sr-users November 2015

sr-users@lists.kamailio.org

80 participants
123 discussions

by jignesh gandhi

Hello All, I have a need to save the Record-Route header from the INVITE and use it as a route header in requests like BYE. PROXY --> Kamailio(as PROXY) --> Asterisk I am using rr module to add a record_route when passing the INVITE to Asterisk. Thanks, Jignesh Gandhi

8 years, 7 months

UDP send: Operation not permitted

by Sebastian Damm

Hi, we have a load balancer which is handling a lot of SIP traffic all day. There's always 20-40 Mbit SIP traffic going through. From time to time we see in our logs messages like these: Sep 16 09:46:28 ecker /usr/sbin/kamailio[25505]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f2d9d6b3ce0,1321,0,46.237.225.126:5060,16): Operation not permitted(1) Sep 16 09:46:38 ecker /usr/sbin/kamailio[25194]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7efc982b8fc8,420,0,82.113.121.183:35794,16): Operation not permitted(1) Sep 16 09:46:40 ecker /usr/sbin/kamailio[25505]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f2d9d6b3ce0,1338,0,5.158.137.9:55067,16): Operation not permitted(1) Sep 16 09:46:44 ecker /usr/sbin/kamailio[25183]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7efc982d9f48,450,0,178.165.131.197:37515,16): Operation not permitted(1) Sep 16 09:46:49 ecker /usr/sbin/kamailio[25643]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f93fb624530,496,0,172.56.7.69:25643,16): Operation not permitted(1) Sep 16 09:46:55 ecker /usr/sbin/kamailio[25335]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f41632cda98,598,0,80.215.234.139:3396,16): Operation not permitted(1) Sep 16 09:46:56 ecker /usr/sbin/kamailio[25345]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f41632f4840,459,0,94.197.120.191:8225,16): Operation not permitted(1) I know that these messages can be produced by iptables blocking the outbound traffic. But our outbound chain looks basically like this: root@ecker:~# iptables-save | grep OUTPUT :OUTPUT DROP [0:0] -A OUTPUT -m state --state RELATED,ESTABLISHED -j ACCEPT -j ACCEPT -A OUTPUT -m state --state INVALID -j DROP -A OUTPUT -o lo -m state --state NEW -j ACCEPT -A OUTPUT -o lo -j ACCEPT -A OUTPUT -m state --state NEW -j ACCEPT We don't have the nf_ct_sip module loaded, syslog doesn't say anything, and even clearing all iptables rules doesn't eliminate those errors. Has anyone ever seen this? It looks like a load thing, because at weekends there are significantly less errors. Thanks, Sebastian

8 years, 8 months

Kamailio TLS

by Kethzer Docteur

Im trying to enable Kamailio TLS but the configuration is not running at all when I cahnge it liike this as follow. Kamailio 3.3 ubuntu 12.04 LTS #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif Please help -- Kethzer Docteur

8 years, 9 months

Append to reply in default action of failure_route

by Efelin Novak

Hi, I would like to append a header to a 'winning' negative reply in failure_route and let the Kamailio do the default action (state fully forward the winning reply). When I use append_to_reply("Foo: bar\r\n"); and then call exit; in failure_route nothing is appended. When I use same append_to_reply then t_relay("505","Error"); and exit; the header is appended. When I use append and t_reply with dialog modul turned on I got a bug I'm solving here '[SR-Users] t_reply in failure route with dialog module'. So my question is how to put a header into a reply when I don't want to alter its code or text? I'm using Kamilio 4.0.4 on Debian 7.1 Thanks for an answer Efelin

8 years, 10 months

kamailio 4.0.1 websocket configuration

by Rupayan Dutta

Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta

8 years, 10 months

SIP Trunking Kamailio directly to external SIP Trunk Provider

by amjadali_bb＠hotmail.com

Hi, I have got two Asterisk servers and both are communicating with the SIP Trunk provider (External Network) directly - the Asterisk servers are connected with the Kamailio and then Kamailio with the Agent Phones. I need some change in this architecture that I want Kamailio to communicate with the SIP Trunk provider (External Network) directly. Can any one please advise if this is possible and how ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/SIP-Trunking-Kamailio-directly-to-e… Sent from the Users mailing list archive at Nabble.com.

8 years, 11 months

why is rtpenengine-delete deleting the whole call?

by Juha Heinanen

in my test sip proxy parallel forks the invite to two contacts and two rtpengine-offers are made with these kind of call-id and via-branch params: "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0" "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1", then the via-branch .0 callee answers and rtpengine-answer is sent to with this kind of call-id and via-branch params: "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0", after that sip proxy deletes the other branch by sending rtpengine-delete with this kind of params: Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Received command 'delete' from 127.0.0.1:40700 Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Dump for 'delete' from 127.0.0.1:40700: { "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1", "received-from": [ "IP4", "127.0.0.1" ], "from-tag": "3322319973f3909e", "command": "delete" } and rtpengine responds: Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] Response dump for 'delete' to 127.0.0.1:40700: { "created": 1448583951, "last signal": 1448583955, "tags": { "3322319973f3909e": { "tag": "3322319973f3909e", "created": 1448583951, "in dialogue with": "0f6759297b8f1711", "medias": [ { "index": 1, "type": "audio", "protocol": "RTP/AVP", "streams": [ { "local port": 53864, "endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10458 }, "advertised endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10458 }, "last packet ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... ": 1448583951, "flags": [ "RTP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } }, { "local port": 53865, "endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10459 }, "advertised endpoint": { "family": "IPv4", "address": "192.98.102.10", "port": 10459 }, "last packet": 1448583951, "flags": [ "RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } } ], "flags": [ "initialized", "send", "recv" ] } ] }, "0f6759297b8f1711": { "tag": "0f6759297b8f1711", ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.0", "created": 1448583951, "in dialogue with": "3322319973f3909e", "medias": [ { "index": 1, "type": "audio", "protocol": "RTP/AVP", "streams": [ { "local port": 53844, "endpoint": { "family": "IPv4", "address": "10.0.2.15", "port": 10144 }, "advertised endpoint": { "family": "IPv4", "address": "10.0.2.15", "port": 10144 }, "last packet": 1448583951, "flags": [ "RTP", "RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 ... Nov 27 02:25:55 lohi rtpengine[3792]: [f33a7e21c57edbf3] ... } }, { "local port": 53845, "endpoint": { "family": "IPv6", "address": "::", "port": 0 }, "advertised endpoint": { "family": "IPv6", "address": "::", "port": 0 }, "last packet": 1448583951, "flags": [ "RTCP", "fallback RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } } ], "flags": [ "initialized", "send", "recv", "rtcp-mux" ] } ] } }, "totals": { "RTP": { "packets": 0, "bytes": 0, "errors": 0 }, "RTCP": { "packets": 0, "bytes": 0, "errors": 0 } }, "result": "ok" } which then results in deletion of the whole call, i.e., also the branch that answered. question: why does rtpengine not do what it is asked to do in rtpengine-delete, i,e. delete the offer corresponding to "call-id": "f33a7e21c57edbf3", "via-branch": "z9hG4bK79d6.d19587ffadd9f72bca61d9578eb12bd9.1"? -- juha

8 years, 11 months

Generating db-tables .html file

by smititelu

Hello, I want to add a new table to the [1]. I managed to create a new *kamailio-*.xml* in "lib/srdb1/schema/", run make and see the resulting **-dbschema.xml* in doc/databases/kamailio, along with the other .sql scripts. Here I got stuck; I can't find the scrip that generates the final .html. I tried grepping the code for some keywords found in page source without success. Also, couldn't find the answer reading [2]. Any ideas how the final .html is generated? Thanks, Stefan [1] http://kamailio.org/docs/db-tables/kamailio-db-devel.html [2] http://www.kamailio.org/dokuwiki/doku.php/development:db-schema

8 years, 11 months

terminate call when dialog when remote contact disappears without BYE

by Thibault Gueslin

I would like to terminate the dialog when one party disconnects without sending a BYE (no more connctivity, client shutdown) I am using TCP connection. Should I terminate the dialog from event_route[usrloc:contact-expired] ? I have tried also to use SIP OPTION parameter from dialog module modparam("dialog", "ka_timer", 10 ) modparam("dialog", "ka_interval", 30 ) but nothing is sent) thibault

8 years, 11 months

Kamailio - Asterisk, problem with SIP trunk mode

by Mickael MONSIEUR

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

8 years, 11 months

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sr-users November 2015