24 Mar
2022
24 Mar
'22
6:40 p.m.
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(a)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(a)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