[SR-Users] Kamalio & Asterisk integration erroneous behavior, setup issue

David Villasmil david.villasmil.work at gmail.com
Fri Aug 30 13:18:00 CEST 2019


The users are registered on kamailio, not asterisk, that’s why you don’t
see them in asterisk.

The voicemail is happening because asterisk doesn’t know where the user
being called is. So I assume kamailio is not forwarding the registration
location to asterisk.

Make a trace with I.e.: sngrep while registering, you should the register
forward happening.

On Fri, 30 Aug 2019 at 09:55, Aristeidis Tsitras <tsitras at gmail.com> wrote:

> new to the area and trying to setup Kamailio with Asterisk in a single
> machine. All users will register to Kamailio's port and in case of need for
> media, it will be forwarded to Asterisk, that is my intention. All of my
> work is based on the following link
> https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
> Here is what i have done:
>
>    - Debian 8, 64 bit machine with mysql and odbc
>    -
>
>
>
>
>
>
>
> *root at kamast: ~ $ lsb_release -a No LSB modules are available. Distributor
>    ID: Debian Description:    Debian GNU/Linux 8.11 (jessie) Release:
>     8.11 Codename:       jessie root at kamast: ~ $ uname -a Linux kamast
>    3.16.0-10-amd64 #1 SMP Debian 3.16.72-1 (2019-08-13) x86_64 GNU/Linux
>    root at kamast: ~ $  *
>    - Kamailio 5.2 installed from Kamailio's deb repository
>    - Asterisk 13LTS installed from source
>    - Used the same passwords such as kamailiorw and asterisk_password,
>    since this is a test system, for proof of concept.
>
> I did import to the mysql>asterisk database 3 users 2200, 2201 and 2202.
> Then created in sip.conf the same 3 users with the same credentials. Then
> on 3 PCs i used softphones (Jitsi, Zoiper) and registered each account to a
> softphone. Problems:
>
>    - Cannot see the users in the Asterisk's cli, sip show peers
>    - I can see users only in Kamailio with kamctl ul show
>    - A call between the extensions goes to voicemail. It never reaches
>    the other destination eg 2200 calls 2201 and in Asterisk's console i am
>    getting a message that 2201 is absent and it goes to voicemail. The same
>    with any other extension.
>
> Attached you can find:
>
>    1. Kamailio.cfg
>    2. Asterisk's sip.conf
>    3. Asterisk's extension.conf
>    4. The import that i have done to mysql for the user creation.
>
>
> I would appreciate if someone could point me to the error and help me fix
> it please?
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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