sr-users March 2015

sr-users@lists.kamailio.org

88 participants
149 discussions

by Daniel-Constantin Mierla

Hello, in order to facilitate easier interaction and updates, references (http://www.kamailio.org/w/references/) and some of the business use cases already listed on kamailio.org are in the process of being moved to a new page: - http://www.kamailio.org/w/used-by/ This offers now the possibility to submit new entries without the interaction with Kamailio admins that could lead to delays. The form is available at: - http://www.kamailio.org/w/used-by-form/ The goals is to show where Kamailio is used as well as increase the visibility of the products, solutions and services offered by the Kamailio community. Anyone is welcome to add own references. Besides a spambot protection of the form, we rely for now on the good manners of human nature, if you notice anything inadequate there, please report to be corrected. Cheers, Daniel -- 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

tsilo module

by Alex Balashov

Hello, I don't understand the tsilo module. What is the practical use case for it? Thanks, -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

9 years, 8 months

Kamailio BLF Issue TCP vs UDP

by Fred Posner

Greetings all: Having a weird issue with BLF relay from Kamailio <-> FreeSWITCH on the EC2 network. Set-up: Natted Client <-> Natted Kamailio <-> Natted FreeSWITCH Kamailio has a public advertised IP (Amazon Cloud) and sends to FreeSWITCH's public IP (Amazon Cloud). Clients register to Kamailio and the registration sent to FreeSWITCH (via uac). What works: When a client registers to kamailio via UDP, all works beautifully. What doesn't work: When a client registers to kamailio via TCP, a NOTIFY is sent with almost no information. Example of UDP traffic: U 2015/03/26 22:12:30.483767 FREESWITCH_IPADDR:5060 -> KAMAILIO_NAT_IPADDR:5060 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKv7X88FmKg04UF. Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Record-Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Max-Forwards: 70. From: <sip:3000@CLIENT_DOMAIN;user=phone>;tag=YF4HMSQNlOOt. To: <sip:3000@CLIENT_DOMAIN>;tag=xew5nmpr9q. Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj. CSeq: 366877453 NOTIFY. Contact: <sip:3000@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: dialog. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=2686. Content-Type: application/dialog-info+xml. Content-Length: 546. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="14" state="full" entity="sip:3000@CLIENT_DOMAIN"> <dialog id="312b3bee-d405-11e4-a8b6-41b65d0bd045" direction="initiator"> <state>confirmed</state> <local> <identity display="3000">sip:3000@CLIENT_DOMAIN</identity> <target uri="sip:3000@CLIENT_DOMAIN"> <param pname="+sip.rendering" pvalue="yes"/> </target> </local> <remote> <identity display="*98">sip:*98@CLIENT_DOMAIN</identity> <target uri="sip:**3000@CLIENT_DOMAIN"/> </remote> </dialog> </dialog-info> U 2015/03/26 22:12:30.483877 KAMAILIO_NAT_IPADDR:5060 -> CLIENT_PUBLIC_IPADDR:42477 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0. Via: SIP/2.0/UDP KAMAILIO_PUBLIC_IPADDR:5060;branch=z9hG4bKfabf.4908255bf1623d250ddf7d51a3b49fb8.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKv7X88FmKg04UF. Record-Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Max-Forwards: 69. From: <sip:3000@CLIENT_DOMAIN;user=phone>;tag=YF4HMSQNlOOt. To: <sip:3000@CLIENT_DOMAIN>;tag=xew5nmpr9q. Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj. CSeq: 366877453 NOTIFY. Contact: <sip:3000@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: dialog. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=2686. Content-Type: application/dialog-info+xml. Content-Length: 546. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="14" state="full" entity="sip:3000@CLIENT_DOMAIN"> <dialog id="312b3bee-d405-11e4-a8b6-41b65d0bd045" direction="initiator"> <state>confirmed</state> <local> <identity display="3000">sip:3000@CLIENT_DOMAIN</identity> <target uri="sip:3000@CLIENT_DOMAIN"> <param pname="+sip.rendering" pvalue="yes"/> </target> </local> <remote> <identity display="*98">sip:*98@CLIENT_DOMAIN</identity> <target uri="sip:**3000@CLIENT_DOMAIN"/> </remote> </dialog> </dialog-info> Example of TCP failure: U 2015/03/26 22:19:45.963196 FREESWITCH_IPADDR:5060 -> KAMAILIO_NAT_IPADDR:5160 NOTIFY sip:3001@CLIENT_PUBLIC_IPADDR:5065;transport=tcp SIP/2.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKjH5Nt8QKUU9Sp. Route: <sip:KAMAILIO_PUBLIC_IPADDR:5160;r2=on;lr=on;ftag=9jSFr02tgvHym;nat=yes>. Route: <sip:KAMAILIO_PUBLIC_IPADDR:5160;transport=tcp;r2=on;lr=on;ftag=9jSFr02tgvHym;nat=yes>. Max-Forwards: 70. From: "3000" <sip:3000@CLIENT_DOMAIN>;tag=9jSFr02tgvHym. To: <sip:3001@CLIENT_DOMAIN>;tag=1dba0f3ceb32436bi0. Call-ID: 0882b1bc-4ea9-1233-f3a2-12e513dfd8f0. CSeq: 73368127 NOTIFY. Contact: <sip:mod_sofia@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: refer;id=101. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: message/sipfrag;version=2.0. Content-Length: 16. . SIP/2.0 200 OK. T 2015/03/26 22:19:45.963407 KAMAILIO_NAT_IPADDR:5160 -> CLIENT_PUBLIC_IPADDR:5065 [AP] NOTIFY sip:3001@CLIENT_PUBLIC_IPADDR:5065;transport=tcp SIP/2.0. Via: SIP/2.0/TCP KAMAILIO_PUBLIC_IPADDR:5160;branch=z9hG4bKe961.8d1f19f1acc0295e5e397afa76910115.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKjH5Nt8QKUU9Sp. Max-Forwards: 69. From: "3000" <sip:3000@CLIENT_DOMAIN>;tag=9jSFr02tgvHym. To: <sip:3001@CLIENT_DOMAIN>;tag=1dba0f3ceb32436bi0. Call-ID: 0882b1bc-4ea9-1233-f3a2-12e513dfd8f0. CSeq: 73368127 NOTIFY. Contact: <sip:mod_sofia@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: refer;id=101. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: message/sipfrag;version=2.0. Content-Length: 16. . SIP/2.0 200 OK. My thoughts is that this is on FreeSWITCH, just not sending a NOTIFY with anything of value. That being said, I'd love to know if (a) anyone else has had this issue or (b) if someone has a different theory. Thanks! -- Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

9 years, 8 months

rtpengine DTLS

by Marc Soda

I'm running a new kamailio/rtpengine instance and, using webrtc, I can't get audio to flow between the browser and rtpengine. Kamailio seems to be handing it off properly. Based on the rtpengine logs and a packet capture it doesn't look like DTLS is being negotiated properly: https://gist.github.com/marcantonio/ea0077a5884e4e4b6b45 I see "SRTCP output wanted, but no crypto suite was negotiated" and I've never seen that one before (the SRTCP part). I have this exact config working elsewhere, but this new setup is not working? Any advice? Thanks, Marc

9 years, 8 months

Siremis error Call to a member function getQuoted()

by canuck15

When you add/update/view Administrator usernames To reproduce Administration > User Management > Click on any existing username. The browser goes blank. Error in apache logs is: PHP Fatal error: Call to a member function getQuoted() on a non-object in /var/www/html/openbiz/bin/data/BizDataSql.php on line 237 This is on the latest Siremis v4.2.0 downloaded today I am using CentOS v6.6, php v5.3.3

9 years, 8 months

active_watchers - dbt_table_check_row(): null value not allowed - field 17

by Alexandru Raileanu

Hi, I have kamailio 4.2 with presence, auth, textdb activated. The scenario is that Kamailio receives a SUBSCRIBE, sends 200ok then a NOTIFY. (like it should do) The problem is what when I kill kamailio and run it again, it gives me this error: 0(30580) ERROR: db_text [dbt_tb.c:474]: dbt_table_check_row(): null value not allowed - field 17 0(30580) DEBUG: db_text [dbt_file.c:480]: dbt_load_file(): error at row=1 col=26 c=1 0(30580) ERROR: db_text [dbt_lib.c:307]: dbt_db_get_table(): could not load database from file [active_watchers] 0(30580) ERROR: db_text [dbt_base.c:460]: dbt_delete(): failed to load table <active_watchers>! The field with null values is record_route, so if I put some value I manage to run kamailio. my conf, maybe I missed some field? #!define DBURL "text:///etc/kamailio/dbtext" #!ifdef WITH_PRESENCE modparam("presence", "db_url", DBURL) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:10.0.37.1:5060" ) modparam("pua_reginfo", "default_domain", "10.0.37.1") modparam("pua_reginfo", "server_address", "sip:10.0.37.1") modparam("pua_reginfo", "outbound_proxy", "sip:10.0.37.1") modparam("pua_reginfo", "publish_reginfo", 1) modparam("pua", "db_mode", 2) modparam("pua", "hash_size", 11) modparam("pua", "db_url", DBURL) modparam("pua", "db_table", "pua") modparam("pua", "min_expires", 0) modparam("pua", "default_expires", 3600) modparam("pua", "update_period", 1000) modparam("pua", "dlginfo_increase_version", 1) modparam("pua", "reginfo_increase_version", 1) modparam("pua", "check_remote_contact", 0) modparam("pua", "fetch_rows", 1000) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif Or as a workaround, can I reset the active_watchers table every time I start kamailio? because *kamdbctl create/drop/init* wipes all the other tables... thanks a lot! Alex

9 years, 8 months

File transferring issue

by Wingsravi R

Dear Kamailio'ns, I am working on kamailio Server (V 4.0) installed on Ubuntu 12.04 system. Currently i am trying file transfer feature using IMSDroid SIP clients. file transferring is fine between two IMSDroid clients but at the receiving end (file recipient) i cont see the sender identity/ sender name. it is just displaying as 'null' (Screen shot is attached). But the full file is transferring to destination address without any problem. I have configured MSRP module related script in my kamailio configuration file (find the attachment). but i guess it is not invoking anyway when file transferring and the file transferring is negotiating client-to-client basis. And also find Sip capture of file transfer (ngrep based) for better understanding. What could be the problem? Why at the file recipient side it is not displaying sender's identity (instead it is 'null')? how can i resolve this issue? Anyone please help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi.

9 years, 8 months

[4.2.x] "socket" feature missing in dispatcher module

by Sven Neuhaus

Hello, in 4.2.2, the "socket" module was added to the documentation of the dispatcher module (see http://kamailio.org/docs/modules/4.2.x/modules/dispatcher.html#dispatcher.e… ), however the code is missing, the feature isn't implemented in 4.2.x. This was in commit 9bb59f922. Please include the patch that adds this feature from the trunk in the 4.2.x branch, as documented. A brief search turned up the commits aae775859f54f84734ba2cfc2945c02c52b95e1f and 2a605383d5c524d9875256c7f95f64216b970331 with this feature, perhaps there are more. Thanks, -Sven Neuhaus

9 years, 8 months

Re: [SR-Users] kamailio asterisk NOTIFY

by Slava Bendersky

Hello Daniel, Look like transactions are not match when it send reply for NOTIFY. I think it should reuse connection. Here SDP output where cseq are match and asterisk does retransmissions until time out. Also I wonder where Path route is reused I see %* instead ; as delimiter between transport and port. Retransmitting #8 (NAT) to 10.18.130.46:5060: NOTIFY sip:191@192.168.88.246:5062;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport Max-Forwards: 70 Route: <sip:inbound@10.18.130.46;lr;received=sip:client_public_ip:5062%3Btransport%3Dtcp> From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp> Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 24/0 (0/0) --- Retransmitting #9 (NAT) to 10.18.130.46:5060: NOTIFY sip:191@192.168.88.246:5062;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport Max-Forwards: 70 Route: <sip:inbound@10.18.130.46;lr;received=sip:99.251.238.38:5062%3Btransport%3Dtcp> right here From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp> Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 24/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport=5060;received=10.18.130.51 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp>;tag=dc299d9ac16f3377ac0b8dfa22eaf0a6-2fe2 Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Mar 23 08:45:47 canlvprx01 /usr/sbin/kamailio[20143]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect client_pub_ip:5063 failed (timeout) 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: Monday, March 23, 2015 4:21:47 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, as you can see the 408 is for another request, because the cseq is 103. The first one has cseq 102, but has no Record-Route header, which is mandated by new RFCs. In kamailio, if you added in request_route when handling the NOTIFY, you can try to add record route via reply_route if the client is not keeping it in reply, kind of following fix: reply_route { if(is_method("NOTIFY")) { if(!is_present_hf("Record-Route")) { insert_hf("Record-Route: < sip:yourserverip:port;transport=...\r\n "); } } Cheers, Daniel On 22/03/15 18:11, Slava Bendersky wrote: Hello Daniel, I still can't understand clearly why when asterisk send NOTIFY to client through kamailio it coming back with with 408. I I tried different case event change Contact:, but no luck can't I see what the issue all routes in first SUBSCRIBE looks ok . Any though or help will be really appreciate. This what I see on asterisk debug Reliably Transmitting (NAT) to 10.18.130.46:5060: NOTIFY sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK5dd3f266 Max-Forwards: 70 Route: <sip:10.18.130.46:5060;lr=on;ftag=3b27a2a409;nat=yes> --->Kamailio route From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2> ;tag=3b27a2a409 Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: b1edde040d695988 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 88 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 5/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.51:5060;rport=5060;branch=z9hG4bK5dd3f266 From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2> ;tag=3b27a2a409 Call-ID: b1edde040d695988 CSeq: 102 NOTIFY Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Server: Media5-fone/4.1.6.3283 Android/5.1 Supported: eventlist, replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6386387f;rport=5060 From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as00936a6d To: <sip:102@192.168.88.243:5066;alias=client_pub_ip~5066~2> ;tag=a3aea18741 Call-ID: 5a66b87a618d1042 CSeq: 103 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:28:33 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, that function might be in textopsx module. Cheers, Daniel On 13/03/15 14:25, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Where possible find updated version of options for textops. [sr_module.c:701]: find_mod_export_record(): find_export_record: <remove_hf_value> not found Specially Record-Route manipulation. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 8:54:26 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, Here paste for first SUBSCRIBE. http://fpaste.org/197545/25118514/ 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: Thursday, March 12, 2015 5:09:17 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, can you get the sip traffic from the network (from the first subscribe of this presence dialog)? It will help to see the Route set. Cheers, Daniel On 12/03/15 00:36, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ 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 Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.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 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 _______________________________________________ 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 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 _______________________________________________ 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 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

msilo module

by Rajat Yadav

Hello i m very new to kamailio.I m trying to develop an application using webrtc and kamailio. In this i tried using msilo module so as to store offline messages but wasnt able to achieve it.After that i searched a lot on internet about working msilo codes but i didnt find it. Can someone please send me a working code of msilo which i can embed in the kamailio configuration file. Thank you.

9 years, 8 months

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sr-users March 2015