24 Jan
2024
24 Jan
'24
11:42 a.m.
Dear List,
I’m trying to set up the dialog module and I keep getting the error "bad sip message
or missing Contact hdr”:
---
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE
2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:219]: populate_leg_info():
bad sip message or missing Contact hdr
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE
2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:952]: dlg_new_dialog(): could
not add further info to the dialog
---
This is for MRCP traffic and the SIP INVITEs do not contain the Contact header. They look
like this:
----
INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ
Max-Forwards: 70
From: <sip:10.10.10.10:5099>;tag=134a3eKD2a8Ua
To: <sip:some.mrcp.server.example.com:8060>
Call-ID: 051df0ff-3541-123d-f090-02f373ff014c
CSeq: 78491159 INVITE
User-Agent: google_stt
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER,
UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 329
v=0
o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10
s=-
c=IN IP4 10.10.10.10
t=0 0
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechrecog
a=cmid:1
m=audio 12034 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 L16/8000
a=sendonly
a=mid:1
----
According to the RFC, the Contact header is not mandatory
(https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t have an easy way
of adding the Contact header to the INVITEs. Is there a way to configure Kamailio’s dialog
module to do without the Contact header? Any pointers appreciated.
Thank you so much
With best wishes,
Unai Rodriguez
24 Jan
24 Jan
12:30 p.m.
Hi Unai,
The CONTACT is used as the R-URI in subsequent replies so upstream
providers parse this from my experience.
You can build one yourself in Kamailio if needed using the textops module -
https://kamailio.org/docs/modules/5.7.x/modules/textops.html#textops.f.appe…
Rough examples:
Native:
insert_hf("Contact: $fu\r\n");
Kemi Python:
KSR.textopsx.append_hf_value("Contact", "<sip:" +
KSR.pv.get("$fU") + "@" +
KSR.pv.get("$si") + ":5060>")
Thanks,
John.
On Wed, 24 Jan 2024 at 11:15, Unai Rodriguez via sr-users <
sr-users(a)lists.kamailio.org> wrote:
Dear List,
I’m trying to set up the dialog module and I keep getting the error "bad
sip message or missing Contact hdr”:
---
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1
78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog
[dlg_handlers.c:219]: populate_leg_info(): bad sip message or missing
Contact hdr
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1
78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog
[dlg_handlers.c:952]: dlg_new_dialog(): could not add further info to the
dialog
---
This is for MRCP traffic and the SIP INVITEs do not contain the Contact
header. They look like this:
----
INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ
Max-Forwards: 70
From: <sip:10.10.10.10:5099>;tag=134a3eKD2a8Ua
To: <sip:some.mrcp.server.example.com:8060>
Call-ID: 051df0ff-3541-123d-f090-02f373ff014c
CSeq: 78491159 INVITE
User-Agent: google_stt
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 329
v=0
o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10
s=-
c=IN IP4 10.10.10.10
t=0 0
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechrecog
a=cmid:1
m=audio 12034 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 L16/8000
a=sendonly
a=mid:1
----
According to the RFC, the Contact header is not mandatory (
https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t
have an easy way of adding the Contact header to the INVITEs. Is there a
way to configure Kamailio’s dialog module to do without the Contact header?
Any pointers appreciated.
Thank you so much
With best wishes,
Unai Rodriguez
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to
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1:32 p.m.
Hi,
Contact is mandatory in INVITEs. Per RFC 3261 § 8.1.1.8 ("Contact"):
The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in the establishment of a
dialog. For the methods defined in this specification, that includes
only the INVITE request.
-- Alex
On 24 Jan 2024, at 05:42, Unai Rodriguez via sr-users
<sr-users(a)lists.kamailio.org> wrote:
Dear List,
I’m trying to set up the dialog module and I keep getting the error "bad sip message
or missing Contact hdr”:
---
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE
2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:219]: populate_leg_info():
bad sip message or missing Contact hdr
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE
2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:952]: dlg_new_dialog(): could
not add further info to the dialog
---
This is for MRCP traffic and the SIP INVITEs do not contain the Contact header. They look
like this:
----
INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ
Max-Forwards: 70
From: <sip:10.10.10.10:5099>;tag=134a3eKD2a8Ua
To: <sip:some.mrcp.server.example.com:8060>
Call-ID: 051df0ff-3541-123d-f090-02f373ff014c
CSeq: 78491159 INVITE
User-Agent: google_stt
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER,
UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 329
v=0
o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10
s=-
c=IN IP4 10.10.10.10
t=0 0
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechrecog
a=cmid:1
m=audio 12034 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 L16/8000
a=sendonly
a=mid:1
----
According to the RFC, the Contact header is not mandatory
(https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t have an easy way
of adding the Contact header to the INVITEs. Is there a way to configure Kamailio’s dialog
module to do without the Contact header? Any pointers appreciated.
Thank you so much
With best wishes,
Unai Rodriguez
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-leave(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:
--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
25 Jan
25 Jan
9:35 a.m.
I tried this (and some variations) to no avail:
request_route {
insert_hf("Contact: $fu\r\n");
...
...
I see the Contact header inserted but after receiving the first SIP INVITE (ie too late).
Is there a way I can instruct the DIALOG module to use a value of Contact that’s been
computed dynamically on the config? Is it possible to modify what Kamailio holds in memory
for the value of Contact before the dialog module processes it? Any other points on how I
could deal with this missing Contact header?
Thank you so much
With best wishes,
Unai Rodriguez
On 24 Jan 2024 at 13:54 +0100, Alex Balashov via sr-users
<sr-users(a)lists.kamailio.org>rg>, wrote:
Hi,
Contact is mandatory in INVITEs. Per RFC 3261 § 8.1.1.8 ("Contact"):
The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in the establishment of a
dialog. For the methods defined in this specification, that includes
only the INVITE request.
-- Alex
On 24 Jan 2024, at 05:42, Unai Rodriguez via
sr-users <sr-users(a)lists.kamailio.org> wrote:
Dear List,
I’m trying to set up the dialog module and I keep getting the error "bad sip message
or missing Contact hdr”:
---
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE
2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:219]: populate_leg_info():
bad sip message or missing Contact hdr
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE
2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:952]: dlg_new_dialog(): could
not add further info to the dialog
---
This is for MRCP traffic and the SIP INVITEs do not contain the Contact header. They look
like this:
----
INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ
Max-Forwards: 70
From: <sip:10.10.10.10:5099>;tag=134a3eKD2a8Ua
To: <sip:some.mrcp.server.example.com:8060>
Call-ID: 051df0ff-3541-123d-f090-02f373ff014c
CSeq: 78491159 INVITE
User-Agent: google_stt
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER,
UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 329
v=0
o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10
s=-
c=IN IP4 10.10.10.10
t=0 0
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechrecog
a=cmid:1
m=audio 12034 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 L16/8000
a=sendonly
a=mid:1
----
According to the RFC, the Contact header is not mandatory
(https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t have an easy way
of adding the Contact header to the INVITEs. Is there a way to configure Kamailio’s dialog
module to do without the Contact header? Any pointers appreciated.
Thank you so much
With best wishes,
Unai Rodriguez
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-leave(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:
--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-leave(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:
6:24 p.m.
On 25 Jan 2024, at 03:35, Unai Rodriguez via sr-users
<sr-users(a)lists.kamailio.org> wrote:
I tried this (and some variations) to no avail:
request_route {
insert_hf("Contact: $fu\r\n");
...
...
I see the Contact header inserted but after receiving the first SIP INVITE (ie too late).
Is there a way I can instruct the DIALOG module to use a value of Contact that’s been
computed dynamically on the config? Is it possible to modify what Kamailio holds in memory
for the value of Contact before the dialog module processes it? Any other points on how I
could deal with this missing Contact header?
I believe that calling msg_apply_changes() will sync the change lumps to the in-memory
message structure.
Whether this will affect the version of the message that `dialog` operates on, I'm not
100% sure...
-- Alex
--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
26 Jan
26 Jan
11:21 a.m.
Hi Alex/All,
So using msg_apply_changes() worked! This is great, thank you! So now that I can see
results when I do "kamctl dialog show”. I’m using algorithm “call load distribution”
with this config:
# cat /etc/kamailio/dispatcher.list
1 sip:host1.example.com:8060;transport=tcp duid=stt_h01
1 sip:host2.example.com:8060;transport=tcp duid=stt_h02
1 sip:host3.example.com:8060;transport=tcp duid=stt_h03
2 sip:host1.example.com:8060;transport=tcp duid=tts_h01
2 sip:host2.example.com:8060;transport=tcp duid=tts_h02
2 sip:host3.example.com:8060;transport=tcp duid=tts_h03
The output of "kamctl dialog show” does not show any obvious information that the
hosts will be load balanced according to the one with the least number of concurrent
sessions.. Is this all I need? Am I missing something or this looks correct? How is the
information that dialog keeps connected to the configuration of the backends?
Thank you so much
With best wishes,
Unai Rodriguez
On 25 Jan 2024 at 18:48 +0100, Alex Balashov via sr-users
<sr-users(a)lists.kamailio.org>rg>, wrote:
On 25 Jan 2024, at 03:35, Unai Rodriguez via
sr-users <sr-users(a)lists.kamailio.org> wrote:
I tried this (and some variations) to no avail:
request_route {
insert_hf("Contact: $fu\r\n");
...
...
I see the Contact header inserted but after receiving the first SIP INVITE (ie too late).
Is there a way I can instruct the DIALOG module to use a value of Contact that’s been
computed dynamically on the config? Is it possible to modify what Kamailio holds in memory
for the value of Contact before the dialog module processes it? Any other points on how I
could deal with this missing Contact header?
I believe that calling msg_apply_changes() will sync the change lumps to the in-memory
message structure.
Whether this will affect the version of the message that `dialog` operates on, I'm
not 100% sure...
-- Alex
--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-leave(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:
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