Modifying SDP as drop-in replacement for overloaded Asterisk box - looking for help/paid consulting fast
22 Sep
2023
22 Sep
'23
3:58 a.m.
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.
Thank you!
Markus
22 Sep
22 Sep
4:45 a.m.
Probably you need to modify only the RURI.
If you looking for a paid consultant you should post to the business
mailing list. Also, you can take a look here:
https://www.kamailio.org/w/business/
-ovidiu
On Thu, Sep 21, 2023 at 21:29 Markus <universe(a)truemetal.org> wrote:
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.
Thank you!
Markus
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4:56 a.m.
Hi Markus,
Can you elaborate upon the way in which you are using Kamailio+RTPEngine "as a
drop-in replacement"? Drop-in replacement for what? Or that is to say, what are you
trying to accomplish here, functionally?
I have the suspicion that what you're doing is probably best accomplished in a
different and more straightforward way. :-)
-- Alex
On Sep 21, 2023, at 8:58 PM, Markus
<universe(a)truemetal.org> wrote:
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted emergency
situation (didn't monitor traffic growth properly and now encountering packet loss
during peak times) as a drop-in replacement for an overloaded Asterisk box in a
call-termination-to-upstream-carrier scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP 1.1.1.1 ->
Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work - carrier is rejecting
it. Carrier is authenticating our calls based on our IP address 2.2.2.2, no username/pass
involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the original
Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier without Kamailio in the path,
and these are the differences I noticed, and probably the things I have to mimick with
Kamailio in order to make it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the resources to
learn it as fast as I must to avoid further packet loss. If there's anyone available
who can help me to get this done today, optionally in exchange for money, I'd be
grateful.
Thank you!
Markus
__________________________________________________________
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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!
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--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
12:11 p.m.
Hi Alex,
I'm trying to replace the Asterisk box with an instance of
Kamailio+RTPEngine because the Asterisk box is heavily overloaded and
calls that are passing through this box are encountering packet loss.
The idea behind it is that the bundle of Kamailio+RTPEngine will be less
CPU-intense than Asterisk and that the machine this bundle runs on would
be able to handle the current call load without packet loss.
With "drop-in replacement" I meant that no changes on the upstream
carrier side can be made for the moment (they're slow), thus I'm having
to use the IP of the Asterisk box for the Kamailio+RTPengine bundle.
The purpose of the Asterisk box (and, once I got it to work,
Kamailio+RTPEngine will be the replacement) is to route SIP voice calls
from several other Asterisk boxes in the LAN to the carrier.
That single overloaded Asterisk box is the gateway to the carrier so to
speak. And its IP 2.2.2.2 which is authorized to send INVITE's towards
the carrier can't get changed on the carrier side for the moment.
Thanks :)
Markus
Am 22.09.2023 um 03:56 schrieb Alex Balashov:
Hi Markus,
Can you elaborate upon the way in which you are using Kamailio+RTPEngine "as a
drop-in replacement"? Drop-in replacement for what? Or that is to say, what are you
trying to accomplish here, functionally?
I have the suspicion that what you're doing is probably best accomplished in a
different and more straightforward way. :-)
6:38 p.m.
So, forgive the silly question, but why do you need to do anything except to engage
RTPEngine? Why rewrite any other SIP headers?
On Sep 22, 2023, at 5:11 AM, Markus via sr-users
<sr-users(a)lists.kamailio.org> wrote:
Hi Alex,
I'm trying to replace the Asterisk box with an instance of Kamailio+RTPEngine because
the Asterisk box is heavily overloaded and calls that are passing through this box are
encountering packet loss. The idea behind it is that the bundle of Kamailio+RTPEngine will
be less CPU-intense than Asterisk and that the machine this bundle runs on would be able
to handle the current call load without packet loss.
With "drop-in replacement" I meant that no changes on the upstream carrier side
can be made for the moment (they're slow), thus I'm having to use the IP of the
Asterisk box for the Kamailio+RTPengine bundle.
The purpose of the Asterisk box (and, once I got it to work, Kamailio+RTPEngine will be
the replacement) is to route SIP voice calls from several other Asterisk boxes in the LAN
to the carrier.
That single overloaded Asterisk box is the gateway to the carrier so to speak. And its IP
2.2.2.2 which is authorized to send INVITE's towards the carrier can't get changed
on the carrier side for the moment.
Thanks :)
Markus
Am 22.09.2023 um 03:56 schrieb Alex Balashov:
--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
Hi Markus,
Can you elaborate upon the way in which you are using Kamailio+RTPEngine "as a
drop-in replacement"? Drop-in replacement for what? Or that is to say, what are you
trying to accomplish here, functionally?
I have the suspicion that what you're doing is probably best accomplished in a
different and more straightforward way. :-)
__________________________________________________________
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:
23 Sep
23 Sep
4:13 p.m.
Hm. To use RTPEngine I need Kamailio, correct? Or can I use RTPEngine
just with Asterisk?
If I need Kamailio: Kamailio in the default configuration is behaving
differently than Asterisk in terms of the contents of the SIP header
fields that it sends by default to the carrier IP (please see my first
E-Mail in this thread where I compared the headers that Kamailio sends
vs. the headers that Asterisk sends). This breaks the connection to the
carrier so to speak (INVITE's not being accepted by the carrier). Thus,
when I replace Asterisk with Kamailio, I need to modify Kamailio's
config in a way that I get it to send exactly the same headers than
Asterisk would.
If I don't need Kamailio: That would be great, but I'm thinking the load
of just Asterisk + RTPEngine would be still higher than Kamailio +
RTPEngine. I don't need any of the Asterisk features really, just need
to forward and receive SIP packets, and I guess Kamailio is performing
much better than Asterisk at this task.
Am 22.09.2023 um 17:38 schrieb Alex Balashov via sr-users:
So, forgive the silly question, but why do you need to
do anything except to engage RTPEngine? Why rewrite any other SIP headers?
On Sep 22, 2023, at 5:11 AM, Markus via sr-users
<sr-users(a)lists.kamailio.org> wrote:
Hi Alex,
I'm trying to replace the Asterisk box with an instance of Kamailio+RTPEngine because
the Asterisk box is heavily overloaded and calls that are passing through this box are
encountering packet loss. The idea behind it is that the bundle of Kamailio+RTPEngine will
be less CPU-intense than Asterisk and that the machine this bundle runs on would be able
to handle the current call load without packet loss.
With "drop-in replacement" I meant that no changes on the upstream carrier side
can be made for the moment (they're slow), thus I'm having to use the IP of the
Asterisk box for the Kamailio+RTPengine bundle.
The purpose of the Asterisk box (and, once I got it to work, Kamailio+RTPEngine will be
the replacement) is to route SIP voice calls from several other Asterisk boxes in the LAN
to the carrier.
That single overloaded Asterisk box is the gateway to the carrier so to speak. And its IP
2.2.2.2 which is authorized to send INVITE's towards the carrier can't get changed
on the carrier side for the moment.
Thanks :)
Markus
Am 22.09.2023 um 03:56 schrieb Alex Balashov:
Hi Markus,
Can you elaborate upon the way in which you are using Kamailio+RTPEngine "as a
drop-in replacement"? Drop-in replacement for what? Or that is to say, what are you
trying to accomplish here, functionally?
I have the suspicion that what you're doing is probably best accomplished in a
different and more straightforward way. :-)
__________________________________________________________
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!
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5:55 p.m.
Hello,
to use rtpengine you need some kind of server like Kamailio, it does not work with
asterisk.
Many people use Kamailio for carrier interconnection, in the end it is (as you already
stated) just a matter of the correct configuration.
Cheers,
Henning
--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://gilawa.com
-----Original Message-----
From: Markus via sr-users <sr-users(a)lists.kamailio.org>
Sent: Samstag, 23. September 2023 15:13
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Cc: Markus <universe(a)truemetal.org>
Subject: [SR-Users] Re: Modifying SDP as drop-in replacement for overloaded
Asterisk box - looking for help/paid consulting fast
Hm. To use RTPEngine I need Kamailio, correct? Or can I use RTPEngine just
with Asterisk?
If I need Kamailio: Kamailio in the default configuration is behaving differently
than Asterisk in terms of the contents of the SIP header fields that it sends by
default to the carrier IP (please see my first E-Mail in this thread where I
compared the headers that Kamailio sends vs. the headers that Asterisk
sends). This breaks the connection to the carrier so to speak (INVITE's not
being accepted by the carrier). Thus, when I replace Asterisk with Kamailio, I
need to modify Kamailio's config in a way that I get it to send exactly the same
headers than Asterisk would.
If I don't need Kamailio: That would be great, but I'm thinking the load of just
Asterisk + RTPEngine would be still higher than Kamailio + RTPEngine. I don't
need any of the Asterisk features really, just need to forward and receive SIP
packets, and I guess Kamailio is performing much better than Asterisk at this
task.
Am 22.09.2023 um 17:38 schrieb Alex Balashov via sr-users:
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So, forgive the silly question, but why do you
need to do anything except to
engage RTPEngine? Why rewrite any other SIP headers?
> On Sep 22, 2023, at 5:11 AM, Markus via sr-users <sr-
users(a)lists.kamailio.org> wrote:
>
> Hi Alex,
>
> I'm trying to replace the Asterisk box with an instance of
Kamailio+RTPEngine because the Asterisk box is heavily overloaded and calls
that are passing through this box are encountering packet loss. The idea
behind it is that the bundle of Kamailio+RTPEngine will be less CPU-intense
than Asterisk and that the machine this bundle runs on would be able to
handle the current call load without packet loss.
>
> With "drop-in replacement" I meant that no changes on the upstream
carrier side can be made for the moment (they're slow), thus I'm having to use
the IP of the Asterisk box for the Kamailio+RTPengine bundle.
>
> The purpose of the Asterisk box (and, once I got it to work,
Kamailio+RTPEngine will be the replacement) is to route SIP voice calls from
several other Asterisk boxes in the LAN to the carrier.
>
> That single overloaded Asterisk box is the gateway to the carrier so to
speak.
And its IP 2.2.2.2 which is authorized to send INVITE's towards the
carrier can't get changed on the carrier side for the moment.
>
> Thanks :)
> Markus
>
>
> Am 22.09.2023 um 03:56 schrieb Alex Balashov:
>> Hi Markus,
>> Can you elaborate upon the way in which you are using
Kamailio+RTPEngine
"as a drop-in replacement"? Drop-in replacement for
what? Or that is to say, what are you trying to accomplish here, functionally?
>> I have the suspicion that what you're
doing is probably best
>> accomplished in a different and more straightforward way. :-)
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mailing list options or unsubscribe:
22 Sep
22 Sep
11:04 a.m.
You can start with the following:
# Change URI(s)
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact:
<sip:$var(contact_username)@2.2.2.2:5060>\r\n");
OR
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:+61123123123@2.2.2.2:5060>\r\n");
and then add the remaining modifications if needed as per your upstream carrier
requirements.
Regards,
Shah Hussain
________________________________
From: Markus <universe(a)truemetal.org>
Sent: Friday, September 22, 2023 8:58 AM
To: sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org>
Subject: [SR-Users] Modifying SDP as drop-in replacement for overloaded Asterisk box -
looking for help/paid consulting fast
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.
Thank you!
Markus
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12:58 p.m.
Wow, cool, thanks! I added this snippet in the
request_route {
part, I hope that's correct. When I try to start Kamailio I get:
kamailio: Not starting : invalid configuration file!
kamailio: 0(2371) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error
searching pvar "cU"
kamailio: 0(2371) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong
char [U/85] in [$cU] at [2 (0)]
kamailio: 0(2371) : <core> [cfg.y:3368]: yyerror_at(): parse error in
config file //etc/kamailio/kamailio.cfg, line 530, column 34-36: Can't
get from cache: $cU
Somehow it doesn't know what $cU is?
It looks like this now:
...
# authentication
route(AUTH);
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:$var(contact_username)@2.2.2.2:5060>\r\n");
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
...
(formatting changed for E-Mail compatibility)
Thanks again,
Markus
Am 22.09.2023 um 10:04 schrieb Shah Hussain Khattak:
You can start with the following:
# Change URI(s)
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact:
<sip:$var(contact_username)@2.2.2.2:5060>\r\n");
OR
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:+61123123123@2.2.2.2:5060>\r\n");
and then add the remaining modifications if needed as per your upstream
carrier requirements.
Regards,
Shah Hussain
------------------------------------------------------------------------
*From:* Markus <universe(a)truemetal.org>
*Sent:* Friday, September 22, 2023 8:58 AM
*To:* sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org>
*Subject:* [SR-Users] Modifying SDP as drop-in replacement for
overloaded Asterisk box - looking for help/paid consulting fast
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
<https://pastebin.com/6PKcRjPU>
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.
Thank you!
Markus
__________________________________________________________
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:
2:15 p.m.
I should have mentioned, please load the following modules:
loadmodule "textops.so"
loadmodule "textopsx.so"
then restart Kamailio.
________________________________
From: Markus <universe(a)truemetal.org>
Sent: Friday, September 22, 2023 5:58 PM
To: sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org>
Cc: Shah Hussain Khattak <shahhusayn(a)msn.com>
Subject: Re: [SR-Users] Modifying SDP as drop-in replacement for overloaded Asterisk box -
looking for help/paid consulting fast
Wow, cool, thanks! I added this snippet in the
request_route {
part, I hope that's correct. When I try to start Kamailio I get:
kamailio: Not starting : invalid configuration file!
kamailio: 0(2371) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error
searching pvar "cU"
kamailio: 0(2371) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong
char [U/85] in [$cU] at [2 (0)]
kamailio: 0(2371) : <core> [cfg.y:3368]: yyerror_at(): parse error in
config file //etc/kamailio/kamailio.cfg, line 530, column 34-36: Can't
get from cache: $cU
Somehow it doesn't know what $cU is?
It looks like this now:
...
# authentication
route(AUTH);
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:$var(contact_username)@2.2.2.2:5060>\r\n");
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
...
(formatting changed for E-Mail compatibility)
Thanks again,
Markus
Am 22.09.2023 um 10:04 schrieb Shah Hussain Khattak:
You can start with the following:
# Change URI(s)
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact:
<sip:$var(contact_username)@2.2.2.2:5060>\r\n");
OR
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:+61123123123@2.2.2.2:5060>\r\n");
and then add the remaining modifications if needed as per your upstream
carrier requirements.
Regards,
Shah Hussain
------------------------------------------------------------------------
*From:* Markus <universe(a)truemetal.org>
*Sent:* Friday, September 22, 2023 8:58 AM
*To:* sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org>
*Subject:* [SR-Users] Modifying SDP as drop-in replacement for
overloaded Asterisk box - looking for help/paid consulting fast
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
<https://pastebin.com/6PKcRjPU>
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.
Thank you!
Markus
__________________________________________________________
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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Edit mailing list options or unsubscribe:
2:24 p.m.
textops.so was already loaded. Now added textopsx.so too but the error
remains the same.
Is $cU a pseudo-variable that should be available within Kamailio by
default? I can't find it here:
https://www.kamailio.org/wiki/cookbooks/5.5.x/pseudovariables
Is it maybe $ct or something else? :)
"$ct - Contact header
$ct - reference to body of contact header"
Thanks again,
Markus
Am 22.09.2023 um 13:15 schrieb Shah Hussain Khattak:
I should have mentioned, please load the following
modules:
loadmodule "textops.so"
loadmodule "textopsx.so"
then restart Kamailio.
------------------------------------------------------------------------
*From:* Markus <universe(a)truemetal.org>
*Sent:* Friday, September 22, 2023 5:58 PM
*To:* sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org>
*Cc:* Shah Hussain Khattak <shahhusayn(a)msn.com>
*Subject:* Re: [SR-Users] Modifying SDP as drop-in replacement for
overloaded Asterisk box - looking for help/paid consulting fast
Wow, cool, thanks! I added this snippet in the
request_route {
part, I hope that's correct. When I try to start Kamailio I get:
kamailio: Not starting : invalid configuration file!
kamailio: 0(2371) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error
searching pvar "cU"
kamailio: 0(2371) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong
char [U/85] in [$cU] at [2 (0)]
kamailio: 0(2371) : <core> [cfg.y:3368]: yyerror_at(): parse error in
config file //etc/kamailio/kamailio.cfg, line 530, column 34-36: Can't
get from cache: $cU
Somehow it doesn't know what $cU is?
It looks like this now:
...
# authentication
route(AUTH);
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:$var(contact_username)@2.2.2.2:5060>\r\n");
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
...
(formatting changed for E-Mail compatibility)
Thanks again,
Markus
Am 22.09.2023 um 10:04 schrieb Shah Hussain Khattak:
You can start with the following:
# Change URI(s)
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact:
<sip:$var(contact_username)@2.2.2.2:5060>\r\n");
OR
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:+61123123123@2.2.2.2:5060>\r\n");
and then add the remaining modifications if needed as per your upstream
carrier requirements.
Regards,
Shah Hussain
------------------------------------------------------------------------
*From:* Markus <universe(a)truemetal.org>
*Sent:* Friday, September 22, 2023 8:58 AM
*To:* sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org>
*Subject:* [SR-Users] Modifying SDP as drop-in replacement for
overloaded Asterisk box - looking for help/paid consulting fast
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
<https://pastebin.com/6PKcRjPU>
<https://pastebin.com/6PKcRjPU <https://pastebin.com/6PKcRjPU>>
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.
Thank you!
Markus
__________________________________________________________
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:
2:42 p.m.
Hmm, try to use: subst_hf() then:
3.15. subst_hf(hf, subexp, flags)
Perl-like substitutions in the body of a header field.
Meaning of the parameters is as follows:
* hf - header field name.
* subexp - substitution expression in the same format as of the 'subst'
function parameter.
* flags - control flags - it has to be one of: a - all headers matching the name; f -
only first header matching the name; l - only the last header matching the name.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE,
BRANCH_ROUTE.
Example 1.15. search_body usage
...
if ( subst_hf("From", "/:test@/:best@/", "a") ) { /*....*/
};
...
For example, from one google search:
subst_hf("Contact",
"/(.*)?<(sip):(.*)@(.*)/<sip:$var(pilot_tn)@xxx.xxx.xxx.xxx:5060;transport=udp>/",
"a");
Sent from my iPhone
On 22 Sep 2023, at 9:25 pm, Markus <universe(a)truemetal.org> wrote:
textops.so was already loaded. Now added textopsx.so too but the error remains the
same.
Is $cU a pseudo-variable that should be available within Kamailio by default? I can't
find it here: https://www.kamailio.org/wiki/cookbooks/5.5.x/pseudovariables
Is it maybe $ct or something else? :)
"$ct - Contact header
$ct - reference to body of contact header"
Thanks again,
Markus
Am 22.09.2023 um 13:15 schrieb Shah Hussain Khattak:
I should have mentioned, please load the following modules:
loadmodule "textops.so"
loadmodule "textopsx.so"
then restart Kamailio.
------------------------------------------------------------------------
*From:* Markus <universe(a)truemetal.org>
*Sent:* Friday, September 22, 2023 5:58 PM
*To:* sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org>
*Cc:* Shah Hussain Khattak <shahhusayn(a)msn.com>
*Subject:* Re: [SR-Users] Modifying SDP as drop-in replacement for overloaded Asterisk box
- looking for help/paid consulting fast
Wow, cool, thanks! I added this snippet in the
request_route {
part, I hope that's correct. When I try to start Kamailio I get:
kamailio: Not starting : invalid configuration file!
kamailio: 0(2371) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error
searching pvar "cU"
kamailio: 0(2371) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong
char [U/85] in [$cU] at [2 (0)]
kamailio: 0(2371) : <core> [cfg.y:3368]: yyerror_at(): parse error in
config file //etc/kamailio/kamailio.cfg, line 530, column 34-36: Can't
get from cache: $cU
Somehow it doesn't know what $cU is?
It looks like this now:
...
# authentication
route(AUTH);
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:$var(contact_username)@2.2.2.2:5060>\r\n");
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
...
(formatting changed for E-Mail compatibility)
Thanks again,
Markus
Am 22.09.2023 um 10:04 schrieb Shah Hussain Khattak:
You can start with the following:
# Change URI(s)
$ru = "sip:" + $rU + "(a)3.3.3.3".3.3";
$tu = "sip:" + $tU + "(a)3.3.3.3".3.3";
$fu = "sip:" + $fU + "(a)2.2.2.2".2.2";
$var(contact_username) = $cU;
# Remove existing Contact header
remove_hf("Contact");
# Insert new Contact header using the stored username
insert_hf("Contact:
<sip:$var(contact_username)@2.2.2.2:5060>\r\n");
OR
# Insert new Contact header using the stored username
insert_hf("Contact: <sip:+61123123123@2.2.2.2:5060>\r\n");
and then add the remaining modifications if needed as per your upstream carrier
requirements.
Regards,
Shah Hussain
------------------------------------------------------------------------
*From:* Markus <universe(a)truemetal.org>
*Sent:* Friday, September 22, 2023 8:58 AM
*To:* sr-users(a)lists.kamailio.org <sr-users(a)lists.kamailio.org>
*Subject:* [SR-Users] Modifying SDP as drop-in replacement for overloaded Asterisk box -
looking for help/paid consulting fast
Hi list,
I'm trying to use Kamailio 4.4.4 with rtpengine in a self-inflicted
emergency situation (didn't monitor traffic growth properly and now
encountering packet loss during peak times) as a drop-in replacement for
an overloaded Asterisk box in a call-termination-to-upstream-carrier
scenario.
My test scenario is to make a call from a SIP softphone to Asterisk IP
1.1.1.1 -> Kamailio/rtpengine IP 2.2.2.2 -> Upstream carrier 3.3.3.3
sngrep on Kamailio box 2.2.2.2 - the following SDP will not work -
carrier is rejecting it. Carrier is authenticating our calls based on
our IP address 2.2.2.2, no username/pass involved.
2023/09/22 02:06:49.216136 2.2.2.2:5060 -> 3.3.3.3:5060
INVITE sip:+32xxxxxxxx@2.2.2.2;user=phone SIP/2.0
Record-Route: <sip:2.2.2.2;lr>
Via: SIP/2.0/UDP
2.2.2.2;branch=z9hG4bKd9c3.d6fa3abe5d52b827e2054de5573028e0.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK473270e8
Max-Forwards: 69
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 20.0.0
Date: Fri, 22 Sep 2023 00:06:50 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: <sip:+61xxxxxxxxx@2.2.2.2;user=phone>
Content-Type: application/sdp
Content-Length: 314
X-SIP: 1.1.1.1
v=0
o=root 1093000903 1093000903 IN IP4 1.1.1.1
s=Asterisk PBX 20.0.0
c=IN IP4 2.2.2.2
t=0 0
m=audio 25742 RTP/AVP 8 9 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:25743
a=ptime:20
I'm comparing this rejected INVITE to a successful INVITE sent by the
original Asterisk box at IP 2.2.2.2 (now Kamailio box) to the carrier
without Kamailio in the path, and these are the differences I noticed,
and probably the things I have to mimick with Kamailio in order to make
it work:
INVITE sip:+32xxxxxxxxx@2.2.2.2;user=phone SIP/2.0
should be
INVITE sip:+32xxxxxxxxx@3.3.3.3;user=phone SIP/2.0
To: <sip:+32xxxxxxxx@2.2.2.2;user=phone>
should be
To: <sip:+32xxxxxxxx@3.3.3.3;user=phone>
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@1.1.1.1>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:+61xxxxxxxxx@2.2.2.2>;tag=as3d75aadd
Contact: <sip:+61xxxxxxxxx@1.1.1.1:5060>
should be
Contact: <sip:+61xxxxxxxxx@2.2.2.2:5060>
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
should be
Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@2.2.2.2:5060
o=root 1093000903 1093000903 IN IP4 1.1.1.1
should be
o=root 1093000903 1093000903 IN IP4 2.2.2.2
My kamailio.cfg can be found here: https://pastebin.com/6PKcRjPU
<https://pastebin.com/6PKcRjPU>
<https://pastebin.com/6PKcRjPU <https://pastebin.com/6PKcRjPU>>
These are the Asterisk boxes I want to originate calls from to Kamailio:
[root@voip30 ~]# kamctl address show
+-----+-----+----------+------+------+-----------+
| id | grp | ip_addr | mask | port | tag |
+-----+-----+----------+------+------+-----------+
| 195 | 1 | 1.1.1.1 | 32 | 0 | voip20.sv |
| 196 | 1 | 1.1.1.2 | 32 | 0 | voip21.sv |
| 197 | 1 | 1.1.1.3 | 32 | 0 | voip22.sv |
| 198 | 1 | 1.1.1.4 | 32 | 0 | voip23.sv |
| 199 | 1 | 1.1.1.5 | 32 | 0 | voip24.sv |
| 200 | 1 | 1.1.1.6 | 32 | 0 | voip25.sv |
| 201 | 1 | 1.1.1.7 | 32 | 0 | voip26.sv |
| 202 | 1 | 1.1.1.8 | 32 | 0 | voip27.sv |
| 203 | 1 | 1.1.1.9 | 32 | 0 | voip28.sv |
+-----+-----+----------+------+------+-----------+
This is the upstream carrier I want Kamailio to proxy calls to:
[root@voip30 ~]# kamctl dispatcher show
dispatcher gateways
+----+-------+------------------+-------+-------+------------+------+
| id | setid | destination | flags | prio. | attrs | desc |
+----+-------+------------------+-------+-------+------------+------+
| 12 | 1 | sip:3.3.3.3:5060 | 0 | 0 | weight=100 | |
+----+-------+------------------+-------+-------+------------+------+
(output manually slightly modified to look properly over E-Mail)
As you might have guessed I'm a Kamailio noob... and don't have the
resources to learn it as fast as I must to avoid further packet loss. If
there's anyone available who can help me to get this done today,
optionally in exchange for money, I'd be grateful.
Thank you!
Markus
__________________________________________________________
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:
426
days inactive
427
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participants (5)
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Alex Balashov -
Henning Westerholt -
Markus -
Ovidiu Sas -
Shah Hussain Khattak