sr-users July 2015

sr-users@lists.kamailio.org

86 participants
151 discussions

Kamailio deployment
by Srujan Kumar K 09 Jul '15

09 Jul '15

Hai, I am connecting my kamailio server to MSC for my solution which has both inbound and outbound. The SIP trunk is not UP and we don't see any packets transmitted between the 2 nodes. We are also trying to load SIPI/SIPT module Kamailio server IP : 172.22.14.12 MSC IP : 172.28.0.68 I have also attached configuration file with this mail. Requesting your help on this. Regards ~Srujan

1 0

Moving from haproxy to db_cluster
by Jan Hazenberg 09 Jul '15

09 Jul '15

Hi, I have a question about the db_cluster module. We are currently using a galera mysql cluster from severalnines, in this setup haproxy is installed on the kamailio nodes handling the database failover. The question is if a database node fails and the TCP connection breaks the sql query will probably fail when using haproxy since it is not aware of the query. Will the db_cluster module send the query to the next available mysql node in the cluster without a failing query? If thats the case i should not notice the failing dbnode in my kamailio script. In other words, will there be a advantage in using the db_cluster module instead of haproxy? Thanks, Jan Hazenberg

2 3

code coverage in kamailio using gcov
by Surendra Pullaiah 09 Jul '15

09 Jul '15

Hello I am trying to implement code coverage tool in kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and it got compiled successfully. My problem is when I am running kamailio it is saying lib64/kamailio/libkcore.so.1: undefined symbol: __gcov_merge_add. Can anyone help me on this. Regards Surendra

1 0

Function sdp_remove_codecs_by_id seems to be not working
by José Seabra 09 Jul '15

09 Jul '15

Hello, I'm using the function sdp_remove_codecs_by_id from sdpops module in order to remove some codecs in INVITE request before send it to freeswitch, but the function doesn't remove the codec, and it doesn't give any error message. I'm using this function in request route. Kamailio version is 4.2.2. INVITE that kamailio receives from phone: INVITE sip:401@teste.d <sip%3A401(a)teste.itcenter.com.pt>emo.pt;user=phone SIP/2.0 Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0 Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060 From: "301" <sip:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt> >;tag=oztyflbzbx To: <sip:401@teste.demo.pt <sip%3A401(a)teste.itcenter.com.pt>;user=phone> Call-ID: 3c3a58a25d63-ghfc5xdg1sn0 CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1 X-Serialnumber: 000413262FA0 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:teste.demo.pt <http://teste.itcenter.com.pt> >;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 391 v=0 o=root 24935823 24935823 IN IP4 192.168.10.147 s=call c=IN IP4 192.168.10.147 t=0 0 m=audio 19410 RTP/AVP 0 8 9 99 3 18 4 101 a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv INVITE that kamailio send to freeswitch after execute sdp_remove_codecs_by_id("18"): INVITE sip:401@teste.demo.pt;user=phone SIP/2.0. Record-Route: <sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>. Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Via: SIP/2.0/UDP 10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0. Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0. Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060. From: "301" <sip:301@teste.demo.pt>;tag=zvjgcz9zs9. To: <sip:401@teste.demo.pt;user=phone>. Call-ID: 3c3a7c84e065-pr2hm0uk9yfz. CSeq: 2 INVITE. Max-Forwards: 68. Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1. X-Serialnumber: 000413262FA0. P-Key-Flags: resolution="31x13", keys="4". User-Agent: snom370/8.4.35. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Call-Info: <sip:teste.itcenter.com.pt>;appearance-index=1. Session-Expires: 3600;refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 403. . v=0. o=root 228603317 228603317 IN IP4 100.64.250.4. s=call. c=IN IP4 100.64.250.4. t=0 0. m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:9 G722/8000. a=rtpmap:99 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:49405. SDP body has no changes related with codecs. Anyone call help please. Thank you BR José Seabra -- Cumprimentos José Seabra

3 6

Textops and Multipart Body - adding ISUP segfaults Kamailio
by andre second 09 Jul '15

09 Jul '15

Hi, I am using Kamailio version: kamailio 4.3.0 (x86_64/linux) c6aa95 on CentOS 6 I am trying to encapsulate ISUP in the INVITE: ...if(has_body("application/sdp")){ set_body_multipart(); if(msg_apply_changes()) { $var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61 6c 6c"; append_body_part("$var(acm)","application/isup; version=itu-t92+","signal; handling=optional"); xlog("L_INFO", "ISUP Changes Applied Succesfully"); }} ...And getting the following error: Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1691]: get_boundary(): boundary is <--unique-boundary-1>...Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1808]: check_boundaries(): last bondary without -- at the end Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1612]: replace_body(): old size body[365] actual[534]Jul 7 14:59:12 ukrcld1000 kamailio: CRITICAL: <core> [data_lump.c:463]: free_lump(): non free-able lump: 0x7f7a33d5ae40 flags=1 and then Kamailio crashes. Can you please recommend? Thanks,Andrei

2 3

usleep with variable
by Antanas Masevicius 09 Jul '15

09 Jul '15

Hello, been struggling to make a random sleep with cfgutils functions. ex: $avp(i:444) = 123; usleep("$avp(i:444)"); gives an error: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio-sp1new.cfg:723 Kamailio version: 4.3 If usleep accepts only numbers, is there any other way to sleep by passing variable? best regards, Antanas -- # <#> Antanas Masevičius Technikos Direktorius UAB Nacionalinis telekomunikacijų tinklas Panerių g. 51, LT-03160 Vilnius *Tel.* 8 5 2056000 *Tiesiog. tel.* 8 700 00031 *Mob. tel.* 8 685 77730 *Faks.* 8 700 00034 *El. p.:* antanas.masevicius(a)ntt.lt <#> # www.ntt.lt <http://www.ntt.lt> | www.spykas.com <http://www.spykas.com>

1 1

MetaSwitch ignores Contact: header
by Markus 08 Jul '15

08 Jul '15

Dear list, I'm running kamailio-4.2.3-3.1.x86_64 in a very simple setup: one Asterisk server behind kamailio. But I'm using dispatcher, the other servers are just down currently. :) Different vendors route DIDs to kamailio's IP and kamailio handles the communication towards Asterisk. Now I have a new vendor who uses MetaSwitch, and there is a problem. All calls get dropped with a BYE from the Asterisk side after 32 seconds because Asterisk never receives the ACK, because kamailio doesn't forward it, because MetaSwitch sends it as ACK sip:number@KAMAILIO-IP instead of ACK sip:number@ASTERISK-IP That means MetaSwitch ignores what we tell it in the Contact: header. Is this MetaSwitch'es "fault" or kamailio's? All other vendors work fine without problems. Therefore I have compared an "old" vendor and the new one. Here is the result that shows the difference: Old vendor - works: OLDVENDOR-IP.5060 > KAMAILIO-IP.5060: INVITE sip:number@KAMAILIO-IP KAMAILIO-IP.5060 > OLDVENDOR-IP.5060: SIP/2.0 100 trying KAMAILIO-IP.5060 > ASTERISK-IP.5060: INVITE sip:number@KAMAILIO-IP ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 100 Trying ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > OLDVENDOR-IP.5060: SIP/2.0 200 OK OLDVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@ASTERISK-IP:5060 KAMAILIO-IP.5060 > ASTERISK-IP.5060: ACK sip:number@ASTERISK-IP:5060 New vendor with MetaSwitch - doesn't work: NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: INVITE sip:number@KAMAILIO-IP:5060 KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 100 trying KAMAILIO-IP.5060 > ASTERISK-IP.5060: INVITE sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 100 Trying ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ... and so on ... Can I do something about this or is a change on MetaSwitch'es side in order? Thank you! Markus

3 3

BYE request with Sipp refused
by Loic Chabert 08 Jul '15

08 Jul '15

Hello everyone, I'm a newbie in kamailio world and make some tests to dispatch call over several asterisk instances. This part works, my call has been distributed over two instances (if i make a call with jitsi for example). Now, i want to test performance of my cluster and use Sipp to chalenge it. All method is working fine, except the BYE method, where kamailio send me back a 404 Not here. This is my SIPP scenario: ... <pause/>  <send retrans="500"> <![CDATA[ BYE sip:[service]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@loicchabert.fr>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> ... I use the classic file of kamailio, with some adjustement for dispatching. Any ideas concerning this problem ? This BYE is not in transaction so he is drop by kamailio ? Thank in advance, Regards.

3 6

Sending a message form within kamailio
by Shouvanik Chakrabarti 08 Jul '15

08 Jul '15

Hello, I am writing a module for kamailio and as part of its functionality I need to generate and send SIP messages. How would I go about sending an SIP message from within kamailio? Any help would be deeply appreciated. Thanks, Shouvanik

3 5

Kamailio as radius server
by Ali Taher 08 Jul '15

08 Jul '15

Hello, I'm wondering if kamailio can act as radius server who's responsible of authentication,authorization and accounting, where kamailio receive accounting record types (start-stop-interim) and write them in a database. Thanks Ali

3 3

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sr-users July 2015