This is the ACK packet that is not getting recognized by JsSIP
ACK sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 SIP/2.0 Via: SIP/2.0/WSS 68.19.59.72:443;branch=z9hG4bKc3f.244f565f3d688006fb9c33138458f554.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEjuMlWEWB0rO.aJgBc1WSPAMG4Z3RjLO.pqWSPlMRFwWEergc** From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19353 ACK Max-Forwards: 69 User-Agent: Asterisk PBX 18.8.0 Content-Length: 0
tryit-jssip.js:8 JsSIP:UA Request-URI does not point to us +40s
------- Original Message -------
On Thursday, March 24th, 2022 at 8:20 PM, Xuo Guoto xuoguoto@protonmail.com wrote:
Hi,
It seems when I paste the message in the web client, it got removed. Now trying again in text mode.
REGISTER sip:erx-staging-q01.mydomain.com SIP/2.0
Via: SIP/2.0/WSS ol3dhprvu7jv.invalid;branch=z9hG4bK3674021
Max-Forwards: 69
To: sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
From: "User" sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com;tag=65u34oje2s
Call-ID: b624vmbvuioma46354gmi5
CSeq: 1 REGISTER
Contact: sip:93he4k0p@ol3dhprvu7jv.invalid;transport=ws;+sip.ice;reg-id=1;+sip.instance="urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.9.0
Content-Length: 0
I hadn't noticed that some text was removed by the client.
X.
------- Original Message -------
On Thursday, March 24th, 2022 at 6:31 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
is the REGISTER without a Contact URI or the message you pasted omitted it?
Cheers,
Daniel
On 22.03.22 10:29, Xuo Guoto wrote:
Hello all,
I am facing an issue with JsSIP not recognizing replies from Kamailio. the call sequence goes as follows:
INVITE -----------------------------><-------------------------------SIP/2.0 100 Trying<-------------------------------SIP/2.0 180 Ringing<-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------BYE404 Not Found ---------------------->
When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us