- sr-users - kamailio.org

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

03 Nov '14

running kamailio from deb http://deb.kamailio.org/kamailio jessie main mysql> show columns; ERROR 1064 (42000): You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '' at line 1 it was a while ago, but i used the included script and followed wiki instructions to create the initial DB from memory it was a simple 1 liner cmd any help is appreciated thanks mysql> show tables; +---------------------+ | … [View More]

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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:… [View More]

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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 ;… [View More]

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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

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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: … [View More]

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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 … [View More]

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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:… [View More]

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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 … [View More]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 [View Less]

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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, --… [View More]

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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 … [View More]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 [View Less]

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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 [… [View More]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 ... [View Less]

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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 … [View More]

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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?

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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: <… [View More]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? [View Less]

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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 … [View More]

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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.… [View More]

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

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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 … [View More]

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 … [View More]

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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/&… [View More]gt; 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 [View Less]

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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.… [View More]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 [View Less]

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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 ./… [View More]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 [View Less]

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(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 … [View More]

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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]: … [View More]

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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 … [View More]

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