: 0.69, 0.21, 0.07 14:09:37 up 5 min, 2 users, load average: 0.71, 0.22, 0.07 14:10:35 up 6 min, 2 users, load average: 0.89, 0.36, 0.13 14:11:07 up 6 min, 2 users, load average: 0.93, 0.42, 0.16 ... I don't have time right now to see if it will rise to infinity, but it just keeps going up. There are no SIP messages being received at all, and there is absolutely nothing going on. The ascent of the load average is much higher with >= 4 threads. I have 8 "CPUs" available from /proc/cpuinfo (quad-core processor with HyperThreading). When I attach strace to the PID of the Async Worker, there's nothing going on except the normal recvfrom() blocking wait: [root@centosity6 kamailio]# kamctl ps | grep Async which: no greadlink in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/) Process:: ID=11 PID=1559 Type=Async Task Worker Process:: ID=12 PID=1560 Type=Async Task Worker [root@centosity6 kamailio]# strace -p 1559 Process 1559 attached - interrupt to quit recvfrom(9, [root@centosity6 ~]# strace -p 1560 Process 1560 attached - interrupt to quit recvfrom(9, Nor do the other Kamailio threads seem to be doing anything unusual. 'ps aux' and 'top' show 0.0% CPU load on the Kamailio processes themselves, and 'iowait' shows the system as 98-99% idle. I suppose it's possible that this is a peculiarity of the testing environment: I am running this on CentOS 6.5 inside VirtualBox, which I think is KVM-based. I have not had a chance to duplicate this experiment in a different kind of execution environment yet. It's possible that the load average is simply being improperly indicated because of the VM environment. -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 6

append_hf on kamctl fifo dlg_bridge destroys ACK
by Kristian F. Høgh 31 Oct '14

31 Oct '14

Hi, Tested on 4.0.6 (+contact header REFER patch) and trunk. When appending header on originating INVITE, the ACK from kamailio gets messed up. I generate call, using myself as proxy kamctl fifo dlg_bridge sip:201@[domain] sip:202@[domain] sip:[kamailio-ip]:5060 In route[tm:local-request], i use append_hf("P-Asserted-Identity:...") If I remove append_hf, the ACK looks OK. Regards, Kristian Høgh Trace on local interface: INVITE sip:201@[domain] SIP/2.0 Via: SIP/2.0/UDP [kamailio-ip];branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 To: <sip:201@[domain]> From: <sip:controller@kamailio.org>;tag=760f7752be879bf1a1346b149d21b85f-8c89 CSeq: 10 INVITE Call-ID: 0b4f5d725846cc40-12109@[kamailio-ip] Max-Forwards: 70 Content-Length: 147 User-Agent: kamailio Contact: <sip:controller@kamailio.org:5060> Content-Type: application/sdp P-Asserted-Identity: <sip:201@[domain]> <snip SDP> SIP/2.0 200 OK Via: SIP/2.0/UDP [kamailio-ip];rport=5060;branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 Record-Route: <sip:[kamailio-ip];lr;ftag=760f7752be879bf1a1346b149d21b85f-8c89;vsf=AAAAAFFfXzREVlRVXUR1WU8dAxFCHAFHQgYCbC5kaw--;did=abe.bd12> Contact: <sip:201@[peer-ip]:43017;rinstance=50924dd7a7927b2f;transport=UDP> To: <sip:201@[domain]>;tag=594df15d From: <sip:controller@kamailio.org>;tag=760f7752be879bf1a1346b149d21b85f-8c89 Call-ID: 0b4f5d725846cc40-12109@[kamailio-ip] CSeq: 10 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.0.19649 r19608 Allow-Events: presence, kpml Content-Length: 157 <snip SDP> ACK sip:201@[peer-ip]:43017;rinstance=50924dd7a7927b2f;transport=UDP SIP/2.0 Via: SIP/2.0/UDP [kamailio-ip];branch=z9hG4bK9d54.ef433c83000000000000000000000000.0 \0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00P\0xeeu\0x00\0x00\0x00\0x00\0x00$\0x12v\0x00\0x00\0x00\0x00\0x00\0xbb\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0xf0\0xf0\0xf0\0xf0\0x00\0x00\0x00\0x00\0x16\0x0075\0x13\0x01\0x00\0x00X\0xfc\0xc2Vk\0x7f\0x00\0x00\0x00\0xfd\0xc2Vk\0x7f\0x00\0x00\0x0a\0x00\0x00\0x00f-8c\0x0b\0xfd\0xc2Vomain\0x00<domain>\0x009@178X\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00\0x00To: <sip:201@[domain]>;tag=594df15d k\0x7f\0x00\0x00\0x17\0x00\0x00\0x00 ACK Route: <sip:[kamailio-ip];lr;ftag=760f7752be879bf1a1346b149d21b85f-8c89;vsf=AAAAAFFfXzREVlRVXUR1WU8dAxFCHAFHQgYCbC5kaw--;did=abe.bd12> User-Agent: kamailio Content-Length: 0 REFER ...

2 4

Susceptibility to POODLE Vulnerability?
by Rainer Piper 31 Oct '14

31 Oct '14

Hi all, is it possible to add in http://kamailio.org/docs/modules/4.2.x/modules/tls.html under the line > 9.1. |tls_method| (string) ... ... If rfc3261 conformance is desired, TLSv1 must be used. For compatibility with older clients SSLv23 is a good option. *Example 1.3. Set |tls_method| parameter* ... modparam("tls", "tls_method", "TLSv1") ... < !!! *a warning **that the use of SSLv3 **susceptibility to POODLE Vulnerability* !!! -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:rainer@sip.soho-piper.de:5072 (pjsip-test) XMPP: rainer(a)xmpp.soho-piper.de

4 6

cSeq increasing
by Yuriy Gorlichenko 30 Oct '14

30 Oct '14

Does it possible increase cSeq manually (for example remove and then append headers?) for UAC module when send INVITE messages with Auth, or kamailio have pseudovar for this header?

4 8

UAC INVITE to Provider
by Yuriy Gorlichenko 30 Oct '14

30 Oct '14

Hello. I use kamailio for calling to porvider. My providr seccefuully registered from UAC module, but when I try to call through it? it back 401 Unauthorised. I send second try with Digest Auth header at INVITE and it receive me 401 too... I register this provider from asterisk and call succesfully Ok. So i get dump from asterisk This is successfull INVITE: INVITE sip:89126975590@sip.provider.com SIP/2.0 Via: SIP/2.0/UDP 17.4.28.7:50600;branch=z9hG4bK5f118681;rport Max-Forwards: 70 From: <sip:gw2@17.4.28.7:50600>;tag=as33192a38 To: <sip:89126975590@sip.provider.com> Contact: <sip:gw2@17.4.28.7:50600> Call-ID: 021088c360a8dbf023bf35560a9daf1e@17.4.28.7:50600 CSeq: 103 INVITE User-Agent: Asterisk PBX 12.6.1 Authorization: Digest username="gw2", realm="provider.com", algorithm=MD5, uri="sip:89126975590@sip.provider.com", nonce="014d80ca", response="67bad8a0c97afc2b6747b471a56bca9f" Date: Wed, 29 Oct 2014 18:50:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 253 v=0 o=root 1098729670 1098729671 IN IP4 17.4.28.7 s=Asterisk PBX 12.6.1 c=IN IP4 17.4.28.7 t=0 0 m=audio 10088 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv Then I get dump from my kamailio (unsuccessfull INVITE) INVITE sip:89126975590@sip.provider.com SIP/2.0 Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as4684d4b9;lr=on> Via: SIP/2.0/UDP sip.myservice.com:5068 ;branch=z9hG4bK600b.1d5ff0fd59d4f3d2a1a06d722c0daa92.2 Via: SIP/2.0/UDP my.aterisk:50600;branch=z9hG4bK2b8d9b09;rport=50600 Max-Forwards: 70 From: <sip:gw2@sip.myservice.com:5068>;tag=as4684d4b9 To: <sip:89126975590@sip.provider.com > Contact:<sip:Vebinar-gw2@sip.myservice.com:5068> Call-ID: 445a7b884aeeab125d91886210c9beb7@sip.myservice.com:50600 CSeq: 102 INVITE User-Agent: SoftSwitch Date: Wed, 29 Oct 2014 22:32:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 Authorization: Digest username="gw2", realm="provider.com", nonce="10129bde", uri="sip:89126975590@sip.provider.com ", response="6d3411b24cbb57ad72271790ec01b453", algorithm=MD5 v=0 o=root 468654998 468654998 IN IP4 1.2.3.4 s=SoftSwitch c=IN IP4 1.2.3.4 t=0 0 m=audio 30104 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30105 I see difference between packetts only at SDP (not inportant things) and at VIA and request route Headers. All other fields identical. So -why Asterisk call successull and Kamailio kall unsuccessfull? What the differense?

2 2

websocket and SIP
by Ricardo Martinez 30 Oct '14

30 Oct '14

Hello. I’m having some problems using websocket to communicate a webRTC client with the SIP world. I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem? Hope that someone could help me. Best Regards, Ricardo Martinez.-

3 5

Verified version of RTPproxy and Kamailio
by Juan Perez 30 Oct '14

30 Oct '14

Can somebody please tell me what would be a verified combination of kamailio and rtpproxy?I mean, for example, kamailio 4.1 and rtpproxy 1.x that can be downloaded from:http://<real_url> I found at least 3 different places where one can download rtpproxy: http://sourceforge.net/p/sippy/rtpproxy/ci/master/tree/ https://github.com/miconda/rtpproxy https://github.com/sippy/rtpproxy I downloaded the one from https://github.com/sippy/rtpproxy and have tested against kamailio 4.1, 4.2 and they are not compatible.It seems that the logi inside the function "send_rtpp_command" is not compatible with the rtpproxy version from https://github.com/sippy/rtpproxy and that causes kamailio to disable rtpproxy and it becomes useless. Second questions: do the feature added to the rtpproxy (parameter -A to use external public IP for devices behind NAT) is compatible with OPENSIPS only? I am about to test with kamailio 3.3, I will update this post later onthank you in advancejp

1 0

kamailio server as Load balancing and Failover
by Nandini madhu 30 Oct '14

30 Oct '14

Dear All, Greetings, I want to implement kamailio server as failover and load balancing. Is there any configuration needed in kamailio.cfg. As I have searched in blogs, For load balancing, Virtual IP is added and also heartbeat software is needed. Please let me Know how to configure kamailio server for load balancing and failover Waiting for response Thank you, Regards, Nandini

3 2

Wrong port in via, from and contact headers
by Eric Koome 30 Oct '14

30 Oct '14

This is my setup: UAC -> firewall (1:1 nat) -> Kamailio:5060 - > asterisk:5080 . | . sip providers I'm running kamailio 4.1.x & ast 11 based on aspitos kamailio configuration tutorial. My problem is port number in via, contact and from headers is 5080 instead of 5060 for calls to provider. Public ip is fine. I have used advertised_port=5060 but no change. Also tried record_route_preset(ip:5060), but no change. I can make calls but I would imagine I would miss acks & and reinvites. Any thoughts on what I'm missing? Eric

1 0

Problem initiating a call with dlg_bridge
by Paul Smith 29 Oct '14

29 Oct '14

I seem to be going round in circles… I am trying to use dlg_bridge() from the dialog module to establish a call between two SIP endpoints. I have tested with Snom phones and linphone soft phone with the same result. I get an outbound call to the first (from) end point, I answer the phone… and then nothing obvious happens. I can get this same result when using kamcmd or from within the kamailio.cfg. For example using kamcmd: kamcmd dlg.bridge_dlg sip:105@mykamailioip sip:106@mykamailioip sip:mykamailioip:5060 I expected this to send an invite from "controller" via my KamailioIP to my registered local subscriber 105 (from), and then send a re-invite to 105 so that 105 creates a call leg to 106 (to). The kamailio.cfg is pretty much the default. I am running 4.2 and rtpengine to proxy the RTP streams. Not sure if it is necessary or useful but I call setflag() at the start of the request_route() to set the Dialog flag. What else should I have included in the kamailio.cfg to make this work? Also is there any way to control the SDP in the initial invite from dlg_bridge. By default I see RTP/AVP, with alaw and ulaw…. I’d like to offer RTP/SAVP swell or instead of RTP/AVP. Thanks

3 7

Trouble with max-forwards
by Mike Dunton 29 Oct '14

29 Oct '14

Readers, I am having issues with passing the max-forward header to my freeswitch service. Here are the sip invites Here is a call coming from a kazoo box though my carrier Into kamailio. Kamailio shows it has Max-Forward after the sanitize check, but when it reaches freeswitch no Max-Forward Header. *KAMAILIO* kamailio[2687]: ERROR: <script>: After Sanitize - [INVITE sip:+1850764####@aio07-bandwidth.tresta-aio.com:5060 <http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/> SIP/2.0 Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.###:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.59;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.85 <sip%3A%2B18507646071(a)67.231.8.85>> Call-ID: 537168748_70701748(a)67.231.9.# <537168748_70701748(a)67.231.9.59>## CSeq: 1289511105 INVITE *Max-Forwards: 49* Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer X-FS-Support: update_display,send_info Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>;privacy=off;screen=no v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 *FREESWITCH* recv 1479 bytes from udp/[10.1.13.123]:5060 at 18:31:26.312854: ------------------------------------------------------------------------ INVITE sip:+1850764###@aio07-bandwidth.tresta-aio.com:5060 <http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/> SIP/2.0 Record-Route: <sip:10.1.13.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8> Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 10.1.13.123;branch=z9hG4bKc892.1b308e66.0 Via: SIP/2.0/UDP 67.231.8.195;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.85;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.59:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.# <sip%3A%2B1NUMBERSCRUBED(a)67.231.9.59>##;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.# <sip%3A%2B18507646071(a)67.231.8.85>##> Call-ID: 537168748_70701748(a)67.231.9.# <537168748_70701748(a)67.231.9.59>## CSeq: 1289511105 INVITE MaxForwards would go here? Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 <http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>>;privacy=off;screen=no X-AUTH-IP: 67.231.8.195 v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 Here is my kamailio default.cfg http://pastebin.com/vxdFe8n0 Can anyone point me in the right direction, and tell me why freeswitch isn't being passed this header? Both kamailio and freeswitch are on the same box in this example. Thanks for your help. Mike

2 1

404 Not here as response to BYE requests
by Marko Seidenglanz 29 Oct '14

29 Oct '14

Hello, We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC Proxy doing the encryption and NAT Traversal. Everything works as expected, except that BYE Requests sent by the WebRTC Client are not forwarded by Kamailio to Asterisk. We use record routing. Instead Kamailio responds with a "404 Not here". INVITE Headers: Kamailio --> Browser: sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 Via: SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK2d4f70db Max-Forwards: 70 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as32dd9e24 To: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com> Contact: <sip:anonymous@10.240.177.13:5060> Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Wed, 29 Oct 2014 10:04:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676 BYE Headers Browser --> Kamailio: BYE sip:anonymous@anonymous.invalid SIP/2.0 Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 , SIP/2.0/UDP 10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com >;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3 To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as32dd9e24 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 0 BYE Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Reason: Q.850;cause=16 BYE Response Kamailio --> Browser: SIP/2.0 404 Not here Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0 , SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37 From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com >;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8 To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as0b12a022 Call-ID: 683def08655393622c55299c1a7c92d5@10.240.177.13:5060 CSeq: 0 BYE Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 In the logs I can see the following messages: [loose.c:113]: find_first_route(): No Route headers found [loose.c:929]: loose_route(): There is no Route HF Does anybody know, why Kamailio may respond with 404 Not here? Do I have to send the BYE request directly to Asterisk? Kind regards, Marko

2 3

Kamailio and RTPProxy
by Juan Perez 29 Oct '14

29 Oct '14

I installed Kamailio 4.2 and the latest rtpproxy. When starting Kamailio I see these lines: Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [sr_module.c:590]: load_module(): load_module: trying to load </usr/local Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [sr_module.c:357]: register_module(): register PV from: rtpproxy Oct 27 21:54:26 ip-10-0-2-68 kamailio: DEBUG: <core> [cfg.lex:1724]: pp_define(): defining id: MOD_rtpproxy Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: ERROR: rtpproxy [rtpproxy.c:1707]: send_rtpp_command(): timeout waiting reply f Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:10.0.2.68:77 Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP Oct 27 21:54:32 ip-10-0-2-68 ./kamailio[3887]: WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy <udp:10 Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: ERROR: rtpproxy [rtpproxy.c:1707]: send_rtpp_command(): timeout waiting reply f Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:10.0.2.68:77 Oct 27 21:54:37 ip-10-0-2-68 ./kamailio[3898]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP I run rtpproxy like this: /usr/local/rtpproxy-install-dir/bin/rtpproxy -l <public ip here> -s udp:10.0.2.68:7722 -F -d DBUG:LOG_LOCAL3 I have these linee in the kamailio.cfg file loadmodule "rtpproxy.so" # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:10.0.2.68:7722") I have tried with localhost, with 127.0.0.1 and get the same results ( I mean configure kamailio to contact rttpproxy with localhost instead of ip and at the same time running rtpproxy with the same reference. I can see the rtpproxy running/listening [root@ip-10-0-2-68 sbin]# netstat -lnp | grep rtpproxy udp 0 0 10.0.2.68:7722 0.0.0.0:* 3608/rtpproxy I can also see these lines: [root@ip-10-0-2-68 sbin]# netstat -np | grep 7722 udp 0 0 10.0.2.68:49772 10.0.2.68:7722 ESTABLISHED 3928/./kamailio udp 0 0 10.0.2.68:36097 10.0.2.68:7722 ESTABLISHED 3897/./kamailio udp 0 0 10.0.2.68:49158 10.0.2.68:7722 ESTABLISHED 3906/./kamailio udp 0 0 10.0.2.68:37129 10.0.2.68:7722 ESTABLISHED 3904/./kamailio udp 0 0 10.0.2.68:39052 10.0.2.68:7722 ESTABLISHED 3926/./kamailio udp 0 0 10.0.2.68:38161 10.0.2.68:7722 ESTABLISHED 3919/./kamailio udp 0 0 10.0.2.68:53144 10.0.2.68:7722 ESTABLISHED 3887/./kamailio udp 0 0 10.0.2.68:39721 10.0.2.68:7722 ESTABLISHED 3905/./kamailio udp 0 0 10.0.2.68:59563 10.0.2.68:7722 ESTABLISHED 3909/./kamailio udp 0 0 10.0.2.68:51884 10.0.2.68:7722 ESTABLISHED 3921/./kamailio udp 0 0 10.0.2.68:46902 10.0.2.68:7722 ESTABLISHED 3930/./kamailio udp 0 0 10.0.2.68:33981 10.0.2.68:7722 ESTABLISHED 3931/./kamailio udp 0 0 10.0.2.68:60221 10.0.2.68:7722 ESTABLISHED 3898/./kamailio udp 0 0 10.0.2.68:35651 10.0.2.68:7722 ESTABLISHED 3900/./kamailio udp 0 0 10.0.2.68:33105 10.0.2.68:7722 ESTABLISHED 3927/./kamailio udp 0 0 10.0.2.68:43349 10.0.2.68:7722 ESTABLISHED 3887/./kamailio udp 0 0 10.0.2.68:39259 10.0.2.68:7722 ESTABLISHED 3899/./kamailio udp 0 0 10.0.2.68:35170 10.0.2.68:7722 ESTABLISHED 3907/./kamailio udp 0 0 10.0.2.68:38760 10.0.2.68:7722 ESTABLISHED 3908/./kamailio In the rtpproxy I see that it is sent data and it replies too, I even got a wireshark capture showing that as well: Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3931_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3927_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3926_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:get_command: received command "3928_2 V" Oct 27 21:54:41 ip-10-0-2-68 rtpproxy[3608]: DBUG:rtpc_doreply: sending reply "20040107#012" But somehow kamailio does not see it: [root@ip-10-0-2-68 sbin]# ./kamctl fifo nh_show_rtpp which: no greadlink in (/sbin:/bin:/usr/sbin:/usr/bin:/) which: no gdb in (/sbin:/bin:/usr/sbin:/usr/bin:/) udp:10.0.2.68:7722:: set=0 index:: 0 disabled:: 1 weight:: 1 recheck_ticks:: 53253287 I even when to the kamailio code and saw something that may have something to do with it but havent tried it yet:, in the implementation of the function "send_rtpp_command" I see that the socket created is using TCP. But I tried configuring kamailio/rtpproxy to use tcp instead but I also got the same results, kamailio seems to fail to acknoledgwe the replies from rtpproxy. I also have SELINUX disabled and the iptables are not running. I dont know what else to check, if somebody can shed some light into this I'd appreciate it. jp2014

1 1

(no subject)
by Yuriy Gorlichenko 29 Oct '14

29 Oct '14

Hello. I use kamailio with last rtpengine and I have 5-7 Seconds voice delay. This happened only for from webphone. But it is not client issue as i see. Wireshark at client side shows that RTP starts as soon I pick up call. So rtp leaves rtpengine and goes to the destination with delay... I use WSS and think that problem at handshake. There are some statisticcs after call finished (as example). You may see that one of streams created after 5 seconds delay. Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] --- Tag 'rsik48leli', created 1:12 ago, in dialogue with 'as7b4cb593' Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34178 <> 8.2.10.25:52463, 340 p, 58032 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34179 <> 8.2.10.25:52468 (RTCP), 10 p, 960 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] --- Tag 'as7b4cb593', created 1:12 ago, in dialogue with 'rsik48leli' Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34194 <> 10.0.1.6:16376, 201 p, 36582 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [ 28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34195 <> 10.0.1.6:16377 (RTCP), 1 p, 78 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] Final packet stats: Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] --- Tag 'as3af30098', created 1:17 ago, in dialogue with 'bqinihbhsf' Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------ Media #1, port 34142 <> 10.0.1.6:17258, 200 p, 35200 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------ Media #1, port 34143 <> 10.0.1.6:17259 (RTCP), 1 p, 78 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] --- Tag 'bqinihbhsf', created 1:12 ago, in dialogue with 'as3af30098' Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------ Media #1, port 34162 <> 8.2.10.25:52453, 216 p, 36768 b, 0 e Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------ Media #1, port 34163 <> 8.2.10.25:52453 (RTCP),

2 1

Ubuntu upstart problem | Kamailio ready before mysqlDB
by Moritz Graf 29 Oct '14

29 Oct '14

Hi There, since 4.2 we had the problem that kamailio did not start after a reboot. A "/etc/init.d/kamailio restart" fixxed it. The problem is quite obvious: kamailio-loglile => Kamailio can't connect to mysqlDB 13:33:44 Oct 29 13:33:44 kamailio-test /usr/sbin/kamailio[1063]: DEBUG: db_mysql [km_my_con.c:94]: db_mysql_new_connection(): opening connection: mysql://xxxx:xxxx@localhost/routing Oct 29 13:33:44 kamailio-test /usr/sbin/kamailio[1063]: ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Oct 29 13:33:44 kamailio-test /usr/sbin/kamailio[1063]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool mysql-logfile => ready for connections 13:33:45 141029 13:33:45 [Note] /usr/sbin/mysqld: ready for connections. Version: '5.5.40-0ubuntu0.12.04.1' socket: '/var/run/mysqld/mysqld.sock' port: 3306 (Ubuntu) So simply mysql is not ready for connnections when kamailio is ready. I "fixxed" this with a "sleep 5;" in the initfile. But this is not what I call a clean solution. Anyone else had the same problem and fixxed it a little cleaner? Maybe converted the init- to a upstart-script? regards, moritz -- Moritz Graf, M.Sc. Betrieb NGN-Plattform

1 0

rtpproxy_manage fails to rewrite SDP
by Igor Potjevlesch 29 Oct '14

29 Oct '14

Hello, I'm working with rtpproxy module for managing calls to a particular route. The thing is that for some calls, the "Connection information" is not rewritten properly whereas another calls, less than 1 second before or after, are properly rewritten. When "cr_route" defines that the call must be route to this carrier, I go to a route in order to make some change: - Keep only PCMA, PCMU, G729a and telephone-event - Depends on the R-URI: o If the uri contains IP addresses or domain name of the carrier: rtpproxy_manage("iefrcoz20","<CARRIER_IP_ADDRESS"); o else : rtpproxy_manage("eifrcoz20"); Kamailio listens on <CARRIER_IP_ADDRESS> and also on an internal one. A dedicated instance of RTPProxy runs with the following option: "-l <INTERNAL_IP_ADDRESS>/<CARRIER_IP_ADDRESS>". The idea is to tell to my customers to send their RTP to the internal IP address, and to tell to the carrier to send the RTP to the IP address that they have attributed to me. For hundred and hundred calls, this mechanism is working fine. But sometimes, no, the SDP remains as the original one. I don't know how to diagnose this. Could it be a saturation of RTPProxy? Regards, Igor.

2 5

content-length mismatch for kamailio 4.2.x
by andrew 29 Oct '14

29 Oct '14

hi all, kamailio 4.2.x has been released, And I used two endpoint to test kamailio server. I install kamailio 4.2.x on centos 6.5 via 'yum'. One endpoint failed to call another, I captured packets via wireshark, and found that invite method is recognized as error sip signal because of content-length' mismatch. I examined the invite packets, and calculated content-length in invite signal. There is no mistake for the value of content-length. I guess it may be a bug in kamailio 4.2.x. But I don't know what causes kamailio to report the mismatch response. who can give me help? I have uploaded the captured packets as an attachment. Below are two snapshots from the captured packets. B.R. andrew

2 1

an issue on terminating dialog for kamailio 4.2.x
by qw 29 Oct '14

29 Oct '14

hi, I install kamailio 4.2.x on centos 6.5 via 'yum'. One endpoint called another, where one dialog is created, and caller and callee can talk with each other. But when the caller hangup the phone(i.e. send bye to the callee), the callee can't receive the bye message, and the dialog can't be terminated normally. I have captured the packets via wireshark, and uploaded kamailio.cfg and the captured files as attachment. In the captured file, PRACK and BYE message can be sent to kamailio server successfully, but kamailio server doesn't forward the two messages to the callee. Why PRACK and BYE message can't be sent to the callee? Looking forward to your help! B.R. Andrew

2 1

Error while executing S-CSCF (Kamailio 4.2.0)
by Kamal Palei 28 Oct '14

28 Oct '14

Hi All I have followed link http://nil.uniza.sk/book/export/html/617 to install P-CSCF/I-CSCF/S-CSCF. Though I have some doubts in configuration , still I have put the configuration to best of my knowledge. When I execute SCSCF, I get below errors. I see some database related errors. As per the followed link, I find some DB configurations and I did accordingly. I do not see any DB configurations for SCSCF, still I see two below lines # Connection URL for the database: #!define DB_URL "con1=>mysql://scscf:secret@localhost/scscf" #!define DB_URL2 "con2=>mysql://scscf:secret@localhost/scscf" in file scscf.cfg Is this really required, or can I comment these two lines. *root@Kamal-E5430:/etc/kamailio/scscf# root@Kamal-E5430:/etc/kamailio/scscf# root@Kamal-E5430:/etc/kamailio/scscf# ./scscf -f kamailio.cfg -Eloading modules under config path: /usr/local/lib64/kamailio/modules/Listening on udp: 192.168.1.114 [192.168.1.114]:4080 tcp: 192.168.1.114 [192.168.1.114]:4080Aliases: *: scscf.group4-imslab11.in:* 0(25982) INFO: <core> [tcp_main.c:4745]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(25984) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(25984) INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available 0(25984) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Access denied for user 'scscf'@'localhost' (using password: YES) 0(25984) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(25984) WARNING: db_cluster [dbcl_data.c:456]: dbcl_init_connections(): cannot connect to database - connection [con2] 0(25984) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Access denied for user 'scscf'@'localhost' (using password: YES) 0(25984) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(25984) WARNING: db_cluster [dbcl_data.c:473]: dbcl_init_connections(): cannot connect to database - connection [con2] 0(25984) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Access denied for user 'scscf'@'localhost' (using password: YES) 0(25984) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(25984) WARNING: db_cluster [dbcl_data.c:456]: dbcl_init_connections(): cannot connect to database - connection [con1] 0(25984) ERROR: db_mysql [km_my_con.c:109]: db_mysql_new_connection(): driver error: Access denied for user 'scscf'@'localhost' (using password: YES) 0(25984) ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool 0(25984) WARNING: db_cluster [dbcl_data.c:473]: dbcl_init_connections(): cannot connect to database - connection [con1] 0(25984) ERROR: <core> [db.c:401]: db_table_version(): error in db_query 0(25984) ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table presentity 0(25984) ERROR: presence [presence.c:322]: mod_init(): error during table version check 0(25984) ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so)ERROR: error while initializing modules 0(25984) INFO: cdp [mod.c:257]: cdp_exit(): CDiameterPeer child stopping ... 0(25982) ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to piperoot@Kamal-E5430:/etc/kamailio/scscf# root@Kamal-E5430:/etc/kamailio/scscf# * Best Regards Kamal

1 1

SCSCF Configuration
by Kamal Palei 28 Oct '14

28 Oct '14

Hi All I am trying to setup P-CSCF/I-CSCF/S-CSCF using Kamailio 4.2.0. I see configurations as below both in icscf.cfg and scscf.cfg files. *#!define HOSTNAME "icscf-1.imscore.org <http://icscf-1.imscore.org>"* *#!define NETWORKNAME "ims.imscore.org <http://ims.imscore.org>"* First line is *#!define HOSTNAME "icscf-1.imscore.org <http://icscf-1.imscore.org>"*Assuming *icscf-1* is the host name , *imscore.org <http://imscore.org> *is the domain name. Coming to second line *#!define NETWORKNAME "ims.imscore.org <http://ims.imscore.org>"* So it looks to me, NETWORKNAME is something like *ims.domainname* *Is this correct?* *OR* *I should put only domain name in NETWORKNAME* Please clarify. Best Regards kamal

1 0

Kamailio 4.2.0 - Error in default configuration
by Kamal Palei 28 Oct '14

28 Oct '14

Hi All I am in process of setting up of P-CSCF using Kamailio 4.2.0. I wanted to disable WITH_ANTIFLOOD (as I am not sure what shall be the content of dispatcher.list file for me, as it is juts a local setup). When I disable WITH_ANTIFLOOD, and try to execute I get below errors. Wherever *ds_is_from_list* is referred in Kamailio.cfg, I am getting error in those places. Can somebody pls suggest a work around when we disable macro WITH_ANTIFLOOD *root@Kamal-E5430:/etc/kamailio/pcscf# kamailio -f kamailio.cfg -Eloading modules under config path: /usr/local/lib64/kamailio/modules/ 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_is_from_list (params 0) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 486, column 23: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_is_from_list (params 0) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 658, column 45: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_is_from_list (params 0) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 672, column 23: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_is_from_list (params 0) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 706, column 37: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_select_dst (params 2) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 846, column 29: unknown command, missing loadmodule? 0(25255) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command ds_select_dst (params 2) 0(25255) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 993, column 29: unknown command, missing loadmodule?ERROR: bad config file (6 errors)* Thanks Kamal

3 7

Htable iterators, am I doing it wrong?
by davy van de moere 28 Oct '14

28 Oct '14

I was giving the htable iterator a go , but I can't get it to work. My relevant config would is: modparam("htable", "htable", "testhtable=>size=4;") in the request route I have this: sht_iterator_start("i1", "testhtable"); while(sht_iterator_next("i1")) { xlog("L_INFO"," $shtitkey(i1) is: $shtitval(i1)\n"); } sht_iterator_end("i1"); I added some values to my testhtable: $ kamcmd htable.dump testhtable { entry: 0 size: 2 slot: { item: { name: x[0] value: 1 } item: { name: x[1] value: 1 } } } But when I have something route over that part I get: Oct 27 15:33:36 kamailio[18828]: INFO: <script>: <null> is: <null> Oct 27 15:33:36 kamailio[18828]: INFO: <script>: <null> is: <null> So it is correct to find 2 items in the htable, but for both the key and value are not found. Tried it on both 4.2 and master. Thx!

2 2

Selecting rtpproxy instance on 200 reply
by Grant Bagdasarian 27 Oct '14

27 Oct '14

Hello, This was originally posted for opensips, but I figured someone here might know the answer as well since kamailio and opensips are "alike". I have the following RTPproxy configuration: modparam("nathelper", "rtpproxy_sock", "1 == udp:127.0.0.1:12221") modparam("nathelper", "rtpproxy_sock", "2 == udp:127.0.0.1:12222") In the main route I decide which instance to select for each INVITE containing SDP: if($ru=~"1.1.1.10") { set_rtp_proxy_set("1"); xlog("L_INFO", "USE RTP PROXY 01 \r\n"); } else if($ru=~"2.2.2.10") { set_rtp_proxy_set("2"); xlog("L_INFO", "USE RTP PROXY 02 \r\n"); } else { t_reply("404", "Not Found"); exit; } rtpproxy_offer(); I’m getting the following error for a 200 OK: ERROR:nathelper:select_rtpp_node: script error -no valid set selected ERROR:nathelper:force_rtp_proxy_body: no available proxies Do I need to do the same decision for 200 OK messages, and call rtpproxy_answer()? Regards, Grant

1 0

3.3.2 dumps core on unresolvable names
by Måns Nilsson 27 Oct '14

27 Oct '14

Hi, I've now twice experienced a core dump in my Kamailio install, where it tries to process an INVITE that is made to a domain name that does not exist. Log: Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "ulndsort.sr.se" Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: ERROR: tm [ut.h:327]: failed to resolve "ulndsort.sr.se" Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) Oct 27 14:49:58 landsort /usr/local/kamailio/sbin/kamailio[40058]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction Oct 27 14:50:02 landsort /usr/local/kamailio/sbin/kamailio[40114]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 14 Oct 27 14:50:02 landsort /usr/local/kamailio/sbin/kamailio[40051]: ALERT: <core> [main.c:785]: child process 40058 exited by a signal 11 Oct 27 14:50:02 landsort /usr/local/kamailio/sbin/kamailio[40051]: ALERT: <core> [main.c:788]: core was generated Backtrack from coredump: (gdb) bt #0 t_unref (p_msg=0x8010af140) at atomic_x86.h:231 #1 0x0000000000509581 in exec_post_script_cb (msg=0x8010af140, type=Variable "type" is not available. ) at script_cb.c:195 #2 0x00000000004297fa in receive_msg ( buf=0x891ae0 "INVITE sip:vaxo-10@ulndsort.sr.se SIP/2.0\r\nVia: SIP/2.0/UDP 134.25.127.234:5060;rport;branch=z9hG4bKPjNOjNgRxGeWilvm07L79hdOmgzVAvYB78\r\nMax-Forwards: 69\r\nFrom: \"Teknisk Utveckling MTU 24\" <sip:mtu-24@"..., len=Variable "len" is not available. ) at receive.c:225 #3 0x00000000004b5579 in udp_rcv_loop () at udp_server.c:544 #4 0x0000000000455654 in main_loop () at main.c:1633 #5 0x0000000000459c33 in main (argc=Variable "argc" is not available. ) at main.c:2546 (gdb) Does this relate to any known issue? Should I just upgrade? FreeBSD 9, by the way. -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 You should all JUMP UP AND DOWN for TWO HOURS while I decide on a NEW CAREER!!

1 0

Kamailio for ARM with rtpproxy-ng
by Marino Mileti 27 Oct '14

27 Oct '14

Hi guys, I've cross compiled Kamailio (4.1.6) for ARM (Wandboard-IMX6). Everything seems works fine, but when I try to enable rtpproxy-ng module and Alignment trap error occurred just during rtpp_test (exchanging ping-ponge message betweek rtpproxy and rtpproxy-ng module of Kamailio) rtpengine[2258]: Got valid command from 127.0.0.1:50470: ping - { "command": "ping" } rtpengine[2258]: Returning to SIP proxy: d6:result4:ponge user.err kernel: [66832.254887] Alignment trap: not handling instruction edc30b08 at [<40b3a130>] user.alert kernel: [66832.262030] Unhandled fault: alignment exception (0x801) at 0x4033a782 local0.err kamailio: ERROR: <core> [daemonize.c:307]: daemonize(): Main process exited before writing to pipe In x86 architecture same packages works correctly without any problem. Does anybody encountered this kind of error? Marino ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-for-ARM-with-rtpproxy-ng-t… Sent from the Users mailing list archive at Nabble.com.

2 5

Kamailio 4.2.0 install issue
by Kamal Palei 27 Oct '14

27 Oct '14

Dear All I compiled Kamailio4.2.0 with below command make group_include="standard standard-dep stable experimental" all to compile all modules. When I install, I get below error touch /usr/local/lib64/kamailio/modules/counters.so install -m 755 counters.so /usr/local/lib64/kamailio/modules LD (gcc) [U kamcmd] kamcmd */usr/bin/ld: error: cannot find -lncurses* collect2: ld returned 1 exit status make[2]: *** [kamcmd] Error 1 ERROR: make --no-print-directory -C ../../utils/sercmd install-if-newer failed make[1]: *** [install-utils] Error 1 make: *** [install-modules] Error 1 am I missing any package here.., please let me know. Thanks Kamal

2 2

Error-While-Running-icscf.cfg
by Anil Kumar 27 Oct '14

27 Oct '14

Hi All, Please can anyone help me to solve this error, occurred while running icscf module . 0(4053) : <core> [cfg.y:3406]: yyerror_at(): parse error in config file /etc/kamailio-icscf/icscf.cfg, from line 52, column 1 to line 53, column 0: syntax error ERROR: bad config file (1 errors) Thanks Anil

2 1

How to setup Kamailio as P-CSCF/S-CSCF/I-CSCF
by Kamal Palei 27 Oct '14

27 Oct '14

Hi All Last few days we have tried to setup Kamailio for IMS server setup. We have not been successful. Now we are planning to use Kamailio 4.2.0 code base (Latest) to setup Kamailio IMS setup. I am looking for a link/document, that gives proper step by step instructions how to setup Kamailio as IMS servers (P-CSCF, I-CSCF and S-CSCF). Can somebody kindly provide the official link or any other working link, that I can follow to setup P-CSCF, I-CSCF ,S-CSCF and HSS. Thanks kamal

2 5

Setting up web sockets and ephemeral auth
by Nolan Darilek 27 Oct '14

27 Oct '14

Having a hard time piecing together lots of pieces out of context. Here's what I want: 1. User logs in to my web app via its own authentication mechanisms. 2. Via an authenticated REST endpoint, my web app connects to the ephemeral auth service and returns a temporary username/password to the client. 3. The client connects to a web socket SIP interface, authenticating with the username/password it was given. 4. I'd also like to support authentication via standard SIP client using the user's default username/password, but that's a bit down the road still. Here is my kamailio-local.cfg. I'm using the web socket example copied from the docs, with a little cleanup and a few optional checks removed for now. loadmodule "db_mongodb.so" #!define DBURL "mongodb://mongo/perceptron" #!define WITH_AUTH #!define WITH_NAT loadmodule "xhttp.so" #modparam("xhttp", "url_match", "^/sip/") tcp_accept_no_cl=yes loadmodule "msrp.so" # Only required if using MSRP over WebSockets loadmodule "websocket.so" loadmodule "auth_ephemeral.so" modparam("auth_ephemeral", "secret", "Kamailio rulez!!11") event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); # xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm=~ "GET" ) { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { # xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { exit; } } xhttp_reply("404", "Not found", "", ""); } When I attempt to validate this, I get: 0(1) ERROR: <core> [pvapi.c:790]: pv_parse_spec2(): error searching pvar "hdr" 0(1) ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char [U/85] in [$hdr( Upgrade)] at [5 (5)] 0(1) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /usr/local /etc/kamailio/kamailio-local.cfg, line 28, column 7-19: Can't get from cache: $h dr(Upgrade) ERROR: bad config file (1 errors) I don't know what that means. I've copied the config directly from the website. I tried quoting "Upgrade" and such in the $hdr lines, but that gives me: 0(1) ERROR: <core> [pvapi.c:790]: pv_parse_spec2(): error searching pvar "hdr" 0(1) ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char ["/34] in [$hdr( "Upgrade")] at [5 (5)] 0(1) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /usr/local /etc/kamailio/kamailio-local.cfg, line 28, column 7-21: Can't get from cache: $h dr("Upgrade") ERROR: bad config file (1 errors) What am I missing here? I also had to comment out the xlog lines because those gave me errors. Do I need another module loaded to use xlog? Finally, does anyone have a working, complete example of SIP over web sockets with ephemeral auth integrated? I'm not sure if I should be checking the ephemeral auth credentials in the web sockets code, the SIP routes or somewhere else. I'm also not sure how I should set up authentication such that users can log in with both their standard credentials or those retrieved via the ephemeral auth module. Thanks.

2 1

Integrating Kamailio with existing IMS
by Muhammad Shahzad 27 Oct '14

27 Oct '14

Hi, I am trying add WebRTC support to existing IMS using kamailio. The idea is to let kamailio handle all webrtc calls alone with diameter backend to IMS for AAA. So, i thought of using auth_diameter and aac module for authentication and accounting. However auth_diameter docs say that the module is obsolete and will work no more. Also i am not sure about how to use diameter support in aac module. So, is there any any tutorial or how-to for this? I am aware of ims_* modules but i think they are meant to be used in case we want kamailio to act as IMS core (P-CSCF, S-CSCF etc.). Thank you.

1 0

best way to cope with peaks in traffic
by davy van de moere 27 Oct '14

27 Oct '14

Gents, assuming we don't use async methods (I know, it's against fashion), eventually you'll end up in having some bottlenecks in your config... Now thanks to a lot of children you can easily work around this, a maximum of 32 children gives you already quite something, and if you use the dispatcher, you can ramp that up to quite some capacity. ;) Now, there are moments where still limits are passed, so what would be the best way to keep your system healthy? I was thinking in the direction of using a hashtable value or $var to keep track of the children in use and e.g. if 31 children are used, refuse to process the 32th request (assuming 32 children are setup). Would this be a valid strategy? Or am I overthinking? Grtz, Davy

2 3

lcr module ping
by Debojit Kakoti 27 Oct '14

27 Oct '14

Hi, My current setup is I have one kamailio server and one asterisk server.In the asterisk server I have two trunk where there are two different PRI provider is used for outbound call. I am using uac_replace_from function to route call to different trunk of the same server. However I want to use lcr module and I can check if particular gateway is up and down using lcr module and route the call. But problem is In my current scenario the gateway is up but one PRI is down for one provider and other one is up.So how to check if one sip trunk is working and other one is not working? Can you suggest me a solution? -- Debojit Kakoti Mobile : +91 9650654870 Email : debojkakoti88(a)gmail.com Skype : deboj1988

2 1

rtpengine not working with append_branch
by Yuriy Gorlichenko 27 Oct '14

27 Oct '14

Hello all. I use rtpengine and rtpproxy-ng module at kamailio for proxying RTP and modifying SDP between endpoints. I use two types of clients - such as WSS based and UDP based clients. I have a trouble with append_branch and rtpengine handling for this packets. I try to implement this logic of my script Logic of my script: --checking location table for rows with needed account --get info from contact at loop for every step --check technology (sip or ws) --append_branch with existing destination for this account --rewrite packet with rtpengine to needed technology after loop forward packets via t_relay At listing bellow rtpproxy_manage have no flag "b" that implement needed functionality. It not included at the script because I tried include it but rtpengine says that ca nnot handle this request. As i understand this flag worked with rtpproxy and rtpproxy-ng apps only. Not with rtpengine. At my test I have 2 endpoints with WS and UDP phones (at fist step ir WS and 2 step it UDP). All calls going form asterisk via UDP. When I do these steps (at my script) I see packet at TCP dump and saw that sended only one packet to UDP but body of packet is WS. Then I saw log of kamailio. I see that at second step packet changed body to WS body (so strange because other steps before and after goes for UDP (as at logic of script)) If I set "b" flag I can call only UDP endpoint, but at log I see as I say that rtpengine can not do thomething with SIP packets and send with t_relay uncghanged UDP packet only. sql_query("ca", "select contact from location where username='$tU'", "ra"); xlog("rows: $dbr(ra=>rows) cols: $dbr(ra=>cols)\n"); if($dbr(ra=>rows)>0){ $var(i)=0; while($var(i)<$dbr(ra=>rows)){ xlog("L_INFO","SQL query return contact {$dbr(ra=>[$var(i),0])} for {$tU} at step {$var(i)}\n"); if ($dbr(ra=>[$var(i),0])=~"transport=ws"){ xlog("L_INFO", "This is a Websocket call to endpoint"); sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); xlog("L_INFO","SQL query return recieved {$var(recieved)} for {$tU}\n"); $du=$var(recieved); xlog("L_INFO", "Request going FROM ASTERISK to WS. Destination is {$du}\n"); xlog("L_INFO","Websocket Destination URI is {$var(recieved)} for {$tU}\n"); rtpproxy_manage("froc+SP"); t_on_reply("REPLY_FROM_WS"); append_branch("sip:$tU@$du"); $var(i) = $var(i) + 1; } else { xlog("L_INFO", "This is a classic UDP call to endpoint"); $du="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); xlog("L_INFO","Classic Destination URI is {$dbr(ra=>[$var(i),0])} for {$tU}\n"); rtpproxy_manage("co"); t_on_reply("MANAGE_CLASSIC_REPLY"); append_branch("sip:$tU@$du"); $var(i) = $var(i) + 1; } #append_branch("sip:$tU@$du"); } } return 1; So it customised schema of standart example seturi("sip:a@example.com <https://e.mail.ru/compose/?mailto=mailto%3asip%253Aa@example.com>"); append_branch("sip:b@example.com <https://e.mail.ru/compose/?mailto=mailto%3asip%253Ab@example.com>"); append_branch("sip:c@example.com <https://e.mail.ru/compose/?mailto=mailto%3asip%253Ac@example.com>"); append_branch("sip:d@example.com <https://e.mail.ru/compose/?mailto=mailto%3asip%253Ad@example.com>"); t_relay(); I need fro advice how to fix this may be with another applications or with another steps. Thanks.

3 12

msilo kamailio 4.2
by nadie 27 Oct '14

27 Oct '14

Hello, excuse me for the question. I compiled kamailio 4.2 from git with msilo. kamailio.cfg ############################## loadmodule "msilo.so" # ------ msilo params ----- modparam("msilo", "db_url", DBURL) modparam("msilo","from_address","sip:registrar@kamailio.org") modparam("msilo","contact_hdr","Contact:registrar@192.168.1.3:5060;msilo=yes\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","*** User $rU is offline!") But now, i don't know where introduce route code. And in the documentation route coude tells "test only, REGISTER ok" Are there code for secure production? Thank you.

1 0

Fwd: DROUTING module slow loading
by Uri Shacked 26 Oct '14

26 Oct '14

Hi, When DROUTING table "dr_gateways" is very big (40000 rows) reloading the module to memory take about 60 seconds !!! My tests show 5000 rows takes 7-11 seconds...... It is very slow. The rules table, with 40000 rows, loads immediately though. Any ideas why? Or how to get the process faster? (I know the gateway table is too big. but, this is the current implementation...) kamailio version - 3.3.6 (48db34) BR, Uri

1 0

new compiler warning in 4.2
by Juha Heinanen 25 Oct '14

25 Oct '14

i build kamailio from latest 4.2 branch and got CC (cc) [M textops.so] textops.o textops.c: In function 'append_multibody_helper': textops.c:1867:2: warning: implicit declaration of function 'get_mixed_part_delimiter' [-Wimplicit-function-declaration] please fix asap. -- juha

1 0

Converting JSON string to normal string
by Muhammad Shahzad 25 Oct '14

25 Oct '14

Hi, Using method json_get_field i am able to extract json string. However, this string comes with quotes which cause a hurdle in assigning to various pseudo variables, for example, -- json_get_field($redis(r=>value), "to_number", "$var(wim_to)"); json_get_field($redis(r=>value), "domain", "$var(wim_domain)"); $uac_req(ruri)="sip:" + $var(wim_to) + "@" + $var(wim_domain); -- This results in $uac_req(ruri) being set as. sip:"49XXXXXXXX"@"sip.mydomain.com" Which is obviously wrong and fails the transaction processing. Is there any way to get rid of these quotes from json strings? One possible way i can think of is using string transformation "s.substr", but there are a lot of json strings i have in config and it is too hectic to transform all of them. Is there any easier way out of this? Thank you.

2 2

Failed to install Kamailio database
by Mahmoud Ramadan Ali 25 Oct '14

25 Oct '14

Hiii everyone, I can not create kamailio database and get this error message... any ideas ? Thanks in advance...

2 6

How do I configure this thing?
by Nolan Darilek 24 Oct '14

24 Oct '14

OK, sorry for the basic question, but I'm overwhelmed. My install gives me a kamailio-basic.cfg, kamailio.cfg and kamailio-oob.cfg. Is there any explanation as to the difference between these? I assume "basic" is a simpler config, but I don't know what it lacks vs. kamailio.cfg, and I have no idea what "oob" means ("out-of-band?") The configs reference a kamailio-local.cfg. My assumption was that I could place config changes here, but I see later that kamailio.cfg changes some parameters. In particular, I set various auth_db parameters in kamailio-local.cfg which kamailio.cfg changes later. So I don't understand when I might want to use kamailio-local. The config files refer me to a cookbook, but unless I'm missing something, the cookbook is a long list of config directives. When I read "cookbook" I assume I'm being directed to something containing recipes like "how to enable web sockets," with step-by-step guides about which config files to edit and what to add. Am I just missing that? Is there a task-oriented getting started page that I'm missing, that would help me set up a web socket-based SIP server? Then, when I hit the inevitable issues, I can resolve specific problems rather than trying to figure this all out from scratch? I see the documentation attached to the web sockets module, but many of these docs include stand-alone config snippets with no indication as to how to integrate them into a larger config. For instance, the ephemeral auth module tells me how to set up an ephemeral auth route but doesn't give me enough context to place the snippet correctly in the 900+-line kamailio.cfg, and actually I suspect it wouldn't work anyway since I see nothing about HTTP in the kamailio.cfg. Sorry if I seem frustrated. I'm willing to learn, I've just been handed a jet engine and a few suggested tweaks, but not enough context to know how to make them. :) I'll be learning more about SIP and what role a SIP router plays, but it'd be helpful to be able to bootstrap a server so I can learn by doing as well. I'd just like to provide audio/video communication for a single SIP domain, authenticating against MongoDB and with ephemeral auth so Javascript clients can get temporary credentials. I don't want voicemail, outbound routing, etc. This seems like it should be simple to achieve, but I don't know how or where to start in a way that would allow me to build up. Thanks.

3 6

Kamailio + RTPEngine behind NAT (STUN Handling)
by Marko Seidenglanz 24 Oct '14

24 Oct '14

Hello, We want to use Kamailio (4.2) + RTPEngine (3.3) behind NAT. Unfortunately the STUN Messages (Binding Requests) sent from remote peer (Google Chrome ) do not arrive at RTPEngine host. I think they get blocked by the firewall. RTPEngine itself does not send any STUN Messages until it get's the first message from the remote peer as Google Chrome has the "active" role in DTLS Agreement. Is there a way to tell RTPEngine to open the ports for remote peer, so that the messages arrive? When RTPEngine is not behind NAT, everything works as expected. Kind regards, Marko

2 1

Dealing with High PDD
by Ryan Brindley 24 Oct '14

24 Oct '14

Hey Community, I've tried to google and view the docs for Kamailio about fr_inv_timer (and other timers), but I don't think they do what I'm wanting. Basically, I want to cancel an INVITE request and progress to the next contact if the current contact has not send a non-100 provisional code or final response after a certain amount of time. Am I missing some obvious setting that I overlooked in my searches or do I have to approach this 'manually' (aka, not just a setting)? Any help is greatly appreciated. Ryan Brindley

2 2

Kamailio, authentication and MongoDB
by Nolan Darilek 24 Oct '14

24 Oct '14

Hello. I'm attempting to set up Kamailio for the first time and have a few questions. 1. The auth_db module specifically calls out MySQL, Postgres and flat file databases, but makes no mention of the new MongoDB module. Is the new MongoDB module supported as well and the docs just haven't been updated to account for that? 2. In looking at the auth_db parameters, I see values for column names, but nothing for a table/collection. How do I specify what table or collection my user data is in? 3. How do I create HA1 strings? The docs state: HA1 strings are MD5 hashes of username, password, and realm. Is that: md5(username+":"+password+":"+realm) or something else? Thanks.

1 1

dispatcher behind NAT - lost ACK
by Nicholas Gill 24 Oct '14

24 Oct '14

Hello sr-users, We have Kamailio (behind NAT) configured acting as a proxy in front of some FreeSWITCH servers. There appears to be something amiss with my inbound (dispatcher) configuration which leads to misaddressing / misdelivery of the ACK after 200 OK (outbound calls appear to be proxied correctly [1]). Calls from the sip provider incoming to Kamailio are distributed by the dispatcher module to the FS server (For testing in this simplified scenario FS and Kamailio are on the same machine): SIP provider (5060) sends an invite to Kamailio(5061), Kamailio uses the dispatcher module to select a backend server and forwards the INVITE to FS(5070) (see inbound-callflow.png; kamailio.ext and kamailio.int are the same machine, just public/private addresses). I notice that at no point is a Via/record-route header for the FS server inserted into the forwarded session. I'm not actually certain this is a requirement, but I can't think of another obvious way that Kamailio could proxy the same session to the same FS server. The 100 Trying seems unremarkable [2], I suspect the 200 OK [3] is problematic. It has been proxied from the FS server, however contains no reference to the FS server address either in the via headers nor record-route (kamailio.int:5070). The incoming ACK [4] then appears to be misdelivered / lost - Kamailio receives it on the private address and forwards it to the public address rather than the FS server. My configuration [5] is built based on the default configuration + examples from the dispatcher module. There are some provisions for FreeSWITCH internal/external profiles made so the configuration listens on 2 different ports. This particular scenario should only use the 5061 port as it involves calls to an external sip provider (briefly 5060 should be proxied to FS:5080 and 5061 should be proxied to FS:5070 and vice-versa). If someone can see an issue with the configuration and/or point to an error in the call flow (i.e. should FS be inserting the Via header?) that would be greatly appreciated. Thanks, -nick [1] Outbound call flow (see also outbound-callflow.png) FS(port 5070) sends an invite to Kamailio(5061) (Kamailio is configured in FS as an outbound proxy), INVITE contains Via header for the FS server, Kamailio forwards to sip provider, and routes all messages back and forth correctly. [2] 100 Trying SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.0;rport=5060 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com> Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 INVITE Server: kamailio (4.1.6 (x86_64/linux)) Content-Length: 0 [3] Proxied 200 OK (Kamailio -> Sip Provider) SIP/2.0 200 OK Via: SIP/2.0/UDP sip.provider.com:5060;rport=5060;branch=z9hG4bKfffb.7c81ee53.0 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK Record-Route: <sip:kamailio.ext:5061;lr=on> Record-Route: <sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr;did=0ec.84ff82b1> From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com>;tag=BFvQmggHrg74m Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 INVITE Contact: <sip:0390156842@kamailio.int:5070;transport=udp> User-Agent: IMX Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-FS-Support: update_display,send_info [4] Incoming ACK (Sip Provider -> Kamailio) ACK sip:0390156842@kamailio.ext:5061;transport=udp SIP/2.0 Record-Route: <sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr> Via: SIP/2.0/UDP sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.2 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK509ZQBKB947aF Route: <sip:kamailio.ext:5061;lr=on> Max-Forwards: 69 From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com>;tag=BFvQmggHrg74m Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 ACK Contact: <sip:gw+Exetel@far.external.ip:5060;transport=udp;gw=Exetel> Proxy-Authorization: Digest username="0384171900", realm="far.external.ip", nonce="xxxx", algorithm=MD5, uri="sip:0390156842@sip.provider.com", response="xxxx" Content-Length: 0 P-hint: rr-enforced [5] Kamailio configuration (based on default configuration). Dispatcher set 1 is FS internal profile (5080) and set 2 is external profile (5070) ... #!define WITH_NAT ... alias="kamailio.int:5060" alias="kamailio.int:5061" alias="kamailio.ext:5060" alias="kamailio.ext:5061" listen=udp:kamailio.int:5060 advertise kamailio.ext:5060 listen=udp:kamailio.int:5061 advertise kamailio.ext:5061 port=5060 port=5061 ... loadmodule "dispatcher.so" modparam("dispatcher", "flags", 2) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #modparam("dispatcher", "ds_ping_interval", 60) ... modparam("nathelper", "sipping_from", "sip:pinger@etellicom.com") ... request_route { ... # Not from FS servers gets dispatched to FS servers. if(!ds_is_from_list("1") and !ds_is_from_list("2")) { route(DISPATCH); } # Requests from FS External profile routed out via specific port. if(ds_is_from_list("2")) $fs="udp:kamailio.int:5061"; ... } route[DISPATCH] { if(dst_port==5061) { $fs="udp:kamailio.int:5061"; # Hash over to uri for test purposes if(!ds_select_dst("2", "2")) { send_reply("404", "No destination"); exit; } } else { # Hash over to uri for test purposes if(!ds_select_dst("1", "2")) { send_reply("404", "No destination"); exit; } } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } ...

2 3

sip response not handled by kamailio
by Marko Seidenglanz 24 Oct '14

24 Oct '14

Hello, We are trying to use Kamailio 4.2 with RTPEngine 3.3 behind NAT. Somehow SIP responses (200 OK) are not handled by Kamailio. The following INVITE is send to the receiving end (WebRTC Client): INVITE sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK865.9db0c023bec2f314e89ae0f18b78d2f0.0 Via: SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK644600e2 Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as68f0a26a To: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com> Contact: <sip:anonymous@10.240.215.73:5060;alias=146.148.113.245~5060~1> Call-ID: 368f6ec901a3552e3b37a70907634f4f@10.240.215.73:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Fri, 24 Oct 2014 08:38:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 738 P-hint: outbound ....... The client response looks like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK865.9db0c023bec2f314e89ae0f18b78d2f0.0;rport=5060;received=104.155.11.255 , SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK644600e2 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as68f0a26a To: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com >;tag=E2VMB39U5NYHCXLZREFGQV2DAXPCZXNQ6J6P Contact: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com >;tag=E2VMB39U5NYHCXLZREFGQV2DAXPCZXNQ6J6P Call-ID: 368f6ec901a3552e3b37a70907634f4f@10.240.215.73:5060 CSeq: 102 INVITE Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Content-Type: application/sdp Content-Length: 938 ....... I have a xlog call as the first statement in kamailio routing script. I can see the request, which get's handled correctly but the response does not appear in kamailio log, though it arrives at the machine. I am totally helpless, what might be wrong. If we use kamailio on a public accessable machine everything works as expected. Maybe anyone might have had similiar problems using kamailio behind NAT. Kind regards, Marko

2 3

RPM Based installs...
by Derrick Bradbury 24 Oct '14

24 Oct '14

Just a question for the RPM maintainers... Is there a reason for some of the modules (such as carrierroute) not being included in the RPMS? It's been bugging me for a while, when I do an installation, I have to go and compile all the sources .... Not that it's an issue, just easier to do it from RPM if available. Thanks! Derrick

2 1

Kamailio Infront of Asterisk with remote PBX
by Kenny Watson 24 Oct '14

24 Oct '14

Hi, I have a few asterisk servers providing some basic SIP trunking and routing. We have remote PBXs trunked onto asterisk which calls come into asterisk and are routing down to extensions on the remote PBX via prefix routing. I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer which Invites come into and are routed out to either SIP phones which are registered or to the remote PBX. I'm looking for some advice as to which modules would be best to use to achieve this as the remote PBXs will be dynamically registered rather than fixed gateways. Please let me know what further information would be helpful. Thanks Kenny Watson

4 7

async workers
by Alex Balashov 24 Oct '14

24 Oct '14

Hi, What is the practical limit to the number of async worker processes? With SIP child processes, it seems to be about the number of available CPUs in /proc/cpuinfo. After that--at least, per my testing--one begins to hit the point of diminishing returns, presumably due to SHM IPC and synchronisation issues. Is the restriction similar in the async execution context? -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

4 8

Crash after async_sleep().
by Vitaliy Aleksandrov 24 Oct '14

24 Oct '14

Hello list. root@proxy:/# kamcmd core.version kamailio 4.1.4 (i386/linux) I'm getting a crash when I'm trying to simulate a ringing UAS from kamailio config. To achieve this I've added the next config actions: > if (is_method("INVITE") && !has_totag() && $rU =~ "^999") { > sl_send_reply("100", "Fake Trying"); > sl_send_reply("180", "Fake Ringing"); > > t_set_fr(120000, 120000); > t_set_retr(65000, 65000); > t_set_auto_inv_100(0); > > async_sleep("60"); > send_reply("480", "Fake Temporary Unavailable"); > > t_release(); #<--- > > exit(); > }; I expect to receive 180 reply immediately and 480 after 60 seconds. The problem appears when a caller cancels a call after 60 seconds which is equal to async_sleep() wait time. I suppose there is a race condition happens between UDP worker that processes CANCEL request and timer process which calls t_continue() (tm's function). Calling t_newtran() explicitly doesn't change the situation. The only way I found to avoid crash is to remove t_release() call. In this case kamailio complains that "script write didn't release transaction" but continues to process new calls. With -DTIMER_DEBUG and -DDBG_QM_MALLOC I got next information: /usr/sbin/kamailio[11307]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0xb1d8c520), called from tm: h_table.c: free_cell(178), first free tm: h_table.c: free_cell(157) - aborting Crash mostly happens because of broken circular list of timers (timer_ln->next == NULL), but one time I saw got abort() during CANCEL request processing when tm tried to call free_cell() for an already freed pointer.

2 1

Re: [SR-Users] One sided or no voice issue with websockets
by Amit Patkar 23 Oct '14

23 Oct '14

Please check ICE server settings. Your browser may be publishing local IP. One way voice or no voice is typical case when client is behind firewall. Regards, Amit Patkar dodul <dodul(a)live.com> wrote: >Hi > >I didn't get any responses from anyone regarding my issue. Can someone please give me some clue what I can do? Or if more information is needed please let know so that I can provide. > > >Sent from my Samsung Galaxy smartphone. > > >-------- Original message -------- >From: Kamrul Khan <dodul(a)live.com> >Date:10-21-2014 17:45 (GMT-06:00) >To: sr-users(a)lists.sip-router.org >Cc: >Subject: [SR-Users] One sided or no voice issue with websockets > >Hi, > >We have a setup with sipml5 to kamailio. It works perfectly within local network. In public network the signaling establishes perfectly, but most of the time we hear no voice, sometimes we hear one sided voice and in rare cases we hear voice from both sides. To fix this issue we configured our nathandler like the below: But, still no luck. Any idea how to fix this? Please HELP!!! > >modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") >modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") >. >. >. >route { >. >. > if (nat_uac_test("115")) { > if(nat_uac_test("64")) > force_rport(); > } > if (method=="REGISTER") { > fix_nated_register(); > add_rcv_param(); > } else { > fix_nated_contact(); > if(nat_uac_test("64")){ > if (!add_contact_alias()) { > xlog("L_ERR", "Error aliasing contact <$ct>\n"); > sl_send_reply("400", "Bad Request"); > exit; > } > } else { > add_rcv_param(); > } > } > } >. >. > >onreply_route { > if (nat_uac_test(64)) { > add_contact_alias(); > } >. >. >} > > > > >_______________________________________________ >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 > >_______________________________________________ >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

3 3

Kamailio behind NAT (Asterisk - Kamailio - RTPEngine - Browser)
by Marko Seidenglanz 23 Oct '14

23 Oct '14

Hello, I have a problem with the following configuration. I want to make calls from Asterisk to a browser using RTPEngine as relay. Everything works fine, if Kamailio is not natted (See kamailio_without_nat.log). If it's address is translated, then 200 OK responses from the browser don't seem to be received by kamailio. I have configured Kamailio (4.2) and RTPEngine (3.3) both with advertisement of the public IP. Asterisk has public IP: 146.148.113.245 Kamilio has public IP: 104.155.11.255 Browser has public IP: 79.241.195.106 The INVITE after SDP Rewrite looks like this: INVITE sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255;lr=on;nat=yes> Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK683.5e75aa8b7f88561a91033a9b611fc0aa.0 Via: SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK5a79a7cf Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as53a7de72 To: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com> Contact: <sip:anonymous@10.240.215.73:5060;alias=146.148.113.245~5060~1> Call-ID: 523c5fda707c565c51b78c586247818e@10.240.215.73:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 23 Oct 2014 15:29:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 738 P-hint: outbound v=0 o=root 985629145 985629145 IN IP4 104.155.11.255 s=Asterisk PBX 12.5.0 c=IN IP4 104.155.11.255 t=0 0 a=ice-lite m=audio 31630 RTP/SAVPF 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:31631 a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:H9uLHKvstgjd58cZKa0LeRaxvf5XNJhaHgxDut7L a=setup:actpass a=fingerprint:sha-1 D9:26:45:E1:D6:E7:01:A4:04:90:F2:15:AF:A3:AD:01:3C:39:9B:7D a=ice-ufrag:YoTvWbLM a=ice-pwd:eifInSr3Oh8ZvMXjlE89mh8kF1OC a=tDndidate:krhMgLYHbAxWidfK 1 UDP 2130706431 104.155.11.255 31630 typ host a=candidate:krhMgLYHbAxWidfK 2 UDP 2130706430 104.155.11.255 31631 typ host The 200 OK like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK683.5e75aa8b7f88561a91033a9b611fc0aa.0;rport=5060;received=104.155.11.255 , SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK5a79a7cf From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as53a7de72 To: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com >;tag=6QTSN6N2SBW2FW49TMGJQJYZMGHQZ7LCUMQT Contact: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com >;tag=6QTSN6N2SBW2FW49TMGJQJYZMGHQZ7LCUMQT Call-ID: 523c5fda707c565c51b78c586247818e@10.240.215.73:5060 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 938 v=0 o=- 1907956949290984340 2 IN IP4 127.0.0.1 s=- t=0 0 m=audio 43590 UDP/TLS/RTP/SAVPF 0 8 101 c=IN IP4 79.241.195.106 a=rtcp:1 IN IP4 0.0.0.0 a=candidate:290309024 1 udp 2122260223 192.168.35.78 43590 typ host generation 0 a=candidate:2416297236 1 udp 1686052607 79.241.195.106 43590 typ srflx raddr 192.168.35.78 rport 43590 generation 0 a=candidate:1607352144 1 tcp 1518280447 192.168.35.78 0 typ host tcptype active generation 0 a=candidate:2416297236 1 udp 1686052607 79.241.195.106 43590 typ srflx raddr 192.168.35.78 rport 43590 generation 0 a=ice-ufrag:ZtbtZieUG9l22iCb a=ice-pwd:kRkmd0XuFdj+CKXVpInmK3yV a=fingerprint:sha-256 ED:DB:B8:D5:4D:38:1F:81:DC:94:9F:EB:6D:07:56:75:57:45:F4:F7:57:4D:C6:89:70:CC:13:6D:35:C0:8B:45 a=setup:active a=mid:audio a=sendrecv a=rtcp-mux a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=msid-semantic: WMS mPbtogqqqsv8DygpyOmxrTkjgW9aGzkbP3Vk Unfortunately Kamailio log does not tell me, that the response get's processed. (See kamailio_behind_nat.log). I traced with TShark and can see that the SIP/200 OK arrive at kamailio host. Any Idea, what might be missing in my routing script? The attachements can be found here: https://drive.google.com/folderview?id=0BxwAyaFvy_7fdHVnWXdFa2hEeHM&usp=sha… I would be very grateful for any help. Regards, Marko

1 0

Kamailio behind NAT (Asterisk - Kamailio - RTPEngine - Browser)
by Marko Seidenglanz 23 Oct '14

23 Oct '14

Hello, I have a problem with the following configuration. I want to make calls from Asterisk to a browser using RTPEngine as relay. Everything works fine, if Kamailio is not natted (See kamailio_without_nat.log). If it's address is translated, then 200 OK responses from the browser don't seem to be received by kamailio. I have configured Kamailio (4.2) and RTPEngine (3.3) both with advertisement of the public IP. Asterisk has public IP: 146.148.113.245 Kamilio has public IP: 104.155.11.255 Browser has public IP: 79.241.195.106 The INVITE after SDP Rewrite looks like this: INVITE sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255;lr=on;nat=yes> Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK683.5e75aa8b7f88561a91033a9b611fc0aa.0 Via: SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK5a79a7cf Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as53a7de72 To: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com> Contact: <sip:anonymous@10.240.215.73:5060;alias=146.148.113.245~5060~1> Call-ID: 523c5fda707c565c51b78c586247818e@10.240.215.73:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 23 Oct 2014 15:29:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 738 P-hint: outbound v=0 o=root 985629145 985629145 IN IP4 104.155.11.255 s=Asterisk PBX 12.5.0 c=IN IP4 104.155.11.255 t=0 0 a=ice-lite m=audio 31630 RTP/SAVPF 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:31631 a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:H9uLHKvstgjd58cZKa0LeRaxvf5XNJhaHgxDut7L a=setup:actpass a=fingerprint:sha-1 D9:26:45:E1:D6:E7:01:A4:04:90:F2:15:AF:A3:AD:01:3C:39:9B:7D a=ice-ufrag:YoTvWbLM a=ice-pwd:eifInSr3Oh8ZvMXjlE89mh8kF1OC a=tDndidate:krhMgLYHbAxWidfK 1 UDP 2130706431 104.155.11.255 31630 typ host a=candidate:krhMgLYHbAxWidfK 2 UDP 2130706430 104.155.11.255 31631 typ host The 200 OK like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK683.5e75aa8b7f88561a91033a9b611fc0aa.0;rport=5060;received=104.155.11.255 , SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK5a79a7cf From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as53a7de72 To: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com >;tag=6QTSN6N2SBW2FW49TMGJQJYZMGHQZ7LCUMQT Contact: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com >;tag=6QTSN6N2SBW2FW49TMGJQJYZMGHQZ7LCUMQT Call-ID: 523c5fda707c565c51b78c586247818e@10.240.215.73:5060 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 938 v=0 o=- 1907956949290984340 2 IN IP4 127.0.0.1 s=- t=0 0 m=audio 43590 UDP/TLS/RTP/SAVPF 0 8 101 c=IN IP4 79.241.195.106 a=rtcp:1 IN IP4 0.0.0.0 a=candidate:290309024 1 udp 2122260223 192.168.35.78 43590 typ host generation 0 a=candidate:2416297236 1 udp 1686052607 79.241.195.106 43590 typ srflx raddr 192.168.35.78 rport 43590 generation 0 a=candidate:1607352144 1 tcp 1518280447 192.168.35.78 0 typ host tcptype active generation 0 a=candidate:2416297236 1 udp 1686052607 79.241.195.106 43590 typ srflx raddr 192.168.35.78 rport 43590 generation 0 a=ice-ufrag:ZtbtZieUG9l22iCb a=ice-pwd:kRkmd0XuFdj+CKXVpInmK3yV a=fingerprint:sha-256 ED:DB:B8:D5:4D:38:1F:81:DC:94:9F:EB:6D:07:56:75:57:45:F4:F7:57:4D:C6:89:70:CC:13:6D:35:C0:8B:45 a=setup:active a=mid:audio a=sendrecv a=rtcp-mux a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=msid-semantic: WMS mPbtogqqqsv8DygpyOmxrTkjgW9aGzkbP3Vk Unfortunately Kamailio log does not tell me, that the response get's processed. (See kamailio_behind_nat.log). I traced with TShark and can see that the SIP/200 OK arrive at kamailio host. Any Idea, what might be missing in my routing script? I would be very grateful for any help. Regards, Marko

1 0

RPC uac.reg_info hangs second time
by Dan Christian Bogos 23 Oct '14

23 Oct '14

Hey Guys, I am in the process of testing the reloading of the uac registrations (thanks Daniel for that), and hit some issue: * when using uac_reg_info twice in a row, second command always hangs. """ root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_info l_uuid 301 { l_uuid: 301 l_username: 301 l_domain: pbx.example.com r_username: 301 r_domain: pbx.example.com realm: asterisk auth_username: 301 auth_password: check123 auth_proxy: sip:172.16.254.75 expires: 360 flags: 0 diff_expires: -1414052722 timer_expires: 0 } root@Dev2:/etc/kamailio# root@Dev2:/etc/kamailio# root@Dev2:/etc/kamailio# root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_info l_uuid 301 ^C root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_info l_uuid 301 """ Do you see anything wrong with my data or did I hit a bug? Since we are here and could not find much documentation around, is there a way to decrease the default 150 seconds blocking reloads from the database? """ root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_reload root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_reload error: 500 - Failed to shift records - check log messages """ Syslog: """ Oct 23 10:32:17 Dev2 /usr/sbin/kamailio[2693]: ERROR: uac [uac_reg.c:369]: uac_reg_ht_shift(): shifting the memory table is not possible in less than 150 """ Thanks in advance for any kind of tip! DanB

2 1

Unix timestamp with millisecond precision
by Grant Bagdasarian 23 Oct '14

23 Oct '14

Hello, I'm currently using the $TS psuedovariable to get the current unix timestamp, but this only returns the timestamp up to a second precision. Is it possible in kamailio to get the current unix timestamp with milliseconds precision? Regards, Grant

2 3

hello. I'm a user for kamailio. I wanna ask something
by 신희준 23 Oct '14

23 Oct '14

Hello I have a question about uac module. When i use uac_req_send function, from tag is made by this function automatically. I wanna make a PRACK request which is in-dialog request, so i don't want to append new from-tag at that situation. I think there some solution, but i can't find.. can you help me?? thank you

1 0

usrloc: Unregister client on TCP connection close
by Camille Oudot 22 Oct '14

22 Oct '14

Hi, I'm looking for a mechanism to unregister clients when the corresponding TCP connection is closed, but had no success so far. I've tried the handle_lost_tcp parameter of usrloc, but it didn't work: the user was not unregistered, whereas the connection was seen as closed by kamailio. Is it still a relevant option? Otherwise, is there any mechanism that would trigger some kind of event_route whenever a connection (TCP, TLS, WS, WSS) is lost or closed? Cheers, -- Camille

3 6

Re: [SR-Users] Connecting IMS and Webrtc!!
by Carsten Bock 22 Oct '14

22 Oct '14

Hi Dionlar, can you elaborate the Memory issue a little bit more? Can you also send me more information, on what OS and what Kamailio-Version you are using? What configuration are you using? If you have a memory issue, feel free to contact sr-users(a)lists.sip-router.org, they should be able to help you as well. Probably no-one of our team (core-networks@) will be able to help you further, as i am taking care of the implementation myself. Thanks, Carsten 2014-10-22 1:27 GMT+02:00 dionlar lang <dionlar.lang(a)gmail.com>: > Hello dear!! > > I am Lang, undergraduate student research at the University Sheik Anta Diop > of Dakar and I have a memory issue focusing on the interconnection between > IMS and Webrtc.Je am writing to ask your proposals because I just read some > of your articles about the subject.Is it you please help me. > > Thank you and I count on you! -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-0 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

Re: [SR-Users] One sided or no voice issue with websockets
by dodul 22 Oct '14

22 Oct '14

Hi I didn't get any responses from anyone regarding my issue. Can someone please give me some clue what I can do? Or if more information is needed please let know so that I can provide. Sent from my Samsung Galaxy smartphone. -------- Original message -------- From: Kamrul Khan <dodul(a)live.com> Date:10-21-2014 17:45 (GMT-06:00) To: sr-users(a)lists.sip-router.org Cc: Subject: [SR-Users] One sided or no voice issue with websockets Hi, We have a setup with sipml5 to kamailio. It works perfectly within local network. In public network the signaling establishes perfectly, but most of the time we hear no voice, sometimes we hear one sided voice and in rare cases we hear voice from both sides. To fix this issue we configured our nathandler like the below: But, still no luck. Any idea how to fix this? Please HELP!!! modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") . . . route { . . if (nat_uac_test("115")) { if(nat_uac_test("64")) force_rport(); } if (method=="REGISTER") { fix_nated_register(); add_rcv_param(); } else { fix_nated_contact(); if(nat_uac_test("64")){ if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } else { add_rcv_param(); } } } . . onreply_route { if (nat_uac_test(64)) { add_contact_alias(); } . . } _______________________________________________ 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

Asynchronous sql operation also for acc module?
by Sebastian Damm 22 Oct '14

22 Oct '14

Hi, I read the article about asynchronous sqlops: http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-4-asynchronous-sql.h… I was wondering whether there are any plans of bringing that feature to the acc module, too. We have seen it in the past that a blocking database server brought our proxy to a halt because the acc module was blocking, too. So this could help a lot if the acc entries were written asynchronously. Best Regards, Sebastian

2 3

NATMANAGE quirks
by aft 22 Oct '14

22 Oct '14

Hi, I'm trying to re packet rtp streams based on Destination IP. I've implemented it like below : route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } xlog("Destination [$du]..."); if ($du == "108.59.14.153") rtpproxy_manage("coz20"); else rtpproxy_manage("coz100"); } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } rtpproxy_manage("coz100"); } #!endif return; } But its not working, and the log comes up like : Oct 21 08:12:33 srv4617 /usr/local/sbin/kamailio[28346]: ERROR: <script>: Destination...... <null>

2 2

clean out obsolete entries in dialog table?
by Måns Nilsson 22 Oct '14

22 Oct '14

Hi, I've got a dialog database table with lots of old entries; some several months old. I'd like to remove them, but if there is a method that keeps things synchronous and checks beyond "DELETE FROM dialog WHERE .... " I'd like to use it. Any pointers? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 I'm a GENIUS! I want to dispute sentence structure with SUSAN SONTAG!!

1 0

Fwd: ACK not forwarding properly
by Sadique Ali 22 Oct '14

22 Oct '14

hi, i want to forward call from VITELITY to sip TWILIO , by using config file (attached one) call gets connected but end automatically after 33 seconds. i could find some miss configurations in sip packet flow with routing ACK ( attached .pcap ) . Please suggest a solution . -regards

1 0

Fwd: ACK not forwarding properly
by Sadique Ali 22 Oct '14

22 Oct '14

---------- Forwarded message ---------- From: Sadique Ali <spalikklpd(a)gmail.com> Date: 21 October 2014 18:55 Subject: ACK not forwarding properly To: sr-users-es(a)lists.sip-router.org hi, i want to forward call from VITELITY to sip TWILIO , by using config file (attached one) call gets connected but end automatically after 33 seconds. i could find some miss configurations in sip packet flow with routing ACK ( attached .pcap ) . Please suggest a solution . -regards

1 0

One sided or no voice issue with websockets
by Kamrul Khan 21 Oct '14

21 Oct '14

Hi, We have a setup with sipml5 to kamailio. It works perfectly within local network. In public network the signaling establishes perfectly, but most of the time we hear no voice, sometimes we hear one sided voice and in rare cases we hear voice from both sides. To fix this issue we configured our nathandler like the below: But, still no luck. Any idea how to fix this? Please HELP!!! modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") . . . route { . . if (nat_uac_test("115")) { if(nat_uac_test("64")) force_rport(); } if (method=="REGISTER") { fix_nated_register(); add_rcv_param(); } else { fix_nated_contact(); if(nat_uac_test("64")){ if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } else { add_rcv_param(); } } } . . onreply_route { if (nat_uac_test(64)) { add_contact_alias(); } . . }

1 0

MSILO module not delivering messages
by aawaise 21 Oct '14

21 Oct '14

I have tried to add msilo module to my kamailio.cfg using the link http://sip-router.1086192.n5.nabble.com/Msilo-not-delivering-message-status… Problem is that messages from a user are received on the server. And they can be seen using siremis. But they are not forwarded when the intended receiver gets online. I have added the outboud_proxy in modparameters of MSILO module #!ifdef WITH_MSILO modparam("msilo", "db_url", "mysql://openser:5mil5My5ql@192.168.1.20/openser") modparam("msilo", "from_address", "sip:$rU@mydomain") modparam("msilo", "contact_hdr", "Contact:<sip:$rU@mydomain>\r\n") modparam("msilo", "content_type_hdr", "Content-Type:text/plain\r\n") modparam("msilo", "offline_message", "User $rU is offline!") modparam ("msilo", "outbound_proxy", "sip:mydomain:6524;transport=tcp") modparam ("msilo", "outbound_proxy", "sip:mydomain") modparam("msilo", "expire_time", 604800) #!endif I have added the code available on the link to my kamailio.cfg. -- View this message in context: http://sip-router.1086192.n5.nabble.com/MSILO-module-not-delivering-message… Sent from the Users mailing list archive at Nabble.com.

2 2

SIP Express Router Contact Header Buffer Overflow
by Kay Montag 21 Oct '14

21 Oct '14

Hi Team, my customer gets the following message. Find attached. I checked the provided link: http://www.iptel.org/ser/sercurity/secalert-002-0_8_10.patch But I get the error: "Page not found" Could you assist me with that issue? Regards Kay Mit freundlichen Grüßen Kay Montag -- avodaq AG, Landsberger Str. 290, D-80687 München Tel.: +49.1803.78 63 78, E-Mail: service(a)avodaq.com First European Cisco Master Unified Communications. Cisco Gold Partner. Cisco Learning Partner. Vorstandsvorsitzender: Andreas Kusch Aufsichtsratsvorsitzender: Klaus Genssen Amtsgericht München, Handelsregister 136037

3 4

Hash Table capacity
by Ricardo Martinez 21 Oct '14

21 Oct '14

Hello list. I have a question about the HTABLE module. Suppose that I have a mysql database with 2 million of records. Is possible to load this table as a hash table compatible with Kamailio!? Hope someone can help me. Thanks, Ricardo Martinez.-

6 15

Handle "MESSAGE" packet
by aawaise 21 Oct '14

21 Oct '14

I want to know how kamailio server handles incoming MESSAGE packets of SIP format ? If a sender sends a MESSAGE packet for receiver, and receiver is offline, how will server handle the packet. Can it hold the packet until the receiver is online and then deliver to receiver ?? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Handle-MESSAGE-packet-tp131509.html Sent from the Users mailing list archive at Nabble.com.

3 4

Re: [SR-Users] kamailio 4.2 ，have some problems when use mysql db
by Daniel-Constantin Mierla 21 Oct '14

21 Oct '14

Are you using 4.1 or 4.2? Because the subject mentioned 4.2 but the version of kamailio is 4.1 Daniel On 20/10/14 11:39, 张顺通 wrote: > -M 12 can not help. all config is same. > > see ka.log in attachment, kamailio version is 4.1.6 > > version: kamailio 4.1.6 (x86_64/linux) 010d57 > > flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, > DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, > DBG_QM_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: 010d57 > > compiled on 16:48:03 Oct 20 2014 with gcc 4.1.2 > > -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 13

send remote command to kamailio
by Vik Killa 20 Oct '14

20 Oct '14

Hello, I've installed kamailio from debian repo: deb http://deb.kamailio.org/kamailio wheezy main deb-src http://deb.kamailio.org/kamailio wheezy main I am unable to use any RPC related module though... apt-cache search doesn't show them...I installed every kamailio-* package just to make sure but still no luck. Basically, i just want to reload a hash remotely Locally I can do this with: kamcmd htable.dump freeswitch_ips Do I need to compile/install kamailio from source in order to use an RPC package? Thanks. /V

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parallel forking
by Yuriy Gorlichenko 20 Oct '14

20 Oct '14

Hello. I try to do parallel fork calls to endpoints that have same username and different destination URI. Logic of my script: checking location table for rows with needed account get info from contact at loop for every step check technology (sip or ws) append_branch with existing destination for this account rewrite packet with rtpengine to needed technology forward packets via t_relay sql_query("ca", "select contact from location where username='$tU'", "ra"); xlog("rows: $dbr(ra=>rows) cols: $dbr(ra=>cols)\n"); if($dbr(ra=>rows)>0){ $var(i)=0; while($var(i)<$dbr(ra=>rows)){ xlog("L_INFO","SQL query return contact {$dbr(ra=>[$var(i),0])} for {$tU} at step {$var(i)}\n"); if ($dbr(ra=>[$var(i),0])=~"transport=ws"){ xlog("L_INFO", "This is a Websocket call to endpoint"); sql_pvquery("ca", "select received from location where contact='$dbr(ra=>[$var(i),0])'","$var(recieved)"); xlog("L_INFO","SQL query return recieved {$var(recieved)} for {$tU}\n"); $du=$var(recieved); xlog("L_INFO", "Request going FROM ASTERISK to WS. Destination is {$du}\n"); xlog("L_INFO","Websocket Destination URI is {$var(recieved)} for {$tU}\n"); rtpproxy_manage("froc+SP"); t_on_reply("REPLY_FROM_WS"); append_branch("sip:$tU@$du"); $var(i) = $var(i) + 1; } else { xlog("L_INFO", "This is a classic UDP call to endpoint"); $du="sip:"+$(dbr(ra=>[$var(i),0]){s.select,1,@}); xlog("L_INFO","Classic Destination URI is {$dbr(ra=>[$var(i),0])} for {$tU}\n"); rtpproxy_manage("co"); t_on_reply("MANAGE_CLASSIC_REPLY"); append_branch("sip:$tU@$du"); $var(i) = $var(i) + 1; } #append_branch("sip:$tU@$du"); } } return 1; So it customised schema of standart example seturi("sip:a@example.com"); append_branch("sip:b@example.com"); append_branch("sip:c@example.com"); append_branch("sip:d@example.com"); t_relay(); At my test I have 2 endpoints with WS and UDP phones (at fist step ir WS and 2 step it UDP). when I do these steps (at my script) I see packet at TCP dump and saw that sended only one packet to UDP but body of packet is WS. Then I saw log of kamailio. I see that at second step packet changed body to WS body (so strange because other steps before and after goes for UDP (as at logic of script)) What I need to do for send parallel INVITES to multiple endpoints correctly? Thanks

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回复：Re: kamailio 4.2 ，have some problems when use mysql db
by shuntongzhang 20 Oct '14

20 Oct '14

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kamailio 4.2 ，have some problems when use mysql db
by 张顺通 20 Oct '14

20 Oct '14

Oct 20 08:14:45 ka[15500]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Oct 20 08:14:45 ka[15500]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Oct 20 08:14:45 ka[15500]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 Oct 20 08:14:45 ka[15500]: INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Oct 20 08:14:45 ka[15500]: ERROR: <core> [db_res.c:114]: db_new_result(): no private memory left Oct 20 08:14:45 ka[15500]: ERROR: db_mysql [km_dbase.c:229]: db_mysql_store_result(): no memory left Oct 20 08:14:45 ka[15500]: ERROR: <core> [db_query.c:139]: db_do_query_internal(): error while storing result Oct 20 08:14:45 ka[15500]: ERROR: <core> [db.c:401]: db_table_version(): error in db_query Oct 20 08:14:45 ka[15500]: ERROR: dispatcher [dispatch.c:669]: init_ds_db(): failed to query table version Oct 20 08:14:45 ka[15500]: ERROR: dispatcher [dispatcher.c:319]: mod_init(): could not initiate a connect to the database Oct 20 08:14:45 ka[15500]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module dispatcher (/lib64/kamailio/dispatcher.so) I need help.thks

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What is Attribute Value Pair (AVP) ?
by Mahmoud Ramadan Ali 19 Oct '14

19 Oct '14

Hiiii everyone, Forgive me for such simple question but i'm newbie to Kamailio. i was reading about the dispatcher module http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html#dispatch… and i want someone to explain to me in a simple way what is AVP , for example the AVP in the *modparam("dispatcher", "grp_avp", "$avp(dsgrp)")* and what the "dsgrp" represents and how to configure the AVP values ... Kinda Regards Thanks in advance...

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Regarding Kamailio usage in ARM/Embedded Linux
by Nandini madhu 19 Oct '14

19 Oct '14

Dear all, Greetings. I am working on Kamailio for Linux environment. But, i don't have any idea regarding usage of Kamailio in arm/embedded Linux environment. Please help me in this regard, how can i port Kamailio into an embedded linux device. Kindly advice. Regards Sermj2012

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pipelimit in 4.2.0 and beyond
by Alex Balashov 19 Oct '14

19 Oct '14

Hi, I was testing the new dynamic pipelimit instantiation functionality heralded in 4.2, and, strangely enough, it doesn't seem to work at all. Every time I call pl_check(), it just blocks the worker thread indefinitely, until I--rather quickly--run out of available workers. I've tried it with db_url (which I gathered to not be mandatory), without a db_url, using every algorithm possible, with a static pipe name, with a PV pipe name, etc. It all yields the same results. Here is my current employment of it: if($(xavp(call=>ingress_cps_limit){s.int}) > 0) { xlog("L_INFO", "DEBUG: PRE PL CHECK $ci\n"); if(!pl_check("1", "RED", "2")) { xlog("L_INFO", "DEBUG: AFTER PL CHECK $ci\n"); pl_drop(); xlog("L_INFO", "Ingress call setup rate limit for this BG " "$(xavp(call=>ingress_cps_limit){s.int}) has " "been exceeded; rejecting call\n"); exit; } xlog("L_INFO", "DEBUG: AFTER PL CHECK $ci\n"); } modparam("pipelimit", "hash_size", 12) modparam("pipelimit", "timer_interval", 5) modparam("pipelimit", "reply_code", 503) modparam("pipelimit", "reply_reason", "Call Setup Rate Exceeded") I never reach "AFTER PL CHECK". Here's what I actually want to do: if($(xavp(call=>ingress_cps_limit){s.int}) > 0) { if(!pl_check("$avp(trunk_id)", "RED", "$(xavp(call=>ingress_cps_limit){s.int})")) { pl_drop(); xlog("L_INFO", "!> " "Ingress call setup rate limit for this BG " "$(xavp(call=>ingress_cps_limit){s.int}) has " "been exceeded; rejecting call\n"); exit; } } I tried upping debug to debug=3 but get nothing insightful that pertains to pipelimit at all, as far as I can see. This is on a stock CentOS 6.5 install, sip-router:master:36e695453762b5ed15e5048fad2c559bb239dad5 compiled from source. Thanks! -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

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Kamailio - Asterisk Integration
by Bugaian A. Vitalie 18 Oct '14

18 Oct '14

Hello list, thanks for creating kamailio and good documentation for it. Also thanks to Daniel for this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I managed to integrate kamailio with asterisk and a2billing and I do have some small issues with some UA's. Generally all soft phones are OK but from some I can not generate calls. Especially it is true for Linksys SPA and Grandstream gxv3140. From 10 phones I have one with this issue, all config is the same on device side and on server/config side, but some are failing to generate calls. These UA will receive calls but will not be able to call. They are registered OK and I see these devices as all other ones in location table. One work around I have found is to update manually the cc_sip_buddy table(from asterisk realtime) with IP address and port and big part of these US's will start working. But still I have some that will not work even after that. What I see when capturing with ngrep is this: 1) Invite with SDP 2) 407 Proxy Authentication Required 3) ACK 4) Invite with SDP 5) 100 trying -- your call is important to us 6) 401 Unauthorized 7) ACK Not all captured calls have all 7 steps some of these drop at 4th step and I mention that all UA-s are showing OK as registered. I use sip.Server == "kamailio (4.1.6 (x86_64/linux))" ; Please suggest some steps to increase that authentication time(i suspect that these UA's do not have sufficient time to authenticate) and help me to understand why this is happening. Thank you in andvance. Vitalie Alex Bugaian.

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Kamailio 4.16 - IMS Registration Failure
by Kamal Palei 18 Oct '14

18 Oct '14

Dear All I have tried a lot and looks getting Kamailio IMS setup is very difficult and no proper documentation available. Though I have setup all CSCFs and FHoSS , still REGISTRATION fails with 483 response. I have setup DNS properly as shown below. *root@necs04:/etc/kamailio/pcscf# root@necs04:/etc/kamailio/pcscf# ping pcscf.group1-imslab11.in <http://pcscf.group1-imslab11.in>PING pcscf.group1-imslab11.in <http://pcscf.group1-imslab11.in> (192.168.1.100) 56(84) bytes of data.64 bytes from localhost (192.168.1.100): icmp_req=1 ttl=64 time=0.037 ms64 bytes from localhost (192.168.1.100): icmp_req=2 ttl=64 time=0.028 ms^C--- pcscf.group1-imslab11.in <http://pcscf.group1-imslab11.in> ping statistics ---2 packets transmitted, 2 received, 0% packet loss, time 1000msrtt min/avg/max/mdev = 0.028/0.032/0.037/0.007 msroot@necs04:/etc/kamailio/pcscf# root@necs04:/etc/kamailio/pcscf# root@necs04:/etc/kamailio/pcscf# root@necs04:/etc/kamailio/pcscf# ping icscf.group1-imslab11.in <http://icscf.group1-imslab11.in>PING icscf.group1-imslab11.in <http://icscf.group1-imslab11.in> (192.168.1.100) 56(84) bytes of data.64 bytes from localhost (192.168.1.100): icmp_req=1 ttl=64 time=0.023 ms64 bytes from localhost (192.168.1.100): icmp_req=2 ttl=64 time=0.029 ms64 bytes from localhost (192.168.1.100): icmp_req=3 ttl=64 time=0.024 ms^C--- icscf.group1-imslab11.in <http://icscf.group1-imslab11.in> ping statistics ---3 packets transmitted, 3 received, 0% packet loss, time 1999msrtt min/avg/max/mdev = 0.023/0.025/0.029/0.004 msroot@necs04:/etc/kamailio/pcscf# root@necs04:/etc/kamailio/pcscf# ping scscf.group1-imslab11.in <http://scscf.group1-imslab11.in>PING scscf.group1-imslab11.in <http://scscf.group1-imslab11.in> (192.168.1.100) 56(84) bytes of data.64 bytes from localhost (192.168.1.100): icmp_req=1 ttl=64 time=0.022 ms64 bytes from localhost (192.168.1.100): icmp_req=2 ttl=64 time=0.022 ms64 bytes from localhost (192.168.1.100): icmp_req=3 ttl=64 time=0.021 ms^C--- scscf.group1-imslab11.in <http://scscf.group1-imslab11.in> ping statistics ---3 packets transmitted, 3 received, 0% packet loss, time 1998msrtt min/avg/max/mdev = 0.021/0.021/0.022/0.005 msroot@necs04:/etc/kamailio/pcscf# * Also dig result is as shown below. *root@necs04:/etc/kamailio/pcscf# root@necs04:/etc/kamailio/pcscf# dig naptr group1-imslab11.in <http://group1-imslab11.in>; <<>> DiG 9.8.1-P1 <<>> naptr group1-imslab11.in <http://group1-imslab11.in>;; global options: +cmd;; Got answer:;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 10919;; flags: qr aa rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 1, ADDITIONAL: 3;; QUESTION SECTION:;group1-imslab11.in <http://group1-imslab11.in>. IN NAPTR;; ANSWER SECTION:group1-imslab11.in <http://group1-imslab11.in>. 86400 IN NAPTR 10 50 "s" "SIP+D2T" "" _sip._tcp.pcscf.group1-imslab11.in <http://tcp.pcscf.group1-imslab11.in>.group1-imslab11.in <http://group1-imslab11.in>. 86400 IN NAPTR 10 50 "s" "SIP+D2U" "" _sip._udp.pcscf.group1-imslab11.in <http://udp.pcscf.group1-imslab11.in>.;; AUTHORITY SECTION:group1-imslab11.in <http://group1-imslab11.in>. 86400 IN NS necs04.group1-imslab11.in <http://necs04.group1-imslab11.in>.;; ADDITIONAL SECTION:pcscf.group1-imslab11.in <http://pcscf.group1-imslab11.in>. 86400 IN A 192.168.1.100necs04.group1-imslab11.in <http://necs04.group1-imslab11.in>. 86400 IN A 192.168.1.100_sip._udp.pcscf.group1-imslab11.in <http://udp.pcscf.group1-imslab11.in>. 86400 IN SRV 0 0 4060 pcscf.group1-imslab11.in <http://pcscf.group1-imslab11.in>.;; Query time: 0 msec;; SERVER: 192.168.1.100#53(192.168.1.100);; WHEN: Fri Oct 17 17:34:22 2014;; MSG SIZE rcvd: 259root@necs04:/etc/kamailio/pcscf# * When I send a REGISTER message from a sip client, I get 483 Too Many Hops. I am totally clueless whats going wrong. Can somebody guide me , where is the fault. On thing I do not understand is dispatcher.list stuff. So I have kept empty disptacher.list file for PCSCF and SCSCF. Can that be an issue. The REGISTER and 483 response looks as below. *REGISTER sip:pcscf.group1-imslab11.in <http://pcscf.group1-imslab11.in> SIP/2.0Via: SIP/2.0/TCP 192.168.1.114:49732;branch=z9hG4bK457115572From: <sip:necs_0@group1-imslab11.in <sip%3Anecs_0(a)group1-imslab11.in>>;tag=1068512852To: <sip:necs_0@group1-imslab11.in <sip%3Anecs_0(a)group1-imslab11.in>>Call-ID: 1874336526CSeq: 16 REGISTERContact: <sip:necs_0@192.168.1.114:49732;transport=TCP;line=a126427bb8e82c9>;+sip.instance="<urn:uuid:2dab9970-55f5-11e4-ae19-41032d8f6492>"Authorization: Digest username="necs_0(a)group1-imslab11.in <necs_0(a)group1-imslab11.in>", realm="pcscf.group1-imslab11.in <http://pcscf.group1-imslab11.in>", nonce=" ", uri="sip:pcscf.group1-imslab11.in <http://pcscf.group1-imslab11.in>", response=" "Max-Forwards: 70User-Agent: NECS IMS client v1.01Expires: 600000Supported: pathSupported: gruuContent-Length: 0* *SIP/2.0 483 Too Many HopsVia: SIP/2.0/TCP 192.168.1.114:49732;branch=z9hG4bK457115572From: <sip:necs_0@group1-imslab11.in <sip%3Anecs_0(a)group1-imslab11.in>>;tag=1068512852To: <sip:necs_0@group1-imslab11.in <sip%3Anecs_0(a)group1-imslab11.in>>;tag=68c95b6520b8472128534dfbca9fcb81.d739Call-ID: 1874336526CSeq: 16 REGISTERServer: Kamailio P-CSCFContent-Length: 0* Kindly somebody help me how I can troubleshoot and fix it. Many thanks Kamal

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Re: [SR-Users] Configuring TLS and WSS with Kamailio
by dodul 18 Oct '14

18 Oct '14

Hi Thanks a lot for your reply. I'll check the cn and let you know. In my configuration I'm listening to tcp port but no tls port? Can that be the reason? Otherthan that if it was rejected because of the certificate shouldn't I get some errors in the log? Every time I'm trying it's getting request timed out. Sent from my Samsung Galaxy smartphone. -------- Original message -------- From: Gonzalo Gasca <gascagonzalo(a)gmail.com> Date:10-17-2014 19:33 (GMT-06:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: ben(a)langfeld.me Subject: Re: [SR-Users] Configuring TLS and WSS with Kamailio Hi Ben, In regards your certificates, please double check the following: a) CN field: In your webrtc client check the URL used for wss. and verify it matches the CN field of the certificate installed in kamailio: Example: In my sipml5 client I configured wss://ramenlabs.io:5063 and my certificate in fact contains in the cn field ramenlabs.io Subject: OU=Domain Control Validated, OU=PositiveSSL, CN=ramenlabs.io openssl x509 -in /etc/certs/sip.192.168.146.133/cert.pem -noout -text b) I have successfully configured Kamailio 4.1 with TLS and WSS using TLS port 5063 Topology: 1) sipml5 --wss--> ngnix -wss-> kamailio (sip registrar) 2) sipml5 --wss---> kamailio Relevant configurations: kamailio.cfg tls.cfg certificates Kamailio: loading modules under /usr/lib/x86_64-linux-gnu/kamailio/modules/ Listening on udp: 172.31.27.85:5060 tcp: 172.31.27.85:5060 tcp: 172.31.27.85:5062 tls: 172.31.27.85:5061 * tls: 172.31.27.85:5063 <http://172.31.27.85:5063>* Aliases: * tls: ip-172-31-27-85.us-west-2.compute.internal:5063* tls: ip-172-31-27-85.us-west-2.compute.internal:5061 tcp: ip-172-31-27-85.us-west-2.compute.internal:5062 tcp: ip-172-31-27-85.us-west-2.compute.internal:5060 udp: ip-172-31-27-85.us-west-2.compute.internal:5060 *: ramenlabs.io:* *: 172.31.27.85:* kamailio.cfg tls.cfg https://github.com/spicyramen/llamato/blob/LlamatoReg/kamailio.cfg https://github.com/spicyramen/llamato/blob/LlamatoReg/tls.cfg openssl s_client -connect 172.31.27.85:5063 where this i my internal IP address I get presented the certificates. HTH -G On Fri, Oct 17, 2014 at 5:10 PM, Kamrul Khan <dodul(a)live.com> wrote: > ok. I have made some changes. rather than getting the TLS configaration > from a file I added this lines: > > #!ifdef WITH_TLS > # ----- tls params ----- > > modparam("tls", "private_key", "/etc/certs/sip.192.168.146.133/key.pem") > modparam("tls", "certificate", "/etc/certs/sip.192.168.146.133/cert.pem") > modparam("tls", "ca_list", "/etc/certs/demoCA/cert.pem") > > now, Im getting different logs which looks good. Getting positive results > from openssl test, > > openssl s_client -connect 192.168.146.133:5061 -tls1 > CONNECTED(00000003) > ^C > > But when I try to connect using my webRTC client or even from web-browsers > im getting timed out. I think im close .. please help me fixing this issue. > > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:275]: fill_missing(): TLSs<default>: tls_method=9 > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:287]: fill_missing(): TLSs<default>: > certificate='/etc/certs/sip.192.168.146.133/cert.pem' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:294]: fill_missing(): TLSs<default>: > ca_list='/etc/certs/demoCA/cert.pem' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:301]: fill_missing(): TLSs<default>: crl='(null)' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:305]: fill_missing(): TLSs<default>: > require_certificate=0 > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:312]: fill_missing(): TLSs<default>: > cipher_list='(null)' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:319]: fill_missing(): TLSs<default>: > private_key='/etc/certs/sip.192.168.146.133/key.pem' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:323]: fill_missing(): TLSs<default>: > verify_certificate=0 > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:326]: fill_missing(): TLSs<default>: verify_depth=9 > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:670]: set_verification(): TLSs<default>: No client > certificate required and no checks performed > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:275]: fill_missing(): TLSc<default>: tls_method=9 > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:287]: fill_missing(): TLSc<default>: > certificate='/etc/certs/sip.192.168.146.133/cert.pem' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:294]: fill_missing(): TLSc<default>: > ca_list='/etc/certs/demoCA/cert.pem' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:301]: fill_missing(): TLSc<default>: crl='(null)' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:305]: fill_missing(): TLSc<default>: > require_certificate=0 > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:312]: fill_missing(): TLSc<default>: > cipher_list='(null)' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:319]: fill_missing(): TLSc<default>: > private_key='/etc/certs/sip.192.168.146.133/key.pem' > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:323]: fill_missing(): TLSc<default>: > verify_certificate=0 > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:326]: fill_missing(): TLSc<default>: verify_depth=9 > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12123]: > INFO: tls [tls_domain.c:673]: set_verification(): TLSc<default>: Server MAY > present invalid certificate > Oct 17 16:35:38 ubuntu /usr/local/kamailio-devel//sbin/kamailio[12136]: > INFO: ctl [io_listener.c:225]: io_listen_loop(): io_listen_loop: using > epoll_lt io watch method (config) > > > ------------------------------ > From: dodul(a)live.com > To: ben(a)langfeld.me; sr-users(a)lists.sip-router.org > Date: Sat, 18 Oct 2014 05:08:20 +0600 > > Subject: Re: [SR-Users] Configuring TLS and WSS with Kamailio > > I dint find any in my installation directory. probably it doesn't come > with standard installation unless you explicitly mention for it... not sure > though > > ------------------------------ > Date: Fri, 17 Oct 2014 19:56:50 -0300 > From: ben(a)langfeld.co.uk > To: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Configuring TLS and WSS with Kamailio > > Out of curiosity, why don't you use the pre-built packaged binaries > instead of building from source? > > On 17 October 2014 19:53, Kamrul Khan <dodul(a)live.com> wrote: > > Hi, > > > Im trying to connect my WebRTC clietn to kamailio via WSS. I > successfully connected it via WS but having trouble connecting it via WSS. > My first issue is I get error messages while i try to compile TLS > module(console log in the end of the document). But, still it creates > tls.so file. So i copied the tls.so to my kamailio modules directory and > then updated my kamailio configuration as below: > > > #!define WITH_TLS > > . > > . > > listen=tcp:127.0.0.1:5061 > > listen=tcp:192.168.146.133:5061 > > . > > . > > #!ifdef WITH_TLS > > enable_tls=yes > > #!endif > > . > > . > > #!ifdef WITH_TLS > > # ----- tls params ----- > > modparam("tls", "config", "/usr/local/kamailio-devel/etc/kamailio/tls.cfg") > > #!endif > > > Now, according to this mailing list, > http://lists.sip-router.org/pipermail/sr-users/2013-March/077182.html : “*when > tls module is installed, a self signed pair of certificate-private key is > generated in /usr/local/etc/kamailio”* > > > In my case, I dont even have the /usr/local/etc/kamailio directory. So, > I followed > http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates to > create my certificate and key. And updated my tsl.cfg, this is how my > tsl.cfg looks like: > > > [server:192.168.146.133:5061] > > method = TLSv1 > > verify_certificate = no > > require_certificate = no > > > private_key = /etc/certs/sip.192.168.146.133/key.pem > > certificate = /etc/certs/sip.192.168.146.133/cert.pem > > ca_list = /etc/certs/demoCA/cert.pem > > > [client:192.168.146.133:5061] > > verify_certificate = yes > > require_certificate = yes > > > > Then I restarted my kamailio server. And i get the following errors in > my kamailio log: > > > Im not sure what wrong im doing. Please help me: > > > Oct 17 15:44:50 ubuntu kamailio: INFO: tls [tls_init.c:385]: > init_tls_compression(): tls: init_tls: disabling compression... > > Oct 17 15:44:50 ubuntu kamailio: WARNING: <core> [socket_info.c:1397]: > fix_hostname(): WARNING: fix_hostname: could not rev. resolve > 192.168.146.133 > > Oct 17 15:44:50 ubuntu kamailio: message repeated 2 times: [ WARNING: > <core> [socket_info.c:1397]: fix_hostname(): WARNING: fix_hostname: could > not rev. resolve 192.168.146.133] > > Oct 17 15:44:50 ubuntu kamailio: INFO: <core> [tcp_main.c:4836]: > init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected) > > Oct 17 15:44:50 ubuntu kamailio: WARNING: <core> [daemonize.c:352]: > daemonize(): pid file contains old pid, replacing pid > > Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: > INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob > > Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: > INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available > > Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: > INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 512 > > Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: > INFO: utils [utils.c:288]: mod_init(): forward functionality disabled > > Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: > INFO: utils [utils.c:197]: pres_db_init(): xcap_auth_status function is > disabled > > Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: > ERROR: tls [tls_init.c:668]: tls_check_sockets(): TLSs< > 192.168.146.133:5061>: No listening socket found > > Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: > ERROR: <core> [sr_module.c:970]: init_mod(): init_mod(): Error while > initializing module tls > (/usr/local/kamailio-devel/lib64/kamailio/modules/tls.so) > > Oct 17 15:44:50 ubuntu /usr/local/kamailio-devel//sbin/kamailio[11013]: : > tls [tls_locking.c:103]: locking_f(): BUG: tls: locking_f (callback): > invalid lock number: 30 (range 0 - 0), called from eng_table.c:227 > > Oct 17 15:44:51 ubuntu kamailio: ERROR: <core> [daemonize.c:307]: > daemonize(): Main process exited before writing to pipe > > > tls module compile log > > > xxx@ubuntu:/usr/local/src/kamailio-4.1/kamailio$ sudo make -C modules/tls > make: Entering directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls' > make: Leaving directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls' > make: Entering directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls' > CC (gcc) [M tls.so] tls_bio.o > CC (gcc) [M tls.so] tls_cfg.o > CC (gcc) [M tls.so] tls_config.o > CC (gcc) [M tls.so] tls_ct_wrq.o > CC (gcc) [M tls.so] tls_domain.o > In file included from tls_domain.c:39:0: > tls_domain.c: In function âload_certâ: > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_domain.c:506:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR("load_cert:"); > ^ > tls_domain.c: In function âload_ca_listâ: > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_domain.c:536:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR("load_ca_list:"); > ^ > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_domain.c:543:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR("load_ca_list:"); > ^ > tls_domain.c: In function âload_crlâ: > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_domain.c:575:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR("load_crl:"); > ^ > tls_domain.c: In function âload_private_keyâ: > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_domain.c:990:5: note: in expansion of macro âTLS_ERRâ > TLS_ERR("load_private_key:"); > ^ > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_domain.c:998:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR("load_private_key:"); > ^ > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_domain.c:1005:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR("load_private_key:"); > ^ > CC (gcc) [M tls.so] tls_dump_vf.o > CC (gcc) [M tls.so] tls_init.o > CC (gcc) [M tls.so] tls_locking.o > CC (gcc) [M tls.so] tls_mod.o > CC (gcc) [M tls.so] tls_rpc.o > CC (gcc) [M tls.so] tls_select.o > CC (gcc) [M tls.so] tls_server.o > In file included from tls_server.c:52:0: > tls_server.c: In function âtls_complete_initâ: > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_server.c:192:3: note: in expansion of macro âTLS_ERRâ > TLS_ERR("Failed to create SSL or BIO structure:"); > ^ > tls_server.c: In function âtls_shutdownâ: > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_server.c:521:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR("SSL error:"); > ^ > tls_server.c: In function âtls_encode_fâ: > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_server.c:798:5: note: in expansion of macro âTLS_ERRâ > TLS_ERR(err_src); > ^ > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_server.c:832:5: note: in expansion of macro âTLS_ERRâ > TLS_ERR(err_src); > ^ > tls_server.c: In function âtls_read_fâ: > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_server.c:1186:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR(err_src); > ^ > tls_util.h:52:6: warning: variable âretâ set but not used > [-Wunused-but-set-variable] > int ret; \ > ^ > tls_server.c:1220:4: note: in expansion of macro âTLS_ERRâ > TLS_ERR(err_src); > ^ > CC (gcc) [M tls.so] tls_util.o > CC (gcc) [M tls.so] tls_verify.o > LD (gcc) [M tls.so] tls.so > make: Leaving directory `/usr/local/src/kamailio-4.1/kamailio/modules/tls' > > _______________________________________________ > 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 > > > > _______________________________________________ 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 > > _______________________________________________ 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 > > _______________________________________________ > 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 > > _______________________________________________ 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

What is Attribute Value Pair (AVP) ?
by Mahmoud Ramadan Ali 18 Oct '14

18 Oct '14

Hiiii everyone, Forgive me for such simple question but i'm newbie to Kamailio. i was reading about the dispatcher module http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html#dispatch… and i want someone to explain to me in a simple way what is AVP , for example the AVP in the *modparam("dispatcher", "grp_avp", "$avp(dsgrp)")* and what the "dsgrp" represents and how to configure the AVP values ... Kinda Regards Thanks in advance...

1 0

Max serial forks
by Muhammad Shahzad 17 Oct '14

17 Oct '14

Hi, After some testing it appears that kamailio can do maximum 12 serial forks using transaction manager (tm module). Is it possible to make it configurable in kamailio.cfg or at least increase it with static higher value (e.g. 15)? Thank you.

3 6

No need for Authentication... Why?
by Manuel Camarg 17 Oct '14

17 Oct '14

>It is a matter of config file, mainly the route[AUTH] -- remove the if >condition about what has to be authenticated and authenticate all the >requests. > >Cheers, >Daniel Perfect, solved the issue :) Regards Manuel

1 0

Auth problem with Jitsi on current kamailio stable
by flow27＠meta-matrix.de 17 Oct '14

17 Oct '14

Hi, I have installed kamailio stable on Debian Wheezy from the repository, followed the main tutorial and pretty much everything looked like it was working at first. I can connect to my sip server but when I try to log in I keep getting a failed login. In the mysql database I have the correct username and domain in the subscriber table. No typos there so then I enabled logging with debug=3 to check the logs but I am not sure how to interprete what I see. It seems like there 2 two login procedures. The first one seems to fail, the second one seems to work since I see a "check_response: Authorization is OK" in there but still on Jitsi I keep getting told my password is incorrect. Here is the log dump of both login procedures from the same login attempt (I only tried to login once and as you see on the timestamp it all happens within the same second) This is the first attempt where the realm is not found even though it is exactly the domain I have in the subscriber mysql table: Oct 17 08:21:17 video /usr/sbin/kamailio[5170]: DEBUG: auth_db [authorize.c:486]: auth_check(): realm [video.mydomain.de] table [subscriber] flags [1] Oct 17 08:21:17 video /usr/sbin/kamailio[5170]: DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm 'video.mydomain.de' not found Oct 17 08:21:17 video /usr/sbin/kamailio[5170]: DEBUG: auth_db [authorize.c:252]: digest_authenticate_hdr(): no credentials Oct 17 08:21:17 video /usr/sbin/kamailio[5170]: DEBUG: auth [challenge.c:127]: get_challenge_hf(): build_challenge_hf: realm='video.mydomain.de' Oct 17 08:21:17 video /usr/sbin/kamailio[5170]: DEBUG: auth [challenge.c:269]: get_challenge_hf(): auth: 'WWW-Authenticate: Digest realm="video.mydomain.de", nonce="VEEK6VRBCb32GLPSTiKlfAo4loEn7BTY"#015#012' Oct 17 08:21:17 video /usr/sbin/kamailio[5170]: DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) Oct 17 08:21:17 video /usr/sbin/kamailio[5170]: DEBUG: <core> [msg_translator.c:204]: check_via_address(): check_via_address(79.212.239.151, 192.168.2.32, 0) Oct 17 08:21:17 video /usr/sbin/kamailio[5170]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list 0x7f2f830daa10 Then a second login attempt takes place where it looks different: Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <REGISTER> Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:video.mydomain.de:5188> Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <REGISTER> Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=10 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [46]; uri=[sip:oliver2@video.mydomain.de] Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body ["oliver2" <sip:oliver2@video.mydomain.de>#015#012] Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-383534-d21acd094f094ec7f76ca01734bb9c80>; state=16 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [receive.c:193]: receive_msg(): preparing to run routing scripts... Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=f59174d Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: siputils [checks.c:103]: has_totag(): no totag Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=(nil) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=60350, isACK=0 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: auth_db [authorize.c:486]: auth_check(): realm [video.mydomain.de] table [subscriber] flags [1] Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: auth [api.c:96]: pre_auth(): auth: digest-algo: parsed value: 0 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:118]: db_new_result(): allocate 56 bytes for result set at 0x7f2f8d09c040 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: db_mysql [km_res.c:68]: db_mysql_get_columns(): 1 columns returned from the query Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:155]: db_allocate_columns(): allocate 8 bytes for result names at 0x7f2f8d09c148 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 4 bytes for result types at 0x7f2f8d09c0d8 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: db_mysql [km_res.c:86]: db_mysql_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7f2f8d0aa448 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: db_mysql [km_res.c:93]: db_mysql_get_columns(): RES_NAMES(0x7f2f8d0aa448)[0]=[password] Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: db_mysql [km_res.c:137]: db_mysql_get_columns(): use DB1_STRING result type Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:184]: db_allocate_rows(): allocate 16 bytes for rows at 0x7f2f8d09c288 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_row.c:119]: db_allocate_row(): allocate 32 bytes for row values at 0x7f2f8d09e480 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_val.c:117]: db_str2val(): converting STRING [xxxxxxx] Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: auth_db [authorize.c:196]: get_ha1(): HA1 string calculated: a4fe9f8e75b2cb7b19da2cbc37b33bcc Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: auth [api.c:211]: auth_check_response(): check_response: Our result = '1aab3cbf895ec5c2491135a0a6869f4e' Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: auth [api.c:218]: auth_check_response(): check_response: Authorization is OK Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:81]: db_free_columns(): freeing 1 columns Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:85]: db_free_columns(): freeing RES_NAMES[0] at 0x7f2f8d0aa448 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:94]: db_free_columns(): freeing result names at 0x7f2f8d09c148 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:99]: db_free_columns(): freeing result types at 0x7f2f8d09c0d8 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:54]: db_free_rows(): freeing 1 rows Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_row.c:97]: db_free_row(): freeing row values at 0x7f2f8d09e480 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:62]: db_free_rows(): freeing rows at 0x7f2f8d09c288 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [db_res.c:136]: db_free_result(): freeing result set at 0x7f2f8d09c040 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: auth [challenge.c:127]: get_challenge_hf(): build_challenge_hf: realm='video.mydomain.de' Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: auth [challenge.c:269]: get_challenge_hf(): auth: 'WWW-Authenticate: Digest realm="video.mydomain.de", nonce="VEEK6VRBCb32GLPSTiKlfAo4loEn7BTY"#015#012' Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [msg_translator.c:204]: check_via_address(): check_via_address(79.212.239.151, 192.168.2.32, 0) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list 0x7f2f830daa10 Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [usr_avp.c:644]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil) Oct 17 08:21:17 video /usr/sbin/kamailio[5172]: DEBUG: <core> [receive.c:296]: receive_msg(): receive_msg: cleaning up Any help on what I did wrong is appreciated... Best regards Oliver

1 0

New: max branches configurable via parameter
by Daniel-Constantin Mierla 17 Oct '14

17 Oct '14

Hello, being brought in discussions many times, including in the recent past days, I replaced the static define of max branches limit with a core parameter. It impacts the destination set stored in core and the uac fields stored in transaction. It can be set now with global parameter: max_branches The old limit was 12, now it can be set in the range of 1 to 31. Few aspects that I would like to discuss to get this refactored to fit the best of common scenarios: 1) the 31 upper limit comes from tm cancelled branches bitmask, which now is stored in an integer. Should be easy to get it to 63 (planned after some testing that proves the concept used now is ok). Would it be a need for even more? 2) would it make sense to specify the max number of branches per transaction, in config, before creating the transaction? The upper limit will be the max_branches value from the core. 3) thinking of common cases of what can be forked a lot, I thought that we can a simplification of 2) by specifying two limits: one for initial requests which are very likely to have many branches (think of initial INVITE via LCR or location) and one for requests within dialog which are likely to have one or very few branches (e.g., replicating BYE to a peer server). Opinions? The main benefit for 2) and 3) would be less memory usage. Testing and feedback of the new max_branches parameter is very appreciated. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

5 4

Kamailio v4.2.0 RPMs and updates to rpm.kamailio.org
by Daniel-Constantin Mierla 17 Oct '14

17 Oct '14

Hello, the RPMs for Kamailio v4.2.0 were generated for Centos (5, 6, 7), Fedora (18, 19, 20), OpenSuse (12.3, 13.1, 13.2) and RedHat (5, 6, 7). Packages are made with OpenSuse build service, but as different build project -- see the details at: - http://www.kamailio.org/wiki/packages/rpms The change was made to have a build project for each release series from now on, making easier to manage and mirror. Given this, the mirror at rpm.kamailio.org has been updated as well. If you used it for 4.1.x or older releases, you have to upgrade the repo path, see: - http://rpm.kamailio.org And again a call for people using RPMs: join the efforts in packaging and managing there repos, we have the infrastructure now, but we are short in time/human resources. If you want to keep the packages up to date, come on board and help a bit. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Mozilla Firefox + SIP Phone voice not established
by Amit Patkar 17 Oct '14

17 Oct '14

Hi I have configured latest Kamailio 4.2.0 + Rtpengine 3.3.0.0+0~mr3.6.0.0 There are 2 SIP clients. Once client is registered as WebRTC client from Mozilla Firefox browser 29.0.1 and second SIP client is desktop SIP phone - PortGo Softphone (portsip.com) When I am trying to call softphone from Firefox, call is getting established, where as voice is not established. RTP is not flowing between Browser and Softphone. Same is working with Google Chrome. Following command is used for negotiation rtpengine_manage("replace-origin replace-session-connection ICE=force"); Log is as follows Oct 16 18:39:14 kamailio rtpengine[28613]: Got valid command from 127.0.0.1:60471: offer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-29.0.1 3618 1 IN IP4 0.0.0.0#015#012s=Doubango Telecom - firefox#015#012t=0 0#015#012a=ice-ufrag:6e5ff262#015#012a=ice-pwd:21a7dd42eb50ab711d7466a39f4994fe#015#012a=fingerprint:sha-256 F2:B7:D6:64:90:66:D7:33:B1:04:F7:EC:8D:78:A1:9F:1D:E1:59:92:31:64:98:D3:6D:A7:DF:60:C9:DA:21:AE#015#012m=audio 3779 UDP/TLS/RTP/SAVPF 109 0 8 101#015#012c=IN IP4 49.248.118.98#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telep ... Oct 16 18:39:14 kamailio rtpengine[28613]: ... hone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=setup:actpass#015#012a=candidate:0 1 UDP 2128609535 192.168.2.18 3779 typ host#015#012a=candidate:1 1 UDP 1692467199 49.248.118.98 3779 typ srflx raddr 192.168.2.18 rport 3779#015#012a=candidate:0 2 UDP 2128609534 192.168.2.18 3780 typ host#015#012a=candidate:1 2 UDP 1692467198 49.248.118.98 3780 typ srflx raddr 192.168.2.18 rport 3780#015#012a=rtcp-mux#015#012", "ICE": "force", "replace": [ "origin", "session-connection" ], "call-id": "1b211056-eb41-7edf-59b5-fa2f51161f32", "receive ... Oct 16 18:39:14 kamailio rtpengine[28613]: ... d-from": [ "IP4", "192.168.2.18" ], "from-tag": "R3okhRsDImBAiNMCZ04e", "command": "offer" } Oct 16 18:39:14 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] Creating new call Oct 16 18:39:14 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] Returning to SIP proxy: d3:sdp735:v=0#015#012o=Mozilla-SIPUA-29.0.1 3618 1 IN IP4 0.0.0.0#015#012s=Doubango Telecom - firefox#015#012t=0 0#015#012a=ice-lite#015#012m=audio 30428 UDP/TLS/RTP/SAVPF 109 0 8 101#015#012c=IN IP4 192.168.7.78#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:30429#015#012a=rtcp-mux#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:X1DlmZvOXeVYgcPNBeiDT15PjuDw01QRjS6K3q2l#015#012a=setup:actpass#015#012a=fingerprint:sha-1 1A:A ... Oct 16 18:39:14 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] ... D:10:CA:48:BC:06:5F:BA:77:A5:72:5B:A8:4F:6B:05:68:03:C2#015#012a=ice-ufrag:pfDmNOv7#015#012a=ice-pwd:y7T3NQJ5BBwHUqtzSjKjk87JmK47#015#012a=candidate:WudoNuJD1FjBzlYN 1 UDP 2130706431 192.168.7.78 30428 typ host#015#012a=candidate:WudoNuJD1FjBzlYN 2 UDP 2130706430 192.168.7.78 30429 typ host#015#0126:result2:oke Oct 16 18:39:15 kamailio rtpengine[28613]: Got valid command from 127.0.0.1:60147: answer - { "sdp": "v=0#015#012o=- 29112514 29112514 IN IP4 192.168.2.18#015#012s=http://www.portsip.com#015#012c=IN IP4 192.168.2.18#015#012t=0 0#015#012m=audio 21666 RTP/SAVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:YBXJIjofPfzbpUUXyR5gPrQi7SXOu1uwCXGvmKfy#015#012a=sendrecv#015#012", "ICE": "force", "replace": [ "origin", "session-connection" ], "call-id": "1b211056-eb41-7edf-59b5-fa2f51161f32", "received-from": [ " ... Oct 16 18:39:15 kamailio rtpengine[28613]: ... IP4", "192.168.2.18" ], "from-tag": "R3okhRsDImBAiNMCZ04e", "to-tag": "2c76015a", "command": "answer" } Oct 16 18:39:15 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] Returning to SIP proxy: d3:sdp555:v=0#015#012o=- 29112514 29112514 IN IP4 192.168.7.78#015#012s=http://www.portsip.com#015#012c=IN IP4 192.168.7.78#015#012t=0 0#015#012a=ice-lite#015#012m=audio 30438 RTP/SAVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:30439#015#012a=setup:active#015#012a=fingerprint:sha-1 1A:AD:10:CA:48:BC:06:5F:BA:77:A5:72:5B:A8:4F:6B:05:68:03:C2#015#012a=ice-ufrag:xzf5G88u#015#012a=ice-pwd:WPxvr1djpERiXBBgHJX3r1bYBo3f#015#012a=candidate:WudoNuJD1FjBzlYN 1 UDP 2130706431 192.168.7.78 30 ... Oct 16 18:39:15 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] ... 438 typ host#015#012a=candidate:WudoNuJD1FjBzlYN 2 UDP 2130706430 192.168.7.78 30439 typ host#015#0126:result2:oke Oct 16 18:39:15 kamailio /usr/local/sbin/kamailio[28806]: NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1413464955;method=INVITE;from_tag=R3okhRsDImBAiNMCZ04e;to_tag=2c76015a;call_id=1b211056-eb41-7edf-59b5-fa2f51161f32;code=200;reason=OK;src_user=5011;src_domain=192.168.7.78;src_ip=192.168.2.18;dst_ouser=9002;dst_user=9002;dst_domain=192.168.2.18 Oct 16 18:39:15 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32 port 30429] SRTCP output wanted, but no crypto suite was negotiated Oct 16 18:39:16 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32 port 30428] SRTP output wanted, but no crypto suite was negotiated Oct 16 18:39:19 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32 port 30428] Confirmed peer address as 192.168.2.18:21666 Oct 16 18:39:20 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32 port 30429] Confirmed peer address as 192.168.2.18:21667 Oct 16 18:39:20 kamailio rtpengine[28613]: Got valid command from 127.0.0.1:36626: delete - { "ICE": "force", "call-id": "1b211056-eb41-7edf-59b5-fa2f51161f32", "received-from": [ "IP4", "192.168.2.18" ], "from-tag": "2c76015a", "command": "delete" } Oct 16 18:39:20 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] Scheduling deletion of call branch '2c76015a' in 30 seconds Oct 16 18:39:20 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] Returning to SIP proxy: d7:createdi1413464954e11:last signali1413464955e4:tagsd8:2c76015ad3:tag8:2c76015a7:createdi1413464954e16:in dialogue with20:R3okhRsDImBAiNMCZ04e6:mediasld5:indexi1e4:type5:audio8:protocol8:RTP/SAVP7:streamsld10:local porti30428e8:endpointd6:family4:IPv47:address12:192.168.2.184:porti21666ee19:advertised endpointd6:family4:IPv47:address12:192.168.2.184:porti21666ee12:crypto suite23:AES_CM_128_HMAC_SHA1_8011:last packeti1413464960e5:flagsl3:RTP6:filled9:confirmede5:statsd7: ... Oct 16 18:39:20 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] ... packetsi228e5:bytesi39216e6:errorsi0eeed10:local porti30429e8:endpointd6:family4:IPv47:address12:192.168.2.184:porti21667ee19:advertised endpointd6:family4:IPv47:address12:192.168.2.184:porti21667ee12:crypto suite23:AES_CM_128_HMAC_SHA1_8011:last packeti1413464960e5:flagsl4:RTCP6:filled9:confirmede5:statsd7:packetsi2e5:bytesi56e6:errorsi0eeee5:flagsl11:initialized4:SDESeeee20:R3okhRsDImBAiNMCZ04ed3:tag20:R3okhRsDImBAiNMCZ04e7:createdi1413464954e16:in dialogue with8:2c76015a6:mediasld5:indexi1e4: ... Oct 16 18:39:20 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] ... type5:audio8:protocol8:RTP/SAVP7:streamsld10:local porti30438e8:endpointd6:family4:IPv47:address12:192.168.2.184:porti3779ee19:advertised endpointd6:family4:IPv47:address12:192.168.2.184:porti3779ee11:last packeti1413464954e5:flagsl3:RTP6:fillede5:statsd7:packetsi0e5:bytesi0e6:errorsi0eeed10:local porti30439e8:endpointd6:family4:IPv67:address2:::4:porti0ee19:advertised endpointd6:family4:IPv67:address2:::4:porti0ee11:last packeti1413464954e5:flagsl4:RTCP6:fillede5:statsd7:packetsi0e5:bytesi0e6:e ... Oct 16 18:39:20 kamailio rtpengine[28613]: [1b211056-eb41-7edf-59b5-fa2f51161f32] ... rrorsi0eeee5:flagsl11:initialized9:DTLS-SRTP3:ICEeeeee6:totalsd3:RTPd7:packetsi228e5:bytesi39216e6:errorsi0ee4:RTCPd7:packetsi2e5:bytesi56e6:errorsi0eee6:result2:oke Oct 16 18:39:20 kamailio /usr/local/sbin/kamailio[28815]: NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1413464960;method=BYE;from_tag=2c76015a;to_tag=R3okhRsDImBAiNMCZ04e;call_id=1b211056-eb41-7edf-59b5-fa2f51161f32;code=200;reason=OK;src_user=9002;src_domain=192.168.7.78;src_ip=192.168.2.18;dst_ouser=5011;dst_user=5011;dst_domain=df7jal23ls0d.invalid Another issue is for reverse scenario. Softphone calling WebRTC client. This option does not work with Mozilla & Chrome. Please help. Am I missing something? Do I need to modify configuration to make this work? -- *Thanks & Regards,* Amit Patkar

1 0

Kamailio v4.2.0 Released - new major version is out
by Daniel-Constantin Mierla 16 Oct '14

16 Oct '14

Kamailio v4.2.0 is out – a new major release, bringing out a consistent set of new features and improvements. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v4-2-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.2.0! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

weight for rtpproxy_sock (string)
by george wu 16 Oct '14

16 Oct '14

How can I specify weight for socket? can I get an example? How to set weight for each socket below? thanks. http://kamailio.org/docs/modules/stable/modules/rtpproxy.html Sockets can be weighted by adding '=#' to a socket where # is an integer. A socket with a weight of 2 will be chosen twice as often as one with a weight of 1. # multiple sets of multiple rtproxies modparam("rtpproxy", "rtpproxy_sock", "1 == udp:localhost:12221 udp:localhost:12222") modparam("rtpproxy", "rtpproxy_sock", "2 == udp:localhost:12225")

1 0

Registrar module and local paths
by Ben Langfeld 16 Oct '14

16 Oct '14

Hello everyone. I'm having a problem with registrations and Path and hope that someone can help. I have a Kamailio instance which is both a WebSocket proxy and a registrar which has the following behaviour: * A REGISTER with a Path header already on it gets recorded in the location table. * I use add_path() on all incoming REGISTERs, and a REGISTER relayed to another registrar reaches that registrar with a Path header added appropriately. Unfortunately, save("location") ignores the path added by add_path(). I'd prefer to keep the proxy and registrar co-located for the short term. Does anyone know if/how I might be able to have the Path header properly stored in the case that it was added by add_path()? Regards, Ben Langfeld

5 8

Kamailio-4.1.6 IMS Setup - Registration Issue
by Kamal Palei 16 Oct '14

16 Oct '14

Dear All I have setup a Kamailio proxy for testing IMS call flows. I have configured as per help provided. On registration, I get 483 response from server. When I searched it in google and help, it was suggested to set the alias name properly. I set the alias name properly. Still I get 483 from Kamailio (used as IMS server) Can somebody help me with what are the possible reasons for this. Thanks kamal

2 3

textopsx issue when modifying same header several times
by Andrew Pogrebennyk 16 Oct '14

16 Oct '14

Hi, we are filtering some method names from the Allow header with kamailio 4.1: depending in the configuration: if (hf_value_exists("Allow", "INFO")) { xlog("L_INFO", "Remove INFO from Allow\n"); exclude_hf_value("Allow", "INFO"); } if (hf_value_exists("Allow", "REFER")) { xlog("L_INFO", "Remove REFER from Allow\n"); exclude_hf_value("Allow", "REFER"); } There is a problem in case multiple rewrites are done on the same header so the resulting message is going to be broken: Input: Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, INFO, REFER OUTPUT: Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, PRACK, The trailing comma is not allowed. So we have to use msg_apply_changes(). But kamailio is quite restrictive on where you can call msg_apply_changes(): 1. it must be done before record_route()* and 2. it is not allowed in the branch route. As for me it's valid use case when you need to do some header manipulations only for some of the branches, isn't? *(1) is further aggravated by the fact that the record_route() for loose-routed requests is usually called quite early in the processing pipeline. Imagine kamailio.cfg would look like: request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # ---> filter Allow route(FILTER_ALLOW_METHODS); # handle requests within SIP dialogs route(WITHINDLG); I don't like the idea of changing the message before loose-routing. Taking (1) and (2) into account, would it be possible to a) fix exclude_hf_value to allow to filter multiple values in a single pass and b) allow msg_apply_changes() in branch route without any ill effects on the record-route headers? Please share your ideas. Thanks, Andrew

1 0

XCAP Jitsi
by Abdul Gafar 16 Oct '14

16 Oct '14

We configure use XCAP in version: kamailio 4.1.6 (x86_64/linux) Error in SIP Client (Jitsi) Resource cannot be read http://192.x.x.x.x:5060/xcap-root/xca-caps/global/index For Null Please Help me , for solve it's solution //Gafar

2 1

Releasing v4.2.0
by Daniel-Constantin Mierla 16 Oct '14

16 Oct '14

Hello, short note to say that packaging of v4.2.0 will start after 14:00 (Berlin time) -- anyone willing to do changes to branch 4.2 should send a short notice on sr-dev and wait for confirmation to avoid conflicts. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

How to setup stun parameters?
by george wu 16 Oct '14

16 Oct '14

Hi, This document is too simple. http://kamailio.org/docs/modules/stable/modules/stun.html How to setup stun parameters? ip/ports? Thanks. George

3 3

Query-for-Dispatcher.list
by Anil Kumar 16 Oct '14

16 Oct '14

Hello All, Please can anyone give some information regarding dispatcher.list file. I am using LAN network and trying to setup a kamailio-ims server. What is the info I need to configure in dispatcher.list ? I see error while running pcscf module. Thanks

2 1

Checking documentation for v4.2
by Daniel-Constantin Mierla 15 Oct '14

15 Oct '14

Hello, in preparation for releasing v4.2.0 on Thursday this week, I spent some time today to publish the documentation for v4.2.x series at the usual places on kamailio.org site. I am writing here to ask everyone having some spare time to check through usual places they used in the past to read documentation on kamailio.org and see if something is missing for 4.2.x or something is wrong (e.g., broken or wrong links). Starting points can be: - http://www.kamailio.org/w/documentation/ - http://www.kamailio.org/wiki/ Report any finding back to mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

rtpengine upgrade fails
by Juha Heinanen 15 Oct '14

15 Oct '14

just to complete the thread, the dependency problem has been fixed and upgrade from 3.3.0.0+0~mr3.5.0.0 to 3.3.0.0+0~mr3.6.0.0 succeeded without issues. -- juha

3 4

Group User Presence
by rastrella 15 Oct '14

15 Oct '14

Hi guys, I've setup my kamailio more or less with all my needs, thanks to all your helps :) Now I've a question regarding presence. My kamailio manage all sip clients that are presents in my home. When I receive call all my phone rings simultaneously and so on... I would like that I can manage presence status of my home in order that...if some of my phones is "on call" the status of my "group" is "on call". It's something like that my "home" presence status is the AND/OR of all my phones presence status... Is there any way to make this? I think that I've to use PUA module, but the problem is that multiple calls are possible so is there any way to understand from script number of ongoing calls? ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Group-User-Presence-tp131478.html Sent from the Users mailing list archive at Nabble.com.

2 1

No need for Authentication... Why?
by Manuel Camarg 15 Oct '14

15 Oct '14

Following this asipto guide: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I've found that no auth is required to create a communication: Invites are sent regardless of a previously done auth with the kam's auth_db I've used this sip checker: http://www.sinologic.net/proyectos/asterisk/checkSecurity/ Result: Uh oh! you allow external calls... Configure better your sip configuration to avoid this calls SIP/2.0 100 trying -- your call is important to us What is missing in the config file explained in the guide? Manuel

2 3

Kamailio 4.0.4 slow memory leak
by Heenan, Timothy Steven 15 Oct '14

15 Oct '14

Greetings, I'm running into a slow memory leak on my kamailio 4.0.4 SIP proxies. I'm observing a steady increase in the memory consumption until there is no more left. Kamailio then starts repeating this in the logs: ERROR: dispatcher [dispatch.c:279]: add_dest2list(): no more memory. What would be the best way to debug this kind of a memory leak? The proxy does not handle any registrations but does route a fair amount of calls. Thank you in advance, Tim Heenan Engineer I - VoIP Wholesale | Windstream timothy.heenan(a)windstream.com<mailto:email@windstream.com> | windstreambusiness.com<http://www.windstreambusiness.com/> o: 847.348.1338 | f: 847.963.0116 ---------------------------------------------------------------------- This email message and any attachments are for the sole use of the intended recipient(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message and any attachments.

2 19

presence module issue with NOTIFY messsage.
by ThanhTruong 14 Oct '14

14 Oct '14

Hi all, I have install kamailio 4.2 latest version with presence module. But I cant get contact status and send message. This is my configure file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.2 - 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/wiki/ # 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 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** 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 ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### 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. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!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 ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!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 ####### 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" pstn.gw_port = "" desc "PSTN GW Port" #!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 ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.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" #!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 # ----------------- 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 ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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 ####### 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); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); 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); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) 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; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); exit; } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); 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()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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 route(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")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # 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 $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); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!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; } # IP authorization and user uthentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$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[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; # ThanhTruong Handle Rtpproxy Header Field if (is_present_hf("Rtpproxy")) { rtpproxy_manage("co"); }else{ xlog("L_ERR", "DO NOT Route call via RTPPROXY ....................\n"); } #rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # 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}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $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")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } and this is message tracer : U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 REGISTER sip:sip.example.com:5060 SIP/2.0. Call-ID: a9fa97851a69f48cf899f393d8840f78@0:0:0:0:0:0:0:0. CSeq: 1 REGISTER. From: "126" <sip:126@sip.example.com>;tag=f171bbda. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-19e7f68f37f97d0596018a16a6f92588. Max-Forwards: 70. User-Agent: Jitsi2.4.4997Mac OS X. Expires: 600. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 401 Unauthorized. Call-ID: a9fa97851a69f48cf899f393d8840f78@0:0:0:0:0:0:0:0. CSeq: 1 REGISTER. From: "126" <sip:126@sip.example.com>;tag=f171bbda. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.3c8d. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-19e7f68f37f97d0596018a16a6f92588;rport=27000;received=113.172.178.53. WWW-Authenticate: Digest realm="sip.example.com", nonce="VDBGtVQwRYklXB5tPKgFK0p+CLOeqnrd". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 REGISTER sip:sip.example.com:5060 SIP/2.0. Call-ID: a9fa97851a69f48cf899f393d8840f78@0:0:0:0:0:0:0:0. CSeq: 2 REGISTER. From: "126" <sip:126@sip.example.com>;tag=f171bbda. To: "126" <sip:126@sip.example.com>. Max-Forwards: 70. User-Agent: Jitsi2.4.4997Mac OS X. Expires: 600. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-ef20767880d63aa0678cf55bfdf23d8f. Authorization: Digest username="126",realm="sip.example.com",nonce="VDBGtVQwRYklXB5tPKgFK0p+CLOeqnrd",uri="sip:sip.example.com:5060",response="7948d5cc188fae8d1fc56ef76668b622". Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 200 OK. Call-ID: a9fa97851a69f48cf899f393d8840f78@0:0:0:0:0:0:0:0. CSeq: 2 REGISTER. From: "126" <sip:126@sip.example.com>;tag=f171bbda. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.70fc. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-ef20767880d63aa0678cf55bfdf23d8f;rport=27000;received=113.172.178.53. Contact: <sip:126@192.168.0.103:20730;rinstance=f422aa5d2e46080f>;expires=1623;received="sip:113.172.178.53:27046", <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600;received="sip:113.172.178.53:27000". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 REGISTER sip:sip.example.com:5060 SIP/2.0. Call-ID: a9fa97851a69f48cf899f393d8840f78@0:0:0:0:0:0:0:0. CSeq: 2 REGISTER. From: "126" <sip:126@sip.example.com>;tag=f171bbda. To: "126" <sip:126@sip.example.com>. Max-Forwards: 70. User-Agent: Jitsi2.4.4997Mac OS X. Expires: 600. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-ef20767880d63aa0678cf55bfdf23d8f. Authorization: Digest username="126",realm="sip.example.com",nonce="VDBGtVQwRYklXB5tPKgFK0p+CLOeqnrd",uri="sip:sip.example.com:5060",response="7948d5cc188fae8d1fc56ef76668b622". Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 200 OK. Call-ID: a9fa97851a69f48cf899f393d8840f78@0:0:0:0:0:0:0:0. CSeq: 2 REGISTER. From: "126" <sip:126@sip.example.com>;tag=f171bbda. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.70fc. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-ef20767880d63aa0678cf55bfdf23d8f;rport=27000;received=113.172.178.53. Contact: <sip:126@192.168.0.103:20730;rinstance=f422aa5d2e46080f>;expires=1623;received="sip:113.172.178.53:27046", <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600;received="sip:113.172.178.53:27000". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 SUBSCRIBE sip:126@sip.example.com SIP/2.0. Call-ID: daaf0a31e8ff211da4b9a26715b97486@0:0:0:0:0:0:0:0. CSeq: 1 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=6b1a44c2. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-ee5ab5bb8d09684994d7c54e4bcf749f. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Event: presence.winfo. Accept: application/watcherinfo+xml. Expires: 3600. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 407 Proxy Authentication Required. Call-ID: daaf0a31e8ff211da4b9a26715b97486@0:0:0:0:0:0:0:0. CSeq: 1 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=6b1a44c2. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.633c. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-ee5ab5bb8d09684994d7c54e4bcf749f;rport=27000;received=113.172.178.53. Proxy-Authenticate: Digest realm="sip.example.com", nonce="VDBGt1QwRYs4FBEAqaT7qgC/bVpE6gSw". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 SUBSCRIBE sip:126@sip.example.com SIP/2.0. Call-ID: 2eaf31c4c83c4ea8d955cf26d7318d08@0:0:0:0:0:0:0:0. CSeq: 1 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=d0c25ea9. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-2b89583dfe1146d7bf4fbb845c046275. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Event: message-summary. Accept: application/simple-message-summary. Expires: 3600. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 407 Proxy Authentication Required. Call-ID: 2eaf31c4c83c4ea8d955cf26d7318d08@0:0:0:0:0:0:0:0. CSeq: 1 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=d0c25ea9. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.b775. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-2b89583dfe1146d7bf4fbb845c046275;rport=27000;received=113.172.178.53. Proxy-Authenticate: Digest realm="sip.example.com", nonce="VDBGt1QwRYs4FBEAqaT7qgC/bVpE6gSw". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: d401f9525044cf69a4e69972c2f4da08@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=7863ae0e. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc2969e12853a7feb1bbe3f27ac09887. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:5060 SIP/2.0 200 Keepalive. Call-ID: d401f9525044cf69a4e69972c2f4da08@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=7863ae0e. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.3e6e. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc2969e12853a7feb1bbe3f27ac09887;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 SUBSCRIBE sip:126@sip.example.com SIP/2.0. Call-ID: daaf0a31e8ff211da4b9a26715b97486@0:0:0:0:0:0:0:0. CSeq: 2 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=6b1a44c2. To: "126" <sip:126@sip.example.com>. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Event: presence.winfo. Accept: application/watcherinfo+xml. Expires: 3600. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-6b9a0437c22274450a94d979f61ba0cd. Proxy-Authorization: Digest username="126",realm="sip.example.com",nonce="VDBGt1QwRYs4FBEAqaT7qgC/bVpE6gSw",uri="sip:126@sip.example.com",response="17330e0333412b150fd08fc7d87f1d8e". Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 200 OK. Call-ID: daaf0a31e8ff211da4b9a26715b97486@0:0:0:0:0:0:0:0. CSeq: 2 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=6b1a44c2. To: "126" <sip:126@sip.example.com>;tag=c182c0c06e577e5881c162aa7d80ef4e-f6fc. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-6b9a0437c22274450a94d979f61ba0cd;rport=27000;received=113.172.178.53. Expires: 3600. Contact: <sip:abc.def.ghi.klm:5060;transport=udp>. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U abc.def.ghi.klm:5060 -> 192.168.0.103:5060 NOTIFY sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi;alias=113.172.178.53~27000~1 SIP/2.0. Via: SIP/2.0/UDP sip.example.com:5060;branch=z9hG4bK474f.1df9eac6000000000000000000000000.0. To: <sip:126@sip.example.com>;tag=6b1a44c2. From: <sip:126@sip.example.com>;tag=c182c0c06e577e5881c162aa7d80ef4e-f6fc. CSeq: 2 NOTIFY. Call-ID: daaf0a31e8ff211da4b9a26715b97486@0:0:0:0:0:0:0:0. Content-Length: 197. User-Agent: kamailio (4.2.0-pre1 (x86_64/linux)). Max-Forwards: 70. Event: presence.winfo. Contact: <sip:abc.def.ghi.klm:5060;transport=udp>. Subscription-State: active;expires=3600. Content-Type: application/watcherinfo+xml. . <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="1" state="full"> <watcher-list resource="sip:126@sip.example.com" package="presence"/> </watcherinfo> U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 SUBSCRIBE sip:126@sip.example.com SIP/2.0. Call-ID: 2eaf31c4c83c4ea8d955cf26d7318d08@0:0:0:0:0:0:0:0. CSeq: 2 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=d0c25ea9. To: "126" <sip:126@sip.example.com>. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Event: message-summary. Accept: application/simple-message-summary. Expires: 3600. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-84cbc7e144aec1d1f05099ebca9da3d2. Proxy-Authorization: Digest username="126",realm="sip.example.com",nonce="VDBGt1QwRYs4FBEAqaT7qgC/bVpE6gSw",uri="sip:126@sip.example.com",response="17330e0333412b150fd08fc7d87f1d8e". Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 404 No voicemail service. Call-ID: 2eaf31c4c83c4ea8d955cf26d7318d08@0:0:0:0:0:0:0:0. CSeq: 2 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=d0c25ea9. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.ef78. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-84cbc7e144aec1d1f05099ebca9da3d2;rport=27000;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: d401f9525044cf69a4e69972c2f4da08@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=7863ae0e. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc2969e12853a7feb1bbe3f27ac09887. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:5060 SIP/2.0 200 Keepalive. Call-ID: d401f9525044cf69a4e69972c2f4da08@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=7863ae0e. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.3e6e. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc2969e12853a7feb1bbe3f27ac09887;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:26832 .... U abc.def.ghi.klm:5060 -> 192.168.0.103:5060 NOTIFY sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi;alias=113.172.178.53~27000~1 SIP/2.0. Via: SIP/2.0/UDP sip.example.com:5060;branch=z9hG4bK474f.1df9eac6000000000000000000000000.0. To: <sip:126@sip.example.com>;tag=6b1a44c2. From: <sip:126@sip.example.com>;tag=c182c0c06e577e5881c162aa7d80ef4e-f6fc. CSeq: 2 NOTIFY. Call-ID: daaf0a31e8ff211da4b9a26715b97486@0:0:0:0:0:0:0:0. Content-Length: 197. User-Agent: kamailio (4.2.0-pre1 (x86_64/linux)). Max-Forwards: 70. Event: presence.winfo. Contact: <sip:abc.def.ghi.klm:5060;transport=udp>. Subscription-State: active;expires=3600. Content-Type: application/watcherinfo+xml. . <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="1" state="full"> <watcher-list resource="sip:126@sip.example.com" package="presence"/> </watcherinfo> U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: d401f9525044cf69a4e69972c2f4da08@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=7863ae0e. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc2969e12853a7feb1bbe3f27ac09887. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:5060 SIP/2.0 200 Keepalive. Call-ID: d401f9525044cf69a4e69972c2f4da08@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=7863ae0e. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.3e6e. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc2969e12853a7feb1bbe3f27ac09887;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27046 -> abc.def.ghi.klm:5060 . . U abc.def.ghi.klm:5060 -> 192.168.0.103:5060 NOTIFY sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi;alias=113.172.178.53~27000~1 SIP/2.0. Via: SIP/2.0/UDP sip.example.com:5060;branch=z9hG4bK474f.1df9eac6000000000000000000000000.0. To: <sip:126@sip.example.com>;tag=6b1a44c2. From: <sip:126@sip.example.com>;tag=c182c0c06e577e5881c162aa7d80ef4e-f6fc. CSeq: 2 NOTIFY. Call-ID: daaf0a31e8ff211da4b9a26715b97486@0:0:0:0:0:0:0:0. Content-Length: 197. User-Agent: kamailio (4.2.0-pre1 (x86_64/linux)). Max-Forwards: 70. Event: presence.winfo. Contact: <sip:abc.def.ghi.klm:5060;transport=udp>. Subscription-State: active;expires=3600. Content-Type: application/watcherinfo+xml. . <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="1" state="full"> <watcher-list resource="sip:126@sip.example.com" package="presence"/> </watcherinfo> U 113.172.178.53:26748 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: c06f95df711ac1cfa07e2a5fbb3add79@0:0:0:0:0:0:0:0. CSeq: 30 OPTIONS. From: "127" <sip:127@sip.example.com>;tag=80c6c9b6. To: "127" <sip:127@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK-353637-39070c742a32ce2fb155d4c63e40abf0. Max-Forwards: 70. Contact: "127" <sip:127@192.168.0.102:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Windows 7. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:5060 SIP/2.0 200 Keepalive. Call-ID: c06f95df711ac1cfa07e2a5fbb3add79@0:0:0:0:0:0:0:0. CSeq: 30 OPTIONS. From: "127" <sip:127@sip.example.com>;tag=80c6c9b6. To: "127" <sip:127@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.32bb. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK-353637-39070c742a32ce2fb155d4c63e40abf0;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 REGISTER sip:sip.example.com:5060 SIP/2.0. Call-ID: fede40e5d6742a43b3dff64b1584b3be@0:0:0:0:0:0:0:0. CSeq: 1 REGISTER. From: "126" <sip:126@sip.example.com>;tag=20b8e68. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-556ed07f63ead6c5d04c34eb3a0a7019. Max-Forwards: 70. User-Agent: Jitsi2.4.4997Mac OS X. Expires: 600. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 401 Unauthorized. Call-ID: fede40e5d6742a43b3dff64b1584b3be@0:0:0:0:0:0:0:0. CSeq: 1 REGISTER. From: "126" <sip:126@sip.example.com>;tag=20b8e68. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.7bd8. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-556ed07f63ead6c5d04c34eb3a0a7019;rport=27000;received=113.172.178.53. WWW-Authenticate: Digest realm="sip.example.com", nonce="VDBGuVQwRY1cVIxjVVNJCKirWxOUzPkN". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 REGISTER sip:sip.example.com:5060 SIP/2.0. Call-ID: fede40e5d6742a43b3dff64b1584b3be@0:0:0:0:0:0:0:0. CSeq: 1 REGISTER. From: "126" <sip:126@sip.example.com>;tag=20b8e68. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-556ed07f63ead6c5d04c34eb3a0a7019. Max-Forwards: 70. User-Agent: Jitsi2.4.4997Mac OS X. Expires: 600. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 401 Unauthorized. Call-ID: fede40e5d6742a43b3dff64b1584b3be@0:0:0:0:0:0:0:0. CSeq: 1 REGISTER. From: "126" <sip:126@sip.example.com>;tag=20b8e68. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.7bd8. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-556ed07f63ead6c5d04c34eb3a0a7019;rport=27000;received=113.172.178.53. WWW-Authenticate: Digest realm="sip.example.com", nonce="VDBGulQwRY6I/6yOqmDKU12BjirB4skO". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 REGISTER sip:sip.example.com:5060 SIP/2.0. Call-ID: fede40e5d6742a43b3dff64b1584b3be@0:0:0:0:0:0:0:0. CSeq: 2 REGISTER. From: "126" <sip:126@sip.example.com>;tag=20b8e68. To: "126" <sip:126@sip.example.com>. Max-Forwards: 70. User-Agent: Jitsi2.4.4997Mac OS X. Expires: 600. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-a04a1326eb5a42f7d6955ef55b22d460. Authorization: Digest username="126",realm="sip.example.com",nonce="VDBGuVQwRY1cVIxjVVNJCKirWxOUzPkN",uri="sip:sip.example.com:5060",response="5a378134b4f1df52e69f1c82fd5e3c96". Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 200 OK. Call-ID: fede40e5d6742a43b3dff64b1584b3be@0:0:0:0:0:0:0:0. CSeq: 2 REGISTER. From: "126" <sip:126@sip.example.com>;tag=20b8e68. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.2bc0. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-a04a1326eb5a42f7d6955ef55b22d460;rport=27000;received=113.172.178.53. Contact: <sip:126@192.168.0.103:20730;rinstance=f422aa5d2e46080f>;expires=1619;received="sip:113.172.178.53:27046", <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>;expires=600;received="sip:113.172.178.53:27000". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:26748 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: 282bf6cf0151acf56b6f4dc3ea57a438@0:0:0:0:0:0:0:0. CSeq: 31 OPTIONS. From: "127" <sip:127@sip.example.com>;tag=746f6c5a. To: "127" <sip:127@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK-353637-dacef276fc1c7c7733dbd968487dc542. Max-Forwards: 70. Contact: "127" <sip:127@192.168.0.102:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Windows 7. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:5060 SIP/2.0 200 Keepalive. Call-ID: 282bf6cf0151acf56b6f4dc3ea57a438@0:0:0:0:0:0:0:0. CSeq: 31 OPTIONS. From: "127" <sip:127@sip.example.com>;tag=746f6c5a. To: "127" <sip:127@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.b5a9. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK-353637-dacef276fc1c7c7733dbd968487dc542;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 SUBSCRIBE sip:126@sip.example.com SIP/2.0. Call-ID: 0cbb6fe545bcbf07d761c9b2a449bd9e@0:0:0:0:0:0:0:0. CSeq: 1 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=79b767b9. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-9b1db4554327d50f02a53dd0514b9bfe. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Event: presence.winfo. Accept: application/watcherinfo+xml. Expires: 3600. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 407 Proxy Authentication Required. Call-ID: 0cbb6fe545bcbf07d761c9b2a449bd9e@0:0:0:0:0:0:0:0. CSeq: 1 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=79b767b9. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.774f. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-9b1db4554327d50f02a53dd0514b9bfe;rport=27000;received=113.172.178.53. Proxy-Authenticate: Digest realm="sip.example.com", nonce="VDBGulQwRY6I/6yOqmDKU12BjirB4skO". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 SUBSCRIBE sip:126@sip.example.com SIP/2.0. Call-ID: e422d82a76d723ae66bf01d2eda0579b@0:0:0:0:0:0:0:0. CSeq: 1 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=7fc9dd97. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-08a4df3326d5c070e556dba514608ed6. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Event: message-summary. Accept: application/simple-message-summary. Expires: 3600. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 407 Proxy Authentication Required. Call-ID: e422d82a76d723ae66bf01d2eda0579b@0:0:0:0:0:0:0:0. CSeq: 1 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=7fc9dd97. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.19b3. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-08a4df3326d5c070e556dba514608ed6;rport=27000;received=113.172.178.53. Proxy-Authenticate: Digest realm="sip.example.com", nonce="VDBGulQwRY6I/6yOqmDKU12BjirB4skO". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: 5e9017d6273c100e1757c4842f833640@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=b71031ed. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-397bcdc29e1933cdeced0d1066f15953. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:5060 SIP/2.0 200 Keepalive. Call-ID: 5e9017d6273c100e1757c4842f833640@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=b71031ed. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.428d. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-397bcdc29e1933cdeced0d1066f15953;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 PUBLISH sip:126@sip.example.com SIP/2.0. Call-ID: ff11d1237ed1f83caaa3f813f8e8b7ac@0:0:0:0:0:0:0:0. CSeq: 19 PUBLISH. From: "126" <sip:126@sip.example.com>;tag=f8e313e0. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-7cce216591fd90840efd2ca6d7c72a1e. Max-Forwards: 70. Content-Type: application/pidf+xml. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Expires: 3600. Event: presence. Content-Length: 427. . <?xml version="1.0" encoding="UTF-8" standalone="no"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:126@sip.example.com"><dm:person id="p3196"><rpid:activities/></dm:person><tuple id="t1156"><status><basic>open</basic></status><contact>sip:126@sip.example.com</contact><note>Online</note></tuple></presence> U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 407 Proxy Authentication Required. Call-ID: ff11d1237ed1f83caaa3f813f8e8b7ac@0:0:0:0:0:0:0:0. CSeq: 19 PUBLISH. From: "126" <sip:126@sip.example.com>;tag=f8e313e0. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.a1dc. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-7cce216591fd90840efd2ca6d7c72a1e;rport=27000;received=113.172.178.53. Proxy-Authenticate: Digest realm="sip.example.com", nonce="VDBGulQwRY6I/6yOqmDKU12BjirB4skO". Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: d401f9525044cf69a4e69972c2f4da08@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=7863ae0e. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc2969e12853a7feb1bbe3f27ac09887. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:5060 SIP/2.0 200 Keepalive. Call-ID: d401f9525044cf69a4e69972c2f4da08@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=7863ae0e. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.3e6e. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc2969e12853a7feb1bbe3f27ac09887;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:26748 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: 282bf6cf0151acf56b6f4dc3ea57a438@0:0:0:0:0:0:0:0. CSeq: 31 OPTIONS. From: "127" <sip:127@sip.example.com>;tag=746f6c5a. To: "127" <sip:127@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK-353637-dacef276fc1c7c7733dbd968487dc542. Max-Forwards: 70. Contact: "127" <sip:127@192.168.0.102:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Windows 7. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:5060 SIP/2.0 200 Keepalive. Call-ID: 282bf6cf0151acf56b6f4dc3ea57a438@0:0:0:0:0:0:0:0. CSeq: 31 OPTIONS. From: "127" <sip:127@sip.example.com>;tag=746f6c5a. To: "127" <sip:127@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.b5a9. Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK-353637-dacef276fc1c7c7733dbd968487dc542;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U abc.def.ghi.klm:5060 -> 192.168.0.103:5060 NOTIFY sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi;alias=113.172.178.53~27000~1 SIP/2.0. Via: SIP/2.0/UDP sip.example.com:5060;branch=z9hG4bK474f.1df9eac6000000000000000000000000.0. To: <sip:126@sip.example.com>;tag=6b1a44c2. From: <sip:126@sip.example.com>;tag=c182c0c06e577e5881c162aa7d80ef4e-f6fc. CSeq: 2 NOTIFY. Call-ID: daaf0a31e8ff211da4b9a26715b97486@0:0:0:0:0:0:0:0. Content-Length: 197. User-Agent: kamailio (4.2.0-pre1 (x86_64/linux)). Max-Forwards: 70. Event: presence.winfo. Contact: <sip:abc.def.ghi.klm:5060;transport=udp>. Subscription-State: active;expires=3600. Content-Type: application/watcherinfo+xml. . <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="1" state="full"> <watcher-list resource="sip:126@sip.example.com" package="presence"/> </watcherinfo> U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 SUBSCRIBE sip:126@sip.example.com SIP/2.0. Call-ID: 0cbb6fe545bcbf07d761c9b2a449bd9e@0:0:0:0:0:0:0:0. CSeq: 2 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=79b767b9. To: "126" <sip:126@sip.example.com>. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Event: presence.winfo. Accept: application/watcherinfo+xml. Expires: 3600. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-e1bb6fe7148ddebb3058273fb7f59a6b. Proxy-Authorization: Digest username="126",realm="sip.example.com",nonce="VDBGulQwRY6I/6yOqmDKU12BjirB4skO",uri="sip:126@sip.example.com",response="943363e96487711789491683418f96cb". Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 200 OK. Call-ID: 0cbb6fe545bcbf07d761c9b2a449bd9e@0:0:0:0:0:0:0:0. CSeq: 2 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=79b767b9. To: "126" <sip:126@sip.example.com>;tag=c182c0c06e577e5881c162aa7d80ef4e-389e. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-e1bb6fe7148ddebb3058273fb7f59a6b;rport=27000;received=113.172.178.53. Expires: 3600. Contact: <sip:abc.def.ghi.klm:5060;transport=udp>. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U abc.def.ghi.klm:5060 -> 192.168.0.103:5060 NOTIFY sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi;alias=113.172.178.53~27000~1 SIP/2.0. Via: SIP/2.0/UDP sip.example.com:5060;branch=z9hG4bKe9e5.7da40db3000000000000000000000000.0. To: <sip:126@sip.example.com>;tag=79b767b9. From: <sip:126@sip.example.com>;tag=c182c0c06e577e5881c162aa7d80ef4e-389e. CSeq: 2 NOTIFY. Call-ID: 0cbb6fe545bcbf07d761c9b2a449bd9e@0:0:0:0:0:0:0:0. Content-Length: 197. User-Agent: kamailio (4.2.0-pre1 (x86_64/linux)). Max-Forwards: 70. Event: presence.winfo. Contact: <sip:abc.def.ghi.klm:5060;transport=udp>. Subscription-State: active;expires=3600. Content-Type: application/watcherinfo+xml. . <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="1" state="full"> <watcher-list resource="sip:126@sip.example.com" package="presence"/> </watcherinfo> U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 SUBSCRIBE sip:126@sip.example.com SIP/2.0. Call-ID: e422d82a76d723ae66bf01d2eda0579b@0:0:0:0:0:0:0:0. CSeq: 2 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=7fc9dd97. To: "126" <sip:126@sip.example.com>. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Event: message-summary. Accept: application/simple-message-summary. Expires: 3600. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc60bef221c1fdc4cf652059445e1a9b. Proxy-Authorization: Digest username="126",realm="sip.example.com",nonce="VDBGulQwRY6I/6yOqmDKU12BjirB4skO",uri="sip:126@sip.example.com",response="943363e96487711789491683418f96cb". Content-Length: 0. . U abc.def.ghi.klm:5060 -> 113.172.178.53:27000 SIP/2.0 404 No voicemail service. Call-ID: e422d82a76d723ae66bf01d2eda0579b@0:0:0:0:0:0:0:0. CSeq: 2 SUBSCRIBE. From: "126" <sip:126@sip.example.com>;tag=7fc9dd97. To: "126" <sip:126@sip.example.com>;tag=5ecb26dc70085b593eff6ba03c18a5c9.7215. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-cc60bef221c1fdc4cf652059445e1a9b;rport=27000;received=113.172.178.53. Server: kamailio (4.2.0-pre1 (x86_64/linux)). Content-Length: 0. . U 113.172.178.53:27000 -> abc.def.ghi.klm:5060 OPTIONS sip:sip.example.com SIP/2.0. Call-ID: 5e9017d6273c100e1757c4842f833640@0:0:0:0:0:0:0:0. CSeq: 1 OPTIONS. From: "126" <sip:126@sip.example.com>;tag=b71031ed. To: "126" <sip:126@sip.example.com>. Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK-393532-397bcdc29e1933cdeced0d1066f15953. Max-Forwards: 70. Contact: "126" <sip:126@192.168.0.103:5060;transport=udp;registering_acc=sip_killermobile_mobi>. User-Agent: Jitsi2.4.4997Mac OS X. Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE. Allow-Events: refer. Content-Length: 0. . I see that SUBSCRIBE message send to wrong ip, it is sent to local IP and my sip phone do not receive it. But i do not know how to fix it. please suggest to get it. thanks all, Thanh

5 11

Kamailio 4.1.6 htable issue
by José Seabra 14 Oct '14

14 Oct '14

Hello there, I upgraded my kamailio version from 4.0.6 to 4.1.6 and now when i try start kamailio it breaks down on htable module initialization, as you can see on logs bellow: 0(1273) DEBUG: <core> [sr_module.c:966]: init_mod(): DEBUG: init_mod: presence_profile 0(1273) DEBUG: <core> [sr_module.c:701]: find_mod_export_record(): find_export_record: found <bind_presence> in module presence [/usr/local/lib64/kamailio/modules/presence.so] 0(1273) DEBUG: presence [event_list.c:351]: search_event(): start event= [ua-profile/7] 0(1273) DEBUG: presence [event_list.c:351]: search_event(): start event= [ua-profile.winfo/0] 0(1273) DEBUG: presence [event_list.c:259]: add_event(): successfully added event: ua-profile - len= 10 0(1273) DEBUG: <core> [sr_module.c:966]: init_mod(): DEBUG: init_mod: htable 0(1273) ERROR: htable [ht_api.c:282]: ht_init_tables(): no more shm for [dlg] 0(1273) ERROR: <core> [sr_module.c:970]: init_mod(): init_mod(): Error while initializing module htable (/usr/local/lib64/kamailio/modules/htable.so) ERROR: error while initializing modules 0(1273) : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7fae605547c0), called from htable: ht_api.c: ht_destroy(343), first free htable: ht_api.c: ht_destroy(343) - aborting 0(1271) ERROR: <core> [daemonize.c:307]: daemonize(): Main process exited before writing to pipe The shared memory allocated for kamailio is 256MB and is the same that i had for version 4.0.1, can this be a bug? Thank you for your support -- Cumprimentos José Seabra

2 3

Re: [SR-Users] how to always use rtpproxy-ng and rtpengine to relay rtp packets
by Paul Smith 14 Oct '14

14 Oct '14

Hi Andrew, I just use git... so not sure on the rpm stuff. However I believe 4.2 is to be released as stable branch tomorrow (see other recent thread by Daniel) so the packages probably aren't far behind. The main reasons I see for using the 4.2 branch are: 1) its going to be stable release very soon 2) the options on rtpengine have been refactored ... makes sense to develop on the new version so that you can move from development to production without amending your kamailio.cfg 3) rtpengine seems to be getting quite alot of rapid progress at the moment and gaining stability and features ... I'd rather be on the latest version. Having said all that the logic in the kamailio.cfg for version 4.1 is the same and should work for you if you want to work from the current stable packages. On 13/10/14 11:10, andrew wrote: > hi, > > thanks for your help. > > rtpengine module is used only in kamalio 4.2.x ------ development > version. where to get its rpm packet? > > B.R. > > andrew > > > At 2014-10-13 16:38:47, "Paul Smith" <paul.smith(a)claritytele.com> wrote: > > Hi Andrew > There is a condition in the NATMANAGE route which tests whether or > not to apply rtpproxy_manage() : > > if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) > return; > > One way to force this is to make sure that FLT_NATS is always set. > That can be done in the NATDETECT route by moving > setflag(FLT_NATS) outside of the if(nat_uac_test) condition and > putting it just before the return statement, so the flag is always > set. > > Another way would be to remove or comment out the if > "!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;" line > from NATMANAGE so that the the rtp_manage() is always called. > > It might be worth your while upgrading to the 4.2 branch which has > renamed the rtpproxy-ng module to rtpengine and made the options > in the call to rtpengine_manage() more readable. > > Depending on why you always want to run the rtp proxy ... you > might also want to set : > > rtpengine_manage(replace-session-sonnection replace-origin ICE=force-relay) > > to ensure that media always goes through your media proxy and no > other connections are negotiated between the endpoints. > > On 13/10/14 08:36, andrew wrote: >> hi, >> >> kamailio 4.1.5 is used with rtpengine, whose older counterpart is >> rtpproxy-ng. I made some changes based on the default >> configuration files(i.e. kamailio.cfg), so that rtpproxy-ng >> module is enabled. Kamailio.cfg has been uploaded. In >> kamailio.cfg, there is one route block related to rtpproxy, i.e. >> route[NATMANAGE], where rtpproxy_manage("co") is called. >> Sometimes, rtpproxy-ng can insert new ip:port candidates in SDP. >> But in some case, rtpproxy-ng doesn't rewrite SDP, so rtpengine >> doesn't relay rtp packets at all. >> >> How to setup kamailio.cfg, so that rtpproxy-ng moudle can always >> rewritten new ip:port candidates, and rtpengine always relays rtp >> packets? >> >> Looking foreward to your reply. Thanks! >> >> B.R. >> >> Andrew >> >> >> >> >> _______________________________________________ >> 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

Re: [SR-Users] Alias database
by Paul Smith 14 Oct '14

14 Oct '14

Thanks for the feedback. On 14/10/14 02:13, Alberto Llamas wrote: > Hello Paul, > > That was amazing. Thanks a lot ! > > On Mon, Oct 13, 2014 at 3:42 AM, Paul Smith > <paul.smith(a)claritytele.com <mailto:paul.smith@claritytele.com>> wrote: > > Hi Alberto > The magic you are looking for is in the "domain" module. The > example configuration files in the distribution have it set up and > ready to use if you add the line "#!define WITH_MULTIDOMAIN" > > The important bit for your scenario is in the example config: > > # ----- 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 > > That section tells the domain module to use the database, and to > include the domains loaded from there when matching the special > keyword "myself". The example config files also set up other > modules (like user location module) to be domain aware as well. > > In this setup you do not need to restart kamailio when you add or > remove domains.... but you do need to tell it to reload the > domains from database as Kamailio holds a cached copy of the list > in memory.... so when you make a database change then do "kamctl > domain reload". You can also run "kamctl domain show" to see what > domains are currently listed in memory. > > > > On 12/10/14 21:38, Alberto Llamas wrote: >> >> Hello kamailio gurus, >> >> Thanks if somebody can help me answering nex question: >> >> I have a kamailio server as registrar server with asterisk >> behind. For register porpouse the UA are using subdomain like: >> >> 100(a)test.mydomain.com <mailto:100@test.mydomain.com> >> 200(a)test2.mydomain.com <mailto:200@test2.mydomain.com> >> >> I have to add each alias in the kamailio.cfg file an restart the >> sevice. >> >> Alias="test.mydomain.com <http://test.mydomain.com>" >> Alias="test2.mydomain.com <http://test2.mydomain.com>" >> >> Is there some option to add it in the database? Then i don't have >> to restart the server each time i have to add/edit/delete each >> subdomain. >> >> Thanks a lot, >> >> Albert >> >> >> >> _______________________________________________ >> 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 > > > > > -- > Alberto Llamas > TelecommunicationsEngineer > Digium Certified Asterisk Professional (dCap) > Linux Administrator >

1 0

Proxy Howto
by Google - One 14 Oct '14

14 Oct '14

Hi All I am working my way through an understanding of SIP proxy configuration - slowly! As I go, I am pulling ideas and thoughts into a HOWTO type tutorial. If anyone feels like commenting, and helping me understand the various different applications (and examples of each), I would be grateful. http://manx.biz/opensource/SIP/SIP-Proxy At the moment, it is the Glossary that I am trying to get my head around... Thanks Phil

2 1

WebClients ( WebSocket ) long audio porblem
by Yuriy Gorlichenko 13 Oct '14

13 Oct '14

Hello. I hawe porblem with based at websocket transport clietns. I use wss and with chain sertificate. When I do call with this sert signalling goes normally but sound starts after fiev seconds of picking up. I debug RTP with tshark at my servers and localize issue at the kamailio. Ussue dissapears when I use sertificate without chain.

2 3

Presence Module Installation without reinstalling Kamailiio
by aawaise 13 Oct '14

13 Oct '14

I am using kamailio server. I want to add xcap server for presence module usage without reinstalling kamailio server. How can it be done. Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Presence-Module-Installation-withou… Sent from the Users mailing list archive at Nabble.com.

2 1

Presence Module Installation Error
by aawaise 13 Oct '14

13 Oct '14

Hello, I am using kamailio-3.1.1 server. I want to add presence module service to it. For that I tried to reinstall server by downloading kamailio-3.1.1 package from official web link. When I include modules "db_mysql mysql tls presence presence_xml xhttp xcap_server" in modules.lst. It gives error while running "make all" command. But when kamailio is installed without addition of presence module with "include modules = tls mysql db_mysql", it installs fine. I tried using kamailio-4.1.0, it gives an error on inclusion of presence module. Any help will be highly regarded. Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Presence-Module-Installation-Error-… Sent from the Users mailing list archive at Nabble.com.

2 1

how to always use rtpproxy-ng and rtpengine to relay rtp packets
by andrew 13 Oct '14

13 Oct '14

hi, kamailio 4.1.5 is used with rtpengine, whose older counterpart is rtpproxy-ng. I made some changes based on the default configuration files(i.e. kamailio.cfg), so that rtpproxy-ng module is enabled. Kamailio.cfg has been uploaded. In kamailio.cfg, there is one route block related to rtpproxy, i.e. route[NATMANAGE], where rtpproxy_manage("co") is called. Sometimes, rtpproxy-ng can insert new ip:port candidates in SDP. But in some case, rtpproxy-ng doesn't rewrite SDP, so rtpengine doesn't relay rtp packets at all. How to setup kamailio.cfg, so that rtpproxy-ng moudle can always rewritten new ip:port candidates, and rtpengine always relays rtp packets? Looking foreward to your reply. Thanks! B.R. Andrew

2 1

Roadmap to release v4.2.0
by Daniel-Constantin Mierla 13 Oct '14

13 Oct '14

Hello, after last days of testing at SIPit 31, I think the master branch is in quite good shape. I hope to get the time for a more detailed report, but we had only one major issue when running some stress tests, affecting websocket connections when using a special crafted websocket packet, fixed already. Otherwise, while the tests here are mainly on SIP interoperability, so not much on particular use cases of kamailio, with the help of Olle we have been in pretty much all multi-party tests as well as plenty of one-to-one interop tests. I have also been running Kamailio master branch on several other private or public servers with live traffic and quite complex config (e.g., usual suspects as openrcs.com and voipuser.org) and haven't noticed any issue. Based on the above, I am considering to branch 4.2 next week on Wednesday (Oct 8), and release about one week later (October 15 or 16). Anyone with other opinions or alternative roadmap? On the other hand, we have to start building migration tutorial and updating what is new in this release: * http://www.kamailio.org/wiki/install/upgrade/stable-to-devel * http://www.kamailio.org/wiki/features/new-in-devel Everyone is encouraged to contribute. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

Alias database
by Alberto Llamas 13 Oct '14

13 Oct '14

Hello kamailio gurus, Thanks if somebody can help me answering nex question: I have a kamailio server as registrar server with asterisk behind. For register porpouse the UA are using subdomain like: 100(a)test.mydomain.com 200(a)test2.mydomain.com I have to add each alias in the kamailio.cfg file an restart the sevice. Alias="test.mydomain.com" Alias="test2.mydomain.com" Is there some option to add it in the database? Then i don't have to restart the server each time i have to add/edit/delete each subdomain. Thanks a lot, Albert

2 1

483 too many hops
by Anil Kumar 13 Oct '14

13 Oct '14

Hi All, I am trying to register an ims-client/user to kamailio proxy, but I am seeing 483 too many hops response. I know one of the reason for this, that is MAX-FORWARD limit has reached. But in my case I feel I have given a proper value (70) , but still some looping is happening in the middle and limit is reached. So can anyone please help me, whats going wrong here. Any configuration changes required ? Thanks Anil

2 3

multiple endpoints call
by Yuriy Gorlichenko 12 Oct '14

12 Oct '14

Hello. I have multiple endpoints registered at my kamailio with one account (for example user1 registerd from norhway, USA, and Russia at one time), so when I call from user2 to user1 I want to ring all endpoints registered by user1 account. Now I can ring only one, first entry at location table. What setting may I use for this? Thanks

2 3

Removing dispatcher entry
by dodul 10 Oct '14

10 Oct '14

Hi, I have a small question. If I remove a server from my dispatcher while it has existing calls, what's gonna happen to the next notifications or requests of the live calls. Will it send them to the same server even though the server is not listed on the dispatcher. Or it will forward them to the other servers listed in the dispatcher. Sent from my Samsung Galaxy smartphone. -------- Original message -------- From: Corey Edwards <tensai(a)zmonkey.org> Date:10-10-2014 10:28 (GMT-06:00) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: Subject: Re: [SR-Users] Problems configuring Kamailio on Centos 6.5 On Thu, Oct 9, 2014 at 9:54 PM, logikanet <webmaster(a)logikanet.com> wrote: > Good evening! > > I've been trying to install Kamailio on a Centos 6.5 cloud , I'm able to > compile and install everything, if I run : service kamailio start it > starts kamailio and if I run service kamailio status I tells me that is > running on PID... which is fine, however, if I tweak the > /usr/local/etc/kamailio/kamailio.conf according to the instructions on > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb > , when I run service kamailio start It says "ok" , but when I run service > kamailio status I get " kamailio dead but subsys locked" , I copied and > pasted the settings on the web-site but they don't work, I also tried used > the original cfg file adding the "#!ifdef WITH_ASTERISK" but It still > doesn't work, I'm not an expert on Kamailio I just want to use it as a > register server behind Asterisk , perhaps if you have a valid conf file > that I could use as a template for kamailio 4.0.1 this is what I have : > Kamailio is pretty good about printing out error logs when it fails to start. The trick is to capture them. I would start by running Kamailio from the command line in no-fork mode. # kamailio -DE Corey _______________________________________________ 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

3 2

Problems configuring Kamailio on Centos 6.5
by logikanet 10 Oct '14

10 Oct '14

Good evening! I've been trying to install Kamailio on a Centos 6.5 cloud , I'm able to compile and install everything, if I run : service kamailio start it starts kamailio and if I run service kamailio status I tells me that is running on PID... which is fine, however, if I tweak the /usr/local/etc/kamailio/kamailio.conf according to the instructions on http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb , when I run service kamailio start It says "ok" , but when I run service kamailio status I get " kamailio dead but subsys locked" , I copied and pasted the settings on the web-site but they don't work, I also tried used the original cfg file adding the "#!ifdef WITH_ASTERISK" but It still doesn't work, I'm not an expert on Kamailio I just want to use it as a register server behind Asterisk , perhaps if you have a valid conf file that I could use as a template for kamailio 4.0.1 this is what I have : #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK # # 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://root:axn0505@200.98.201.4/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://root:axn0505@80.241.222.146/asterisk" #!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 ####### 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 = "80.241.222.146" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "200.98.201.4" 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" #!else mpath="/usr/lib64/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" #!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", "name") modparam("auth_db", "password_column", "sippasswd") 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 ####### 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(); #!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("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($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 $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); 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 -- *Giuliano Bellini* *Logikanet Webmaster* *Logikanet.com* | webmaster(a)logikanet.com | +55 (41) 3012-0232 /direct/ | +1 206-279-5207 /direct/ www.logikanet.com <http://www.logikanet.com> |

2 1

Re: [SR-Users] Crash Kamailio 4.1.4
by Igor Potjevlesch 10 Oct '14

10 Oct '14

Hello, We updated this morning Kamailio in 4.1.4 with your patch. Kamailio crashed again this afternoon. here an extract from /var/log/messages : Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2] ua=[(innovaphone IP6010/9.00 hotfix24 [9.061271/9061271/300])], INVITE from 'untrusted' host Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2], INVITE from an authorized SIP trunk Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: WARNING: <script>: time=[Wed Jun 25 13:49:01 2014] call id=[f4653f6fe909d3118f86009033290024(a)A.B.C.D] call seq=[929405] contact header=[<sip:0987654321@A.B.C.D:2057;transport=UDP>] from uri=[sip:0123456789@domain;user=phone] from tag=[16632949] request's method=[INVITE] request's uri=[sip:0123456789@domain;user=phone] to uri=[sip:0123456789@domain;user=phone] to tag=[<null>] sip message id=[46275] process id=[20259] ip source=[A.B.C.D] flags=[2], INVITE from an authorized SDA for current SIP trunk Jun 25 13:49:01 /usr/local/sbin/kamailio[20259]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri 0123456789 was rewritten to sip:0123456789@GW, carrier 1, domain 1 Jun 25 13:49:02 /usr/local/sbin/kamailio[20259]: : <core> [mem/q_malloc.c:140]: qm_debug_frag(): BUG: qm_*: fragm. 0x7f12803cb450 (address 0x7f12803cb480) beginning overwritten(abcdefed)! And this is the btfull : #0 0x0000003d6f6328a5 in raise () from /lib64/libc.so.6 No symbol table info available. #1 0x0000003d6f634085 in abort () from /lib64/libc.so.6 No symbol table info available. #2 0x0000000000546d3c in qm_debug_frag (qm=0x7f1280275010, f=0x7f12803cb450) at mem/q_malloc.c:142 #3 0x0000000000548b26 in qm_free (qm=0x7f1280275010, p=0x7f12803cb480, file=0x6276a0 "<core>: parser/parse_ppi_pai.c", func=0x627a00 "free_pai_ppi_body", line=102) at mem/q_malloc.c:464 #4 0x000000000056e5e6 in free_pai_ppi_body (pid_b=0x7f12803cb480) at parser/parse_ppi_pai.c:102 #5 0x000000000054fee0 in clean_hdr_field (hf=0x7f1274c3c238) at parser/hf.c:126 #6 0x00007f127cb6dde6 in acc_onreply (t=0x7f1274c157f0, req=0x7f1274c3ac08, reply=0x7f12804a6d70, code=200) at acc_logic.c:501 #7 0x00007f127cb6e30a in tmcb_func (t=0x7f1274c157f0, type=512, ps=0x7fff0b015580) at acc_logic.c:573 #8 0x00007f127ed68478 in run_trans_callbacks_internal (cb_lst=0x7f1274c15860, type=512, trans=0x7f1274c157f0, params=0x7fff0b015580) at t_hooks.c:290 #9 0x00007f127ed6868a in run_trans_callbacks_with_buf (type=512, rbuf=0x7f1274c158b0, req=0x7f1274c3ac08, repl=0x7f12804a6d70, flags=200) at t_hooks.c:336 #10 0x00007f127ed9ac06 in relay_reply (t=0x7f1274c157f0, p_msg=0x7f12804a6d70, branch=0, msg_status=200, cancel_data=0x7fff0b0158e0, do_put_on_wait=1) at t_reply.c:2001 #11 0x00007f127ed9d0b7 in reply_received (p_msg=0x7f12804a6d70) at t_reply.c:2499 #12 0x000000000045d837 in do_forward_reply (msg=0x7f12804a6d70, mode=0) at forward.c:777 #13 0x000000000045e0f8 in forward_reply (msg=0x7f12804a6d70) at forward.c:860 #14 0x00000000004a58e7 in receive_msg (buf=0x924600 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP A.B.C.D;branch=z9hG4bK512b.82b197888826f6b60c0c63b79801294d.0;received=A.B.C.D\r\nVia: SIP/2.0/UDP A.B.C.D:2057;branch=z9hG4bK-129F259C;rport=2057\r\nCall-ID: cb0"..., len=1124, rcv_info=0x7fff0b015c60) at receive.c:273 #15 0x000000000053c9a8 in udp_rcv_loop () at udp_server.c:536 #16 0x000000000046d42b in main_loop () at main.c:1617 #17 0x0000000000470533 in main (argc=7, argv=0x7fff0b015f98) at main.c:2545 It seems to be the same problem but in a different source. Can you help me? Regards, Igor. 2014-06-12 17:46 GMT+02:00 Igor Potjevlesch <igor.potjevlesch(a)gmail.com>: > Hello, > > we didn't set the early media parameter . its '0' by default, isn't it? > > regards, > > Igor > > > > > 2014-06-12 17:03 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, >> >> if you get a record for 180 response, then you have also the early_media >> parameter set for acc module, isn't it? >> >> In the morning I pushed a patch that should fix this issue. Use latest >> release 4.1.4 and see if works fine. Report the results to know that it is >> gone or not. >> >> Cheers, >> Daniel >> >> >> On 12/06/14 16:55, Igor Potjevlesch wrote: >> >> Hello, >> >> We don't use $ai in xlog nor in other process. only in ACC. >> >> parameters for ACC are : >> modparam("acc", "db_flag", FLT_ACC) >> modparam("acc", "db_missed_flag", 3) >> modparam("acc", "db_url", DBURLWO) >> modparam("acc", "db_extra", >> "src_user=$fU;username=$Au;src_domain=$fd;src_ip=$si;src_pai=$ai;" >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") >> >> For the 3781-4b1-572014182635-OGNAJ-1-A.B.C.D there is a code 180 >> ringing in the INVITE in ACC table. >> >> regards, >> >> Igor >> >> >> 2014-06-11 23:01 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> Few more things... >>> >>> Are you recording 1xx events? Can you check to see if there is another >>> record in acc table for the same call? You can search by call-id >>> 3781-4b1-572014182635-OGNAJ-1-A.B.C.D >>> >>> Eventually, send also the parameters you set for acc module. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 11/06/14 19:25, Daniel-Constantin Mierla wrote: >>> >>> Hello, >>> >>> so you don't print $ai in xlog() statements nor use it in any >>> assignments or other functions besides acc parameter? >>> >>> Cheers, >>> Daniel >>> >>> On 11/06/14 19:19, Igor Potjevlesch wrote: >>> >>> Hello, >>> >>> We do not access to the P-asserted-identity in our configuration but >>> we added the field PAI in the db base ACC ( for INVITE, ACK and BYE) . >>> I dont know if it's in request_route, failure_route or branch_route . >>> >>> This is the print : >>> >>> (gdb) p mem_block >>> $3 = (struct qm_block *) 0x7f6a6bef1010 >>> (gdb) p shm_block >>> $4 = (struct qm_block *) 0x7f6a5666a000 >>> >>> Regards, >>> >>> Igor >>> >>> >>> 2014-06-11 18:02 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>> >>>> Hello, >>>> >>>> cloning to shm for tm seems ok. Can you tell where you access >>>> P-Asserted-Identity header, via variables? Does it happen in request_route, >>>> failure_route or branch_route? >>>> >>>> Can you print from gdb, any frame: >>>> >>>> p mem_block >>>> p shm_block >>>> >>>> I want to see if parsed filed point to shm or pkg memory. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 11/06/14 17:37, Daniel-Constantin Mierla wrote: >>>> >>>> Hello, >>>> >>>> at least I narrowed it down a bit. It is empty also in the clone stored >>>> in transaction, so it happens either during cloning or before. I will have >>>> to check these parts. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 11/06/14 17:00, Igor Potjevlesch wrote: >>>> >>>> Hello, >>>> >>>> This is the result, always for frame 5 : >>>> >>>> (gdb) p *t->uas.request->pai >>>> $1 = {type = HDR_PAI_T, name = { >>>> s = 0x7f6a60cd34b8 "P-Asserted-Identity: \"0987654321\" >>>> <sip:0987654321@A.B.C.D>\r\nContact: <sip:0987654321@A.B.C.D:5060>\r\nAllow: >>>> INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, >>>> REFER, UPD"..., len = 19}, body = { >>>> s = 0x7f6a60cd34cd "\"0987654321\"<sip:0987654321@A.B.C.D>\r\nContact: >>>> <sip:0987654321@A.B.C.D:5060>\r\nAllow: INVITE, BYE, REGISTER, ACK, >>>> OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE\r\nSupported: >>>> path,"..., len = 43}, len = 66, parsed = 0x7f6a6d81da88, next = >>>> 0x7f6a60cd3f10} >>>> >>>> (gdb) p *((p_id_body_t*)(t->uas.request->pai->parsed)) >>>> $2 = {id = 0x0, num_ids = 0, next = 0x0} >>>> >>>> *Did *you find one thing in common between the 2 occurrences? Do you >>>> have any ideas about what is the cause of this pai reset? >>>> >>>> Regards, >>>> >>>> Igor >>>> >>>> >>>> >>>> >>>> 2014-06-11 16:09 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com> >>>> : >>>> >>>>> Hello, >>>>> >>>>> in the same frame 5, can yo get: >>>>> >>>>> p *t->uas.request->pai >>>>> p *((p_id_body_t*)(t->uas.request->pai->parsed)) >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> >>>>> On 10/06/14 18:35, Igor Potjevlesch wrote: >>>>> >>>>> Hello, >>>>> >>>>> Here is the results : >>>>> >>>>> (gdb) frame 5 >>>>> #5 0x00007f6a687e9b43 in acc_onreply (t=0x7f6a60d16ff8, >>>>> req=0x7f6a60cd2c10, reply=0x7f6a6c119aa8, code=200) at acc_logic.c:471 >>>>> 471 acc_db_request(req); >>>>> >>>>> >>>>> >>>>> -- >>>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>> >>>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>> >>>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >

2 24

uri2dst: bad_uri
by Yuriy Gorlichenko 10 Oct '14

10 Oct '14

Hello. We use 2 kamailio behind load balanser that have domain name of our system . uac settings like this: modparam("uac","restore_mode","none") modparam("uac","reg_db_url", DBURL) modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 20) modparam("uac", "reg_retry_interval", 10) modparam("uac", "reg_contact_addr", "sip.service.com:5068") modparam("uac","auth_realm_avp","$avp(s:realm)") modparam("uac","auth_username_avp","$avp(s:uname)") modparam("uac","auth_password_avp","$avp(s:passwd)") We have a trouble with registering trunks at porvider. When UAC try to register trunk We see fails like this: tm [ut.h:272]: uri2dst2(): ERROR: uri2dst: bad_uri: sip.service.com ERROR: tm [uac.c:266]: t_uac_prepare(): t_uac: no socket found ERROR: uac [uac_reg.c:746]: uac_reg_update(): failed to send request for [TRUNKNAME] What we try: advertised_address=sip.service.com alias=sip.service.com alias=kam2.service.com alias=Kam1.extern.ip.addr alias=Kam2.extern.ip.addr alias=load.balanser.ext.ipaddr These settings not resolved our issue. We have another trouble and think that it depended issues: We can call to external numbers through trunks, but call very slow. (10-15 seconds), and some operators (LIKE AT&T dropped our calls to voicemail systems and somethink like that) Our question is how to successfully register providers at UAC with this scheme. thanks for advices.

2 2

uac_from_replace does not restore when changing display name only
by Klaus Darilion 10 Oct '14

10 Oct '14

Hi! It seems that uac_replace_from does restore the original URI only if the >From URI is changed. Only changing the Display-name, eg: uac_replace_from("$avp(pad)",""); does not add the RR-cookie, thus the original will not be restored (e.g. on responses). Is this on purpose or a bug? regards Klaus PS: The workaround is: uac_replace_from("$avp(pad)","$fu");

1 1

Unconditional Call forwarding
by Rajesh Sharma 10 Oct '14

10 Oct '14

Hi . I have installed Kamailio 4.1.6 and basic registration and proxy server functionalities are working fine. Now I want to simulate a Unconditional call forwarding scenario with 181-Call is being forwarded is reported to originator from server. Thanks

3 3

Kamailio With RTP proxy BYE not receiving
by balu 10 Oct '14

10 Oct '14

Hi I am using kamailio with rtp proxy module. I have 2 questions /issues . 1. When caller or callee ends the call the other end call is not disocnnecting . UA is pjsip based and behind NAT router. Present call flow is pjsipUA (LAN_ip)----->Router (Publicip)-------->Kamailio_with_RTP proxy----> ThridParty SIP Server UA local ip : 192.168.2.11 UA public IP : 89.78.92.23 Kamailio Public ip: 94.50.203.32 Third party Sip server : 76.42.89.25 Here When I disconnect call from either side , it is not disconnecting other side . 2. My second requirement is , how can I define port of third party server . for example if have 3 or 4 sip servers with different sip registration ports other tahn 5060 How can I route registration requests coming from UAs to different ports of third party servers. Please bear my ignorance I am new to kamailio .Hope some experts will help me here . Attached kamailio config and SIP trace taken from kamailio server Thank you

2 1

cr_next_domain not found
by mark＠brightvoip.co.uk 10 Oct '14

10 Oct '14

Hi, I am trying to use the carrierfailureroute table functionality via the cr_next_domain carrierroute function. I am getting the message: "failed to find command cr_next_domain" when I try to start Kamailio I am compiling Kamailio from latest 4.1.6 source, using: make include_modules"db_mysql carrierroute" cfg make group_include="db mysql standard" all make install I have installed libconfuse-devel and the tm module is loaded. Would welcome any helpful suggestions. Thanks, Mark Hall --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

2 1

dlg_set_timeout_by_profile not working on kamailio 4.1.5
by Jose Ferreira 10 Oct '14

10 Oct '14

I'm trying to use dlg_set_timeout_by_profile and it doesn't do nothing on Kamailio 4.1.5 If i try to use the dlg_set_timeout, it also doesn't work, but it gives the following error CRITICAL: dialog [dlg_timer.c:205]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f9825ac6880 tl->next=(nil) tl->prev=(nil) ERROR: dialog [dlg_hash.c:1094]: update_dlg_timeout(): failed to update dialog lifetime If i set the timeout with the $avp, it only works before dlg_manage(), but also with a Warning WARNING: dialog [dlg_handlers.c:1245]: dlg_onroute(): inconsitent dlg timer data on dlg 0x7f9da0dc7f08 [4066:9857] with clid 'bba73ace3ee94b95bf5b1782406047bd' and tags '6460df6b8b3947ba96b1cf65330bf524' 'm2pmrUpHN06Fc' The dialog module configurations are: modparam("dialog", "db_url", DBURL) modparam("dialog", "dlg_flag", FLT_DLG) modparam("dialog", "db_mode", 0) modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "profiles_with_value", "user ; domain ; permissions") modparam("dialog", "profiles_no_value", "inbound ; outbound ; all") modparam("dialog", "send_bye", 1) modparam("dialog", "default_timeout", 21600) modparam("dialog", "ka_timer", 0) modparam("dialog", "ka_interval", 0) modparam("dialog", "timeout_avp", "$avp(dlgtimeout)") I'm very interested to use dlg_set_timeout_by_profile, any idea about what is wrong with that ? Regards, *José Ferreira | Technical Manager* M. +351 91 775 7166 | jferreira(a)wavecom.pt Wavecom, Soluções Rádio, SA Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 234 919 190 | F. +351 234 919 191 *www.wavecom.pt <http://www.wavecom.pt/>* [image: WavecomSignature]

2 1

UAC call to AT&T
by Yuriy Gorlichenko 10 Oct '14

10 Oct '14

Hello. we use UAC module for trunk registration at kamailio. We have successfull routing and calling through our trunks to many providers and have just one problem with AT&T endpoints. When callee at AT&T endpoint hook call (I see OK from provider as response to our INVITE from AT&T endpoint) and then kamailio sends CANCEL to endpoint, but then kam sends ACK to OK... So after this strange scenario I successfully hear invite to leave voice mail to AT&T endpoint... What I saw: I look at codecs- at OK i See ulaw codec, so at my INVITE ulaw presents too. With other providers calls succesfully established. Does Anyone have an experiense with calling to AT&T from Kamailio. P.S. calls through our trunk to AT&T from asterisk establishes succesfully too.

2 1

Serial forking & q value
by Marino Mileti 10 Oct '14

10 Oct '14

Hi guys, I've tried with success to implement serial forking based on q value (i've followed documentation) with the istruction: append_branch("sip:b@example.com", "0.5"); Now my questions are: is it possible to import URI from db? is it possible to import q value from db too? I would like to change the q value based, for example, on "night & day"; so it would be nice if I can store it on DB and then load it dinamically when i need.For the first question I think that it's possible with dbaliases for example...but I don't know how get a q value for each contact :) Marino

2 1

Best way to handle 100 but no 180/183?
by Tobias 09 Oct '14

09 Oct '14

Dear Kamailio users, I have an issue with some upstream carriers quickly replying with 100 Trying but then basically times out. I'm looking for a good solution to distinguish between the 100 reply and any other replies, and alter the timeout values. My goal is to be able to re-route calls where the 180/183 is not received within a few seconds, and if the 180/183 is received I'd like to give the call a much longer timeout. I realize there is a "fr_inv_timer" that would do this, but I would not like to set that to low since I fear it would affect many other call types, i.e. if a carrier would send 183 once but for some reason does not resend on regular intervals. I'm on Kamailio 4.1 and my current settings are:fr_timer : 2000fr_inv_timer : 120000restart_fr_on_each_reply : 1 The only solution I can think of is to update the fr_inv_timer in an on_reply-route if the reply code is 180/183 but I'm not sure that would even work? Is that the way to go or is there another way? Thanks in advance!/Tobias

3 4

Setting Kamailio User-Agent dynamically
by Errol Samuels 09 Oct '14

09 Oct '14

Hello All, I am using Kamailio in front of Freeswitch and Kamailio is handling the registrations then forwarding to Freeswitch and everything is working for the most part. However, there is niggling pain in the neck issue where the user's User-Agent shows in Freeswitch as kamailio (4.0.6 (x86_64/linux)) instead of the user's Real User-Agent I am trying to get Kamailio to pass the real User-Agent of the user without success. I am able to set a static User-Agent by doing this in the global section. *user_agent_header = "User-Agent: We love beer";* However, I am unable to set it dynamically for example: *user_agent_header = "User-Agent: " + $ua;* With is one Kamailio doesn't even start so my syntax is probably wrong. I also tried playing around with the textops module so I have ran out of ideas Any ideas? Thanks in advance. regards, Errol

4 17

KAMAILIO not updating Registration status to DB
by Varghese Paul 09 Oct '14

09 Oct '14

Hi all, We are experiencing a strange problem in our Kamilio server. Some of the webrtc endpoint registrations are not updated to the Database and there is no issue with the registration. Once we flushed the registration for this endpoint then it is updating to the DB for the new Register Packet. Flushed the Registration for the endpont : kamctl ul rm <USERNAME> Kamalio Version: 4.1.2 modparam("usrloc", "db_url", DBURL) *modparam("usrloc", "db_mode", 1)* modparam("usrloc", "nat_bflag", 3) modparam("usrloc", "use_domain", 1) Is there any other settings to avoid this issue ? Regards Varghese Paul

2 2

duplicate information in siptrace table
by Fred 09 Oct '14

09 Oct '14

Hello, I use Kamailio 4.1.3 with siptrace module. In the sip_trace table I see duplicate information, one line with empty and one line with filled traced_user column. The empty lines are useless for me, I would like only entries with traced_user information. Here the siptrace part of kamailio.cfg: #### Parameters for Siptrace module modparam("siptrace", "db_url", DBURL) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_sl_acks", 0) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("siptrace", "trace_delayed", 0) ###### SIP Trace Config in main route if (!is_method("OPTIONS")) { if (!ds_is_from_list("2")) { $avp(s:user) = $fU; } else { $avp(s:user) = $rU; }; if (is_avp_set("$avp(s:user)")){ sip_trace(); #setflag(22); }; }; Has anyone a tip for me how I can get rid of the empty traced_user lines? Regards Fred

2 1

Re: [SR-Users] Segfault in 4.1.6
by David Wilson 09 Oct '14

09 Oct '14

Hi Daniel, Thanks for the response. Reviewing the logs again, the first occurrence was on 25Sep, after upgrading to 4.1.6+precise on 24Sep. Previous upgrades were to: 4.1.5+precise 18Aug 4.1.4+precise June Traffic patterns are fairly uniform over that period. I've installed the nightly (4.1.6+0~20141004025841.74+precise) and we'll see how that goes until the next release. Cheers, Dave > Date: Thu, 02 Oct 2014 09:02:14 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] Segfault in 4.1.6 > Message-ID: <542CF876.6060900(a)gmail.com> > Content-Type: text/plain; charset=utf-8; format=flowed > > Hello, > > I see it crashed in dialog, it was a fix for it after 4.1.6 for the > cleanup timer procedure, affecting some particular cases with dialog > states (e.g., dialogs not answered even after many minutes since > creation). It was there before 4.1.6 (most probably from 4.1.0), not > introduced by that version. What you were using before 4.1.6? > > You can get latest version from branch 4.1 either via GIT or via debian > nightly builds of 4.1 (if using debian). > > Otherwise, yes, a coredump with a gdb backtrace will be very helpful. > > Cheers, > Daniel > > On 02/10/14 08:31, David Wilson wrote: > > Hi all, > > > > I?ve had Kamailio running nicely for nearly a year, but have hit a segfault twice since upgrading to 4.1.6 earlier this week. > > > > The log shows: > > > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1222]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 23 > > Oct 2 11:28:03 localhost kernel: [11259.904645] kamailio[1206]: segfault at 58 ip 00007fa56fba0f86 sp 00007fffea594ca0 error 4 in dialog.so[7fa56fb63000+54000] > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: ALERT: <core> [main.c:777]: handle_sigs(): child process 1206 exited by a signal 11 > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: ALERT: <core> [main.c:780]: handle_sigs(): core was not generated > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: INFO: <core> [main.c:792]: handle_sigs(): INFO: terminating due to SIGCHLD > > Oct 2 11:28:03 localhost /usr/sbin/kamailio[1220]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received > > ?. > > > > The first two lines above seem to be the important parts. Nothing was logged for about 10s prior to this extract. An earlier instance of the issue showed almost the same details: > > > > Oct 1 20:39:12 localhost /usr/sbin/kamailio[1219]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 23 > > Oct 1 20:39:12 localhost kernel: [554502.639227] kamailio[1207]: segfault at 58 ip 00007fd62c5c6f86 sp 00007fffd77c9890 error 4 in dialog.so[7fd62c589000+54000] > > > > I?m trying to catch a core next time it occurs, no core is available yet. > > > > Is this a new issue in 4.1.6? > > Any suggestions about the likely cause? > > > > Cheers, > > Dave. > >

2 1

GIT branch 4.2 has been created
by Daniel-Constantin Mierla 08 Oct '14

08 Oct '14

Hello, short note to say that the branch 4.2 has been created, as announced on mailing list a while ago. This is the official branch to be used for releasing any of the 4.2.x versions. The first one, v4.2.0, is expected to be out in about one week. The version used now in master branch is prefixed with 4.3.0, which is expected to be the next future major release. With this change, practically commits with new features to be part of 4.3.0 can be pushed to master branch from now on. Any fix that is applicable for both master and 4.2 branches has to be committed first in master branch, then cherry-picked or merged to branch 4.2. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Multiple kamailio sharing single Registration DB
by Varghese Paul 08 Oct '14

08 Oct '14

Hi all, How i can setup multiple kamailio servers should share the single registration DB. Requirement: I have two kamailio instances Kamailio-A and Kamalio-B ,both are sharing the same DB which is postgre*SQL *which is hosted in some other server. When a USER AGENT A send a register packet to Kamailio-A it is updating the registration status in the DB. But in kamailio-B if am doing a lookup(location) i am getting -1 as return value. kamctl ul show <USER_AGENT> 404 AOR not found 1. How kamailio retrieving the registrations status ? Is it from the DB or from the internal cache ? 2. How can i read the registration status from the DB always ? Let me know if any one need any more information. Regards Varghese Paul

2 1

chane source IP for REGISTER to provider
by Yuriy Gorlichenko 08 Oct '14

08 Oct '14

Hello. We have multiple kamailio behind load balancer. We use UAC to send REGISTER procedure to provider. Some providers drop wrong ip. So We have loadbalanser with external IP and have kamailio servers with external IPs too. So we need to send REGISTER packets to porovide with source IP of our Load balancer. Now kamailio send REGISTER pakets with its external IP. Question is how to send REGISTER with ip adress of loadbalanser? (allias with it dn and ip already exists at config) (listet already had it ip too) thanks/

2 1

Kamailio -topoh- and Via headers
by Gonzalo Gasca 08 Oct '14

08 Oct '14

I'm using Kamailio as SIP Registrar using Websockets. My topology looks like this: Sip client (sipml5) ---> wss ---> Nginx ---> ws ---> Kamailio 4.1.5 When I look into my SipMl5 application in the Register Message 200 OK from Kamailio I see the Nginx private IP address 172.31.22.2 Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=37111;received=172.31.22.2;branch=z9hG4bKtv75otkzmPVsdNWevweLt4TN9JnLnQ0p How can I remove private IP Address in Via header to achieve topology hiding? >From Kamailio logs: Oct 6 00:34:21 ip-172-31-27-85 /usr/sbin/kamailio[8941]: DEBUG: registrar [reply.c:374]: build_contact(): created Contact HF: Contact: <sips:gogasca@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;expires=200#015#012 Oct 6 00:34:21 ip-172-31-27-85 /usr/sbin/kamailio[8941]: DEBUG: sl [sl.c:288]: send_reply(): reply in stateless mode (sl) Oct 6 00:34:21 ip-172-31-27-85 /usr/sbin/kamailio[8941]: DEBUG: <core> [msg_translator.c:204]: check_via_address(): check_via_address(172.31.22.2, df7jal23ls0d.invalid, 0) O Version: kamailio 4.1.5 (x86_64/linux) # ------ topoh -------- modparam("topoh", "mask_key", "opencall") modparam("topoh", "mask_ip", "<Public IP Address of Kamailio>") modparam("topoh", "vparam_prefix", "llamato") modparam("topoh", "mask_callid", 1) modparam("topoh", "sanity_checks", 1)

2 6

Re: [SR-Users] Presence PUBLISH error
by Marino Mileti 08 Oct '14

08 Oct '14

Thanks Daniel, I'm trying to use sqlite (mysql is too "heavy" for my environment) but usually I receive a lot of commit failed because database is locked (especially in presentity.c and publish.c) Is there any way to manage, maybe i've missed some sqlite configuration parameters? ----Messaggio originale---- Da: miconda(a)gmail.com Data: 6-ott-2014 18.17 A: "Marino Mileti"<marino.mileti(a)alice.it>, "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] Presence PUBLISH error hello, It seems that the column type and value type are mismatching, but they can still be compatible. The fact is that db_text was not designed to handle large data values, so not tested with presence. Eventually you may want use sqlite (or other db module) instead or go into db_text and update as needed. Cheers, Daniel On 06/10/14 15:14, Marino Mileti wrote: Hi guys, with Kamailio 4.1.6 standard configuration and dbtext as db module, when server receives a publish message on the log there's: ERROR: db_text [dbt_base.c:391]: dbt_insert(): incompatible types v[5] - c[7]! ERROR: presence [presentity.c:402]: update_presentity(): inserting new record in database ERROR: presence [publish.c:464]: handle_publish(): when updating presentity and the server reply with 500 Server Internal Error. My conf is standard I've only used DBURL to point to my dbtext files: I've enable presence and presence_xml module # ----- presence params ----- modparam("presence", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence", "db_table_lock_type", 0) modparam("presence_xml", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence_xml", "force_active", 1) Any idea? Marino _______________________________________________ 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

rtpengine with rejected re-invites to new RTP ports
by Jeff Pyle 08 Oct '14

08 Oct '14

Hello, Given the following scenario with Kamailio and rtpengine in the middle: - call establishes with G.711 RTP - b-leg re-invites to T.38, indicating a different port number then he is using for G.711 - a-leg refuses the re-invite with a 488 - call continues using G.711 on original port numbers as if re-invite never happened Will Kamailio and rtpengine handle it? Today I have a functional configuration with rtpproxy on another SIP proxy. It handles every scenario I have tried except this one. rtpproxy sees the new ports in the re-invite and adjusts its session accordingly. If the re-invite is rejected, the old media ports are no longer valid through rtpproxy, and the call fails. Is there an approach I can take with Kamailio and rtpengine to allow this scenario to succeed? Regards, Jeff

2 2

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis
by cpcnetworking 07 Oct '14

07 Oct '14

Thx, exactly what I was looking for!

1 0

tcp_read_data() errors in syslog
by Petr.Wozniak＠seznam.cz 07 Oct '14

07 Oct '14

Hello All, I have installed kamailio with TLS module together with siremis on server and all seems works fine except that sometimes appear in syslog file the following errors: /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection timed out (110) /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:1281]: tcp_read_req(): ERROR: tcp_read_req: error reading /usr/sbin/kamailio[3268]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection reset by peer (104) Some part of syslog please you find enclosed. I don't know where I should search for the cause of these errors and how to debug and remove them. I don't know if these errors are caused by incorrect kamailio configuration or are caused on client's side. Thank you for your opinions and help in advance Regards, Petr

3 2

Adding Dynamic Routing Module management to Siremis
by cpcnetworking 07 Oct '14

07 Oct '14

I'm using Kamailio 4.15 with Siremis 4.1.0 both are great BTW! I'd like to add management of the Kamailio Dynamic Routing Module to Siremis web manager Any pointers on how to do this; I'm not familiar with OpenBiz Thx in advance

2 1

multiple endpoints call
by Marino Mileti 07 Oct '14

07 Oct '14

Take a look at : modparam("registrar", "max_contacts", x) to have up to "x" contacts for AOR and in your REGISTRAR route if your save("location") has some extra parameters (for example the 0x04 used to restrict one single record in loaction table ----Messaggio originale---- Da: ovoshlook(a)gmail.com Data: 6-ott-2014 22.02 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: [SR-Users] multiple endpoints call Hello. I have multiple endpoints registered at my kamailio with one account (for example user1 registerd from norhway, USA, and Russia at one time), so when I call from user2 to user1 I want to ring all endpoints registered by user1 account. Now I can ring only one, first entry at location table. What setting may I use for this? Thanks

1 0

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis
by cpcnetworking 07 Oct '14

07 Oct '14

ok thanks

1 0

tcp_read_data() errors in syslog
by Petr.Wozniak＠seznam.cz 06 Oct '14

06 Oct '14

Hello All, I have installed kamailio with TLS module together with siremis on server and all seems works fine except that sometimes appear in syslog file the following errors: /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection timed out (110) /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:1281]: tcp_read_req(): ERROR: tcp_read_req: error reading /usr/sbin/kamailio[3268]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection reset by peer (104) Some part of syslog please you find enclosed. I don't know where I should search for the cause of these errors and how to debug and remove them. I don't know if these errors are caused by incorrect kamailio configuration or are caused on client's side. Thank you for your opinions and help in advance Regards, Petr

1 0

Presence PUBLISH error
by Marino Mileti 06 Oct '14

06 Oct '14

Hi guys, with Kamailio 4.1.6 standard configuration and dbtext as db module, when server receives a publish message on the log there's: ERROR: db_text [dbt_base.c:391]: dbt_insert(): incompatible types v[5] - c[7]! ERROR: presence [presentity.c:402]: update_presentity(): inserting new record in database ERROR: presence [publish.c:464]: handle_publish(): when updating presentity and the server reply with 500 Server Internal Error. My conf is standard I've only used DBURL to point to my dbtext files: I've enable presence and presence_xml module # ----- presence params ----- modparam("presence", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence", "db_table_lock_type", 0) modparam("presence_xml", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence_xml", "force_active", 1) Any idea? Marino

2 1

Problem when I want install kamailio on CENTOS 7 x64
by ANTHONY HESNAUX 06 Oct '14

06 Oct '14

Hello all, I want to install kamailio on CENTOS 7 x64 but I have this problem when I execute "kamdbctl create » Can you help me ?? Thank YOU Anthony Console display: [root@localhost kamailio-4.0.1]# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. INFO: creating database kamailio ... Warning: Using a password on the command line interface can be insecure. INFO: granting privileges to database kamailio ... Warning: Using a password on the command line interface can be insecure. INFO: creating standard tables into kamailio ... Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. ERROR 1101 (42000) at line 2: BLOB/TEXT column 'body' can't have a default value ERROR: Creating core tables failed at msilo! [root@localhost kamailio-4.0.1]#

2 1

BYE issue - Kamailio +RTP proxy
by balu 06 Oct '14

06 Oct '14

Dear experts , I am using kamailio with rtp proxy module. I have 2 questions /issues . 1. When caller or callee ends the call the other end call is not disocnnecting . UA is pjsip based and behind NAT router. Present call flow is pjsipUA (LAN_ip)----->Router (Publicip)-------->Kamailio_with_RTP proxy----> ThridParty SIP Server UA local ip : 192.168.2.11 UA public IP : 89.78.92.23 Kamailio Public ip: 94.50.203.32 Third party Sip server : 76.42.89.25 Here When I disconnect call from either side , it is not disconnecting other side . 2. My second requirement is , how can I define port of third party server . for example if have 3 or 4 sip servers with different sip registration ports other tahn 5060 How can I route registration requests coming from UAs to different ports of third party servers. Please bear my ignorance I am new to kamailio .Hope some experts will help me here . Attached kamailio config and SIP trace taken from kamailio server Thank you INVITE sip:18792356789@76.42.89.25:5060 SIP/2.0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p. Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25 Contact: <sip:test@192.168.2.11:58429;ob;alias=89.78.92.23~58429~1> Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12140 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9 Content-Type: application/sdp Content-Length: 369 P-hint: outbound v=0 o=- 3620097527 3620097527 IN IP4 94.50.203.32 s=pjmedia c=IN IP4 94.50.203.32 t=0 0 a=X-nat:0 m=audio 60822 RTP/AVP 104 18 0 8 96 c=IN IP4 94.50.203.32 a=rtcp:60823 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=nortpproxy:yes SIP/2.0 407 Proxy Authentication Required CSeq: 12140 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p. From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Proxy-Authenticate: DIGEST realm="sip.testcalls.com", nonce="141110871219020226209383237537" Content-Length: 0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> ACK sip:18792356789@76.42.89.25:5060 SIP/2.0 Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12140 ACK Content-Length: 0 INVITE sip:18792356789@76.42.89.25:5060 SIP/2.0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.4f21;nat=yes> Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25 Contact: <sip:test@192.168.2.11:58429;ob;alias=89.78.92.23~58429~1> Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12141 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9 Proxy-Authorization: Digest username="test", realm="sip.testcalls.com", nonce="141110871219020226209383237537", uri=" sip:18792356789@76.42.89.25:5060", response="e98f243028e20a3d864dc54149db8ab1" Content-Type: application/sdp Content-Length: 369 P-hint: outbound v=0 o=- 3620097527 3620097527 IN IP4 94.50.203.32 s=pjmedia c=IN IP4 94.50.203.32 t=0 0 a=X-nat:0 m=audio 47574 RTP/AVP 104 18 0 8 96 c=IN IP4 94.50.203.32 a=rtcp:47575 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=nortpproxy:yes SIP/2.0 183 Session Progress CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 200 OK CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv ACK sip:76.42.89.25:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.3f67e202ad7510de3a935364e562755c.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPjpDeX-U.AQN7jSbWJ7Tqg4Sbl0Lby6nkE Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12141 ACK Content-Length: 0 BYE sip:test@192.168.2.11:58429;ob;alias=89.78.92.23%7E58429%7E1 SIP/2.0 Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP 76.42.89.25:5060;branch=z9hG4bK190938140255193499457203 From: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Content-Length: 0 Max-Forwards: 70 SIP/2.0 200 OK CSeq: 1 BYE Via: SIP/2.0/UDP 76.42.89.25:5060 ;branch=z9hG4bK190938140255193499457203;rport=5060 From: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Server: kamailio (4.1.3 (i386/linux)) Content-Length: 0 ----------- config file below ----------- #!KAMAILIO # check_via=no rev_dns=no dns=no #!define WITH_NAT #!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 ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### #!ifdef WITH_MYSQL #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif #!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 ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!define WITH_DEBUG #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=2 memlog=2 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes #sreekanth commented no auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" listen=udp:94.50.203.32:8304 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=3074 #!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 ####### 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" pstn.gw_port = "" desc "PSTN GW Port" #!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 ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.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 "mangler.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 # ----------------- 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) modparam("rr", "append_fromtag", 1) #sreekanth changed above value from 0 to 1 # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(i:42)") #sreekanth changed above $avp(RECEIVED) to $avp(i:42) 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 #added by sreekanth #!define DLG_FLAG 28 #!define CC_FLAG 29 loadmodule "dialog.so" modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) loadmodule "rtimer.so"; #!ifdef CNXCC_CHANNEL modparam("rtimer", "timer", "name=ta;interval=1;mode=1;") modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT") #!endif loadmodule "cnxcc.so" modparam("cnxcc", "dlg_flag", CC_FLAG) modparam("cnxcc", "credit_check_period", 1) #check every 1 second #sreekanth-End ####### 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 { #setflag(DLG_FLAG); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); if (is_method("INVITE")) { route(RELAY); } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) 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")) { sl_send_reply("100", "Trying"); 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==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); #t_newtran(); #t_reply("200", "OK"); # exit; } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } 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 route(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(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # 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 $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); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!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_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$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 { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } #if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) #{ # rtpproxy_manage(); # return; #} #from here sreekanth commented #if (is_request()) { #rtpproxy_manage("co"); } #if (is_reply()) { #rtpproxy_manage("z50"); } #if (is_request()) { # if (!has_totag()) { # if(t_is_branch_route()) { # if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { # add_rr_param(";nat=yes"); # } # } # } # } #sreekanth till here if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # 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}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($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|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $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")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

1 0

Simplest config possible [cont]
by White, Phil 06 Oct '14

06 Oct '14

Hi All, I've had an 'interesting' weekend reading RFCs and playing with kamailio. I'm making progress - however, I would be grateful for a little clarification. I have sorted out a very basic stateless redirect proxy (config file below). I have cut it down to the absolute basic minimum - and it mostly appears to do what I would expect it to do. It does allow me to call any registered extension from my SIP softphone. Problem 1: I have a Grandstream Budgetone 200, which will happily receive calls, but will not initiate any. Looking at a ngrep, it is ignoring the redirect, and sends out a flood of UDP INVITEs (also see below for details) Question: Is this the fault of my configuration, or the Grandstream behaving incorrectly? Problem 2: I am having difficulty in working out which module and command I require to perform a function. Could someone please indicate the INVITE route in this file for a stateful proxy? Many thanks, Phil --- ## WARNING - DO NOT USE THIS IN A PRODUCTION SERVER. FOR TESTING & EDUCTION ONLY debug=2 fork=no log_stderror=yes listen=192.168.1.254:5060 ####### Modules Section ######## mpath="/usr/lib/kamailio/modules/" loadmodule "pv.so" loadmodule "sl.so" loadmodule "usrloc" modparam("usrloc", "db_mode", 0) loadmodule "xlog.so" loadmodule "registrar.so" ####### Routing Logic ######## route{ if(method=="REGISTER") { if (!save("location")) { xlog("L_ERR", "**REGISTER: save failed \n"); sl_reply_error(); } else { xlog("L_NOTICE", " REGISTER rcvd $si:$sp \n"); }; }; if(method=="INVITE") { lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; sl_send_reply("302","Redirect"); }; if(method=="PUBLISH" || method=="SUBSCRIBE"){ xlog("L_INFO","REJECTION. We do not process $rm requests. \n"); route(No); }; } route[No] { sl_send_reply(503, "Service Unavailable"); } --- U 2014/10/05 16:38:04.925388 192.168.123.16:5060 -> 192.168.123.254:5060 INVITE sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKaf202530a743a053. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP>. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: replaces, timer, 100rel, path. Session-Expires: 180. Min-SE: 90. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 INVITE. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 407. <...SNIP...> U 2014/10/05 16:38:04.925895 192.168.123.254:5060 -> 192.168.123.16:5060 SIP/2.0 302 Redirect. Via: SIP/2.0/UDP 192.168.123.16:5060 ;branch=z9hG4bKaf202530a743a053;rport=5060. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>;tag=71d9875c03b727cbe0217d2db6ac2a49.422e. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 INVITE. Contact: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>. Server: My SIP Server. Content-Length: 0.U 2014/10/05 16:38:04.954405 192.168.123.16:5060 -> 192.168.123.254:5060 ACK sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKaf202530a743a053. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>;tag=71d9875c03b727cbe0217d2db6ac2a49.422e. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: path. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 ACK. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Length: 0. . U 2014/10/05 16:38:04.975398 192.168.123.16:5060 -> 192.168.123.254:5060 INVITE sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKa55f284fa742ad32. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=9a25392722276c54. To: <sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP>. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: replaces, timer, 100rel, path. Session-Expires: 180. Min-SE: 90. Call-ID: f2989990cfa8525c(a)192.168.123.16. CSeq: 1445 INVITE. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 407. . . . etc...

4 5

how to always use rtpproxy-ng in kamailio
by andrew 05 Oct '14

05 Oct '14

Hi, I have one question about kamailio. rtpproxy-ng is installed with kamailio. When client A initiates one call to client B, SDP in invite/200ok are rewritten, where ice candidates for rtpproxy-ng are added. Sometimes rtp packets are relayed through rtpproxy-ng, but in some cases rtp packets pass through route created by ICE algorithm. How to set kamailio, then rtpproxy-ng is always used? B.R. andrew

2 1

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis
by cpcnetworking 04 Oct '14

04 Oct '14

Anybody have some insight or pointers? Thx

2 1

channel parameter on SDP
by Kelvin Chua 03 Oct '14

03 Oct '14

Hi guys, my provider is being a dick. they do not accept doubango's a=rtpmap:0 PCMU/8000/1 what they wanted was a=rtpmap:0 PCMU/8000 is there a way of changing this on kamailio? Kelvin Chua

2 1

Routing calls to a subscribed softphone
by Will Ferrer 03 Oct '14

03 Oct '14

Hi I was wondering if any one had any advice or examples for me of how to get a call to be routed to a subscribed softphone. We have 2 boxes in our testing deployment, a load balancer / sbc and a call processing box. Calls come in to the sbc, and then are passed to the call processing box. The call is analyzed and the branch uri is rewritten to a destination address when applicable for the call (this is how we handle routing of calls to certain numbers -- we do this utilizing custom code and a custom db). This works just fine when the destination sip uri is phone number (in which case we do lcr) or if the destination goes to a remote address. However when the destination is a subscriber: sip:[subscriber_name]@[our_domain_name], the call is passed back to the sbc, which passes it to the callprocessing box, back and forth until a too many hops error occurs. The subscriber I am trying to send the call too does show up under "kamctl ul show". I feel like there is something basic I must be missing here. Does any one have any advice for me? Thank you very much in advance. All the best. Will Ferrer

2 5

Enable SipTrace
by Bruno Emer 03 Oct '14

03 Oct '14

Hello! I just got some kamailio servers working with websockets using jssip in my clients and now I want to get the SIP Messages to better understand how it flows between my servers. I have enabled the "siptrace" module, and I can now save information on my database, but the problem is that I can't read the SIP Messages there, as it saves some strings in my database. The SIP messages appears like this store in the sip_trace table: \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 I am currently using postgres as my database, and to get this I just have: - Define "WITH_SIP_TRACE" in the beginning of my configuration file: #!KAMAILIO #!define WITH_PGSQL ##!define WITH_DEBUG #!define WITH_SIP_TRACE - loadmodule: #!ifdef WITH_SIP_TRACE loadmodule "siptrace.so" #!endif - Configure SIP_TRACE params: #!ifdef WITH_SIP_TRACE # ---- sip_trace params ---- modparam("siptrace", "db_url", "postgres://kamailio:kamailiorw@localhost /kamailio") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) #!endif - Get the messages: request_route { sip_trace(); setflag(22); route(REQINIT); ... Does anyone knows if is there something wrong in this configuration, or what can I do to fix this and have the SIP messages stored in my database? Thanks! *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

2 2

Re: [SR-Users] Kamailio & queue
by Robert Boisvert 03 Oct '14

03 Oct '14

Only with rtpproxy. On Fri, Oct 3, 2014 at 5:25 AM, Marino Mileti <marino.mileti(a)alice.it> wrote: > mohqueue module is also compatible with rtpproxy-ng or only with rtppoxy? > > Because during startup i've noticed this error: > mohqueue : Unable to load rtpproxy_stream2uac > > > > > >There was a typo in the parameter name. If you use default values, just > remove the modparam. > >Otherwise, I pushed the patches to master and 4.1 branches. So you would > need to upgrade to latest git version. > >Cheers, > >Daniel > >>On 02/10/14 10:15, Marino Mileti wrote: > >>I don't need any IVR so I would like to try this module and following doc > i've setup dbtable name parameters: > >>modparam ("mohqueue", "db_qtable", "mohqueues") > >>modparam ("mohqueue", "db_ctable", "mohcalls") > >>but when start Kamailio says: > >>ERROR: <core> [modparam.c:156]: set_mod_param_regex(): > set_mod_param_regex: parameter <db_qtable> of type <1> not found in module > <mohqueue> > >>What's wrong? > > 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 > > > > --- > Questa e-mail è priva di virus e malware perché è attiva la protezione > avast! Antivirus. > http://www.avast.com > > > _______________________________________________ > 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

[OT] About next Kamailio World, May 27-29, 2014, in Berlin
by Daniel-Constantin Mierla 03 Oct '14

03 Oct '14

Hello, we have preliminarily booked May 27-29, 2014 for the next Kamailio World event, hosted again by FhG Fokus Institute in Berlin. It was really hard to find a time frame during April and May 2015, because of different dates for Catholic and Orthodox Easters, 1st of May as well as quite some many public holidays in Germany. I am writing to see if anyone is aware of any major blocker for that time frame. Write me directly and we will see if anything can be done, selected the best options based on constraints. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Kamailio & queue
by Marino Mileti 03 Oct '14

03 Oct '14

I don't need any IVR so I would like to try this module and following doc i've setup dbtable name parameters: modparam ("mohqueue", "db_qtable", "mohqueues") modparam ("mohqueue", "db_ctable", "mohcalls") but when start Kamailio says: ERROR: <core> [modparam.c:156]: set_mod_param_regex(): set_mod_param_regex: parameter <db_qtable> of type <1> not found in module <mohqueue> What's wrong? ----Messaggio originale---- Da: miconda(a)gmail.com Data: 2-ott-2014 9.44 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org>, "Marino Mileti"<marino.mileti(a)alice.it> Ogg: Re: [SR-Users] Kamailio & queue Thats true if you need to answer and do some IVR, otherwise you can hold the INVITE or other request (until it is cancelled or times out). Specific on a call queue, may worth checking the module: - http://kamailio.org/docs/modules/stable/modules/mohqueue.html Holding the invite in config file is possible with: - t_suspend() and then the (index, label) identification of transaction -- see tmx module - add (index,label) to mqueue - when the routing of invite has to be resumed, fetch from the queue and do t_continue() Cheers, Daniel

2 2

simplest config file possible - not working!
by White, Phil 02 Oct '14

02 Oct '14

Sorry Guys, I'm very new to this. Taking a dive 'under the hood', to try to learn what is going on, and to see what I can achieve. I have 1 sip device ( voip telephone) on the network, constantly firing out REGISTER requests (I have turned off my asterisk server!). That's good. I have a terminal window open, running tcpdump. That's good too - I can see what is going on. I also have kamailio - giving me absolutely nothing. It's not complicated config file. In fact it is the simplest I could do. However, it logs nothing - and i am expecting it to. Can someone put me out of my misery, and point out the glaring error I have? How can I send messages to STDERR while testing? Cheers, Phil ---- ## WARNING - FOR TESTING & EDUCTION ONLY ####### Global Parameters ######### debug=5 fork=no log_stderror=yes port=5060 ####### Modules Section ######## # set module path mpath="/usr/local/lib/kamailio/modules/" loadmodule "sl.so" ####### Routing Logic ######## route{ if(method=="REGISTER") { log(1,"this SIP request is a REGISTER message\n"); sl_send_reply("404", "No registrar"); }; if (af!=INET6) { log(1,"Message received over IPv4 link\n"); }; if(proto==UDP) { log(1,"SIP message received over UDP\n"); }; if(status=="200") { log(1,"this is a 200 OK reply\n"); }; if(uri==myself) { log(1,"the request is for local processing\n"); }; }

3 5

RFC: updating default values
by Daniel-Constantin Mierla 02 Oct '14

02 Oct '14

Hello, I am opening this discussion to decide if there is need to adjust some of the default values we have in Kamailio. Many of them were set even like 10 years ago, so they might not be very actual anymore. The main goal is to get the best possible settings for common usage. To start with, here are some values that should be reviewed: - default private memory is 4MB - if the config is not that small, it might not be sufficient free pkg to play with (e.g., for sql_query() result, storage of $var(...) values). Should it be increased to 8MB or other value? Debian/Centos have startup script that sets its value to 8 via command line - default shared memory is 32MB - for a decent deployment with tm, location, lcr/dispatcher, permissions, and anti-flood, it might leave not much free space. Should we double it or set to a different value - usrloc - db_ops_ruid should enabled (1) - seems stable, without it there are problems updating/deleting location records when UA changes the call-id for same contact address. - usrloc - hash_size - now is 9, which results in 512 slots, meaning is ok for few thousands of registered users, for more, performance will decrease when doing save/lookup location -- should it be made 10 (1024 slots for internal hash table)? - auth_db - load_credentials defaults to 'rpid', meaning that the query to get the password will retrieve also the rpid column. I haven't see rpid being used that much lately (replaced by PAI/PPI). I would make this parameter empty by default to avoid querying for an extra column that is likely to be empty. Perhaps there are more, I just wanted to get started. Reply with your comments to above list as well as add new items you thinks their default values should be adjusted. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 6

Problem with locally generated BYE in dialog timeout.
by Julia Boudniatsky 02 Oct '14

02 Oct '14

Hello, We use dialog module in our configuration. uac_replace_from/to is called after dlg_manage(). restore_mode is auto. URIs are modified automatically in all subsequent local requests, exclude BYE generated in dlg timeout. In this case original URIs are sent to both caller and calee sides. If call going through only one kamailio server, all works fine. Problem starts with a chain of servers with from/to manipulation. When timeout occurs in one kamailio server, calee kamailio server receives BYE with original (not modified) URIs and URI restore gets wrong parsing result. Call clears in caller side and stay connected in caller. Whether there is any solution? BR , Julia.

3 4

Kamailio & queue
by Marino Mileti 02 Oct '14

02 Oct '14

Hi, Is it possible to use kamailio to manage a very simple queue? All calls inbound go directly in queue and then the system to to extract calls and route to agent. A very simple call center. I've seen that there's a mohqueue module...is there any conf example? Many thanks

3 2

event_route[core:receive-parse-error] question
by Julia Boudniatsky 02 Oct '14

02 Oct '14

Hello, I try to use event_route[core:receive-parse-error] in 4.1.6 event_route[core:receive-parse-error] { xlog("L_WARN", "Event-parse-error: $rm from $avp(inc_carrier)/n$mb/n"); } When core parse error occurs , *log message is not printed*. What is wrong? Thank you, Julia

2 3

Segfault in 4.1.6
by David Wilson 02 Oct '14

02 Oct '14

Hi all, I’ve had Kamailio running nicely for nearly a year, but have hit a segfault twice since upgrading to 4.1.6 earlier this week. The log shows: Oct 2 11:28:03 localhost /usr/sbin/kamailio[1222]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 23 Oct 2 11:28:03 localhost kernel: [11259.904645] kamailio[1206]: segfault at 58 ip 00007fa56fba0f86 sp 00007fffea594ca0 error 4 in dialog.so[7fa56fb63000+54000] Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: ALERT: <core> [main.c:777]: handle_sigs(): child process 1206 exited by a signal 11 Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: ALERT: <core> [main.c:780]: handle_sigs(): core was not generated Oct 2 11:28:03 localhost /usr/sbin/kamailio[1175]: INFO: <core> [main.c:792]: handle_sigs(): INFO: terminating due to SIGCHLD Oct 2 11:28:03 localhost /usr/sbin/kamailio[1220]: INFO: <core> [main.c:843]: sig_usr(): INFO: signal 15 received …. The first two lines above seem to be the important parts. Nothing was logged for about 10s prior to this extract. An earlier instance of the issue showed almost the same details: Oct 1 20:39:12 localhost /usr/sbin/kamailio[1219]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 23 Oct 1 20:39:12 localhost kernel: [554502.639227] kamailio[1207]: segfault at 58 ip 00007fd62c5c6f86 sp 00007fffd77c9890 error 4 in dialog.so[7fd62c589000+54000] I’m trying to catch a core next time it occurs, no core is available yet. Is this a new issue in 4.1.6? Any suggestions about the likely cause? Cheers, Dave.

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Relaying ACK to Asterisk
by Igor Potjevlesch 01 Oct '14

01 Oct '14

Hello, I'm having trouble with this scenario (Kamailio and Asterisk are working on the same server, Asterisk listens on 4060 instead of 5060): the UAC sends an ACK request with the following R-URI: sip:955*95%23@ <sip:955*95%23@%3cIP_ASTERISK/KAMAILIO%3e:4060> <IP_ASTERISK/KAMAILIO>:4060. When I'm doing a capture on loopback interface, I just see an ACK request from IP <IP_ASTERISK/KAMAILIO>:5060 to IP <IP_ASTERISK/KAMAILIO>:5060. So the ACK seems to loop inside Kamailio. What could explain that the good port defined by the UAC is deleted? Regards, Igor.

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rtpengine alt stream question
by Alex Balashov 30 Sep '14

30 Sep '14

Hello, I recently encountered SDP attributes that look like this: Media Attribute (a): alt:1 3 : njfxofkg bavpjfxg 192.168.2.59 41978 Media Attribute (a): alt:2 2 : fpukfyaj dqmiarjx 192.168.111.1 41978 Media Attribute (a): alt:3 1 : euvrwenk jctmhavh 192.168.238.1 41978 I was not familiar with these. However, RFC 4796 says: alt: the media stream is taken from the alternative source. A typical use case for this is an event where the ambient sound is separated from the main sound. The alternative audio stream could be, for example, the sound of a jungle. Another example is the video of a conference room, while the main stream carries the video of the speaker. This is similar to the 'live' role in H.239. RFC 6064 http://tools.ietf.org/html/rfc6064#section-4.4 speaks to this a little as well. This does not really make any clearer to me what these mean in this scenario, and unfortunately I cannot find out from the user. My question is: I assume that rtpengine (both Kamailio and daemon-side) does not support this, correct? Are there any plans to in the future? Can anyone with more experience with video, perhaps, give some insight into the meaning and application of this attribute? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

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R: Re: R: Re: RTPPROXY & BRANCH
by Marino Mileti 30 Sep '14

30 Sep '14

Unfortunately rtpengine doesn't work in this way. At the end of the calls this is the output log: Final packet stats: Tag 'Fw3D7R0', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30224 <> 192.168.10.20:7078 , 540 p, 92880 b, 0 e Media #1, port 30225 <> 192.168.10.20:7079 (RTCP), 3 p, 324 b, 0 e Media #2, port 30256 <> 192.168.10.20:9078 , 0 p, 0 b, 0 e Media #2, port 30257 <> 192.168.10.20:9079 (RTCP), 3 p, 264 b, 0 e Tag 'qWE6Gsh', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30140 <> 192.168.10.50:7078 , 533 p, 91068 b, 0 e Media #1, port 30141 <> 192.168.10.50:7079 (RTCP), 5 p, 444 b, 0 e Media #2, port 30170 <> 192.168.10.50:9078 , 0 p, 0 b, 0 e Media #2, port 30171 <> 192.168.10.50:9079 (RTCP), 1 p, 88 b, 0 e Tag 'TTPyT~Hdw', created 0:41 ago, in dialogue with 'Fw3D7R0' Media #1, port 30206 <> 172.20.11.208:7078 , 1070 p, 183736 b, 0 e Media #1, port 30207 <> 172.20.11.208:7079 (RTCP), 4 p, 496 b, 0 e Media #2, port 30240 <> 172.20.11.208:9078 , 4188 p, 1435946 b, 0 e Media #2, port 30241 <> 172.20.11.208:9079 (RTCP), 4 p, 400 b, 0 e 192.168.10.x clients are natted..it seems that rtpengine open 2 ports (for example video) for each receiver (30256 & 30170) and 1 port for the caller (30240). But on the INVITE of Kamailio only video port 30170 is offered to receivers, instead on caller side there are 2 distinct 183s message that offer 30190 & 30240. It's a little bit strange because some of these port doesn't appear in the log of rtpengine. At the end I can see video only on one receiver I don't know if the problem is on Kamailio (rtpproxy-ng module) or in the rtpengine :) > Without rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted > client so no needs of rtpproxy) > - B offers port b1,b2 (183) > - C offers port c1,c2 (182). > - A starts to send audio/video RTP to B on port b1,b2 > - A starts to send audio/video RTP to C on port c1,c2 > > With rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to Kamailio.... > - Kamailio contact rtpproxy because B&C are natted clients > - rtpproxy check callid and offer offers port k1,k2.... > - Kamailio sends INVITE to B offering k1,k2 > - Kamailio sends INVITE to C offering k1,k2 > - B offers port b1,b2 (183) > - C offers port c1,c2 (182) > - Kamailio sends 183 to A (for B leg) offering p1,p2 > - Kamailio sends 183 to A (for B leg) offering p3,p4 > - A starts to stream on p1,p2,p3,p4 but only one receiver can see the video > (B or C depends who will be the first:)) > > I don't know if it depends on that B & C receives same ports; i don't know > if rtpproxy is able to "duplicate" stream received from A to all "receiver" If A sends two streams, there is no need for duplication. A sending to p1 should be forwarded to B (b1) and A sending to p3 should be forwarded to C (c1). Both should be able to receive media sent by A. I believe that's what rtpengine does (but I haven't tested it). The reverse direction might be more confusing, but a final 200 OK with SDP should fix that. cheers _______________________________________________ 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

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Call Group versus MAX_BRANCHES limit
by João Vitor Arruda 30 Sep '14

30 Sep '14

Hi folks, I have a question related with the limited number of branches being 12 in config.h #define MAX_BRANCHES 12 /*!< maximum number of branches per transaction */ I am trying to implement a Call Group that consists in trying each member of the group (that can result in a parallel forking when lookup() is used) in sequence (similar to serial forking). Currently I have written code that is similar to the one described here: http://www.kamailio.org/dokuwiki/dokuphp/tutorials:avpops#serial_forking The only difference is that for each member I call lookup() and then in the failure_route(), I pull the next member from the stack and call lookup() again. The code works fine when used for groups with less than 12 members. However, call groups with 12 or more members fails as I can never reach remaining extensions once MAX_BRANCHES limit is reached. I've tried another approach using the functions described here: http://kamailio.org/docs/modules/4.1.x/modules/tm.html#tm.serial_forking but it too uses a new branch for every group member until the 12 limit is reached out. I also tried to use the functions "remove_branch(index)" and "clear_branches()" (both of which are poorly documented. In fact the only reference I ever found for these functions was here: http://www.kamailio.org/wiki/features/new-in-3.2.x#functions). Unfortunately, I wasn't able to prevent the MAX_BRANCHES limit from being reached. Ultimately, my goal is to have a limitless Call Group. Do you have any suggestions? (other than increasing the hard coded MAX_BRANCHES limit) Thank you, Joao Arruda

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Re: [SR-Users] Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 30 Sep '14

30 Sep '14

Hi,I ran the code on my server (as Daniel suggested) and then tried to send message from one user to other but no traffic was detected.ngrep d any qt W byline "sip" port 5060interface: anyfilter: (ip or ip6) and ( port 5060 )match: sipAlso, after I tried to install the PRESENCE module and having restarted the systems today, I noticed that the response that I am getting on Jitsi is "403 Not Relaying" and not "403 Not Allowed". I am not sure what caused the change.I hope someone can help me figure it out.Cheers!Gaurav Original message From:"DanielConstantin Mierla"< miconda(a)gmail.com >Date: 30 Sep 14 16:19:30Subject: Re: [SRUsers] Kamailio with Jitsi Presence troubleTo: ; "Kamailio (SER) Users Mailing List"Hello,you should grab the SIP traffic on kamailio server and send it to the mailing list. In this way we can spot why you get the 403 Not Allowed. You can use:ngrep d any qt W byline "sip" port 5060Cheers, DanielOn 30/09/14 08:50, Gaurav Kumar wrote:Hello all!I tried sendin g this earlier too but since I was not subscribed tothe mailing list, it bounced back.I am trying to setup a secure videoconferencing setup for mysmall office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used it on aUbuntu 12.04 setup and want to setup a SIP server for very fewusers (less than 10, at most) to use it through Jitsi. I am tryingto follow the tutorials available at: http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://www… am able to log into Jitsi on independent machines, both behindand without a firewall. But the trouble I am facing is that theusers do not appear online to each other. If I try to send a textmessage nonetheless, I get the "403 Not allowed" error.At first, I tried the config file for Kamailio provided on thefirst link. It had some problems due to it being an old versionand the config file on the 2nd link help ed me out. I suspect thereis some problem with the PRESENCE module but I do not know whatsince it does not give any error whatsoever. I added the "#!defineWITHPRESENCE" line and installed the Presence module but to noavail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00bfor Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a weeknow. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now! SIP Express Router (SER) and Kamailio (OpenSER) srusers mailing list srusers(a)lists.siprouter.org http://lists.siprouter.org/cgibin/mailman/listinfo/srusersDanielConstantin Mierla http://twitter.com/#!/miconda http://www.linkedin.com/in/micondaGet Yourself a cool, short @in.com Email ID now!

1 0

Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 30 Sep '14

30 Sep '14

Hello all!I tried sending this earlier too but since I was not subscribed to the mailing list, it bounced back.I am trying to setup a secure videoconferencing setup formy small office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used it on aUbuntu 12.04 setup and want to setup a SIP server for very few users(less than 10, at most) to use it through Jitsi. I am trying to followthe tutorials available at:http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://… able to log into Jitsi on independent machines, both behind andwithout a firewall. But the trouble I am facing is that the users do notappear online to each other. If I try to send a text messagenonetheless, I get the "403 Not allowed" error.At first, I triedthe config file for Kamailio provided on the first link. It had someproblems due to it being an old version and the config fil e on the 2ndlink helped me out. I suspect there is some problem with the PRESENCEmodule but I do not know what since it does not give any errorwhatsoever. I added the "#!define WITHPRESENCE" line and installed thePresence module but to no avail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00b for Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a week now. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now!

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Uri transformations stopped working after upgrade from 3.3 to 4.1.6.
by Julia Boudniatsky 30 Sep '14

30 Sep '14

Hello, Some Uri transformations stopped working after upgrade from 3.3 to 4.1.6. *To: <sip:01412127775678;cic=012;csel=noind@10.25.153.150:5060 <http://noind@10.25.153.150:5060>;user=phone>* *In v3.3* $tu({uri.params}) return: cic=012;csel=noind $tu({uri.param,cic}) return: 012 *In v4.1.6* $tu({uri.params}) return : user=phone $tu({uri.param,cic}) return : Best regards, Julia

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Re: [SR-Users] Questions about multiple contacts in kamailio
by Federico Cabiddu 30 Sep '14

30 Sep '14

Hi, you can achieve this by using the flag 0x04 when calling save() function from registrar module: http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1965144 Regards, Federico On 30 Sep 2014 10:54, "qw" <applemax82(a)163.com> wrote: Hi, I have one question about contact in kamailio. Sometimes, one user may register multiple contacts in kamailio server, where contacts are represented as ip:port. For example, the user registers at first, and lose connection to the internet later. After one minute, the user connects the internet again, but can't get service from kamailio server. Then the user need to registers with new contact. Now there are two contacts in kamailio, i.e. old one and new one. How can I remove the old contact, and make sure there is only one contact in kamailio server. Looking forward to your help! B.R. andrew _______________________________________________ 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

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Questions about multiple contacts in kamailio
by qw 30 Sep '14

30 Sep '14

Hi, I have one question about contact in kamailio. Sometimes, one user may register multiple contacts in kamailio server, where contacts are represented as ip:port. For example, the user registers at first, and lose connection to the internet later. After one minute, the user connects the internet again, but can't get service from kamailio server. Then the user need to registers with new contact. Now there are two contacts in kamailio, i.e. old one and new one. How can I remove the old contact, and make sure there is only one contact in kamailio server. Looking forward to your help! B.R. andrew

1 0

Kamailio -PSTN Gateway
by Zaka Ul Isam 30 Sep '14

30 Sep '14

Hello Folks: Please help with above, I have browsed and tried various suggestions on this list without much luck! I think problem can be reduced to three questions ;) 1: Apart from DEFINE WITH PSTN directive, do I need certain modules to be compiled? (DIALPLAN, CARRIERROUTE, LCR aso) If yes, then we ought to put up a list of modules required for each task/ activity..... 2: How Kamailio core decides the call routing? i.e: based on dialed digits as specified in dialplan . # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00355)[0][4-9]{3,20}$")) return; && #!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 = "10.1.6" desc "PSTN GW Address" pstn.gw_port = "5080" desc "PSTN GW Port" #!endif 3: If Kamailio is listening on say wlan0 interface 10.42.0.1:5060 (Ubuntu HOT SPOT) can it route to PSTN Gateway/ Softswitch on P2P1 (Eth0) without Bridging? Put another way, does kernel route towards the specified interface or Kamailio is capable of routing based on active routing cache? Thanking you in advance, Zak ________________________________ Albtelecom & Eagle Mobile invite you to protect together the Environment. Please do not print the e-mail unless really necessary. Albtelecom & Eagle Mobile iu ftojnë të mbrojmë së bashku Mjedisin. Lutemi të mos e printoni këtë komunikim elektronik nëse nuk është vërtet i nevojshëm. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the addressee/s. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited. If you receive this e-mail in error, please notify the sender immediately and delete it! Albtelecom makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message may belong to sender alone and may not necessarily reflect the opinions of Albtelecom. Albtelecom shall bear no liability for any loss or damage caused by software or e-mail viruses. Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset.

2 2

kamailio call limit
by Yuriy Gorlichenko 30 Sep '14

30 Sep '14

Hello. I try to test with SIPp my stak of kamailio->asterisk. I run SIPp with 200 calls/sec and see only 68 at maximum active calls at server. When I set 500 calls/sec with limit 1000 I see 68 active connections again. So when I try test SIPp to asterisk without Kam i see wright maximum of active channels/calls. My question - does Kamailio have limits ffor calls? If no- help me to understand please what is going on with my server. Thanks.

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Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:51, Marino Mileti wrote: > Without rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted > client so no needs of rtpproxy) > - B offers port b1,b2 (183) > - C offers port c1,c2 (182). > - A starts to send audio/video RTP to B on port b1,b2 > - A starts to send audio/video RTP to C on port c1,c2 > > With rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to Kamailio.... > - Kamailio contact rtpproxy because B&C are natted clients > - rtpproxy check callid and offer offers port k1,k2.... > - Kamailio sends INVITE to B offering k1,k2 > - Kamailio sends INVITE to C offering k1,k2 > - B offers port b1,b2 (183) > - C offers port c1,c2 (182) > - Kamailio sends 183 to A (for B leg) offering p1,p2 > - Kamailio sends 183 to A (for B leg) offering p3,p4 > - A starts to stream on p1,p2,p3,p4 but only one receiver can see the video > (B or C depends who will be the first:)) > > I don't know if it depends on that B & C receives same ports; i don't know > if rtpproxy is able to "duplicate" stream received from A to all "receiver" If A sends two streams, there is no need for duplication. A sending to p1 should be forwarded to B (b1) and A sending to p3 should be forwarded to C (c1). Both should be able to receive media sent by A. I believe that's what rtpengine does (but I haven't tested it). The reverse direction might be more confusing, but a final 200 OK with SDP should fix that. cheers

1 0

Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:23, Marino Mileti wrote: > The problem isn't on 183s but on the multiple INVITE that Kamailio sends to > clients behind rtpengine. Rtpengine open new ports for answer but on INVITE > the rtpengine ports are the same...This happens because for all these > clients the callid is still the same..so for rtpengine there's no > difference...for this reason I can see early media only on one of the > receivers I still don't understand what difference an RTP proxy is supposed to make for the invite. Your offering client only opens one endpoint and sends that out. Without RTP proxy, all receiving clients see the same endpoint and they will all send their early media to that same endpoint. With RTP proxy, the same thing happens. What's the difference? cheers

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Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:29, Marino Mileti wrote: > Wow! Do you have an example of how to do that? How I have to modify my > kamailio.conf in order to instructs rtpproxy to user from-tag & to-tag in > this way? You don't have to do anything, tags are already included in all the messages. cheers

1 0

Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:08, Marino Mileti wrote: > But with from-tag and To-tag it's possible to instruct rtpengine to generate > new couple of ports for each branch of a call? In the source code of > rtpengine it seems that it check only the callid parameter Yes it will. The call-id is only a vague umbrella under which the from/to tags are collected. All media parameters, ports etc are associated with the tags, and a single call can contain any number of different from/to tags, all with their own separate parameters. cheers

2 1

R: Re: R: Re: RTPPROXY & BRANCH
by Marino Mileti 29 Sep '14

29 Sep '14

No no. The video will be sent by the caller user to all the callees. I'l try to explain better. My scenario is: - A make a call to a group... B & C are group member...so Kamailio is able to call them in parallel using alias.. - B & C receive the INVITEs & replies with two separate 183 with SDP (in SDP they specified where they are able to receive audio&video) - A receives two 183...& starts to send its RTP video stream to B & C (early media) - Once B or C answers the call the other leg is cancelled.. Everything is working fine but if B & C are behind NAT rtpproxy is activated and during INVITE for B &C rtprpoxy doesn't generate a couple of new ports for each of them but it uses always the same ports. So the only fastest client (B or C) get the video. I don't want to change IP between 183 & 200, i would like only that rtpptoxy sends INVITE for B & C with differents port.Is it possible to implement this scenario or there's some turnarund? ----Messaggio originale---- Da: frank(a)carmickle.com Data: 25-set-2014 16.22 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] R: Re: RTPPROXY & BRANCH On Sep 25, 2014, at 10:09 AM, Marino Mileti <marino.mileti(a)alice.it> wrote:Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) I don't understand. Are you saying that you have clients that when they receive an invite sent video with 183? How do you want to composite the video to show to the caller? It is not RFC3261 compliant to change IP and port from 183 to 200. Of course you can reinvite after the 200. Most B2BUAs require you to ignore early media and generate something locally to send to the caller or just send them 180. Maybe if you explain your use case someone can help you. --FC

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Re: [SR-Users] rtpproxy & extra_id_pv
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/26/14 16:57, Marino Mileti wrote: > Hello, > >> On Friday 26 September 2014 16:44:44 Marino Mileti wrote: >>> Hi guys, >>> I've seen that setting the parameter extra_id_pv, every branch should >>> be a different callid.. >>> How can i set this parameter? I've tried with : >>> modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)") >>> >>> but in the INVITE message the callid is still the same for all >>> branches. Any suggest? > >> Did you assign a value to $avp(extra) in the script, before calling any of > the rtpproxy functions? > >> Did you use the 'b' parameter in the call to rtpproxy_*() functions? >> -- >> Alex Hermann > > Yes, i've assigned a value in the script, just to try I've assigned $rU and > on the log of rtpengine I can see the value of "via-branch" valorized > correctly. I've also used the 'd' parameter in rtpproxy_manage() function. > > I've also checked on the source code of rtpengine and the part regarding the > "parse & get" of via-branch value is commented...I'm using the master branch > of rtpengine. > > This is the piece of code of call_interfaces.c where the via-branch > parameter is commented: > //bencode_dictionary_get_str(input, "via-branch", &viabranch); The via-branch is currently unused in rtpengine, even though it does appear in the control messages. From-tag and to-tag are sufficient to identify a unique dialogue within a call. cheers

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R: Re: RTPPROXY & BRANCH
by Marino Mileti 29 Sep '14

29 Sep '14

Because I've more than 1 client behind NAT (1,2,3 mobile phones) and I would like to reach all of them in parallel mode. I can't use for all of them same ports because all mobile clients have early media (the receive video media before they answer) So at the moment this scenario is not possible? Is there any possible turnaround? ----Messaggio originale---- Da: rfuchs(a)sipwise.com Data: 24-set-2014 1.15 A: <sr-users(a)lists.sip-router.org> Ogg: Re: [SR-Users] RTPPROXY &amp; BRANCH On 23/09/14 04:35 AM, marino.mileti(a)alice.it wrote: > I've a problem with rtpproxy during a parallel ring scenario. > I've two client behind NAT (192.168.10.20 & 192.168.10.50) and when I > try to call them in parallel mode (ringall) rtpproxy module sends in to > the INVITE the same RTP ports. > > Is it possible to manage rtpproxy in order to generate a couple of new > port for each branch? Not at the moment. The logic is that one endpoint advertised by the client is mapped to one endpoint advertised by rtpengine. Why would you like a separate set of ports? cheers _______________________________________________ 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

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rtpproxy_offer and fix_nated_sdp in one route
by Sebastian Damm 29 Sep '14

29 Sep '14

Hi, I switched from rtpproxy module to the rtpproxy-ng module lately, and noticed a strange behavior. In my branch route to the device, I have two statements: fix_nated_sdp("1"); rtpproxy_offer(); The first command appends a line with "direction:active" to the SDP. The second one puts the RTP proxy in the stream. This worked all well with the old rtpproxy module. But with the new rtpproxy-ng module, I get an empty line after the SDP body, just before the "direction:active" line in the SDP, which makes the packet invalid. I tried changing the order of both statements, but without any difference. What I saw is, that the old module just sent some basic parameters to the rtpproxy and got only IP and port back. The new module sends the complete SDP to the rtpengine and gets back the "fixed" SDP. Has anyone ever seen this? Is there a way to fix it? Best Regards, Sebastian

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event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Event-parse-error: $rm from $avp(inc_carrier)/n$mb/n"); } corelog=1 debug=0 kamailio 4.1.6 from source Wrong header "From" is simulated by SIPP. In log I receive only ERROR:<core> [parser/parse_from.c:79]: parse_from_header() ...... Thank you, Julia

1 0

Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 29 Sep '14

29 Sep '14

Hello all!I am trying to setup a secure videoconferencing setup for my small office. After a lot of convincing, my bosses have allowed me to create a setup and have given me a live IP. I have used it on a Ubuntu 12.04 setup and want to setup a SIP server for very few users (less than 10, at most) to use it through Jitsi. I am trying to follow the tutorials available at:http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://… am able to log into Jitsi on independent machines, both behind and without a firewall. But the trouble I am facing is that the users do not appear online to each other. If I try to send a text message nonetheless, I get the "403 Not allowed" error.At first, I tried the config file for Kamailio provided on the first link. It had some problems due to it being an old version and the config file on the 2nd link helped me out. I suspect there is some problem with the PRESENCE module but I do not know what since it does not give any error whatsoever. I added the "#!define WITHPRESENCE" line and installed the Presence module but to no avail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00b for Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a week now. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now!

1 0

event_route[core:receive-parse-error] is not executed 4.1.6
by Julia Boudniatsky 29 Sep '14

29 Sep '14

Hello, I try to use an event_route[core:receive-parse-error] { xlog("L_WARN", "Parse-error: $rm from $avp(inc_carrier)/n$mb/n"); }

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Does kamailio reloads the modules per sip message ?
by Rahul MathuR 29 Sep '14

29 Sep '14

Hello, We have observed a strange behavior in corex module that it gets loaded at every sip packet which arrives to kamailio. We put a static variable and saw that it gets re-initialized to 0 everytime any sip packet comes to it. Could you please tell me how to stop it and load it just once. Thanks in advance ! -- Warm Regds. MathuRahul

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DBTEXT Engine & ALIAS_DB
by Marino Mileti 29 Sep '14

29 Sep '14

Hi to all, I've setup FS with dbtext module, and now I'm trying to make a alias_db_lookup(dbaliases) but it doesn't find any alias.Same conf with mysql db works great. Is it possible to have alias with dbtext engine? Marino

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tm.t_uac_wait documentation?
by Juha Heinanen 29 Sep '14

29 Sep '14

i could not find any documentation about tm.t_uac_wait rpc command. is it documented somewhere? -- juha

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tm.t_uac_wait does not use advertise address
by Juha Heinanen 29 Sep '14

29 Sep '14

based on more tests, it appears that tm.t_uac_wait does not use advertise address (if given on listen line) when it substitutes !! (SUBST_CHARs) in request headers. i'll open a bug report on it. -- juha

2 1

cnxcc serial fork
by Kelvin Chua 29 Sep '14

29 Sep '14

hi carlos, i have a serial forking scenario to several providers as failover. and i used cnxcc_set_max_credit to control the session credit/time. just wanted to make sure it does what i think it should do so i made a little experiment. the logic goes this way. 1. cnxcc_set_max_credit -> $100, cost per second -> $1 2. call will fail, try the next provider 3. cnxcc_set_max_credit -> $200, cost per second -> $2 what i am expecting to see is, if i do kamctl kamcmd cnxcc.active_clients i would see $200 as the max_amount. but in kamailio 4.1.5, i still see $100, should it be updated to $200? is there an override in cnxcc_set_max_credit when called twice or after failure_route? Kelvin Chua

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rtpproxy & extra_id_pv
by Marino Mileti 26 Sep '14

26 Sep '14

Hi guys, I've seen that setting the parameter extra_id_pv, every branch should be hae different callid.. How can i set this parameter? I've tried with : modparam("rtpproxy", "extra_id_pv", "$avp(extra_id)") but in the INVITE message the callid is still the same for all branches. Any suggest?

2 2

Kamailio server crashing
by aawaise 26 Sep '14

26 Sep '14

Hello, I installed kamailio and replaced kamailio.cfg with a customized kamailio.cfg and replaced kamctlrc file also. And restarted the kamailio server. Server gets started ok and after a minute or two, I get the error, " '/usr/local/kamailio/sbin' doesnot belong to any package ". I am facing this problem for quite some time now. Any solution to this will be very helpful. Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-server-crashing-tp130923.h… Sent from the Users mailing list archive at Nabble.com.

2 1

Bash Code Injection and 'exec' module
by Seudin Kasumovic 25 Sep '14

25 Sep '14

Hi kamailio users, we are witnesses of new discovered bug in bash: Bash Code Injection Vulnerability via Specially Crafted Environment Variables (CVE-2014-6271) https://access.redhat.com/node/1200223 As exec module exports all SIP headers in environment so it's was easy to push bash command. There is attached simple kamailio test config file. With sipp we sent header to output 123 into file /tmp/123 like this: User-Agent: () { :;}; echo 123 > /tmp/123 Debug output from kamailio is: 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTENT_LENGTH=135 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTENT_TYPE=application/sdp 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_ALLOW=INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH * 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_USER_AGENT=() { :;}; echo 123 > /tmp/123* 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_SUBJECT=Performance Test 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_MAX_FORWARDS=70 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CONTACT=< sip:T00157@198.51.100.2:5060> 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CSEQ=1 INVITE 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_CALLID= 1-5394(a)198.51.100.2 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_TO=+442033998806 <sip:+442033998806@orange.voip> 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_FROM=+442033998833 <sip:T00157@orange.voip>;tag=5394SIPpTag001 5(30147) DEBUG: exec [exec_hf.c:278]: print_hf_var(): SIP_HF_VIA=SIP/2.0/UDP 198.51.100.2:5060;branch=z9hG4bK-5394-1-0 5(30147) DEBUG: exec [exec_mod.c:175]: w_exec_msg(): executing [/bin/true] ls /tmp shows new created file !!! I created simple patch to fix this issue in exec module based on suggestion from RedHat until you fix your bash what is recommended. -- Seudin Kasumovic

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Re: [SR-Users] Kamailio - Separated Servers
by Bruno Emer 24 Sep '14

24 Sep '14

Hello Guys! Good news about my tests here! I found some configurations relative to Outbound proxy and I get this working. Now, I can have two outbound proxies, one registrar and one database. I can register one user in the Outbound Proxy1 and one user in Outbound Proxy 2 and place a call between these users. But now I'm facing another problem! The ACK messages are not being delivered, so the session is being dropped after a few seconds. I think that the problem is in the Contact field. Trying to investigate it now! I'm not sure about how to change the message to deliver the ACK. Did someone had this problem before? In anycase, here is the routing configuration that I'm using in my Oubound proxy: request_route { if (($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } route(REQINIT); if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } route(WITHINDLG); t_check_trans(); if (is_method("REGISTER")) { remove_hf("Route"); add_path(); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } else { if (is_method("INVITE|SUBSCRIBE|ACK")) record_route(); if (@via[2] == "") { # From client so route to registrar... if ($rU == $null) { sl_send_reply("484", "Address Incomplete"); exit; } remove_hf("Route"); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } else { # From registrar so route using "Route:" headers... if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: xlog("L_ERR", "in request_route\n"); sl_reply_error(); exit; } } t_on_failure("FAIL_OUTBOUND"); } } route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } route[REQINIT] { 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; } } route[WITHINDLG] { if (has_totag()) { if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: if (is_method("ACK")) { if ( t_check_trans() ) { record_route(); route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not Found"); } } else { if (is_method("NOTIFY")) { record_route(); } route(RELAY); } exit; } } onreply_route { if (($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (!t_check_trans()) { drop; } if ($rm == "REGISTER" && $rs >= 200 && $rs <= 299) { remove_hf("Flow-Timer"); if ($(hdr(Require)[*])=~"outbound") insert_hf("Flow-Timer: FLOW_TIMER\r\n", "Call-ID"); } } failure_route[FAIL_OUTBOUND] { if (t_branch_timeout() || !t_branch_replied()) { send_reply("430", "Flow Failed"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } Thanks again!!! *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2014-09-22 12:40 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hi, Daniel and Frank! > > Thanks for your support! > Now, I changed my scenario just a lil bit to make it easy to understand > and deploy. I'm doing the following: > > 1. Created a server using the "Outbound" module configuration example, > but add WebSocket support in it. (IP Address 192.168.1.16) > 2. Created a server running as Proxy and Registrar with WebSockets > support. (IP Address 192.168.1.15) > 3. Registered two users (bob and alice(a)enterprise.com) using WebRTC. > These users sent they Register requests to the Oubount server > (192.168.1.16). > 4. I'm also using an external postgres database to store the user > informations. In my proxy/registrar server, I've configurated > "modparam("usrloc", "db_mode", 3)", so it will not store data in memory. > > > After register the users in my scenario, I can see the following > information stored in my database (output of kamctl db show location): > > id | ruid | username | domain | > contact | received | path | > expires | q | callid | cseq | last_modified | > flags | cflags | user_agent | socket | methods | > instance | reg_id > > -----+----------------------+----------+----------------+------------------------------------------------+-----------------------+------+---------------------+----+------------------------+------+---------------------+-------+--------+-------------------+---------------------+---------+-------------------------------------------------+-------- > 277 | uloc-54203f71-10bd-2 | alice | enterprise.com | > sip:v4s0prah@o35ctmdnnruv.invalid;transport=ws | sip:192.168.1.16:5060 | > | 2014-09-22 15:31:21 | -1 | udb4shpe4dnm1jsl6016s5 | 82 | > 2014-09-22 15:26:21 | 0 | 0 | JsSIP 0.4.0-devel | udp: > 192.168.1.15:80 | 783 | > <urn:uuid:75c837d1-740a-4c62-baa0-56a06149699e> | 1 > 278 | uloc-54203f71-10bc-2 | bob | enterprise.com | > sip:lr78s270@0mkf6fgtprlh.invalid;transport=ws | sip:192.168.1.16:5060 | > | 2014-09-22 15:34:02 | -1 | 5mpl9q4bkuqsnpk8r316j5 | 82 | > 2014-09-22 15:29:02 | 0 | 0 | JsSIP 0.4.0-devel | udp: > 192.168.1.15:80 | 783 | > <urn:uuid:e92ef911-1ae4-45b1-bfc9-b5752b0f2fe1> | 1 > > > So, I can assume that both users are registered and ready to start > sessions. > > Now, when I try to start a new session between then (let's say bob calling > alice) what I get in the endpoint is the following message: > > SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL) > Via: SIP/2.0/WS > 0mkf6fgtprlh.invalid;rport=33154;received=192.168.1.11;branch=z9hG4bK7711221 > To: <sip:alice@enterprise.com>;tag=94c1b96a0e6eb467885b5057e1e49f9e.1970 > From: <sip:bob@enterprise.com>;tag=on056iipql > Call-ID: 5e1pmj2fppbgg13doi4k > CSeq: 3989 INVITE > Server: kamailio (4.1.5 (x86_64/linux)) > Content-Length: 0 > > What I understood about this problem is that the Proxy server is trying to > forward the INVITE to the Outbound server and this server does not know > that to do with this message - like forward it to alice. > > I'm stucked in this step. > > Here you can find the configuration files that I'm using: > > Outbound Server: > https://mega.co.nz/#!Wlw2RL7S!G2YOP-1QbSCeNQddaAe3kyQMqz8yteZlZsAUpjZX4v4 > > Proxy Server: > https://mega.co.nz/#!qxAXTbbb!4zBjRtmADdl3rPCN2Yidd3xnZEjFahW3GSMb00w41xc > > > > Thanks again! > > Bruno Emer > > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2014-09-18 18:04 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > > Ok, Daniel! >> >> I am trying exactly what you said, but for some reason the proxy server >> can't forward the message back to the WebSocket server. I am getting >> "487 Unresolvable destination". Appears that my Proxy Server can't get data >> from usrloc. >> >> When I register an user using WebRTC I can see that the system stores the >> information in kamailio database and in the field "contact" it puts a >> string. The Proxy server is looking in database and just find this >> information. Actually, I'm trying to understand the entire process that >> envolves passing information from an Edge Server to a Proxy server that >> sends the Register information to another Registrar server (as I described >> before). The database is shared between all the servers, and they are in >> the same network. >> >> I just read the information in another topic, as you said, but I think >> that my case is a lil bit different from it. I you agree, I can post my >> configuration files here. Maybe this can help another users with the same >> problem! >> >> Thanks again!!! >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> <https://www.facebook.com/bruno.emer.5> >> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >> <https://twitter.com/brunoemer_> >> <https://www.flickr.com/photos/122070309@N03/> >> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >> >> 2014-09-17 9:13 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >> Hello! >>> >>> I am relatively new to Kamailio and I'm trying to create a new >>> enviroment using it in my company. I am thinking about use Amazon to host >>> the servers and use OpsWorks to automatically escalate then if necessary. >>> To accomplish this, my idea is to separate the servers, using one dedicated >>> server to run as WebSocket, one to run as proxy and one as a Registrar. >>> I'll be using just one database to store informations to all of my servers. >>> With this, if I need more resources later, I can just create new servers >>> with the specific roles (WebSockets, Proxy, Registrar). >>> >>> By now, the idea is clear, but the point is that I don't know how to >>> separate the WebSockets server from the proxy server. Actually, I can do >>> this, but when I have one agent using a regular softphone and one agent >>> using WebSockets (with JSSIP) they are not able to establish a session if >>> the softphone user starts it. Now, I want to know if is there a way to use >>> two websockets servers, register users using both of then and start >>> sessions between then, with a separated proxy and registrar. >>> >>> Has anyone done this before? Is possible to use kamailio like this? >>> >>> >>> Thanks. >>> >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >> >> >

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