- sr-users - kamailio.org

htable lock - unlock
by Marino Mileti 15 Dec '16

15 Dec '16

Hi guys, I'm going out of mind J I've this piece of script inside an "on_reply_route" that captures only 200OK of INVITE (i would like to save in ht the contact of who has answered a call) sht_lock("pht=>$ci"); if ($sht(pht=>$ci)!=$null){ xlog("L_INFO","Before assignment.."); $sht(pht=>$ci)=$var(myvar); xlog("L_INFO","Assigned done..."); sht_unlock("pht=>$ci"); } else{ xlog("L_INFO","Skipping.."); } sht_unlock("pht=>$(hdr(Call-ID))"); The situation is that I never see the "Assigned done" message in log file... Is this the right use of sht_lock/sht_unlock method? Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

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Kamailio 4.3 Rewriting From and db_alias values
by Gonzalo Gasca Meza 15 Dec '16

15 Dec '16

Hi forum, This is the scenario: VM <- Kamailio -> SP1 ----- SP2 ---- PhoneA (john) PhoneB PhoneA is configured in *dbaliases* table: alias_username, alias_domain, username, domain 11111111, sp1.com, john, sp2.com When PhoneB dials 1111111, Kamailio converts 111111(a)sp1.com to john(a)sp2.com Now I need to provide VM services to PhoneA. (Opposite direction) PhoneA calls voicemail but the calling number is john(a)sp2.com I need to have SP1 use the 11111111(a)sp1.com format How can I overwrite the From? Pseudo algorithm ------------------------ If destination is VM and from domain ($fU) is sp2.com: Find user URI: ( In this case john extracted from john(a)sp2.com in dbaliases) If user is found: rewrite From URI: use alias_username@alias_domain I'm using default kamailio.cfg and calls successfully are forwarded to VM Thank you

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Question on multiple 200OK received after forking
by Marino Mileti 14 Dec '16

14 Dec '16

Hi guys, I've a Kamailio instance listening on both eth0 and wlan0. Kamailio forks call to two branches, so both clients rings together. In case (very very very rare J) of simultaneous answers is there any way to kill/hangup/destroy/cancel the "slowest" leg? I've tried to arrange a semaphore with lock/unlock capabilities of cfgutils, I've used the drop command but client still sending the 200OK because it's waiting for an ACK L Any help will be very appreciated J Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

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question on call cancel
by Kelvin Chua 14 Dec '16

14 Dec '16

I have a peculiar scenario, maybe you guys have an easy solution: 1. a call comes in through an SBC (which is also kamailio) 2. SBC passes call to core kamailio server 3. call reaches 180 ringing state 4. client suddenly changes from wifi to 4G (using 4G IP, VIA headers are all updated) 5. client sends cancel and receives cancelling msg from SBC 6. SBC sends cancel to kamailio core 7. kamailio core sends back Request Terminated after termination the issue is, this message was sent to the wifi IP not the 4G when checking the network dumps, looks like this was because the INVITE was received on kamailio core from the old wifi IP. Is there maybe a way to update the transaction perhaps? Kelvin Chua

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Multi-Domain dispatcher
by Satish Patel 14 Dec '16

14 Dec '16

We have currently dispatcher running for single domain, but in future we have more domain coming so i want to do multi domain dispatching for example dispatcher redirect request for foo.com to foo.registar and bar.com to bar.registar SIP server currently we have following code in dispatcher logic. what should we do to handle multi-domain if(!ds_select_dst("1", "2")) { # if we are here that means no destination is available. We notify the user by 404 and exit the script. xlog("L_NOTICE", "No destination available!"); send_reply("404", "No destination"); exit; }

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regular expression question
by Satish Patel 14 Dec '16

14 Dec '16

I am trying to block SIP scanner so i am trying to use following logic only allow "My-UserAgent" and block rest but its throwing error, but if i use =~ regular expression which works! why negative match doesn't work? if($ua !~ "(My-UserAgent") { xlog("L_INFO","On more scriptkiddie, coming from $si, blocking"); exit; } what could be wrong here?

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event for new tcp connection from client
by Andy 14 Dec '16

14 Dec '16

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.

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Kamailio TM - t_set_fr
by José Seabra 13 Dec '16

13 Dec '16

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

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Accessing SDP changes
by Pete Kelly 13 Dec '16

13 Dec '16

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

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Tsilo iOS Push Notifications with Kamailio/Asterisk Integration
by Benjamin Fitzgerald 12 Dec '16

12 Dec '16

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

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DTMF Using INFO
by Nahum Nir 12 Dec '16

12 Dec '16

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

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msilo timestamp
by Slava Bendersky 12 Dec '16

12 Dec '16

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.

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Error using Siremis 4.3.0 with Kamailio 4.4 and PHP 7
by Diogenes Aviles 09 Dec '16

09 Dec '16

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

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Asterisk Security Advisory (AST-2016-009)
by Matthew Jordan 09 Dec '16

09 Dec '16

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

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IP-based security + registrations
by JLL 09 Dec '16

09 Dec '16

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!

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Siremis manual to learn
by Saranga Nagesh 08 Dec '16

08 Dec '16

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

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Action: restructuring the source code tree
by Daniel-Constantin Mierla 07 Dec '16

07 Dec '16

Hello, today I am going to do the restructuring of the source code tree as discussed previously. - http://lists.sip-router.org/pipermail/sr-dev/2016-December/037715.html If anyone has patches to push using existing structure, do it before 12:00UTC today. If you need more time, write to sr-dev and let's see how we can find an easy solution. Once I finish with the restructuring, I will send another notification, from that moment the commits can go on as usual. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Send SIP Info within a dialog using $uac_req(method)="INFO"
by Jonathan Hunter 07 Dec '16

07 Dec '16

Hi Guys, Is it still the case that when using uac_req_send, you cant send withing a specific dialog?I can modify call-id, but I presume tags may be more of a problem?See old post below from 2015; > I am familiar with uac_req_send. but how do I send it with in a > specific dialog and with data in the INFO req ? sending a new request inside a dialog is not possible with uac_req_send(). It is not easy over all because you change the sequence order (CSeq value). Practically, you need to track how many requests you sent from the middle to update (and restore in reply) when caller or callee sends a new request. dialog module can track changes in CSeq for requests sent to callee, being used now for authentication of INVITE to another provider, when Kamailio adds the credentials. But for more you would need to extend the dialog module. I just need to send a SIP info within an established dialog to stop some function up stream, so wondered if this is still a blocker? Many thanks Jon

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Error using Siremis 4.3.0
by Diogenes Aviles 07 Dec '16

07 Dec '16

I'm using Siremis 4.3.0 and PHP7. Installation was OK, but running siremis I have some problems in 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 Regards -- Diogenes

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Getting destination ip
by David Villasmil 06 Dec '16

06 Dec '16

Hello Guys, I'm wondering how do I get destination IP on a reply? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Re: [SR-Users] Local messages are malformed in Kamailio 4.4.X
by g.Helio 06 Dec '16

06 Dec '16

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Limiting Registration Contacts
by Colin Morelli 05 Dec '16

05 Dec '16

Hey all, I know Kamailio's registrar module has a max_contacts parameter that will limit the number of active contacts for an AOR. However, is there any mechanism to control what happens when that number is hit? When using a web client where every page refresh registers a new contact with Kamailio, it would be useful to automatically roll off the oldest registrations and replace to make room for the new ones. I know this can be manually accomplished with sqlops, but I was hoping something like this already existed. Best, Colin

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INFO: modules purple and iptrtpproxy moved to obsolete folder
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, be aware that the modules purple and iptrtpproxy were moved to obsolete folder. They were not compiling for several years and nobody showed interest in updating them. If anyone wast to rescue them, she/he is welcome to get them back and upgrade the code to work with current versions of their dependencies. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

INFO: kcore internal lib removed
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, mainly for those having private modules or other components for kamailio, take note that the internal library kcore has been removed. It was supposed to be a termporary place where to keep some kamailio specific functions at the time of merging with ser code back in 2008. The main code was related to internal statistics, which was re-implemented as fast counters several years ago, in libkcore being only some wrappers. You have to include now directly the counters.h from core. There were several files with some str operations helpers, not they are exported via strutils.h from core. The sst parser was used only by sst module and moved to it, same being done with some stats helper functions used only by snmpstats module. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

kamailio bridge internal and external network
by zhang90501 05 Dec '16

05 Dec '16

My kamailio listen at internal ip and external ip(advertise parameter), it send message to freeswitch from internal ip and to client from external ip. How can I configure my kamailio to auto generate correct record route header?

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