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
$ct - reference to body of contact header"
Thanks again,
Markus
Am 22.09.2023 um 13:15 schrieb Shah Hussain Khattak:
I should have mentioned, please load the following modules:
loadmodule "textops.so"
loadmodule "textopsx.so"
then restart Kamailio.
------------------------------------------------------------------------
*From:* Markus <[email protected]>
*Sent:* Friday, September 22, 2023 5:58 PM
*To:* [email protected] <[email protected]>
*Cc:* Shah Hussain Khattak <[email protected]>
*Subject:* Re: [SR-Users] Modifying SDP as drop-in replacement for
overloaded Asterisk box - looking for help/paid consulting fast
Wow, cool, thanks! I added this snippet in the
request_route {
part, I hope that's correct. When I try to start Kamailio I get:
kamailio: Not starting : invalid configuration file!
kamailio: 0(2371) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error
searching pvar "cU"
kamailio: 0(2371) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong
char [U/85] in [$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 + "@3.3.3.3";
$tu = "sip:" + $tU + "@3.3.3.3";
$fu = "sip:" + $fU + "@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 + "@3.3.3.3";
$tu = "sip:" + $tU + "@3.3.3.3";
$fu = "sip:" + $fU + "@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:[email protected]:5060>\r\n");
and then add the remaining modifications if needed as per your upstream
carrier requirements.
Regards,
Shah Hussain
------------------------------------------------------------------------
*From:* Markus <[email protected]>
*Sent:* Friday, September 22, 2023 8:58 AM
*To:* [email protected] <[email protected]>
*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:[email protected];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:[email protected]>;tag=as3d75aadd
To: <sip:[email protected];user=phone>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected];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:[email protected];user=phone SIP/2.0
should be
INVITE sip:[email protected];user=phone SIP/2.0
To: <sip:[email protected];user=phone>
should be
To: <sip:[email protected];user=phone>
From: "61xxxxxxxxx" <sip:[email protected]>;tag=as3d75aadd
should be
From: "61xxxxxxxxx" <sip:[email protected]>;tag=as3d75aadd
Contact: <sip:[email protected]:5060>
should be
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
should be
Call-ID: [email protected]: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
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