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:
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?
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--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
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