Incoming call problem in Elastix with CITIC SIP Trunk
Hi,We've setup a Elastix and try to connect to CITIC SIP trunk. CITIC told us that they don't have any Elastix/Asterisk connection information can provided for us.
They do allow our Elastix IP to access to their gateway and give us a phone number say 88888888. ( No user name and password provided )
We have create a SIP trunk in Elastix but only able to make a outgoing call - Incoming call doesn't work.
The config for PEER Details
type=peer
insecure=invite
host=202.0.174.19
disallow=all
context=incoming_calls
allow=alaw&ulaw
No idea how to setup USER Details.
Any help/suggestion is much appreciated.
Thank you.
[attach]4151[/attach] [i=s] 本帖最後由 角色 於 2018-7-6 15:15 編輯 [/i]
I do believe that you to have to provide your fixed IP address to CITIC in order to connect their SIP server.
In general we have to register the SIP information in Asterisk box in order to receive incoming calls. I have modified the link by removing the user and password as follows:[code]
register => 852xxxxyyyy@202.0.174.19/852xxxxyyyy
[/code]Since I am not familiar with Elastic box, please find somewhere to put the above information over there.
The incoming call is defined in your settings "Context=incoming_calls". Thanks for your prompt response but it still doesn't work.:L
[attach]4152[/attach] 1. Please check whether it can be registered or not.
2. When making incoming calls, please check the log to see what happens. (In general, you have to input something like /usr/sbin/asterisk -rvvv, where the path may be incorrect, please check yours). It seems not able to register at all. Log as below;
[2018-07-06 16:08:57] NOTICE[2391]: chan_sip.c:15285 sip_reg_timeout: -- Registration for '85288888888@202.0.174.19' timed out, trying again (Attempt #2)
[2018-07-06 16:08:57] WARNING[2391]: chan_sip.c:24251 handle_response: Host '202.0.174.19:5060' does not implement 'REGISTER' Please check with the following link to see whether it helps:
[url]https://community.freepbx.org/t/how-do-i-register-a-trunk-without-username/9288[/url]
1) register string is modified as[code]
@202.0.174.19
[/code]2) If the above does not work, just remove any registration string.
3) when you make incoming calls, please observe what happens in the Asterisk log. Solved.
Change to below seems make it works.
type=friend
insecure=very
host=202.0.174.19
disallow=all
allow=alaw&ulaw
Really appreciated your help. Great place to learn and ask for help.
Have a nice weekend. Great to hear that you have solved the problem.
I forgot the above settings is used to tied two Asterisk boxes using IP address.
Also thank you for sharing the settings to other members. Please provide a complete settings of both inbound and outbound calls such that other members may consider to subscribe the CITIC SIP Trunk service. Below is the working config;
PEER Details?:
type=friend
insecure=very
host=202.0.174.19
disallow=all
allow=alaw&ulaw
USER Details
type=friend
reinvite=yes
port=5060
insecure=very
host=202.0.174.19
fromdomain=202.0.174.19
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=alaw&ulaw
qualify=300 how about login config
[quote]Below is the working config;
PEER Details?:
type=friend
insecure=very
host=202.0.174.19
disallow=all ...
[size=2][color=#999999]taihung 發表於 2018-7-7 20:07[/color] [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=45141&ptid=7207][img]http://www.telecom-cafe.com/forum/images/common/back.gif[/img][/url][/size][/quote] As per his information, we do not need to have any user credential as the peer SIP
server binds the customer IP address.
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