sr-users December 2016

sr-users@lists.kamailio.org

71 participants
101 discussions

event for new tcp connection from client

by Andy

similar to tcp closed, can this be achieved when a client is connect to tcp -- View this message in context: http://sip-router.1086192.n5.nabble.com/event-for-new-tcp-connection-from-c… Sent from the Users mailing list archive at Nabble.com.

7 years, 11 months

Kamailio TM - t_set_fr

by José Seabra

Hello there, I'm using t_set_fr(0,2000) on calls to registered phones but by any reason this isn't working, the call only enters on the failure route after 30 seconds in case of no response from SBC/Phone, anyone has any idea what can be the reason for that? if(lookup("location")) { xlog("L_INFO", "routing to registered phone ruid=$ruid R=$ru - ID=$ci \n"); t_set_fr(0, 2000); t_on_failure("ASBC_FAILURE"); t_on_branch("ASBC_BRANCH"); route(RELAY); exit; } My RELAY route: route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } version: kamailio 4.4.2 (x86_64/linux) Thanks -- Cumprimentos José Seabra

7 years, 11 months

Accessing SDP changes

by Pete Kelly

Hi I am using Kamailio 4.4. and calling the function sdp_remove_transport in order to remove some SRTP media lines. The function works (of course!) however I would like to get access to the modified SDP within the cfg. I thought this would do the trick: sdp_remove_transport("RTP/SAVP"); sdp_get("$avp(foo)"); xlog("$avp(foo)"); However $avp(foo) does not contain the newly modified SDP. Is there a way for me to access it? I have been looking at things like t_save_lumps() and msg_apply_changes(). Any pointers or confirmation that this is or isn't possible would be very welcome! Thanks Pete

7 years, 11 months

Tsilo iOS Push Notifications with Kamailio/Asterisk Integration

by Benjamin Fitzgerald

Hello, my goal is to address iOS 10's removal of periodic TCP connections in iOS apps with their replacement PushKit, a push-based alternative all VoIP apps should implement. In order to have iOS apps receive SIP calls in the background, I believe I need to use tsilo to suspend the INVITE transaction send a push notification to the device. I have been following the presentation here ( http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-…) however I am not sure how to integrate it with Asterisk ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ). My questions can be summarized below: (1) What changes need to be made from Federico's presentation to use with the Kamailio/Asterisk integration? (2) How can I modify the config to send push notifications for EVERY invite? This is to address the scenario where a device goes offline before the SIP registration expires and a call is received within that window. iOS is very aggressive about killing background applications so it's likely as soon as user leaves the VoIP app, the device will become unreachable. Thanks for any help. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us

7 years, 11 months

DTMF Using INFO

by Nahum Nir

Hello, I tried Googling but didn'tfind anything. Is it possible to support DTMF using INFO? Thanks, Nir

7 years, 11 months

msilo timestamp

by Slava Bendersky

Hello Everyone, What format stored inc_time in msilo module ? I am trying convert it to human readable format, but time looks UTC SELECT FROM_UNIXTIME($(xavp(dbquery[$var(i)]=>inc_time)), '%M %D %Y %h:%i:%s') Slava.

7 years, 11 months

Error using Siremis 4.3.0 with Kamailio 4.4 and PHP 7

by Diogenes Aviles

I'm using Siremis 4.3.0 and PHP7. Installation was OK, but running siremis I have some problems in page "Administration-User Management". When I select an user there is the following error: [2016-12-06 23:18:47 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/siremis-4.3.0/ openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/siremis-4.3.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/siremis-4.3.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/siremis-4.3.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 110 I verified and the file "/var/www/siremis-4.3.0/openbiz/bin/Resource.php" exist. But I don't have any idea where the error is. Some help will be appreciate. Regards -- Diogenes

7 years, 11 months

Asterisk Security Advisory (AST-2016-009)

by Matthew Jordan

Hey all - The Asterisk project just released a security advisory for a security vulnerability in which Asterisk using chan_sip with a proxy can allow for unauthenticated calls. This affects all supported versions of Asterisk (11, 13, 14). Since that may be relevant to those on this mailing list who are not also on the asterisk-users mailing list, I thought it prudent to mention it here as well. A description of the vulnerability follows: Description The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you. The announcement can be seen here: http://lists.digium.com/pipermail/asterisk-announce/2016-December/000662.ht… Thanks again to Walter Doekes for reporting the vulnerability and providing the patch to fix it. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org

7 years, 11 months

IP-based security + registrations

by JLL

Hello, I would like to know if anyone can help me accomplish the following: I have a Kamailio server (4.2.3 on CentOS 7 x64) and 5 Asterisk servers (1.6.2.13 . I know, it's old but we have a lot of custom modules written for it). I need for certain IP's to be "trusted" (able to send INVITE's to K from the internet - these would be our origination providers), but any other IP's that come along would need to register before sending an INVITE (the REGISTER would be sent to any of the Asterisk servers for validation, if this is possible). Can this be done? Thank you!

7 years, 11 months

Siremis manual to learn

by Saranga Nagesh

Hi, I have installed kamailio and Siremis in centos but I don't know how to use Siremis. Can you forward me manual or link to learn about Siremis. Thanks & Best Regards, Nagesh Saranga

7 years, 11 months

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sr-users December 2016