[SR-Users] Re: Is there any way to modify the Refer-To header username?
Hi Henning,
Thanks a lot for the suggestion, it helped me to fix my issue. I used the
replace function and it worked perfectly for my use-case!
4.7. replace(re, txt)
Replaces the first occurrence of re with txt.
Meaning of the parameters is as follows:
*
re - Regular expression.
*
txt - String.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE,
BRANCH_ROUTE.
Example 1.7. replace usage
...
replace("server", "Kamailio SIP Proxy");
...
Regards,
Shah Hussain
From: Henning Westerholt
Sent: Thursday, May 9, 2024 5:36 PM
To: Kamailio (SER) - Users Mailing List
Cc: Shah Hussain Khattak
Subject: RE: Is there any way to modify the Refer-To header username?
Hello,
there are other functions in textops module, e.g. subst(..) that can be used to
do replacements.
Have a look to this module for more details and examples. The approach of
deleting and adding it new works as well, of course.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: Shah Hussain Khattak via sr-users
Sent: Donnerstag, 9. Mai 2024 08:16
To: Kamailio (SER) - Users Mailing List
Cc: Shah Hussain Khattak
Subject: [SR-Users] Is there any way to modify the Refer-To header username?
Hello Experts,
I wanted to check, is there any way to modify the username part of the Refer-To
header in Kamailio? For example, I am getting the following header:
REFER-TO: >
the username part is missing.
Can I add a username inside it? For example, change it to
REFER-TO: >
REFER-TO: >
I could not find any solution in the mailing list, the only way was to remove
the header and then re-add it.
Thank you!
Regards,
Shah Hussain
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[SR-Users] Is there any way to modify the Refer-To header username?
Hello Experts, I wanted to check, is there any way to modify the username part of the Refer-To header in Kamailio? For example, I am getting the following header: REFER-TO: the username part is missing. Can I add a username inside it? For example, change it to REFER-TO: REFER-TO: I could not find any solution in the mailing list, the only way was to remove the header and then re-add it. Thank you! Regards, Shah Hussain __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Need guidance on Teams interop with Kamailio
Hi,
Thanks for the response. I managed to fix this by just adding a check in my
route block based on incoming BYE message and simply relayed it:
if (is_method("BYE") && $ua=~"FreeSwitch BOX/v1\\.0") {
t_relay();
exit;
}
This fixed my issue.
Some other minor tweaks that I applied:
For blind and attended call transfers from MS Teams, I had to disable REFER
support towards MS Teams, then you will be able to perform any kind of transfer
from the Teams client.
Teams require SBC FQDN in the CONTACT header of the INVITE and 200 OK message.
To address the ringback tone issue for inbound calls, I converted the 183
response with SDP into a 180 response without SDP and relayed it towards my
PSTN route.
After all, configuring Kamailio with MS Teams was a good learning experience;
it works like a charm and opens door to many interesting interop scenarios.
Regards,
Shah Hussain
From: Kjartan S via sr-users
Sent: Friday, March 15, 2024 4:23 PM
To: sr-users@lists.kamailio.org
Cc: Kjartan S
Subject: [SR-Users] Re: Need guidance on Teams interop with Kamailio
Your Kamailio is using the RURI for routing the BYE and not RR header.
My tip is to xlog your Kamailio config in order to see if the subsequent BYE
request will run the loose_route function inside your route[WITHINDLG].
if (loose_route()) {
if (is_method("BYE")) {
xlog("L_INFO","Running loose route for subsequent BYE request from $si : $sp
\n");
}
I also recommend not to remove any record route headers when engaging with MS
Teams - so do not remove any RR header and make sure your FreePBX include
record route header originally received from Teams.
Regarding the call disconnect: Might be that the corresponding ACK or the 200
OK (depending on Teams being UAC/UAS`) is not handled from MS Teams side.
Again, make sure not to replace to much of the headers received from Teams
apart from what is mentioned in the MS Teams Kamailio guide.
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[SR-Users] Need guidance on Teams interop with Kamailio
Hello Experts,
I'm not an expert in Kamailio, but thanks to some excellent online resources, I
managed to set up a basic interconnection between Teams and Kamailio.
So far, outbound calls, as well as blind and attended call transfers, are
working smoothly. However, I'm facing a challenge with call disconnection for
outbound calls. Specifically, when an outbound call is made and the called
party hangs up, the BYE message isn't being relayed properly towards the Teams
side.
In my routing configurations, I remove the Record-Route header received from
Teams and add a new Record-Route header with Kamailio's IP before forwarding it
to my IP PBX (FreeSwitch). For the BYE message to be processed correctly,
Microsoft Teams requires the original contact in the Request URI (RURI) and the
Route header to match the initial Record-Route included in the INVITE.
Currently, my BYE message appears as follows:
BYE
sip:api-du-a-auea.pstnhub.microsoft.com:443;x-i=5e96d703-b1e7-449c-aebf-a87bfa628176;x-c=f61b674bc597513ab8d888dbe7c560fb/d/8/9a6c508a9cbd44ebb892b62c33ba9b67
SIP/2.0
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bKeab5.b657e3455b6d11ed6975c69882632a27.0
Via: SIP/2.0/UDP x.x.x.x;received=x.x.x.x;rport=5060;branch=z9hG4bKK3HKQBUvX42ya
Max-Forwards: 69
From: ;tag=j5gptBSXXrXDc
To: "XYZ Pty Ltd"
;tag=3a5f0c9e32bc4eb1bffed29dc954118f
Call-ID: f61b674bc597513ab8d888dbe7c560fb
CSeq: 80555743 BYE
User-Agent: Unknown/v1.0
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Reason: MVTSLocal;cause=10;text="BYE received"
Content-Length: 0
Kamailio conducts a lookup for api-du-a-auea.pstnhub.microsoft.com but ends up
sending the BYE message to an incorrect IP.
Below is a snippet of my routing block configurations:
### Routing Logic
/* Main SIP request routing logic
* - processing of any incoming SIP request starts with this route
* - note: this is the same as route { ... } */
request_route {
# per request initial checks
route(REQINIT);
#check who is the sender
route(INITCHECK);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# 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();
}
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
}
if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# update $du to set the destination address for proxying
#$du = "sip:" + $rd + ":9";
route(RELAY);
exit;
}
route[INITCHECK] {
if(from_uri =~ ".*microsoft.com")
{
setflag(FROM_TEAMS);
$du = "sip:" + "MY_PBX_IP" + ":" + "5060";
//added by SHK
remove_hf("Record-Route"); // Remove existing Record-Route
header
append_hf("Record-Route: \r\n");
route(HANDLE_RTP_FROM_TEAMS);
}
if(from_uri =~ ".*" + "MY_PBX_IP")
{
setflag(FROM_PBX);
$du="sip:sip.pstnhub.microsoft.com;transport=tls";
route(HANDLE_RTP_FROM_PBX);
}
}
#Manage RTP & transcoding comming from Teams to PBX
route[HANDLE_RTP_FROM_TEAMS] {
t_on_reply("PBX_REPLY_TO_TEAMS");
if (has_body("application/sdp"))
{
rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA
replace-origin replace-session-connection ICE=remove media-address=HOST_IP");
// record_route();
t_relay_to_udp("MY_PBX_IP","5060");
}
else
{
rtpengine_manage();
}
}
#Manage RTP & transcoding comming from PBX to Teams
route[HANDLE_RTP_FROM_PBX] {
t_on_reply("TEAMS_REPLY_TO_PBX");
if (has_body("application/sdp"))
{
rtpengine_manage("SRTP codec-mask=all ICE=force
codec-transcode=PCMA replace-origin replace-session-connection
media-address=ADVERTISE_IP");
$rd = "sip.pstnhub.microsoft.com";
$td = "kamsbc.xyzdomain.com";
$fd = "kamsbc.xyzdomain.com";
// record_route();
#Set
[SR-Users] Iterating Through Multiple IP Peers for DID Call Termination in Kamailio
Hello Team,
I hope you're all doing well. I'm reaching out to request your insights on the
following scenario.
Overview:
I have successfully configured Kamailio to authenticate IP addresses, primarily
for PBXs and customers unable to register. With this setup, I can flawlessly
receive incoming traffic and forward it to our internal FreeSwitch servers.
Current issue:
Now, I'm focusing on managing traffic in the opposite direction—specifically
for DIDs. I have a DID table set up in Kamailio that associates DID numbers
with their respective IP peers. Upon receiving a call, a database lookup is
performed to find the destination number, after which the call is terminated at
the customer's IP.
I'm facing difficulty in handling situations where a customer has multiple IP
peers (for redundancy). My aim is to try each of these IPs sequentially when
terminating a specific DID call towards them. The current logic I'm employing
for the INVITE lookup looks like this:
# Database lookup for INVITEs
if (is_method("INVITE")) {
sql_query("didrouting_db", "SELECT route_to FROM did_routing WHERE
did_number='$rU'", "result");
if ($dbr(result=>rows) > 0) {
$var(route_to) = $dbr(result=>[0,0]);
xlog("L_INFO", "Routing DID $rU to $var(route_to)\n");
$du = "sip:" + $var(route_to) + ":5060";
} else {
$du = "sip:" + FS_IP + ":5060";
}
}
The above logic works fine for a single IP.
I intend to use a comma-separated list of IPs in my route_to column (for
example 192.168.1.2,192.168.1.3, ...) and then iterate through these IPs one by
one. I'm having trouble coming up with a workable logic or loop to achieve
this. Would you be able to offer any guidance or suggestions?
Please also recommend if there is a better approach available to handle this
scenario.
Thank you in advance for your valuable input.
Regards,
Shah Hussain
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[SR-Users] Re: Modifying SDP as drop-in replacement for overloaded Asterisk box - looking for help/paid consulting fast
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):(.*)@(.*)//",
"a");
Sent from my iPhone
On 22 Sep 2023, at 9:25 pm, Markus 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
$ct - reference to body of contact header"
Thanks again,
Markus
Am 22.09.2023 um 13:15 schrieb Shah Hussain Khattak:
I should have mentioned, please load the following modules:
loadmodule "textops.so"
loadmodule "textopsx.so"
then restart Kamailio.
*From:* Markus
*Sent:* Friday, September 22, 2023 5:58 PM
*To:* sr-users@lists.kamailio.org
*Cc:* Shah Hussain Khattak
*Subject:* Re: [SR-Users] Modifying SDP as drop-in replacement for overloaded
Asterisk box - looking for help/paid consulting fast
Wow, cool, thanks! I added this snippet in the
request_route {
part, I hope that's correct. When I try to start Kamailio I get:
kamailio: Not starting : invalid configuration file!
kamailio: 0(2371) ERROR: [pvapi.c:828]: pv_parse_spec2(): error
searching pvar "cU"
kamailio: 0(2371) ERROR: [pvapi.c:1032]: pv_parse_spec2(): wrong
char [U/85] in [$cU] at [2 (0)]
kamailio: 0(2371) : [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: \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: \r\n");
OR
# Insert new Contact header using the stored username
insert_hf("Contact: \r\n");
and then add the remaining modifications if needed as per your upstream carrier
requirements.
Regards,
Shah Hussain
*From:* Markus
*Sent:* Friday, September 22, 2023 8:58 AM
*To:* sr-users@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:+32@2.2.2.2;user=phone SIP/2.0
Record-Route:
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: "61x" ;tag=as3d75aadd
To:
Contact:
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,
[SR-Users] Re: Modifying SDP as drop-in replacement for overloaded Asterisk box - looking for help/paid consulting fast
I should have mentioned, please load the following modules:
loadmodule "textops.so"
loadmodule "textopsx.so"
then restart Kamailio.
From: Markus
Sent: Friday, September 22, 2023 5:58 PM
To: sr-users@lists.kamailio.org
Cc: Shah Hussain Khattak
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: [pvapi.c:828]: pv_parse_spec2(): error
searching pvar "cU"
kamailio: 0(2371) ERROR: [pvapi.c:1032]: pv_parse_spec2(): wrong
char [U/85] in [$cU] at [2 (0)]
kamailio: 0(2371) : [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: \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:
> \r\n");
> OR
> # Insert new Contact header using the stored username
> insert_hf("Contact: \r\n");
>
> and then add the remaining modifications if needed as per your upstream
> carrier requirements.
>
>
> Regards,
> Shah Hussain
>
> *From:* Markus
> *Sent:* Friday, September 22, 2023 8:58 AM
> *To:* sr-users@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:+32@2.2.2.2;user=phone SIP/2.0
> Record-Route:
> 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: "61x" ;tag=as3d75aadd
> To:
> Contact:
> 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:
> 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:+32x@2.2.2.2;user=phone SIP/2.0
> should be
> INVITE sip:+32x@3.3.3.3;user=phone SIP/2.0
>
> To:
> should be
> To:
>
> From: "61x" ;tag=as3d75aadd
> should be
> From: "61x" ;tag=as3d75aadd
>
> Contact:
> should be
> Contact:
>
> Call-ID: 3f31e1622a72b6d17f24e42362f4f1d0@1.1.1.1:5060
> should be
> Call-ID:
