22 Mar
2022
22 Mar
'22
10:29 a.m.
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 OK
ACK --------------------------------
<-------------------------------SIP/2.0 200 OK
ACK --------------------------------
<-------------------------------SIP/2.0 200 OK
ACK --------------------------------
<-------------------------------SIP/2.0 200 OK
ACK --------------------------------
<-------------------------------SIP/2.0 200 OK
ACK --------------------------------
<-------------------------------SIP/2.0 200 OK
ACK --------------------------------
<-------------------------------SIP/2.0 200 OK
ACK --------------------------------
<-------------------------------BYE
404 Not Found ----------------------
When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us
From another thread with similar issue at https://groups.google.com/g/sip_js/c/uiaXS_qc2n8
it could be that JsSIP is not recognizing the GRUU is pointing towards it.
In the INVITE message I can see a line "Record-Route:
<mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
<mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
<
<mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
<mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
mailto:;expires=600;received=mailto:;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33
<
<mailto:sip:9747815015@erx-staging-q01.mydomain.com
<mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
<mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
<mailto:sip:9747815015@erx-staging-q01.mydomain.com
<mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
<mailto:sip:9747815015@erx-staging-q01.mydomain.com
<mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
<mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
<mailto:sip:9747815015@erx-staging-q01.mydomain.com
<mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
<
mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
<mailto:sip:9747815015@erx-staging-q01.mydomain.com
<mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
<
mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
<mailto:sip:9747815015@erx-staging-q01.mydomain.com
<mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
<mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
<mailto:sip:9747815015@erx-staging-q01.mydomain.com
24 Mar
24 Mar
2:01 p.m.
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 OK
ACK -------------------------------->
<-------------------------------SIP/2.0 200 OK
ACK -------------------------------->
<-------------------------------SIP/2.0 200 OK
ACK -------------------------------->
<-------------------------------SIP/2.0 200 OK
ACK -------------------------------->
<-------------------------------SIP/2.0 200 OK
ACK -------------------------------->
<-------------------------------SIP/2.0 200 OK
ACK -------------------------------->
<-------------------------------SIP/2.0 200 OK
ACK -------------------------------->
<-------------------------------BYE
404 Not Found ---------------------->
When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does
not point to us
From another thread with similar issue at
https://groups.google.com/g/sip_js/c/uiaXS_qc2n8 it could be that
JsSIP is not recognizing the GRUU is pointing towards it.
In the INVITE message I can see a line "Record-Route: " Is this normal?
I am at a loss to figure this out, and any help or hint to find out
what could be wrong here would be very helpful.
The SIP messages are as follows, as seen from JsSIP:
Send ->
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.ice;reg-id=1;+sip.instance="";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
Receive <-
SIP/2.0 200 OK
Via: SIP/2.0/WSS
ol3dhprvu7jv.invalid;branch=z9hG4bK3674021;rport=23618;received=145.15.191.170
To:
<sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com>;tag=ee7bee7ecdf2759680598685ea71a5eb.d5e40000
From: "User"
<sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com>;tag=65u34oje2s
Call-ID: b624vmbvuioma46354gmi5
CSeq: 1 REGISTER
Contact:
;expires=600;received="sip:145.15.191.170:23618;transport=ws";pub-gruu="sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
<mailto:;expires=600;received=>;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33";temp-gruu="sip:uloc-623966d2-1b2b74-79-435e33f9@erx-staging-q01.mydomain.com
<mailto:;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33>;gr";+sip.instance="";reg-id=1
Server: TLS Kamailio Server
Content-Length: 0
Receive <-
INVITE sip:93he4k0p@ol3dhprvu7jv.invalid;transport=ws SIP/2.0
Record-Route:
Record-Route:
Via: SIP/2.0/WSS
68.19.159.72:443;branch=z9hG4bKc3f.2a0c8eb470c9427d6def7b8eaa3e3f8b.0
Via: SIP/2.0/UDP
127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSI6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEKIMxWr3BN6O.gqMlq1WxerMJ4ZW6aqO.MszSV6z.PwMlqAgc**
From: "User2"
<sip:9747815015@erx-staging-q01.mydomain.com>;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9
To: <sip:Stg-CQD0r2-10020005@10.10.1.9>
Contact:
Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b
CSeq: 19353 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK,
BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 69
User-Agent: Asterisk PBX 18.8.0
Content-Type: application/sdp
Content-Length: 881
Send ->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS
68.19.159.72:443;branch=z9hG4bKc3f.2a0c8eb470c9427d6def7b8eaa3e3f8b.0
Via: SIP/2.0/UDP
127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSI6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEKIMxWr3BN6O.gqMlq1WxerMJ4ZW6aqO.MszSV6z.PwMlqAgc**
To: <sip:Stg-CQD0r2-10020005@10.10.1.9>
From: "User2"
<sip:9747815015@erx-staging-q01.mydomain.com>;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9
Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b
CSeq: 19353 INVITE
Supported: timer,gruu,ice,replaces,outbound
Content-Length: 0
Send ->
SIP/2.0 180 Ringing
Record-Route:
Record-Route:
Via: SIP/2.0/WSS
68.19.159.72:443;branch=z9hG4bKc3f.2a0c8eb470c9427d6def7b8eaa3e3f8b.0
Via: SIP/2.0/UDP
127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSI6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEKIMxWr3BN6O.gqMlq1WxerMJ4ZW6aqO.MszSV6z.PwMlqAgc**
To: <sip:Stg-CQD0r2-10020005@10.10.1.9>;tag=6gc6gshfkb
From: "User2"
<sip:9747815015@erx-staging-q01.mydomain.com>;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9
Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b
CSeq: 19353 INVITE
Contact:
<sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33>
Supported: timer,gruu,ice,replaces,outbound
Content-Length: 0
Send ->
SIP/2.0 200 OK
Record-Route:
Record-Route:
Via: SIP/2.0/WSS
68.19.159.72:443;branch=z9hG4bKc3f.2a0c8eb470c9427d6def7b8eaa3e3f8b.0
Via: SIP/2.0/UDP
127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSI6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEKIMxWr3BN6O.gqMlq1WxerMJ4ZW6aqO.MszSV6z.PwMlqAgc**
To: <sip:Stg-CQD0r2-10020005@10.10.1.9>;tag=6gc6gshfkb
From: "User2"
<sip:9747815015@erx-staging-q01.mydomain.com>;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9
Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b
CSeq: 19353 INVITE
Contact:
<sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33>
Session-Expires: 1800;refresher=uas
Supported: timer,gruu,ice,replaces,outbound
Content-Type: application/sdp
Content-Length: 947
Receive <-
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
68.19.159.72:443;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
Receive <-
BYE
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
68.19.159.72:443;branch=z9hG4bK93f.8fbee7f4747c4d72cc73b403fe51c081.0
Via: SIP/2.0/UDP
127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCBKq36PfMBysO.cA3631Wx3fMmuJ36FuO.PuWEFsgxc4MSKB3A**
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: 19354 BYE
Reason: Q.850;cause=16
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 +3s
Send ->
SIP/2.0 404 Not Found
Via: SIP/2.0/WSS
68.19.159.72:443;branch=z9hG4bK93f.8fbee7f4747c4d72cc73b403fe51c081.0
Via: SIP/2.0/UDP
127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCBKq36PfMBysO.cA3631Wx3fMmuJ36FuO.PuWEFsgxc4MSKB3A**
To: <sip:Stg-CQD0r2-10020005@10.10.1.9>;tag=6gc6gshfkb
From: "User2"
<sip:9747815015@erx-staging-q01.mydomain.com>;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9
Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b
CSeq: 19354 BYE
Content-Length: 0
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* sr-users(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender!
Edit mailing list options or unsubscribe:
* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Online
March 28-31, 2022 (Europe Timezone)
* https://www.asipto.com/sw/kamailio-advanced-training-online/
3:50 p.m.
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
4:40 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
5:16 p.m.
New subject: Resolved: JsSIP:UA Request-URI does not point to us
Did some more digging into the source code of JsSIP. The condition that triggers the error
message is:
https://github.com/versatica/JsSIP/blob/3ab1fa7c8e09231c41ca21657bf96232390…
/**
* Request reception
*/
receiveRequest(request)
{
const method = request.method;
// Check that request URI points to us.
if (request.ruri.user !== this._configuration.uri.user &&
request.ruri.user !== this._contact.uri.user)
{
logger.debug('Request-URI does not point to us');
if (request.method !== JsSIP_C.ACK)
{
request.reply_sl(404);
}
return;
}
If I understand correctly if ruri.user is neither user in config or user in contact the
request gets rejected.
R-URI is:
ACK
sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33
SIP/2.0
The SIP URI configured in tryjsip is sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
This does not match, but if I change the SIP URI to
sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com (all small case) it works fine. I am
not sure if this is a bug or violates the any RFC
Sent with ProtonMail secure email.
------- Original Message -------
On Thursday, March 24th, 2022 at 9:10 PM, Xuo Guoto <xuoguoto(a)protonmail.com>
wrote:
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
771
days inactive
773
days old
4 comments
2 participants
participants (2)
-
Daniel-Constantin Mierla -
Xuo Guoto