4 Sep
2019
4 Sep
'19
12:54 p.m.
The call is never successful in your asterisk.
There is a REGISTER from kamailio to asterisk that fails, and so the INVITE
will not work either.
Try verifying all the authentication info, such as user/passwords, domains,
etc.
The you should try registering straight on asterisk. If (when) that works,
the try registering on kamailio.
And btw, if you didn’t realize that by asterisk’s log, you should probably
read
http://www.siptutorial.net/SIP/
About that public ip, your UA (Softphone) is probably configured to us STUN
to discover its own IP address (which you want when registering/calling on
a public registrar), you should disable that.
On Wed, 4 Sep 2019 at 07:07, Aristeidis Tsitras <tsitras(a)gmail.com> wrote:
HI. Attached you can find the following:
- tcpdump, named eth0.pcap
- asterisk traces
- kamailio's current config
- sip.conf
- extensions.conf
The attempt was to call from 2200 to 2201 and vice versa.
IPs:
1. System: 192.168.1.220
2. Windows PC 192.168.1.7
3. Android phone 192.168.1.124
Another thing that i noticed is that when i have the adsl attached on the
network then on the pcap file i see the public IP of the router something
like 2200(a)62.38.100.150 instead of 2200(a)192.168.1.7. When i unplug the
adsl from the router, then i get 2200(a)192.168.1.7. This happens only for
the windows PC.
Also there was an attempt from a second windows PC in the network to
register, but i could not see anything coming through. It is the extension
2202 and the IP is 192.168.1.8
I would appreciate any help available, please.
Στις Παρ, 30 Αυγ 2019 στις 4:09 μ.μ., ο/η David Villasmil <
david.villasmil.work(a)gmail.com> έγραψε:
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
Log into asterisk’s cli and see what it has to
say.
On Fri, 30 Aug 2019 at 14:07, Aristeidis Tsitras <tsitras(a)gmail.com>
wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
attached you can find the pcap.
IPs
- 192.168.1.220. Kamailio at port 5060 and Asterisk at port 5080
- 192.168.1.7. Zoiper softphone in Windows PC. extension 2200
- 192.168.1.124. Softphone in Android phone. extension 2201
2200 calls 2201. 2200's softphone says call established, probably
voicemail, but 2201 never receives the call.
Thanks in advance for your help
Στις Παρ, 30 Αυγ 2019 στις 2:19 μ.μ., ο/η David Villasmil <
david.villasmil.work(a)gmail.com> έγραψε:
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)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@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@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@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(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users