[SR-Users] pua dialogindo doesn't not generate PUBLISH when routes is written on kemi framework(python)

[Zhan Bazarov]

Hi everyone. I have an issue, when I loaded the kemi python engine - module
pua_dialoginfo stopped sending publishes... Has anyone faced the same
issue?
kamailio version is 5.4.5
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[SR-Users] Presence - issue with NOTIFY "terminated" state

[Zhan Bazarov]

Hello everyone! I faced some unfortunately issue with presence module. Let me
describe setup scheme a bit: I'm publishing dialog states via pua_dialoginfo
from one kamailio server(for calls proccessing itself) to another one
kamailio(standalone kamailio just for handling presence,blf,mwi etc) 

So, about issue, let's say we have device A which has outgoing subscription
to extension which belongs to device B and device C(both devices has same
extension but different usernames, webrtc+sip phone, effectively its the
same Phone, I'm using pubruri_avps to change entities from usernames to
extension) . And, in some cases,for examle, when thirtparty is making call
to this extension(B,C devices) and making CANCEL after couple seconds - I
see the PUBLISH from callserver to presence server with state "terminated',
I see 200ok response.. but presence server isn't sending NOTIFY to device A
about state "terminated", everything is good with "trying" and "early"
states...

That problem is happening 1 of 50 tries.. I've tried a lot of things to fix
that - still are not succeed.

p.s. everything is good from logs/sip point of view...  And also, when I
disable the webrtc device - I'm not able to reproduce that issue anymore.
Looks like something wrong with case, when you have multiple different
username behind the same presentity. 

Maybe someone had same issue and may help me. Thanks! 



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Re: [SR-Users] Called with SDP without DTLS fingerprint from WebRTC to SIP

[Zhan Bazarov]

and also:
xlog("IS REQUEST");
if ($proto =~ "ws") {
xlog("WEB --> SIP");
$xavp(r=>$T_branch_idx) =
$xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove
RTP/AVP";
rtpengine_manage($xavp(r=>$T_branch_idx));
if (route(FROMASTERISK)) {
xlog("FROM INTERNAL");
} else {
xlog("FROM EXTERNAL");
rtpengine_manage("
direction=external direction=internal ICE=remove loop-protect");
}

You don't need to do rtpengine_manage twice... But in ur code its executing
twice - I believe thats the root cause of that issue...



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Re: [SR-Users] Called with SDP without DTLS fingerprint from WebRTC to SIP

[Zhan Bazarov]

Hi, check which protocol which r u using on hardphones... If its UDP - it
wouldn't be work - because in ur route block u r just returning from this
route...

This code block working fine for TCP, but won't be work for UDP.. 

if (is_request()) {
if(has_totag()) {
if (check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}

if (check_route_param("rtp=bridge")) {
setbflag(FLB_BRIDGE);
}

if (check_route_param("rtp=ws")) {
setbflag(FLB_RTPWS);
}
}
}


in your case u r just returning from this route:
if (!isbflagset(FLB_BRIDGE)) {
return;
}
because there are no any flags configured yet, I mean, kamailio doesn't now
anything about connection - because there is no any connection in case when
u r using UDP, because UDP is connection stateless 



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[SR-Users] Send_notify_request within cluster scenario(path usrloc)

[Zhan Bazarov]

Hello everyone. I have an issue with presence server within kamailio
cluster(round-robin DNS, I'm using registrar Path scenario)

So, when both of users are located at the same server - everything works
fine. But when user which is SUBSCRIBED through another one server -
presence module is ignoring path column into usrloc - that leads NAT
traversal impossible...

 So, I've tried using DMQ for handle this: 
 I executed dmq_t_replicate()before each PUBLISH message - in that case I
was able to get PUBLISH on each of server in cluster. But BLF lamps are
confusing in that case - because a lot of notifies coming to the device at
the same time with different dialog states

Guys, I need a help. Any Ideas how I can handle this? 

Best regards, 
Zhan



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[SR-Users] Keep one registration per device(src_ip+port)

[Zhan Bazarov]

Hello. 
I'd like to keep one registration per device in usrloc table, for other
cases I want to store and maintain one contact per AoR. 
Does kamailio support it out of the box?

For second case save("location", "0x01"); working fine.  But how about first
case? for example, if one of device located at home, and second located at
office. - there are two different network and NAT - how I can make both
devices ringing when INVITE is sent? If I do classic multiAor scenario - it
can lead to issues caused by multiple contacts in 200OK response for a
single device and devices are confusing sometimes.

So, what I did now:
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
add_contact_alias();
}

[REGISTRAR] {
if (nat_uac_test(64)) {
if (!save("location", "0x04")) {
sl_reply_error();
}
} else {
if (!save("location")) {
sl_reply_error();
}
}
}

modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 3)
modparam("usrloc", "use_domain", MULTIDOMAIN)
modparam("usrloc", "xavp_contact", "usrlocattrs")
modparam("usrloc", "desc_time_order", 1)
modparam("usrloc", "db_timer_clean", 1)
modparam("usrloc", "close_expired_tcp", 1)
modparam("usrloc", "db_insert_update", 1)
modparam("usrloc", "expires_type", 1)
modparam("usrloc", "matching_mode", 1)

modparam("registrar", "method_filtering", 1)
modparam("registrar", "max_expires", 3530)
modparam("registrar", "min_expires", 1800)
modparam("registrar", "received_avp", "$avp(s:rcv)")
modparam("registrar", "gruu_enabled", 0)
modparam("registrar", "use_path", 1)
modparam("registrar", "path_check_local", 1)
modparam("registrar", "path_mode", 1)
modparam("path", "use_received", 1)


How do u think, maybe modparam("registrar", "append_branches", 0) will make
some help? 



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Re: [SR-Users] How to add parameter ;user=phone in RURI

[Zhan Bazarov]

Hi, try to do smth like:

 if (!has_totag()){
$ru = $ru + ";user=phone"
}



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Re: [SR-Users] kamailio and rtpengine in docker

[Zhan Bazarov]

Hi. Share please docker's run options and network topology. 



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Re: [SR-Users] Polycom looses registration within cluster scenario.

[Zhan Bazarov]

Actually I've fixed the issue by adding this:
fix_nated_register();
add_contact_alias();
istead of 
fix_nated_contact();

and then On the second kam(if request coming from another kam) I'm doing
this:
handle_ruri_alias();





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Re: [SR-Users] Polycom looses registration within cluster scenario.

[Zhan Bazarov]

Hi. How did you deal with that? 
From my side I've tried to use path and outbound on the polycom side.. Now
polycom is not jumbing between proxies but still loosing registration...



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[SR-Users] Polycom looses registration within cluster scenario.

[Zhan Bazarov]

Hello. We faced to an issue with polycom, namely - polycom phones looses
registration within SRV/DNS based cluster scenario.. 
So we have round-robin cluster of SIP-proxy instances behind the same SRV/A
records(each instance has the same weight). The initial registration works
fine, but after the polycom sends a SUBSCRIBE request to another one
sipProxy instance(because of round-robin scenario) - polycom stops sending
re-register by expires which we are providing in 200Ok message... 

So the idea is to keep polycom located on the instance where initial
register request came to.

But polycom is sending SUBSCRIBE in shuffle(not to the server where
registration is located)

We can solve this issue by  keeping registration on lines not on main SIP
configuration itself, otherwise polycom looses registration from cluster at
all. But it lead to rejecting some calls by polycom with '400 bad request.. 
We can solve it by disabling validation. But it leads to additional things
like: polycom has multiple registrations on the device itself..  I mean, for
example, first one registration isn’t expired, and polycom has two
registrations at the same time...


any ideas? How we can avoid this issue right? 
Thanks. 



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Re: [SR-Users] Kamailio presence when account not equal extension

[Zhan Bazarov]

Thanks for u reply!
In additional - extensions may be duplicated... I have some kind of
accound_id. And same extension may exist in other account.. 



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Re: [SR-Users] Kamailio presence when account not equal extension

[Zhan Bazarov]

P.S. Subscribe goes on extension.. 



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[SR-Users] Kamailio presence when account not equal extension

[Zhan Bazarov]

Hello, I'm using presence and pua modules, it works perfectly when username
is equal to extension... 
But now I have case when username is not equal to the extension..
Like:
extension: 
username: aythnbghhgt

How I can implement this? Any ideas? 



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[SR-Users] Kamailio presence when account not equal extension

[Zhan Bazarov]

Hello, I'm using presence and pua modules, it works perfectly when username
is equal to extension... 
But now I have case when username is not equal to the extension..
Like:
extension: 
username: aythnbghhgt

How I can implement this? Any ideas? 



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[SR-Users] How to implement simple presence in kamailio active-active cluster.

[Zhan Bazarov]

Hello! We have next setup, which I've desribed in:
http://sip-router.1086192.n5.nabble.com/Bad-request-when-INVITE-relayed-to-another-kamailio-td183788.html

Presence works fine in setup with single kamailio, without any loaded
presence modules... 
So, I  want to something wtih SUBSCRIBE and NOTIFY methods... But when I
making a relay on second kamailio instance I receive 407 proxy auth required
from this instance... 

So, its how I'm trying to implement this:

part of code from route[LOCATION]
case -2:
if (is_method("SUBSCRIBE|NOTIFY")) {
$ru = "sip:" + $tU + "@" +
"ip_of_second_kam:5078";
t_relay("ip_of_second_kam", 5078);
exit;
} else {
send_reply("405", "Method Not Allowed");
exit;
}


Any ideas how I can do that?
Thanks.



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Re: [SR-Users] Bad request when INVITE relayed to another kamailio

[Zhan Bazarov]

Thanks Alex. I found the issue! That was in device side

https://knowledgebase-iframe.polycom.com/kb/viewContent.do?externalId=15474



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Re: [SR-Users] Bad request when INVITE relayed to another kamailio

[Zhan Bazarov]

hello! Thanks for reply! 

So, now I have two INVITE requests with interval in few seconds:

INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0
Record-Route:

Record-Route: 
Record-Route: 
Via: SIP/2.0/TCP
some.proxy.net:5078;branch=z9hG4bK24ec.940ea1b24f9ae5d68a6f97928661dbb6.0
Via: SIP/2.0/UDP
some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK24ec.73c81b0f32164242a9bf0beb0f96f7bb.0
Via: SIP/2.0/UDP
10.199.240.176:7060;received=10.199.240.176;rport=7060;branch=z9hG4bKPj726e453b-eb1b-4e07-b49d-45703bc42e38
From: "Rick"
;tag=1c95c3be-0c2a-44ef-8715-28735586ad95
To: 
Contact: 
Call-ID: 63cf568c-eef4-4262-8e6b-db3f6caedfbb
CSeq: 23870 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, MESSAGE, REFER
Supported: timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "Rick" 
Max-Forwards: 68
User-Agent: Awesome Calling Platform 3.0
Content-Type: application/sdp
Content-Length:   463

v=0
o=- 128351188 128351188 IN IP4 34.206.126.53
s=Asterisk
c=IN IP4 34.206.126.53
t=0 0
m=audio 30010 RTP/AVP 0 101
a=maxptime:150
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:30011
a=ptime:20
a=ice-ufrag:Cw3k5LF5
a=ice-pwd:SyzlcKXcD2CnOT4hOw72yyvq4c
a=candidate:aUPmOZbVuTuCQ7B9 1 UDP 2130706431 34.206.126.53 30010 typ host
a=candidate:aUPmOZbVuTuCQ7B9 2 UDP 2130706430 34.206.126.53 30011 typ host 


2020/01/26 19:07:01.211105 37.17.41.5:25171 -> 10.199.240.19:5078
SIP/2.0 400 Bad Request
Via: SIP/2.0/TCP
some.proxy.net:5078;branch=z9hG4bK24ec.940ea1b24f9ae5d68a6f97928661dbb6.0
Via: SIP/2.0/UDP
some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK24ec.73c81b0f32164242a9bf0beb0f96f7bb.0
Via: SIP/2.0/UDP
10.199.240.176:7060;received=10.199.240.176;rport=7060;branch=z9hG4bKPj726e453b-eb1b-4e07-b49d-45703bc42e38
From: "Rick"
;tag=1c95c3be-0c2a-44ef-8715-28735586ad95
To: ;tag=F6209FFC-FB2B5291
CSeq: 23870 INVITE
Call-ID: 63cf568c-eef4-4262-8e6b-db3f6caedfbb
Record-Route:
,
,

User-Agent: PolycomVVX-VVX_450-UA/5.8.0.13851
Accept-Language: en
Content-Length: 0


*And few seconds later: *

2020/01/26 19:07:06.187829 10.199.240.19:5078 -> 37.17.41.5:25171
INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0
Record-Route:

Record-Route: 
Record-Route: 
Via: SIP/2.0/TCP
some.proxy.net:5078;branch=z9hG4bK32d3.45058b373afc3bd5897903a83bc0de7d.0
Via: SIP/2.0/UDP
some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK32d3.257af0eab041dbd2a240416d1c6e454f.0
Via: SIP/2.0/UDP
10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPjc442ef42-b769-4fdc-81cf-33453aeaae67
From: "Rick"
;tag=b44068d2-d1af-4b56-87c0-2ea334784c94
To: 
Contact: 
Call-ID: f9f4006d-8df9-4501-a77d-d3f7ba45c327
CSeq: 14602 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, MESSAGE, REFER
Supported: timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "Rick" 
Max-Forwards: 68
User-Agent: Awesome Calling Platform 3.0
Content-Type: application/sdp
Content-Length:   461

v=0
o=- 95944032 95944032 IN IP4 34.206.126.53
s=Asterisk
c=IN IP4 34.206.126.53
t=0 0
m=audio 30082 RTP/AVP 0 101
a=maxptime:150
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:30083
a=ptime:20
a=ice-ufrag:qgZIRgJo
a=ice-pwd:mT7rDx0qrjOkoC4Yzlodqfl7ma
a=candidate:aUPmOZbVuTuCQ7B9 1 UDP 2130706431 34.206.126.53 30082 typ host
a=candidate:aUPmOZbVuTuCQ7B9 2 UDP 2130706430 34.206.126.53 30083 typ host



but this one was succeed

2020/01/26 19:07:07.349559 37.17.41.5:25171 -> 10.199.240.19:5078
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 
some.proxy.net:5078;branch=z9hG4bK32d3.45058b373afc3bd5897903a83bc0de7d.0
Via: SIP/2.0/UDP
some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK32d3.257af0eab041dbd2a240416d1c6e454f.0
Via: SIP/2.0/UDP
10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPjc442ef42-b769-4fdc-81cf-33453aeaae67
From: "Rick"
;tag=b44068d2-d1af-4b56-87c0-2ea334784c94
To: "1030_DEV" ;tag=63F0AFCC-6C807D61
CSeq: 14602 INVITE
Call-ID: f9f4006d-8df9-4501-a77d-d3f7ba45c327
Contact: 
Record-Route:
,
,

User-Agent: PolycomVVX-VVX_450-UA/5.8.0.13851
Allow-Events: conference,talk,hold
Accept-Language: en
Call-Info: ;appearance-index=1
Content-Length: 0





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Re: [SR-Users] Bad request when INVITE relayed to another kamailio

[Zhan Bazarov]

Hello! Thanks for reply!
Removing unnecessary headers didn't solve issue.
Switching on TCP didn't solve it neither.



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[SR-Users] Bad request when INVITE relayed to another kamailio

[Zhan Bazarov]

We have Kamailio-cluster via route53(round-robin) some-domain.net

we have two kamailio with public IP's

phone1 is registered on kam1
phone2 is registered on kam2

when we are calling from phone1 to phone2 callflow looks:

phone1 => kam1 => asterisk => kam1 => t_relay(address of second
kamailio:5078) => kam2 => phone2

it works perfectly, but in case when we are using polycom as phone2 - we are
getting 404 response from polycom...


*Invite from second kamailio
*
2020/01/20 10:31:21.799327 10.199.240.19:5078 -> 37.17.41.5:49811
INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0
Record-Route:

Record-Route: 
Record-Route: 
Via: SIP/2.0/TCP
some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0
Via: SIP/2.0/UDP
some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0
Via: SIP/2.0/UDP
10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c
From: "Penny"
;tag=ba402508-a640-409f-ba30-dffdfe499f43
To: 
Contact: 
Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5
CSeq: 22619 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, MESSAGE, REFER
Supported: timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "Penny" 
Max-Forwards: 68
User-Agent: Awesome Calling Platform 3.0
Content-Type: application/sdp
Content-Length:  463


*Response from POLYCOM
*
2020/01/20 10:31:22.054766 37.17.41.5:49811 -> 10.199.240.19:5078
SIP/2.0 400 Bad Request
Via: SIP/2.0/TCP
some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0
Via: SIP/2.0/UDP
some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0
Via: SIP/2.0/UDP
10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c
From: "Penny"
;tag=ba402508-a640-409f-ba30-dffdfe499f43
To: ;tag=8BC58304-83D9B045
CSeq: 22619 INVITE
Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5
Record-Route:
,
,

User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848
Accept-Language: en
Content-Length: 0



Any ideas how to fix it?



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[SR-Users] Call Pickup and Call Parking

[Zhan Bazarov]

Hello,
What is the Best way to implement call parking and call pickup in setup
Kamailio(active-active) and dispatcher with a few asterisk round-robin?

Could someone share some best practices, please?

Thank you!
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Re: [SR-Users] Unresolvable destination for new added users...

[Zhan Bazarov]

modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 3)
modparam("usrloc", "user_column", "username")
modparam("usrloc", "contact_column", "contact")
modparam("usrloc", "expires_column", "expires")
modparam("usrloc", "q_column", "q")
modparam("usrloc", "callid_column", "callid")
modparam("usrloc", "cseq_column", "cseq")
modparam("usrloc", "methods_column", "methods")
modparam("usrloc", "cflags_column", "cflags")
modparam("usrloc", "user_agent_column", "user_agent")
modparam("usrloc", "received_column", "received")
modparam("usrloc", "socket_column", "socket")
modparam("usrloc", "path_column", "path")
modparam("usrloc", "ruid_column", "ruid")
modparam("usrloc", "instance_column", "instance")
modparam("usrloc", "use_domain", 1)
modparam("usrloc", "matching_mode", 0)
modparam("usrloc", "timer_interval", 30)




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Re: [SR-Users] Unresolvable destination for new added users...

[Zhan Bazarov]

USRLOC DB has been configured by default... Nothing new. What kind of
information might be helpful for you?
In additional - both users registered successfully, I see both of them in
kamctl ul show - both on my x-lite and zoiper...



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[SR-Users] Unresolvable destination for new added users...

[Zhan Bazarov]

I can't make call through kamailio between new added users.

I get 478 ... But it works for older users (who was added couple weeks ago
e.g.) SIP flow, location and subscriber table the same in both cases.

Could you please shed any light on this issue?

*2019/10/02 11:11:20.719834 10.2.1.198:5068  ->
10.4.1.14:50600 *

SIP/2.0 *478* Unresolvable destination (478/SL)

Via: SIP/2.0/UDP
asterisk_ip:asterisk_port;branch=z9hG4bK18626d14;rport=50600;received=10.4.1.14

From: "Admin Admin" ;tag=as09b1f1f3

To: ;tag=a1384aaeda4d71b5edb13882669b943e.8468

Call-ID: *20e023254f1b3c942a61e47d63badfd6@10.4.1.14:50600
*

CSeq: 102 INVITE

Server: MS Lync

Content-Length: 0
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[SR-Users] Using $shv within current call only

[Zhan Bazarov]

Hello,
we ran into a trouble with sharing variable into reply route...

We are going to match asterisk's uniqueid with rtpengine records like this:

route[from_asterisk]{
$var(UniqueId)=$hdr(X-UniqueId);
$var(rtp_flags) = $var(rtp_flags) + "
label={uid:$var(UniqueId),stream:outgoing} record-call=on";

 rtpengine_manage($var(rtp_flags));
}

it nicely work for outgoing request, but doesn't for reply

reply_route[from_pstn]{
 rtpengine_manage("replace-origin replace-session-connection ICE=remove
RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing");
}
and it doesn't work... The one possible way what we found its using $shv
instead of $var/$avp... But its unsecure way, cuz it may confus our
bussines logic...

Do you have any idea how to implement this, how we can assign $shv within
current call only?
Many thanks!
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[SR-Users] Share variable into reply route.

[Zhan Bazarov]

Hello,

we ran into a trouble with sharing variable into reply route...

We are going to match asterisk's uniqueid with rtpengine records like this:

route[from_asterisk]{
$var(UniqueId)=$hdr(X-UniqueId);
$var(rtp_flags) = $var(rtp_flags) + "
label={uid:$var(UniqueId),stream:outgoing} record-call=on";

 rtpengine_manage($var(rtp_flags));
}

reply_route[from_pstn]{
 rtpengine_manage("replace-origin replace-session-connection ICE=remove
RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing");
}
and it doesn't work... The one possible way what we found its using $shv
instead of $var... But its unsecure way, cuz it may confus our bussines
logic...

Do you have any idea how to implement this?
Many thanks!
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[SR-Users] kamailio+asterisk VoiceMail implementation

[Zhan Bazarov]

Hello,

what's best practice for VoiceMail implementation in Kamailio+Asterisk's
scheme?

should I forward register into asterisk? Or I can keep register on
kamailio?
Could you, please, shed any lights on this
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[SR-Users] kamailio doesn't reply on internal interface

[Zhan Bazarov]

Hello, we have interesting issue...
Some time kamailio comes unreachable on private interface for asterisk, but
port checked via telnet and everything is ok...
ps. connection via public ip works good

root@kamailio:/usr/local/etc/kamailio# lsof -i :5068 | grep LISTEN
kamailio 6014 root   13u  IPv4 8341589  0t0  TCP kamailio.local.ip:5068
(LISTEN)
kamailio 6014 root   14u  IPv4 8341590  0t0  TCP
kamailio.public.ip:5068 (LISTEN)
root@kamailio:/usr/local/etc/kamailio#

telnet from asterisk machine:

root@asterisk:/etc/asterisk# telnet kamailio.local.ip 5068

Trying kamailio.local.ip...

Connected to kamailio.local.ip.

Escape character is '^]'.


But peer is unreachable in asterisk... More than, kamailio comes reachable
when I change transport from TCP to UDP, or when I restart kamailio
daemon(but it situation repeats every day)..

Any ideas how I can fix it?
thanks!
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[SR-Users] Kamailio auth via http post.

[Zhan Bazarov]

Hello,

Its possible to made authentication via http request to third party Web-API
instead of storing users credentials into Dataqbase?

Now I looking on auth_ephemeral, but not sure is that my case... Don't see
when I have to put API url..

Could anyone, please, shed any light on this issue?
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Re: [SR-Users] Wrong ACL to Provider kamailio+asterisk+uac

[Zhan Bazarov]

Thanks. It works well. But retransmission timeout problem still exists... I
suppose it was not wrong ACK problem.
But many thanks!!

вс, 4 авг. 2019 г. в 19:46, Henning Westerholt :

> Hello Zhan,
>
> you need to also execute it also for the ACK request. I just tested it on
> a test setup, I just placed it here for a quick test (refer to the default
> cfg):
>
> ...
>
> uac_replace_from("","sip:bat...@gotham.org"
> );
> uac_replace_to("","sip:ro...@gotham.org" );
>
> # handle requests within SIP dialogs
> route(WITHINDLG);
> ...
> Works just fine:
>
> 2019/08/04 18:43:21.727990 XXX.XXX.178.27:5060 -> XXX.XXX.167.107:5060
> ACK sip:custome...@xxx.xxx.64.8:5096;ob SIP/2.0
> Via: SIP/2.0/UDP
> XXX.XXX.178.27;branch=z9hG4bK8cfc.358f1178964b526f50452ba03db8ee3e.0
> Via: SIP/2.0/UDP
> XXX.XXX.167.107;rport=5060;branch=z9hG4bK8cfc.85decb742f8d4df6ca0216d3065be210.0
> v: SIP/2.0/UDP
> XXX.XXX.64.8:5095;received=XXX.XXX.64.8;rport=5095;branch=z9hG4bKPjpwl1uGwlM1NDPbYauTk6DazXWBA1yrzV
> Max-Forwards: 68
> f: sip:bat...@gotham.org;tag=TVDG0gI90HONs9GOc0dw7ykSYFgyWUWZ
> t: sip:ro...@gotham.org;tag=ufPqkX.gDlvL.ammPR9QkQ6beeh4fbOL
> i: E3JNM13kkWFd8gCbt0EAvbGRow6zU8OX
> CSeq: 19753 ACK
> Route: 
> l:  0
>
> Cheers,
>
> Henning
>
> Am 04.08.19 um 18:22 schrieb Zhan Bazarov:
>
> Hello,
> Thanks, I will try!
> Should I add this into the same route?
> I execute uac_replace into route[FROMASTERISK] if method is invite.
>
> вс, 4 авг. 2019 г. в 19:15, Henning Westerholt :
>
>> Hello Zhan,
>>
>> I obvious don't know your cfg details. But I would try to just call the
>> uac_replace_from/ .._to in the main request_route and see if it works at
>> all in your routing logic. Also add some xlog messages to see that it is
>> executed. You can also output with $rm the SIP request method, with $ru the
>> request URI etc.. (see the cookbook for pseudo-variables for all that are
>> available).
>>
>> Cheers,
>>
>> Henning
>> Am 03.08.19 um 22:40 schrieb Zhan Bazarov:
>>
>> Hello,
>> It didn’t make any sense, unfortunately...
>> In additional - kamailio relay right 200 to Asterisk, and asterisk send
>> ACK with wrong from/to(initial values like it was before uac_replace) and
>> then kamailio doesn’t change it to right from/to values.
>>
>> Any idea, please.
>>
>> сб, 3 авг. 2019 г. в 19:17, Henning Westerholt :
>>
>>> Hello Zhan,
>>>
>>> there is no need to execute a branch_route on an ACK. The ACK should be
>>> just routed "normally" through Kamailio. Have a look to the default cfg,
>>> for example. The ACK in a normal call setup is just a "normal" request.
>>>
>>> Cheers,
>>>
>>> Henning
>>> Am 02.08.19 um 19:27 schrieb Zhan Bazarov:
>>>
>>> Thanks!
>>> Rly appreciate your help!
>>>
>>> So, I can try to execute the same branch_route for ACK method?
>>> I have this branch route:
>>>
>>> branch_route[MODIFY]{
>>> uac_replace_from("","$avp(from)");
>>> uac_replace_to($avp(to));
>>> }
>>>
>>> Now I execute this when request goes from asterisk through kamailio to
>>> carriers...
>>>
>>> Should I execute this branch route into ACK method from asterisk too?
>>>
>>> пт, 2 авг. 2019 г. в 20:11, Henning Westerholt :
>>>
>>>> Hello Zhan,
>>>>
>>>> possible reasons:
>>>>
>>>> - the ACK is not routed through your proxy (missing record-route for
>>>> the INVITE)
>>>>
>>>> - the user agent mangles the Route header, therefore the uac module
>>>> can't does it work correctly (look to the kamailio error logs if you spot
>>>> something)
>>>>
>>>> Have a look to your SIP e.g. with "ngrep -qt -d any -W byline port
>>>> 5060" on your proxy.
>>>>
>>>> Cheers,
>>>>
>>>> Henning
>>>> Am 02.08.19 um 19:02 schrieb Zhan Bazarov:
>>>>
>>>> Hello,
>>>> Thanks for reply! Yes I execute uac_replace_to/from in branch_route in
>>>> INVITE method... But it doesn't work for ACK method...
>>>>
>>>> пт, 2 авг. 2019 г. в 18:11, Henning Westerholt :
>>>>
>>>>> Hello Zhan,
>>>>>
>>>>> have a look to the "uac" module, there are functions to re

Re: [SR-Users] Wrong ACL to Provider kamailio+asterisk+uac

[Zhan Bazarov]

Hello,
Thanks, I will try!
Should I add this into the same route?
I execute uac_replace into route[FROMASTERISK] if method is invite.

вс, 4 авг. 2019 г. в 19:15, Henning Westerholt :

> Hello Zhan,
>
> I obvious don't know your cfg details. But I would try to just call the
> uac_replace_from/ .._to in the main request_route and see if it works at
> all in your routing logic. Also add some xlog messages to see that it is
> executed. You can also output with $rm the SIP request method, with $ru the
> request URI etc.. (see the cookbook for pseudo-variables for all that are
> available).
>
> Cheers,
>
> Henning
> Am 03.08.19 um 22:40 schrieb Zhan Bazarov:
>
> Hello,
> It didn’t make any sense, unfortunately...
> In additional - kamailio relay right 200 to Asterisk, and asterisk send
> ACK with wrong from/to(initial values like it was before uac_replace) and
> then kamailio doesn’t change it to right from/to values.
>
> Any idea, please.
>
> сб, 3 авг. 2019 г. в 19:17, Henning Westerholt :
>
>> Hello Zhan,
>>
>> there is no need to execute a branch_route on an ACK. The ACK should be
>> just routed "normally" through Kamailio. Have a look to the default cfg,
>> for example. The ACK in a normal call setup is just a "normal" request.
>>
>> Cheers,
>>
>> Henning
>> Am 02.08.19 um 19:27 schrieb Zhan Bazarov:
>>
>> Thanks!
>> Rly appreciate your help!
>>
>> So, I can try to execute the same branch_route for ACK method?
>> I have this branch route:
>>
>> branch_route[MODIFY]{
>> uac_replace_from("","$avp(from)");
>> uac_replace_to($avp(to));
>> }
>>
>> Now I execute this when request goes from asterisk through kamailio to
>> carriers...
>>
>> Should I execute this branch route into ACK method from asterisk too?
>>
>> пт, 2 авг. 2019 г. в 20:11, Henning Westerholt :
>>
>>> Hello Zhan,
>>>
>>> possible reasons:
>>>
>>> - the ACK is not routed through your proxy (missing record-route for the
>>> INVITE)
>>>
>>> - the user agent mangles the Route header, therefore the uac module
>>> can't does it work correctly (look to the kamailio error logs if you spot
>>> something)
>>>
>>> Have a look to your SIP e.g. with "ngrep -qt -d any -W byline port 5060"
>>> on your proxy.
>>>
>>> Cheers,
>>>
>>> Henning
>>> Am 02.08.19 um 19:02 schrieb Zhan Bazarov:
>>>
>>> Hello,
>>> Thanks for reply! Yes I execute uac_replace_to/from in branch_route in
>>> INVITE method... But it doesn't work for ACK method...
>>>
>>> пт, 2 авг. 2019 г. в 18:11, Henning Westerholt :
>>>
>>>> Hello Zhan,
>>>>
>>>> have a look to the "uac" module, there are functions to rewrite the
>>>> From and To header for passing messages. It works together with the rr
>>>> module.
>>>>
>>>> Cheers,
>>>>
>>>> Henning
>>>> Am 02.08.19 um 13:30 schrieb Zhan Bazarov:
>>>>
>>>> Hello!  Please, is it possible to change from/to headers in ACK message
>>>> to provider? this is exactly our case, please, look it when u get a
>>>> chance.
>>>> http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html#a130052
>>>>
>>>> ___
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>>>>
>>>> --
>>>> Henning Westerholt - https://skalatan.de/blog/
>>>> Kamailio services - https://skalatan.de/services
>>>>
>>>> --
>>> Henning Westerholt - https://skalatan.de/blog/
>>> Kamailio services - https://skalatan.de/services
>>>
>>> --
>> Henning Westerholt - https://skalatan.de/blog/
>> Kamailio services - https://skalatan.de/services
>>
>> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
>
>
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Re: [SR-Users] Wrong ACL to Provider kamailio+asterisk+uac

[Zhan Bazarov]

Hello,
It didn’t make any sense, unfortunately...
In additional - kamailio relay right 200 to Asterisk, and asterisk send ACK
with wrong from/to(initial values like it was before uac_replace) and then
kamailio doesn’t change it to right from/to values.

Any idea, please.

сб, 3 авг. 2019 г. в 19:17, Henning Westerholt :

> Hello Zhan,
>
> there is no need to execute a branch_route on an ACK. The ACK should be
> just routed "normally" through Kamailio. Have a look to the default cfg,
> for example. The ACK in a normal call setup is just a "normal" request.
>
> Cheers,
>
> Henning
> Am 02.08.19 um 19:27 schrieb Zhan Bazarov:
>
> Thanks!
> Rly appreciate your help!
>
> So, I can try to execute the same branch_route for ACK method?
> I have this branch route:
>
> branch_route[MODIFY]{
> uac_replace_from("","$avp(from)");
> uac_replace_to($avp(to));
> }
>
> Now I execute this when request goes from asterisk through kamailio to
> carriers...
>
> Should I execute this branch route into ACK method from asterisk too?
>
> пт, 2 авг. 2019 г. в 20:11, Henning Westerholt :
>
>> Hello Zhan,
>>
>> possible reasons:
>>
>> - the ACK is not routed through your proxy (missing record-route for the
>> INVITE)
>>
>> - the user agent mangles the Route header, therefore the uac module can't
>> does it work correctly (look to the kamailio error logs if you spot
>> something)
>>
>> Have a look to your SIP e.g. with "ngrep -qt -d any -W byline port 5060"
>> on your proxy.
>>
>> Cheers,
>>
>> Henning
>> Am 02.08.19 um 19:02 schrieb Zhan Bazarov:
>>
>> Hello,
>> Thanks for reply! Yes I execute uac_replace_to/from in branch_route in
>> INVITE method... But it doesn't work for ACK method...
>>
>> пт, 2 авг. 2019 г. в 18:11, Henning Westerholt :
>>
>>> Hello Zhan,
>>>
>>> have a look to the "uac" module, there are functions to rewrite the From
>>> and To header for passing messages. It works together with the rr module.
>>>
>>> Cheers,
>>>
>>> Henning
>>> Am 02.08.19 um 13:30 schrieb Zhan Bazarov:
>>>
>>> Hello!  Please, is it possible to change from/to headers in ACK message
>>> to provider? this is exactly our case, please, look it when u get a
>>> chance.
>>> http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html#a130052
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing 
>>> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>> --
>>> Henning Westerholt - https://skalatan.de/blog/
>>> Kamailio services - https://skalatan.de/services
>>>
>>> --
>> Henning Westerholt - https://skalatan.de/blog/
>> Kamailio services - https://skalatan.de/services
>>
>> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
>
>
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Re: [SR-Users] Wrong ACL to Provider kamailio+asterisk+uac

[Zhan Bazarov]

And also, forget to say: it happens after 200ok from carrier...
I mean, the first one transaction kamailio uac <=> carrier is ok.
invite =>
<= 407
invite =>
<=183
<=180



and then 200Ok when PSTN side answer on call.

<= 200 ok
=> wrong ACK (with initial to/from, which they was before uac_replace)



пт, 2 авг. 2019 г. в 20:27, Zhan Bazarov :

> Thanks!
> Rly appreciate your help!
>
> So, I can try to execute the same branch_route for ACK method?
> I have this branch route:
>
> branch_route[MODIFY]{
> uac_replace_from("","$avp(from)");
> uac_replace_to($avp(to));
> }
>
> Now I execute this when request goes from asterisk through kamailio to
> carriers...
>
> Should I execute this branch route into ACK method from asterisk too?
>
> пт, 2 авг. 2019 г. в 20:11, Henning Westerholt :
>
>> Hello Zhan,
>>
>> possible reasons:
>>
>> - the ACK is not routed through your proxy (missing record-route for the
>> INVITE)
>>
>> - the user agent mangles the Route header, therefore the uac module can't
>> does it work correctly (look to the kamailio error logs if you spot
>> something)
>>
>> Have a look to your SIP e.g. with "ngrep -qt -d any -W byline port 5060"
>> on your proxy.
>>
>> Cheers,
>>
>> Henning
>> Am 02.08.19 um 19:02 schrieb Zhan Bazarov:
>>
>> Hello,
>> Thanks for reply! Yes I execute uac_replace_to/from in branch_route in
>> INVITE method... But it doesn't work for ACK method...
>>
>> пт, 2 авг. 2019 г. в 18:11, Henning Westerholt :
>>
>>> Hello Zhan,
>>>
>>> have a look to the "uac" module, there are functions to rewrite the From
>>> and To header for passing messages. It works together with the rr module.
>>>
>>> Cheers,
>>>
>>> Henning
>>> Am 02.08.19 um 13:30 schrieb Zhan Bazarov:
>>>
>>> Hello!  Please, is it possible to change from/to headers in ACK message
>>> to provider? this is exactly our case, please, look it when u get a
>>> chance.
>>> http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html#a130052
>>>
>>> ___
>>> Kamailio (SER) - Users Mailing 
>>> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>> --
>>> Henning Westerholt - https://skalatan.de/blog/
>>> Kamailio services - https://skalatan.de/services
>>>
>>> --
>> Henning Westerholt - https://skalatan.de/blog/
>> Kamailio services - https://skalatan.de/services
>>
>>
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Re: [SR-Users] Wrong ACL to Provider kamailio+asterisk+uac

[Zhan Bazarov]

Thanks!
Rly appreciate your help!

So, I can try to execute the same branch_route for ACK method?
I have this branch route:

branch_route[MODIFY]{
uac_replace_from("","$avp(from)");
uac_replace_to($avp(to));
}

Now I execute this when request goes from asterisk through kamailio to
carriers...

Should I execute this branch route into ACK method from asterisk too?

пт, 2 авг. 2019 г. в 20:11, Henning Westerholt :

> Hello Zhan,
>
> possible reasons:
>
> - the ACK is not routed through your proxy (missing record-route for the
> INVITE)
>
> - the user agent mangles the Route header, therefore the uac module can't
> does it work correctly (look to the kamailio error logs if you spot
> something)
>
> Have a look to your SIP e.g. with "ngrep -qt -d any -W byline port 5060"
> on your proxy.
>
> Cheers,
>
> Henning
> Am 02.08.19 um 19:02 schrieb Zhan Bazarov:
>
> Hello,
> Thanks for reply! Yes I execute uac_replace_to/from in branch_route in
> INVITE method... But it doesn't work for ACK method...
>
> пт, 2 авг. 2019 г. в 18:11, Henning Westerholt :
>
>> Hello Zhan,
>>
>> have a look to the "uac" module, there are functions to rewrite the From
>> and To header for passing messages. It works together with the rr module.
>>
>> Cheers,
>>
>> Henning
>> Am 02.08.19 um 13:30 schrieb Zhan Bazarov:
>>
>> Hello!  Please, is it possible to change from/to headers in ACK message
>> to provider? this is exactly our case, please, look it when u get a
>> chance.
>> http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html#a130052
>>
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>>
>> --
>> Henning Westerholt - https://skalatan.de/blog/
>> Kamailio services - https://skalatan.de/services
>>
>> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
>
>
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Re: [SR-Users] Wrong ACL to Provider kamailio+asterisk+uac

[Zhan Bazarov]

Hello,
Thanks for reply! Yes I execute uac_replace_to/from in branch_route in
INVITE method... But it doesn't work for ACK method...

пт, 2 авг. 2019 г. в 18:11, Henning Westerholt :

> Hello Zhan,
>
> have a look to the "uac" module, there are functions to rewrite the From
> and To header for passing messages. It works together with the rr module.
>
> Cheers,
>
> Henning
> Am 02.08.19 um 13:30 schrieb Zhan Bazarov:
>
> Hello!  Please, is it possible to change from/to headers in ACK message to
> provider? this is exactly our case, please, look it when u get a chance.
> http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html#a130052
>
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>
> --
> Henning Westerholt - https://skalatan.de/blog/
> Kamailio services - https://skalatan.de/services
>
>
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[SR-Users] Wrong ACL to Provider kamailio+asterisk+uac

[Zhan Bazarov]

Hello!  Please, is it possible to change from/to headers in ACK message to
provider? this is exactly our case, please, look it when u get a chance.
http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html#a130052
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[SR-Users] kamailio webrtc 183 early media.

[Zhan Bazarov]

Hello! I have setup kamailio+rtpengine+asterisk.. And I have issue with
early media for WEBrtc endpoints... As I understood right, for this case I
need to use rel100(PRACK) for sending early media... But I didn't find any
example for solve this issue... Can you help me with it? It would be
greatly appreciated!
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[SR-Users] Large PDD

[Zhan Bazarov]

Hello, I have kamailio+asterisk platform. Everything works fine. But,
sometimes, for unclear reasons, the PDD(8000 and more) rises to an incoming
call(at the stage when the call goes to an asterisk).
I can not figure it out myself (SIP is clean, hardware metrics are ok)
and it happens in random order (under load and without it).

Anybody have ideas, what in can the reason?
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