I'm trying to place an outbound call with RingCentral's web phone but am getting a "subscriber not found" message after a few seconds. Incoming calls work fine.
I doublechecked the auth and only get green checkmarks there.
The ringcentral dashboard even shows me the call was successful but there is no indication on the frontend that it ever worked.
I am in Sandbox mode on Ringcentral
Any ideas on what could be the culprit?
https://github.com/ringcentral/ringcentral-web-phone
SIP/2.0 404 Not Found
Via: SIP/2.0/WSS bgnnpnllhaei.invalid;branch=z9hG4bK2664025;received=184.66.244.45
To: <sip:+1###########sip.devtest.ringcentral.com>;tag=10.28.20.49-5070-c12bed41-9dc6-45ca-
From: <sip:1##########*101#sip.devtest.ringcentral.com>;tag=vr1es6q3l5
Call-ID: c1fjtb48lnq8e8h84tl6
CSeq: 7146 INVITE
Contact: <sip:+1###########104.245.63.102:8083;transport=wss>
p-rc-api-ids: party-id=p-a1c1431292a17z1862d48bf09z5beaf0000-1;session-id=s-a1c1431292a17z1862d48bf09z5beaf0000
Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO
Reason: SIP;cause=404;text="Subscriber not found"
Content-Length: 0
Related
I keep running into this error after setting up the webphone using ringcentral-call-js.
The webphone will work for a few phone calls but eventually run into this. I have no idea what could be causing it and I don't find any information online about it.
SIP/2.0 603 Too Many Contacts
Via: SIP/2.0/WSS femcvfqh8p1f.invalid;branch=z9hG4bK6653901;received=24.108.116.162
To: <sip:1##########*101#sip.devtest.ringcentral.com>;tag=7BvYdcd7fcr
From: <sip:1##########*101#sip.devtest.ringcentral.com>;tag=vbdr8qojhl
Call-ID: 8frudde185qca0cgkkbrg1
CSeq: 3632 REGISTER
Content-Length: 0
I have migrated my website and the email records to a new server (other provider). Everything was ok except that now when I want to send a message from my email (my email direction is the same), one of my clients can not receive my messages. I chatted with my client and his mails are ok, he is receiving mails without problems, as he said.
I reported the problem to my Hosting provider and they have changed the mail Exchanger from remote to local but it didn't finish with the problem. Someone knows what could be happening?
This is part of the message that appears:
"
This message was created automatically by mail delivery software.
A message that you sent could not be delivered to one or more of its
recipients. This is a permanent error. The following address(es) failed:
peter#thisismyclientsdirection.com
No Such User Here
peter#thisismyclientsotherdirection.com
No Such User Here
Reporting-MTA: dns; cherry.theserversite.pro
Action: failed
Final-Recipient: rfc822;peter#thisismyclientsdirection.com
Status: 5.0.0
Action: failed
Final-Recipient: rfc822;peter#thisismyclientsotherdirection.com
Status: 5.0.0
Return-path: <comercial#mydomain.com>
Received: from [71.13.252.126] (port=58531 helo=[10.145.123.217])
by cherry.theserversite.pro with esmtpsa (TLS1.2) tls TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
(Exim 4.93)
(envelope-from <comercial#mydomain.com>)
id 1UZ6w-00EaD5-0h; Mon, 19 Oct 2020 13:38:33 -0400
To: peter#thisismyclientsdirection.com, peter#thisismyclientsotherdirection.com
From: comercial#mydomain.com
Subject: =?UTF-8?Q?Reenv=c3=ado_-_cotizaciones_mantenimiento?=
MIME-Version: 1.0
Content-Type: multipart/mixed;
Content-Language: es-ES
X-Antivirus: Avast (VPS 201019-2, 19/10/2020), Outbound message
X-Antivirus-Status: Clean
X-Exim-DSN-Information: Due to administrative limits only headers are returned
"
Thanks,
I found the solution. The problem was that the other websites (the domains of my client) are inside my server and in the Email Routing section (it is in the Cpanel) of those websites, the domains were targeted as "Local server" or not targeted. Despite it, the email server for those domains don't were in my server then the system was confused. I just changed the target to "Remote servers" for both domains (the domains of my client) and the problem disappear.
I hope this explanation could be useful for other developer.
Anyway thank you,
TheJohnny
I'm trying to make a SIP request to a SIP server, using pjsua, a SIP client by pjsip (version 2.10, 2020-02-14). Starting the client this way:
pjsua-x86_64-apple-darwin19.4.0 --id sip:addreessee#sever_host_name:5061;transport=tcp --no-udp
Using the "S" command to send an arbitrary REQUEST, typing a SIP method (I tried with MESSAGE and others) to use in the request and than adding as destination URI "sip:sever_host_name:5061"
The result is:
Destination URI: sip:addreessee#sever_host_name:5061
13:48:02.121 pjsua_core.c .TX 342 bytes Request msg MESSAGE/cseq=53264 (tdta0x7f96c501cca8) to UDP sever_host_name:5061:
MESSAGE sip:addresse#sever_host_name:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;rport;branch=z9hG4bKPjI-s3KUBrnruOqLAKEtCOLnJ.jJPKmoDe
Max-Forwards: 70
From: <sip:addreessee#server_host_name>;tag=1lsf1PY19Qc4fk-8IhoqTV9plx3kX0yC
To: <sip:addreessee#server_host_name>
Call-ID: -X2iZRlerEaevvVvOZlAX5STQnBaGuN2
CSeq: 53264 MESSAGE
Content-Length: 0
So the request is sent over UDP transport layer, not TCP. Can anyone tell me what am I doing wrong?
You should add ;transport=tcp to your request URI each time.
You can read more here (link)
I am getting SIP/2.0 500 Service Unavailable, I created a sip trunk from nexmo to my server. The status is 200 ok. but when i call on that trunk through my mobile then my server is getting SIP response 500 "Service Unavailable" back from 119.XX.XX.X:5060. For detailed log please go through the attachment! Any help will be appreciated.
Did you set your FreePBX server like this:
host=sip.nexmo.com
type=friend
insecure=port,invite
qualify=yes
allow=ulaw,alaw
dtmfmode=rfc2833
fromuser=APIKEY
secret=APISECRET
Register String
APIKEY:APISECRET#sip.nexmo.com
I am using Xlite at one end for sending INVITE.
If i use Kamailio 4.0.1 as outbound proxy,in the call flow it adds lr=on as mentioned below WIRESHARK trace :
Record-Route:
Via: SIP/2.0/UDP 10.44.104.149;branch=z9hG4bK0ecf.1bd4c266.0
Via: SIP/2.0/UDP 10.44.104.160:5998;branch=z9hG4bK-d8754z-829f7d43eed09018-1---d8754z-;rport=5998
and after that the pbx sends 503 response for the INVITE.
but as per RFC 3665 for the call flow ,the lr should be blank as :
Record-Route:
is there any configuration change needed in the Kamailio to meet REcord Route as per RFC 3665 ie lr without On value.
You have to set parameter enable_full_lr for rr module to 0, see:
http://kamailio.org/docs/modules/stable/modules/rr.html#idp21848