sr-users March 2011

sr-users@lists.kamailio.org

83 participants
123 discussions

by Jeremy McNamara

I am attempting to add a Reason header with Q.850 result codes. Unfortunately, this does not work: if (is_method("CANCEL")) { append_hf("Reason: Q.850; cause=16\r\n"); if (t_check_trans()) { t_relay(); } exit; } If I have to use 3.1.x, does it support sending Q.850 codes? Jeremy McNamara

13 years, 8 months

Kamailio setup

by roman＠dmytriv.com

Hello everyone!!! What location I use for chrooted directory? Thanks in advance Roman

13 years, 8 months

Re: [SR-Users] Call subscriber online

by Daniel-Constantin Mierla

Hello, shouldn't the call go to location service before relaying to subscriber B? Is B at a fix address an port and that is local host port 5060? Are you doing all in your computer for testing purposes, because otherwise an application bound to localhost (like could be the softphone B) cannot really communicate with the inter/intra-network? Cheers, Daniel On 3/11/11 4:50 PM, Stefano Larosa wrote: > > Hi, > > I'm new on Kamailio 3.0 > > This is the scenario I would like to build: > > 1 Subscriber A -> 2 kamailio -> 3 asterisk -> 4 Kamailio -> 5 > Subscriber B > > Everything is working fine until the last step > > This is the code that manage the call from asterisk to kamailio > > /if(is_method("INVITE") && (src_ip==80.169.xx.xx) )/ > > / {/ > > / route(TOPROXYUSER);/ > > / }/ > > And this is the code that should end the call the the subscriber > > route[TOPROXYUSER] { > > xlog("L_NOTICE", "$mi route[$rm][0] $fu > -> $ru START PROCESSING MESSAGE\n"); > > rewritehostport("127.0.0.1:5060"); > > if (is_method("BYE|CANCEL")) { > > route(FAIL_ONE); > > } else if (is_method("INVITE")){ > > route(RELAY); > > }; > > exit; > > } > > Thank you, > > Stivu. > > > _______________________________________________ > 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://www.asipto.com

13 years, 8 months

Registrar module creates an extra branch

by Efelin Novak

Hi Folks, I have a problem with registrar module (I suppose). When A calls to B, is some specific situation, B receives two INVITEs. This mostly (not only) happens when one account is registered on two or more UAs. In this situation extra INVITE is sent only to the one that is second in the list (kamctl ul show ACCOUNT). kamctl ul show Alice: Contact:: <sip:orlvuzyd@192.168.28.72:52129 >;q=0;expires=1916;flags=0x0;cflags=0x40;socket=<udp:8.8.8.8:5060 >;methods=0xFFFFFFFF;received=<sip:8.8.8.8:52129>;user_agent=<Blink 0.2.5 (Linux)> Contact:: <sip:Alice@192.168.28.51:5060 >;q=;expires=3061;flags=0x0;cflags=0x40;socket=<udp:8.8.8.8:5060 >;methods=0x129F;received=<sip:8.8.8.8:5060 >;user_agent=<Linksys/PAP2T-3.1.15(LS)> My situation. 1. Alice calls to BOB. 2. kamailio looks up in location table for BOBs registrations 3. in DB there are two rows (two devices(linksys PAP2T and Blink(blink is second one))) 4. kamailio send 3 INVITES ( linksys, blink and blink). Every INVITE has different via.branch (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.1) (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.2) (Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bKe2d7.009b1df2.3) All my registrar parameters are set to default. I have no idea whether this is a bug or my missconfiguration. Thanks for replies. Kind regards, Efelin Novak

13 years, 8 months

usuarios para kamailio y asterisk

by M. ASUNCION MERINO RODRIGUEZ

Hola en la actualidad estoy montando un servidor kamailio que trabaja con un servidor asterisk. Entiendo que lo mejor seria que tanto kamailio como asterisk trabajaran con los mismos usuarios, por eso el servidor kamailio autentica los usuarios contra un servidor radius. Este servidor radius, acceder a un directorio ldap, para validar a los usuarios cuyo usuario es el login de correo y la contraseña correspondiente. Mi pregunta es, como puedo manejar usuarios como "login" si en asterisk existen usuarios creados como números de telefono? debería hacer que los usuarios de radius fueran numeros de telefono? Cualquier ayuda es bienvenida Gracias

13 years, 8 months

Paging from asterisk to accounts with Multiple Registrations...

by Derrick Bradbury

Hi all... interesting problem... I am using kamailio as the registration server for a cluster of asterisk servers. I am allowing multiple registrations on the same account to allow for parallel forking on a normal call... The problem is set paging... when I send the page to the account that has multiple registrations, the parallel fork works properly and the first phone to respond gets "control" of the call. Unfortunately there is an auto answer field in the invite. So simple case: one account, two phones.. invite gets sent to both phones, both phones answer... BUT one phone grabs "control" of the call, and the other phone no longer receives any more commands... so it never hangs up as it doesn't receive a "BYE". Is there any way to turn off the parallel fork behavior, and keep sending all packets to both (or more) phones? Any thoughts are appreciated! Thanks, Derrick

13 years, 8 months

Re: [SR-Users] decimal fraction problem

by Daniel-Constantin Mierla

Hello, cc-ing to the mailing list is very important because even it is an email to show the previous answer was good, that will help other people with similar problem that search on web and read the mailing list archive to know the proposed solution worked and they can use it without asking again on mailing list if it was ok or not. Thanks, Daniel On 3/14/11 12:07 PM, 侯旭光 wrote: > got it > > thanks > > cheers > > 在 2011年3月14日下午5:22，Daniel-Constantin Mierla <miconda(a)gmail.com> 写道： >> Hello, >> >> are you using kamailio 3.1.x? If not, you have to upgrade, the >> $branch(...) variable was updated to be writable starting with this version. >> >> I played last week with it in a need of combining serial forking with >> parallel forking and all is ok with assigning values to $branch(...). >> >> Cheers, >> Daniel >> >> On 3/14/11 5:25 AM, 侯旭光 wrote: >>> sorry to bother again >>> >>> >>> $(branch(q)[-1]) = $var(q); >>> >>> this script line doesn't work and the pv $branch() aren't >>> writable,just readable . index -1 is not accessable either. >>> >>> if append_branch() function doesn't take the q value parameter,the >>> $branch(q) just return NULL (which I think is the default value >>> Q_UNSPECFIED=-1) >>> >>> I find a function set_ruri_q() in dset.c but I don't know how to call >>> it in the configure file. >>> >>> >>> 2011/2/23 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>> Hello, >>>> >>>> On 2/21/11 10:28 AM, 侯旭光 wrote: >>>> >>>> Hello >>>> I need to add q value while using function append_branch(),but the function >>>> only takes decimal fraction as the parameter. >>>> What if I want to use pv to add q value? >>>> The $var and $avp just have string and integer type. >>>> Thanks a lot! >>>> >>>> do: >>>> >>>> km_append_branch("$var(branchuri)"); >>>> $(branch(q)[-1]) = $var(q); >>>> >>>> $var(q) has to hold an integer value that represents the decimal fraction >>>> value multiplied with 100 (so if q should be 0.5, then $var(q) = 50). >>>> >>>> Cheers, >>>> Daniel >>>> >>>> -- >>>> Daniel-Constantin Mierla >>>> http://www.asipto.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 >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> -- Daniel-Constantin Mierla http://www.asipto.com

13 years, 8 months

change in list setup - default is now reply to list

by Henning Westerholt

Hello, for your information - i just did a small change in the mailling list configuration for sr-dev and sr-users, now all replies goes per default to the list instead of the sender. This should help to keep the discussion on the list, especially if new people join. :-) In the past a lot of "Please always include the list in your reply" reminders were necessary.. If you choose "reply to all" then the mail still goes to the list and the sender, so no change in this case. Regards, Henning

13 years, 8 months

Siremis - Auto update dialplan on save?

by Russell Feeser

To Siremis Admin, I'm a bit of a php rookie, and even more of a noob with openbiz cubi, but I've burnt a day and a half trying to implement a new button on the SER Menu > Server Services > Dialplan List page, and curious if you could give some words of advice concerning my plight. Ideally, I would like dialplan to auto update the Kamailio mySQL database with the dp_reload feature; I am simply sick of bouncing Kamailio with a "kamctl restart" from the command line each time I implement a change to the dialplan in Siremis. I have successfully enabled this feature on the drop-down menu in SER Menu > Command Services > MI Commands, however, I don't want to have to navigate to the MI Commands page each time I wish to refresh dialplan. In siremisMICommands.xml, I was able to activate dp_reload with the following code: (taken from siremisMICommands.xml) <?xml version="1.0" standalone="no"?> <PluginService Name="siremisMICommands" Package="asipto" Class="siremisMICommands"> <MIConfig name="MIConfig" type="udp" mode="rich"> <Local name="local" address="127.0.0.1" port="8044" timeout="3.0"/> <Remote name="remote" address="127.0.0.1" port="8033"/> <MICommands> <cmd name="ps" title="List Processes" command="ps"/> <cmd name="uptime" title="Show Uptime" command="uptime"/> <cmd name="getstatsall" title="Get All Statistics" command="get_statistics all"/> <cmd name="which" title="Get All MI Commands" command="which"/> <cmd name="dp_reload" title="Reload Dailplan" command="dp_reload"/> <------ Code I added, WORKS! </MICommands> </MIConfig> </PluginService> Here's where I'm stuck: I can't seem to get a new button to implement dp_reload in the DialplanListForm.xml. I'm not sure if I need to write a function to call dp_reload, or if I can simply activate dp_reload from the DialplanListForm.xml. The code I have written creates a button on the SER Menu > Server Services > Dialplan List page, but thats as far as I've gotten. My code currently looks like this: (taken from DialplanListForm.xml) <Element Name="btn_refresh" Class="Button" Text="Activate" CssClass="button_gray_m" Description="Refresh dialplan in Kamailio without restarting Kamailio"> <EventHandler Name="dp_reload" Event="onclick" EventLogMsg="" Function="" ContextMenu="Activate"/> </Element> Anything you can say would be most helpful, as this is a feature I'm sure many using Siremis would appreciate having. Best, -- R. Z. Feeser 717.460.8095 www.RZFeeser.com www.NaturaGardens.com

13 years, 8 months

pseudo variables available in on_reply route

by Asgaroth

Hi All, I have a requirement to perform some processing based on the source and destination addresses on a message in on_reply route. I can get source ip address using $si pseudo variable, but I cant seem to access the destination ($dd). Is there any way I can access destination ip/domain of message in on_reply route? Thanks for any tips.

13 years, 8 months

← Newer
1
...
6
7
8
9
10
11
12
13
Older →

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 March 2011