sr-users April 2015

sr-users@lists.kamailio.org

89 participants
144 discussions

How to dump or log the value of $avp(ice_priority) with xlog
by Filip Malenka 28 Apr '15

28 Apr '15

Hi all, I am adding the modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)") to my kamailio.cfg and I would like to know, what this "$avp(ice_priority)" returns, either 1, or 2 or anything else.. I tried with xlog xlog("The ICE priority is: $avp(ice_priority)"); and placed it right before the rtpproxy_manage() function call in my routing and did a grep on /var/log/syslog but I only get a "<null>" as returned value when attempting to make a call: Apr 25 17:47:03 localhost /usr/sbin/kamailio[19600]: ERROR: <script>: The ICE priority is: <null> I expected values 1 or 2 according to the rtpproxy module docu. http://kamailio.org/docs/modules/4.0.x/modules/rtpproxy.html#idp18746912 Thank you. Filip.

3 2

Expect the kamailio's sip account is unregistered state when the client app is shutdown
by xuefeng zhang 28 Apr '15

28 Apr '15

Hi Daniel, When the client's app is closed,the sip account may be registered state,because the expired time may have some rest time.I want to implement when the client's app is closed,the account also unregister. How can I implement this function?Would you give me some advices? Thanks! Xuefeng Zhang

6 17

subst_hf with capture groups
by Igor Potjevlesch 28 Apr '15

28 Apr '15

Hello, I can't figure out how to handle this case: subst_hf("P-Asserted-Identity", "/(.*\")0(.*)(.*\")/\133\2\3/", "a"); I'd like to substitute with \1 followed with 33 but it looks like the substitution tries to be done with \133. The capture group \133 doesn't exist. I tried with the following syntax: /${1}33 but ${1} is interpreted as a string. Someone can help me? Thank you. Regards, Igor.

2 3

Kamamilio with PjSIP on Asterisk
by Scott Pabin 28 Apr '15

28 Apr '15

Fellow Masters of the Universe, I'm familiar with using Kamailio to handle many tasks, such as SIP registration for Asterisk. However, that has all been using Asterisk's SIP protocol. I'm setting up a new project and I'd like to base it on PjSIP in Asterisk. What has been your experience and what specific tasks can I offload to Kamailio in this scenario? Are there any limitations with using PjSIP that were not an issue with SIP? What do I want it to do? For now, the typical, registrations and load balancing. Initially, two Kamailio sending traffic to two Asterisk, all sharing a common MySQL cluster. I also have a fair amount of traffic that may not require any Asterisk involvement, such as connecting calls from my carriers to my clients or client to client calls. Thanks for your thoughts guys.

1 0

add feature to dmq to allow for HA (heartbeat or pacemaker)
by Vik Killa 27 Apr '15

27 Apr '15

Hello, I was looking to implement HA with 2 Kamailio machines. I am aware of the dmq module and I thought that would be a great place to start. The problem I see is that dmq module does not have any "failure" callback mechanism. For instance, if KamA sends dmq ping to KamB, and KamB does not respond (perhaps because it's down), then nothing happens. I looked into using the dmq API and checked the dmq code to see how send_message() function works, and I dont see any indication that there is a failure callback. I would be a great improvement to add this sort of functionality to dmq and then other modules or code can be written to use the API to detect when another Kam box is up or down. This could lead to another module like dmq_ha (I'm open to writing this) Thanks, V

3 3

append_branch. How to disable original URI
by Yuriy Gorlichenko 27 Apr '15

27 Apr '15

Hello. I thry to integrate redis for location module and first at all that I do - dublicate location to redis. First At all I create analog of lookup procedure that use location but from redis. I take values from location and create branches by mannualy. All works good but branch route create dublicate of first branch. We talk about it already and I cnow that branch route creates first original Request and then create branches. So my question is how to disable creation of original URI? My cfg part of creation branches is: First of all I create massive of needed endpoints and then create branches as bellow. I create it with different сucles "while" because websockets not blocked when creates dublicate INVITE, but some UDP endpoints can not take call because answer to kamailio 482 reply and CANCELs call. It works fine when I logged on with wesocket device and UNP at one time. But when I logget with 2 UDP devices only this algorithm not worked. Thanks for help. $var(k)=0; xlog("L_INFO", "request URI is $ru"); while ($var(k)<= $var(j)){ if ($(avp(device_contact)[$var(k)])=~"device"){ xlog("L_INFO", "This is a classic UDP call to endpoint"); if ($(avp(device_received)[$var(k)])==""){ xlog("L_INFO", "Received string is EMPTY"); $du="sip:"+$(avp(device_contact[$var(k)]){s.select,1,@}); } else { xlog("L_INFO", "Received string is {$avp(device_received)[$var(k)]}"); $du=$(avp(device_received)[$var(k)]); } $var(UDP_contact)="sip:"+$(avp(device_contact[$var(k)]){s.select,1,@}); append_branch("sip:$tU@$(du{s.select,1,:})","0.3"); xlog("L_INFO","Classic Destination URI is {$(avp(device_contact[$var(k)]){s.select,1,@})} for {$tU}}. Destination is {$du}\n"); } $var(k) = $var(k) + 1; } $var(k)=0; xlog("L_INFO", "request URI is $ru"); while ($var(k)<=$var(j)){ if ($(avp(device_contact)[$var(k)])=~"transport=ws"){ xlog("L_INFO", "This is a classic UDP call to endpoint"); xlog("L_INFO", "Received string is {$avp(device_received)[$var(k)]}"); $du=$(avp(device_received)[$var(k)]); append_branch("sip:$tU@$(du{s.select,1,:})","0.7"); xlog("L_INFO","Classic Destination URI is {$(avp(device_contact[$var(k)]){s.select,1,@})} for {$tU}}. Destination is {$du}\n"); } $var(k) = $var(k) + 1;

2 1

ims_qos routing
by Federico Cabiddu 26 Apr '15

26 Apr '15

Hi, I've been recently playing, with success, with kamailio IMS modules and the examples *cscsf configuration files. Now I'm trying to implement qos but I am stuck with diameter routing. Basically I can route Rx requests only if I use "rx_forced_peer". In the other cases (Realm routing or default routing if I have no realm-specific configuration) cdp module seems not to be able to find a route. What I have in the logs is "Checking if peer XXX handles application 16777236 for vendord 0" (realm routing) or "get_routing_peer(): No connected DefaultRoute peer found for app_id 16777236 and vendor id 0." (for the default route). Looking at get_routing_peer function (cdp module) it seems that the function extracts the vendor_id either from grouped Vendor-Specific-Application-Id AVP or from Vendor-Id AVP if in the request the Auth-Application-Id AVP is present. In this case, even if Vendor-Specif-Application-Id is present, it's not taken in count by the routing logic. In the AAR generated by ims_qos there is the Auth-Application-Id AVP but not the Vendor-Id, so the vendor_id for which cdp module looks for is "0" which is not found in any of the connected peer. Am I missing something (probably)? Thank you and have a nice weekend. Federico

1 0

siptrace for keep-alive OPTIONS
by Koray Vatansever 24 Apr '15

24 Apr '15

Hi, It seems that siptrace module doesn't trace requests which are sent out by dialog module (i.e. keep-alive OPTIONS and dialog timeout BYE). Is there any easy way to do that? any suggestion?

2 1

Re: [SR-Users] Multi party body with content type as application/resource-lists+xml couldnt able to decode
by Daniel-Constantin Mierla 24 Apr '15

24 Apr '15

Hello, you have to say what you tried and what was the failure to be able to help. Pasting the INVITE will be useful as well. Cheers, Daniel On 24/04/15 08:25, Surendra Pullaiah wrote: > > HI Daniel, > > > > Can you please share , why kamailio couldn’t able to > extract the body, when INVITE comes with multi party body (sdp + > resource-lists). Is there any other way to extract the same. > > > > Please share your views… > > Regards > > Surendra > > > > _______________________________________________ > 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

1 0

Re: [SR-Users] drouting
by Keith 24 Apr '15

24 Apr '15

Hi, I am using drouting but having issues when trying to route numbers in E164 in the prefix. Any ideas how to allow for a + in the prefix field in the DB? Thanks Keith

2 1

← Newer
1
2
3
4
5
6
7
8
...
15
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users April 2015