[Feb 20 11:03:36] NOTICE[28271]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '"7708xxxxx" <sip:7708xxxxx@172.20.xx.xx;user=phone>' failed for '172.16.x.x:5060' (callid: isbc78i6wj7ctwcmcsjewn5s77w6s8sttcic@10.18.5.64) - No matching endpoint found
[Feb 20 11:03:16] NOTICE[28271]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '"07720xxxxx" <sip:07720xxxxx@172.20.xx.xx;user=phone>' failed for '172.16.x.x:5060' (callid: isbc7cewntite75esentt6t6578uieemspun@10.18.5.64) - No matching endpoint found
In here 7708xxxxx and 07720xxxxx both are customer numbers. In Every time I need to create endpoint for play the IVR. Can I fix this issue from Asterisk? Or how fix this?
It appears that the Asterisk CLI is indicating that incoming calls from specific numbers (e.g., “7708xxxxx” and “07720xxxxx”) are failing because there are no matching endpoints found. This typically happens when Asterisk receives a call for which it doesn’t have a corresponding endpoint definition.
To resolve this issue, you need to ensure that you have endpoint configurations in your Asterisk configuration that match the SIP addresses (URIs) from which these calls are originating. You can create endpoint definitions for these numbers so that Asterisk can properly handle incoming calls from them.
Navigate to your Asterisk configuration directory, usually located at /etc/asterisk/.
Open or create the file where you define your SIP endpoints. This file is typically named pjsip.conf or sip.conf, depending on the SIP channel driver you’re using (PJSIP or SIP).
Add an endpoint configuration for each SIP address (URI) from which you expect incoming calls. The configuration should include the necessary parameters such as type, context, host, secret, etc. Here’s a basic example:
[7708xxxxx]
type=endpoint
context=my-incoming-context ; Define the context where incoming calls will be handled
host=dynamic ; Set to 'dynamic' if the IP address can change
secret=yourpassword ; Set a secure password for authentication
Hello @danishhafeez , I already created endpoints. But the calls not went through out the IVR path. The Sip line directly configure with my Linux server ( which is installed the Asterisk).
first of all they are giving error of no matching endpoint. so you have to check your endpoint again may be it has bug.
second: SIP/2.0 401 Unauthorized
its mean issue of caller id may be your sip provider not allowed this caller id to send call.contact from your sip provider
Generally for SIP providers you match based on source IP address using a “type=identify” section. If you aren’t doing that, then that would explain it.
Endpoint: 01120xxxxx Unavailable 0 of inf
OutAuth: 01120xxxxx/Admin
InAuth: 01120xxxxx/Admin
Aor: 01120xxxxx 10
Contact: 01120xxxxx/sip:172.16.x.x 5699d0500e NonQual nan
we made discussion with our sip provider they said like they do not have any authentication to reach they end point and also they advice us if we do configuration from asterisk we have to do trunk configuration to reach the end point.
we can ping endpoint ip 172.16.x.x from our Linux which install at asterisk
we can not understand why we can not reach the endpoint from our side?
this is sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
01120xxxxx/01120xxxxx 172.16.x.x Auto (No) No 5060 OK (21 ms)