sr-users March 2015

sr-users@lists.kamailio.org

88 participants
149 discussions

Can following scenario be implemented using Kamilio

by Jayanth Acharya

Would like to know if Kamilio (in it's sip-proxy role) could be configured to somehow take note of the 'Contact:' header of a 302 response from B-party, to forward the INVITE to C-party (specified by the updated Contact URI) ? Alternatively, is there a way for the B-party SIP UAS to get the proxy (Kamilio) to forward the INVITE (from A-party) to C-party ? Does Kamilio have to be configured to act in a specific / non-default role ? In doing so, B-party receives no further SIP messages for this particular dialog. thanks, J

9 years, 8 months

Re: [SR-Users] kamailio asterisk

by Slava Bendersky

Hello Daniel, Is notify should follow subscribe routes ? Here SUBSCRIBE completely look normal to me. <--- SIP read from UDP:10.18.130.46:5060 ---> SUBSCRIBE sip:10102@networklab.ca SIP/2.0 Record-Route: <sip:kamailio_pub_ip:5081;transport=tls;lr=on;ftag=08c99307d3;nat=yes> Accept: application/simple-message-summary Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKdee2.b83c2028dc57c641d2b8cad968347d6a.0;i=2 Via: SIP/2.0/TLS 192.168.0.16:5063;rport=5063;received=client_pub_ip;branch=z9hG4bK0c54032bd8905d262 Max-Forwards: 69 From: "10102" <sips:10102@networklab.ca>;tag=08c99307d3 To: <sips:10102@networklab.ca> Call-ID: cea703eb8c21f709 CSeq: 1096440084 SUBSCRIBE Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Allow-Events: message-summary Authorization: Digest username="10102",realm="networklab.ca",nonce="29ac01f1",uri="sips:10102@networklab.ca",response="0298c2a93bdd4969a3cd0b757d671dbb",algorithm=MD5 Contact: <sips:10102@client_pub_ip:5063>;audio Event: message-summary Expires: 3600 Supported: eventlist, replaces, timer User-Agent: Media5-fone/4.1.6.3283 Android/4.1.2 Content-Length: 0 Path: <sip:outbound@10.18.130.46;lr;received=sip:client_pub_ip:5063%3Btransport%3Dtls> P-hint: outbound Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 3, 2015 1:50:03 AM Subject: Re: [SR-Users] kamailio asterisk Hello, I pushed a fix for it -- the issue was in a LM_DBG() trying to print te address for local socket, but it was not set yet, it was affecting only when running in debug mode with debug=3. I pushed the patch to branch 4.2, so if you want the fix, then you have to install from git branch 4.2. For REGISTER, you need to add the Path header (see path module), otherwise the OPTIONS and INVITE sent out to the phone are not going via Kamailio. You have to check the version of your Asterisk to be one that supports Path. Cheers, Daniel On 27/02/15 14:40, Slava Bendersky wrote: Hello Daniel, Here paste for gdb http://fpaste.org/191338/25043949/ I got REGISTER and SUBSCRIBE start working correctly I see on asterisk correct record routes and sip traffic flow, but when asterisk or client ( soft phone) send OPTIONS or NOTIFY can't get properly relay it. This is SUBSCRIBE route. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TLS 192.168.88.252:5062;received=Client public ip;branch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> Record-Route: <sip:PUBLIC_KAMAILIO_IP:5081;transport=tls;r2=on;lr=on;ftag=a185d974ec;nat=yes> From: "Slava Bendersky" <sips:10101@networklab.ca> ;tag=a185d974ec To: <sips:10101@networklab.ca> ;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060> ;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, February 27, 2015 6:42:33 AM Subject: Re: [SR-Users] kamailio asterisk Hello, I asked for the wrong command, as the bt full was already sent before -- I wanted the output from gdb for: p *tcpconn Daniel On 27/02/15 04:10, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here bt full from back trace. http://fpaste.org/191207/50064491/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, February 26, 2015 9:56:48 PM Subject: Re: [SR-Users] kamailio asterisk Hello Daniel, I tried $rz option on top of request route and that where I see wrong request uri like sip:sips : . And as far I can see it happenes only for SUBSCRIBE INVITE and NOTIFY. if($rz=="sips") { $ru = "sip" + $(ru{s.substr,4,0}); } Slava. Sent from mobile device typos are expected. From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Feb 25, 2015 1:04 PM To: Slava Bendersky Cc: sr-users Subject: Re: [SR-Users] kamailio asterisk Hello, On 25/02/15 17:19, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, substr you suggested didn't worked. See my previous email. your previous email didn't say anything about the results. That's why I asked. Be sure you don't have those spaces that are in the email you wrote. Also, I had more parenthesis in the parameter of the subst_uri(). Or you can try the alternative with: if($rz=="sips") { $ru = "sip" + $(ru{s.substr,4,0}); } I asked for more details from the backtrace to confirm that what I found is the cause for the crash in this case -- see one of my previous emails from today. Cheers, Daniel BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

9 years, 8 months

REmove Header line with tags

by Yuriy Gorlichenko

I need to remove all header line witht tags but remove_hf() removes only Header:value All tags after ";" stay at the packet as garbage and next Header moves up As there ACK sip:12345678900@phone.provider.com SIP/2.0 Via: SIP/2.0/UDP sip.server.com:5068 ;branch=z9hG4bKcb7a.9f70ce5990dd153cd0eaa6f1762c059c.0.cs102 *<There var another Via header that I removed with remove_hf_re but tags is stiil there and Route header mover up>*;rport=1578;received=23.101.134.111Route: <sip:54.241.2.206;lr=on;ftag=as38f20f7b;did=ce8.10a> Max-Forwards: 70 From: "New User" <sip:00987654321@phone.provider.com>;tag=as38f20f7b To: <sip:12345678900@phone.provider.com>;tag=B4r1B569001DB Contact: <sip:contact@10.0.1.17:50600;alias=23.101.134.111~1578~1> Call-ID: 473fc8a64d7d96aa515780dc6237a6d5@10.0.1.17:50600 CSeq: 103 ACK User-Agent: Some Device Content-Length: 0

9 years, 8 months

Re: [SR-Users] kamailio asterisk

by Slava Bendersky

Hello Daniel, Here paste for gdb http://fpaste.org/191338/25043949/ I got REGISTER and SUBSCRIBE start working correctly I see on asterisk correct record routes and sip traffic flow, but when asterisk or client ( soft phone) send OPTIONS or NOTIFY can't get properly relay it. This is SUBSCRIBE route. <--- Transmitting (NAT) to 10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.46;branch=z9hG4bKf852.e0223f39c2bbad8366fdf1b7cb22b336.0;i=8;received=10.18.130.46;rport=5060 Via: SIP/2.0/TLS 192.168.88.252:5062;received=Client public ip;branch=z9hG4bK0bbe1f7d27257bba9;rport=5062 Record-Route: <sip:10.18.130.46;r2=on;lr=on;ftag=a185d974ec;nat=yes> Record-Route: <sip:PUBLIC_KAMAILIO_IP:5081;transport=tls;r2=on;lr=on;ftag=a185d974ec;nat=yes> From: "Slava Bendersky" <sips:10101@networklab.ca>;tag=a185d974ec To: <sips:10101@networklab.ca>;tag=as00757d3e Call-ID: b08adb1ad1804a83 CSeq: 236711034 SUBSCRIBE Server: FPBX-2.11.0(11.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:10101@10.18.130.50:5060>;expires=3600 Content-Length: 0 Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, February 27, 2015 6:42:33 AM Subject: Re: [SR-Users] kamailio asterisk Hello, I asked for the wrong command, as the bt full was already sent before -- I wanted the output from gdb for: p *tcpconn Daniel On 27/02/15 04:10, Slava Bendersky wrote: Hello Daniel, Here bt full from back trace. http://fpaste.org/191207/50064491/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, February 26, 2015 9:56:48 PM Subject: Re: [SR-Users] kamailio asterisk Hello Daniel, I tried $rz option on top of request route and that where I see wrong request uri like sip:sips : . And as far I can see it happenes only for SUBSCRIBE INVITE and NOTIFY. if($rz=="sips") { $ru = "sip" + $(ru{s.substr,4,0}); } Slava. Sent from mobile device typos are expected. From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Feb 25, 2015 1:04 PM To: Slava Bendersky Cc: sr-users Subject: Re: [SR-Users] kamailio asterisk Hello, On 25/02/15 17:19, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, substr you suggested didn't worked. See my previous email. your previous email didn't say anything about the results. That's why I asked. Be sure you don't have those spaces that are in the email you wrote. Also, I had more parenthesis in the parameter of the subst_uri(). Or you can try the alternative with: if($rz=="sips") { $ru = "sip" + $(ru{s.substr,4,0}); } I asked for more details from the backtrace to confirm that what I found is the cause for the crash in this case -- see one of my previous emails from today. Cheers, Daniel BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

9 years, 8 months

Kamailio as a SIP softphone to WebRTC/SIP-over-websockets proxy?

by Rion Carter

I'm pretty new to SIP, RTP/SRTP, WebRTC and Websockets, so I hope this question is coherent. I have a group of SIP Softphones that need to connect to a WebRTC/SIP-over-Websockets server. Can Kamailio be configured to let me do this? Any examples, tutorials or documentation would be appreciated. I'm trying to determine how feasible this task is. :) Thanks!

9 years, 8 months

db_mysql errors

by jaflong jaflong

I am getting errors when I run the following code if (sql_xquery("mydb", "select * from account where account = 'demo'", "res") == 1) { xlog("L_INFO", "my number: $xavp(res=>number)\n"); } else { xlog("L_WARN", "Connection forbidden from $si\n"); sl_send_reply("403", "Forbidden"); exit; } sql_result_free("res"); } 10(4033) ERROR: <core> [db_ut.c:225]: db_str2time(): Error during time conversion 10(4033) ERROR: <core> [db_val.c:159]: db_str2val(): error while converting datetime value from string 10(4033) ERROR: db_mysql [km_row.c:68]: db_mysql_convert_row(): failed to convert value 10(4033) ERROR: db_mysql [km_res.c:190]: db_mysql_convert_rows(): error while converting row #0 10(4033) ERROR: db_mysql [km_res.c:219]: db_mysql_convert_result(): error while converting rows 10(4033) ERROR: db_mysql [km_dbase.c:252]: db_mysql_store_result(): error while converting result 10(4033) ERROR: <core> [db_query.c:188]: db_do_raw_query(): error while storing result10(4033) ERROR: sqlops [sql_api.c:454]: sql_exec_xquery(): cannot do the query Any help please

9 years, 8 months

kamctl: command not available

by Thomas Baeumer

Hi there, I want to use kamctl to monitor the state of a sip-proxy. I just have no idea which commands are available or should i use "kamctl mi" or "kamctl fifo" commands. Google dont give helpful links for that. Im am using kamailio 4.2.3+wheezy on a Debian wheezy and kamctrl dont work: # kamctl fifo uptime Control engine 'FIFO' loaded entering fifo_cmd uptime 500 command 'uptime' not available FIFO command was: :uptime:kamailio_receiver_19959 # kamctl fifo which Control engine 'FIFO' loaded entering fifo_cmd which 500 command 'which' not available FIFO command was: :which:kamailio_receiver_19985 # kamctl mi which Control engine 'FIFO' loaded entering fifo_cmd which 500 command 'which' not available FIFO command was: :which:kamailio_receiver_20016 here is my config: # grep -i fifo kamailio.cfg loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_user", "kamailio") modparam("mi_fifo", "fifo_mode", 0640) # ls -la /tmp/kam* srw-r----- 1 kamailio root 0 Feb 27 12:39 /tmp/kamailio_ctl prw-r----- 1 kamailio kamailio 0 Feb 27 14:25 /tmp/kamailio_fifo The only known command what is working is to look into a hashtable: # kamctl mi sht_dump foo Control engine 'FIFO' loaded entering fifo_cmd sht_dump foo Entry:: 135 sip:0049xxxxxxxxxxx@sip.domain.tld:: 1 Entry:: 205 sip:0049yyyyyyyyyyy@sip.domain.tld:: 1 FIFO command was: :sht_dump:kamailio_receiver_20144 foo any idea what is wrong? and is there a documentation how to use "kamcrl mi" and "kamctl fifo"? Thanks Thomas

9 years, 8 months

Re: [SR-Users] Kamailio + SIP Over WebSocket examples

by Daniel-Constantin Mierla

Hello, sr-users is the mailing list more appropriate for such discussions that the docs. However, you wrote to -owner address which is not the mailing list discussions address but mailing list control address. For the topic, look at the readme of the websocket module, it gives a sample for starting with webrtc. On github (search for), Carlos Ruiz Diaz has some more advanced examples. Cheers, Daniel On 02/03/15 06:48, vineet wrote: > Hi Geeks, > > I was trying to search the good article, blog or tutorials to get > working Kamailio + WEBRTC model i have found some but those are not > worth it to me because I am beginner on it. Folk if anyone has > experience about it or having an idea about it, please refer or > forward to me. > > Thanks In advance. > > Best Regards, > -Vineet > -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

9 years, 8 months

Kamailio + SIP Over WebSocket examples

by vineet

Hi Geeks, I was trying to search the good article, blog or tutorials to get working Kamailio + WEBRTC model i have found some but those are not worth it to me because I am beginner on it. Folk if anyone has experience about it or having an idea about it, please refer or forward to me. Thanks In advance . Best Regards, -Vineet

9 years, 8 months

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sr-users March 2015