sr-users July 2016

sr-users@lists.kamailio.org

82 participants
125 discussions

Kamailio blocked when using with Radius accounting when connection is lost with the database.

by Mx R

Hi there. I have the following scheme. 1. SIP server is running kamailio with module "acc", radiusclient-ng and freeradius. 2. DB server with MSSQL. The problem is the following, if the connection is lost with the database, no one can make calls. Even after the connection to the database is restored. We have to restart the radius and kamailio. And only after that can make calls. Kamailio logs rc_send_server: no reply from RADIUS server localhost: 1813 ERROR: acc [acc.c: 720]: acc_rad_request (): radius-ing failed Radius logs. Error: rlm_sql_unixodbc: SQL down 08S01 [unixODBC] [FreeTDS] [SQL Server] Write to the server failed Below are links to a similar problem. http://lists.opensips.org/pipermail/users/2012-March/021100.html http://lists.opensips.org/pipermail/users/2012-March/021102.html http://lists.opensips.org/pipermail/users/2012-March/021105.html http://lists.opensips.org/pipermail/users/2012-March/021128.html

8 years, 4 months

Simple Kamailio configuration for Asterisk load balancing?

by Tickling Contest

Hello, I am beginning to front my Asterisk cluster with Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for this. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for Kamailio with the following features: (a) Support an arbitrarily large number of Asterisk servers (say, upto 10). (b) Offload SIP registration to a realtime/mysql DB used by the PBXs. Please also let me know if I should have to change my Asterisk PBX config in any way for this to happen. On a very related note, while I appreciate the fact that you need to _really_ understand SIP to configure Kamailio, it should be possible to get this setup by just running a script or GUI. It is difficult to find information on how to load balance multiple PBXs behind Kamailio. Thanks!

8 years, 4 months

[rtprpoxy LB]

by Noriyuki Hayashi

Hi, Thank you for be used great KAMAILIO. Regarding rtpproxy LB. I configured as below. modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.136:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.137:7722") modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.1.138:7722") modparam("rtpproxy", "rtpproxy_disable_tout", 20) modparam("rtpproxy", "rtpproxy_tout", 2) modparam("rtpproxy", "rtpproxy_retr", 2) I could not restart kamailio when one rtpproxy server is down. Does anyone tell this solution by the parameter or somthing ? I am looking forward from anyone. Thank you so much. Noriyuki Hayashi

8 years, 4 months

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail

by tegamjg

El 05/07/2016 11:36, Daniel Tryba <d.tryba(a)pocos.nl> escribió: > > Please keep the mailinglist in the loop, so everybody might benefit from > our ramblings :) > > > Still there are few things i dont understand, i am not using asterisk > > just as a voicemail server since they are actually handling also the > > calls passing first from kamailio and being load balanced to those > > asterisk boxes. May i still use call forwarding as you are using it? > > (Both asterisk have a shared storage with a clustered filesystem, so > > both will be able to see voice messages) > > Yes I think so. I use a seperate machine for voicemail but I see no > problem with other uses (I used to use it for playback of messages and > transcoding ebtween incompatible endpoints). > > By using the prefixes in kamailio to the username in $ru I have in the > extensions.conf: > > exten => _tovm-.,1,NoOp(leave voicemail) > exten => _tovm-.,n,Answer() > exten => _tovm-.,n,Set(CHANNEL(language)=nl) > exten => _tovm-.,n,Voicemail(${EXTEN:5},us) > exten => _tovm-.,n,Playback(Goodbye) > exten => _tovm-.,n,Hangup() > > exten => _getvm-.,1,NoOp(read voicemail) > exten => _getvm-.,n,Set(CHANNEL(language)=nl) > exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) > exten => _getvm-.,n,Hangup() > > > The other question is that i actually though that you need asterisk to > > have users configured in sipusers realtime table to associate their > > mailboxes, which i dont have since those users are stored in the > > subscriber table of kamailio. So am i still able to configure > > voicemail like you are doing it by syncing with the voicemail table?, > > i really hope so haha > > I forgot that fact. So yes I have a realtime sip users list (with > host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio > username, no password (this machine is not directly accessible from > outside)) That is what i was afraid of, so you are not handling kamailio users in the subscribe table of kamailio database but in sipusers table of asterisk database, is that correct? > > You might not be able to have endpoints able to subscribe to > notifications due to this. I baked something inspired by: > http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… > that appears to work for me. > > _______________________________________________ > 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

8 years, 4 months

Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail

by Daniel Tryba

Please keep the mailinglist in the loop, so everybody might benefit from our ramblings :) > Still there are few things i dont understand, i am not using asterisk > just as a voicemail server since they are actually handling also the > calls passing first from kamailio and being load balanced to those > asterisk boxes. May i still use call forwarding as you are using it? > (Both asterisk have a shared storage with a clustered filesystem, so > both will be able to see voice messages) Yes I think so. I use a seperate machine for voicemail but I see no problem with other uses (I used to use it for playback of messages and transcoding ebtween incompatible endpoints). By using the prefixes in kamailio to the username in $ru I have in the extensions.conf: exten => _tovm-.,1,NoOp(leave voicemail) exten => _tovm-.,n,Answer() exten => _tovm-.,n,Set(CHANNEL(language)=nl) exten => _tovm-.,n,Voicemail(${EXTEN:5},us) exten => _tovm-.,n,Playback(Goodbye) exten => _tovm-.,n,Hangup() exten => _getvm-.,1,NoOp(read voicemail) exten => _getvm-.,n,Set(CHANNEL(language)=nl) exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) exten => _getvm-.,n,Hangup() > The other question is that i actually though that you need asterisk to > have users configured in sipusers realtime table to associate their > mailboxes, which i dont have since those users are stored in the > subscriber table of kamailio. So am i still able to configure > voicemail like you are doing it by syncing with the voicemail table?, > i really hope so haha I forgot that fact. So yes I have a realtime sip users list (with host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio username, no password (this machine is not directly accessible from outside)) You might not be able to have endpoints able to subscribe to notifications due to this. I baked something inspired by: http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.… that appears to work for me.

8 years, 4 months

dispatcher module not fetching attrs col from DB

by Jayesh Nambiar

Hi, I'm trying to use the dispatcher module with algo 9 for weighted distribution. I have a column attrs in the dispatcher table with value in the format "weight=90" for my destination addresses. But somehow the ds_list command always shows attrs as blank. I checked that kamailio doesn't fetch the attrs column at all; neither while starting up nor while calling ds_reload. This is the query that kamailio sends to my DB while starting up or calling the fifo ds_reload: select `setid`,`destination`,`flags`,`priority` from `dispatcher` I dont even see an attr column modparam in the documentation although kamailio didnt throw an error when I defined attr_col modparam for dispatcher module. Is there a different way to tell kamailio to fetch the attrs from the DB?? I'm using the 4.4.1 version. Thanks for any suggestion or pointers to solve this. - Jayesh

8 years, 4 months

Kamailio v4.2.8 Released

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v4.2.8 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.7. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.8 (or to 4.3.x/4.4.x series). Important: this version marks the end of planned releases from branch 4.2. From now on the focus is on maintaining the branches 4.4 and 4.3 for stable releases. For more details about version 4.2.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/07/kamailio-v4-2-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.4, at this moment with its latest release v4.4.2. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

8 years, 4 months

[SR-USERS] Kamailio issues with Asterisk Voicemail

by TEG AMJG

Dear list: I am quite new to Kamailio and i am really struggling to get some answer to some questions i have been trying to get answers by reading documentation and other questions or solutions around internet. Even though i am integrating Kamailio with two Asterisk boxes with the module dispatcher, i didnt follow the quite popular guide of Asterisk Kamailio Realtime Integration ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). The reason why i didnt is that it actually behaved weird since i wanted Kamailio to be a stateful proxy and also wanted Kamailio to act as a Registrar Server and Location Server using subscriber and location tables of kamailio database, keeping both Asterisk strictly as Media Servers. Also i have been able to solve some NAT Traversal issues using RTPProxy and not using those Asterisk boxes with their nat or qualify parameters to solve NAT issues. Most of my solution started following this guide ( http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html) which let me handle registration and location with Kamailio and also changed it putting kamailio behind NAT. Now to the real question, is there anyway to include Asterisk voicemail functionality with the solution I already mentioned? the question is because, as I far as I know, Asterisk needs to associate mailboxes to its users. So i believe that i have to practically change most of my solution and find a way to make Kamailio register and locate users with the Realtime integration of Asterisk and map the registration accordingly to the table fields. But it destroys most of my NAT and register solution since i also configured one of the private IPs as a peer in Asterisk sort of like in that guide [Kamailio] type=friend host=192.168.1.244 port=5060 disallow=all allow=gsm allow=g729 allow=alaw allow=ulaw Sorry for making the question so long and to be quite ignorant with some of the concepts involving Kamailio with Asterisk Thanks Alejandro

8 years, 4 months

Increase diversion counter

by Igor Potjevlesch

Hello, I'm looking for the best way to increase the Diversion counter. If my Kamailio is the first to generate a call forward I can manage this with static value into suffix params. But, if the Diversion counter is already set, I don't see how to handle this properly. Someone has already work on this? Regards, Igor.

8 years, 4 months

msgpack/format - new module?!

by Anonim Stefan

Hi, I'm thinking of creating a new module called "msgpack" or "format" which will export config msgpack_serialize() and msgpack_deserialize() for a given input. Is a new module the way to go, or do you think of a better place for this? Thanks, Stefan

8 years, 4 months

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sr-users July 2016