Re: [SR-Users] sngrep TLS connection

2020-12-08 Thread me.proje...@yahoo.com
 There are two options
1- use SIMPDUMP module to dup all SIP packets in Log file SIPDUMP Module


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SIPDUMP Module


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2- use SIPTRACE module and broadcast SIP traffic from kamailio to any local 
port and then capture through sngrep -p BROACST_PORT
SipTrace Module


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SipTrace Module


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loadmodule "siptrace.so"
# check IP and port of your capture nodemodparam("siptrace", "duplicate_uri", 
"sip:127.0.0.1:9060")modparam("siptrace", "hep_mode_on", 0)modparam("siptrace", 
"trace_to_database", 0)modparam("siptrace", "trace_flag", 
22)modparam("siptrace", "trace_on", 1)# modparam("siptrace", "hep_version", 
3)modparam("siptrace", "xheaders_write", 1)
during code use  setflag(22); sip_trace();
This will broadcast SIP packet to local 9060 port and u can capture that 
through sngrep.
I hope it will help.





On Tuesday, December 8, 2020, 01:08:44 PM GMT+5, Daniel-Constantin Mierla 
 wrote:  
 
  
Hello,
 
there were some limitations in sngrep for capturing tls traffic:
 
  * https://github.com/irontec/sngrep/issues/112
 
Not sure if there was any work afterwards to improve. The best place to ask in 
on sngrep project.
 
 
Anyhow, I want to add that if you use Kamailio, then you can load sipdump 
module and get the traffic stored in pcap files -- it requires master branch:
 
  
*https://www.kamailio.org/docs/modules/devel/modules/sipdump.html#sipdump.p.mode
 
For stable branches, sipdump can store the traffic in text files.
 
Cheers,
 Daniel
 
 On 08.12.20 08:37, Yuriy Gorlichenko wrote:
  
 
As I remember sngrep doesn't work with TLS 1.3, so may be it is your case. 
  On Tue, 8 Dec 2020, 08:01 Agiftel,  wrote:
  
Hi all, does anyone was able to use sngrep (with -k option) to decrypt TLS
 connections?
 I'am doing what help says; used private key is the correct one; but no
 traffic is seen on sngrep.
 If i use not ecrypted call i can see everything, so sngrep is working good.
 
 Any hint?
 
 regards
 
 
 
 
 --
 Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
 
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Re: [SR-Users] android client SIP TLS client autentication + kamailio

2020-12-02 Thread me.proje...@yahoo.com
 Hi, I have used it. Some older version of Android I think before Android 4.0 
but not sure, does not support TLS so you have to work with older version of 
TLS or SSL.
Security with HTTPS and SSL | Android Developers


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Security with HTTPS and SSL | Android Developers


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How to enable TLS 1.2 support in an Android application (running on Android 4.1 
JB)


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How to enable TLS 1.2 support in an Android application (running on Andr...

As per the docs in Android for SSLSocket and SSLContext, TLS v1.1 and v1.2 
protocols are supported in API level ...
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Which Library you are using on Android ?






On Wednesday, December 2, 2020, 07:37:02 PM GMT+5, beer Ll 
 wrote:  
 
 HelloHas anybody use a android SIP client with TLS client authentication 
enabled with kamailio?Do you have suggestion ?
Many thanks
Luca



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[SR-Users] MSTeams Kamailio with PBX Hold Refer issue

2020-12-02 Thread me.proje...@yahoo.com
Hi, I have been trying to integrate MSTeams with PBX Through Kamailio. Calls 
are working now. Hold has Issues. MSTeams uses REFER to manage this all with 
NOTIFY. 
I am sending t_send_reply("202", "Accepted");
when receive Refer now I need to send NOTIFY I think I can use t_uac_send() 
NOTIFY I am unable to send t_uac_send properly on when recertifications refer 
from MSTeams.
Any one with the experience on this ?
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Re: [SR-Users] What is the lifecycle of Branch Flag (setbflag)

2020-11-29 Thread me.proje...@yahoo.com
 Thanks a lot for all this :)
On Sunday, November 29, 2020, 11:43:29 PM GMT+5, Alex Balashov 
 wrote:  
 
 The flags only apply to branches created by Kamailio, not incoming 
branches.

Relaying anything creates a new branch; changing the RURI does not 
affect that.

On 11/29/20 1:40 PM, me.proje...@yahoo.com wrote:
> Hi Alex, Thanks for the reply. I think branch is identified by RURI... 
> if we change the RURI and relay it to other hop then would it be new 
> Branch ? If we set BFlag in Request Route would it be available this 
> changed RURI for this transaction ?
> 
> On Sunday, November 29, 2020, 11:21:48 PM GMT+5, Alex Balashov 
>  wrote:
> 
> 
> It will be available within a particular branch, which is a subset of a
> single transaction.
> 
> On 11/29/20 1:20 PM, me.proje...@yahoo.com 
> <mailto:me.proje...@yahoo.com> wrote:
>  > Hi,
>  >
>  > What is the lifecycle of Branch Flag? If its set in Request Route will
>  > it be available till end of Dialog or It will be available during
>  > Transaction only ?
>  >
>  > Thanks
> 
>  >
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> 
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
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Re: [SR-Users] What is the lifecycle of Branch Flag (setbflag)

2020-11-29 Thread me.proje...@yahoo.com
 Hi Alex, Thanks for the reply. I think branch is identified by RURI... if we 
change the RURI and relay it to other hop then would it be new Branch ? If we 
set BFlag in Request Route would it be available this changed RURI for this 
transaction ?
On Sunday, November 29, 2020, 11:21:48 PM GMT+5, Alex Balashov 
 wrote:  
 
 It will be available within a particular branch, which is a subset of a 
single transaction.

On 11/29/20 1:20 PM, me.proje...@yahoo.com wrote:
> Hi,
> 
> What is the lifecycle of Branch Flag? If its set in Request Route will 
> it be available till end of Dialog or It will be available during 
> Transaction only ?
> 
> Thanks
> 
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[SR-Users] What is the lifecycle of Branch Flag (setbflag)

2020-11-29 Thread me.proje...@yahoo.com
Hi,
What is the lifecycle of Branch Flag? If its set in Request Route will it be 
available till end of Dialog or It will be available during Transaction only ?
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[SR-Users] Kamailio Remove Remove "Allow" features from header

2020-11-26 Thread me.proje...@yahoo.com
Hi,
I want to remove some "Allow" features from my Kamailio SBC like I want to keep 
following only 
Allow: OPTIONS, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER,

How can I achieve that?
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Re: [SR-Users] ACK Relay Issue Transaction Not Found

2020-11-24 Thread me.proje...@yahoo.com
 Wow great. Thanks for your help :)

On Tuesday, November 24, 2020, 10:54:25 PM GMT+5, Federico Cabiddu 
 wrote:  
 
 The outgoing (relayed) e2e ACK is not captured by the sipcapture module in 
"transaction" capture mode (the default), because of tm module internals.You 
have to explicitly capture it in the onsend_route.
onsend_route {
    if ($rm == 'ACK') {
        sip_trace();
    }
}
Cheers,
Federico
On Tue, Nov 24, 2020 at 6:47 PM Alex Balashov  wrote:

On 11/24/20 12:35 PM, Antony Stone wrote:

> On Tuesday 24 November 2020 at 18:28:44, Alex Balashov wrote:
> 
>> Hiddenly? :-)
> 
> I like the word - I can think of several contexts it would be useful in, and
> the standard way of expressing the concept is so much less concise.

It's true; the closest and comparably terse adverb I can readily conjure 
is "surreptitiously", which implies a more actively deliberate and 
nefarious level of packet forwarding.

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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
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Re: [SR-Users] ACK Relay Issue Transaction Not Found

2020-11-24 Thread me.proje...@yahoo.com
 Hiddenly means without being captured in SIP_TRACE module 
On Tuesday, November 24, 2020, 10:29:58 PM GMT+5, Alex Balashov 
 wrote:  
 
 Hiddenly? :-)

-- Alex

On 11/24/20 12:22 PM, me.proje...@yahoo.com wrote:

> Thanks for your reply.
> I can't see the ACK being relayed to MSTeams in network capture through 
> in Relay Route
> 
> if(is_method("ACK|BYE")) {
> sip_trace();
> setflag(22);
> }
>    but call has been established successfully. is ACK is being relayed 
> hiddenly by Kamailio ?
> 
> On Tuesday, November 24, 2020, 09:35:32 PM GMT+5, Alex Balashov 
>  wrote:
> 
> 
> Hi,
> 
> If you take a look at the documentation for t_check_trans(), you'll see
> that it behaves in very different ways depending on the type of message
> it is handling, and is supposed to return a negative/false value in some
> scenarios or abort script execution -- context is very important:
> 
> https://kamailio.org/docs/modules/stable/modules/tm.html#tm.f.t_check_trans 
> <https://kamailio.org/docs/modules/stable/modules/tm.html#tm.f.t_check_trans>
> 
> There are two types of ACKs - hop-by-hop ACKs, which acknowledge
> negative final replies, and end-to-end ACKs, which are for successful
> final 2xx replies and meant to be relayed by Kamailio end-to-end, in the
> manner of a normal in-dialog request.
> 
> An ACK of this nature is a separate request and a separate transaction,
> though its CSeq is identical to that of the INVITE transaction it is
> meant to acknowledge. However, a 2xx final reply will close an INVITE
> transaction, subject to `max_inv_lifetime` and `wt_timer`[1] and all
> that stuff:
> 
> The documentation speaks to this point:
> 
>      "For end-to-end ACKs (ACKs to 2xx responses for forwarded
>      INVITE transactions) it will return true if the corresponding
>      INVITE transaction is found and still active and false if not.
> 
>      Note that the e2e ACK matching is more of a hint then a
>      certainty. A delayed e2e ACK might arrive after the
>      transaction wait time elapses, when the INVITE transaction
>      no longer exists and thus would not match anything. There
>      are also cases when tm would not keep all the information
>      needed for e2e ACK matching (since this is not needed for
>      a stateful proxy and it requires additional memory,
>      tm will not keep this information unless needed by
>      some other module or callbacks)."
> 
> So, in other words, you should not expect t_check_trans() to return true
> for an end-to-end ACK. It is reasonable to expect that in the vast
> majority of cases, it will return false. The ACK should be relayed in
> the same manner as any other loose-routed in-dialog request, e.g. a
> reinvite or a BYE. If you follow the logic of the stock configuration
> that Kamailio ships with, this will happen correctly by default.
> 
> -- Alex
> 
> [1] 
> https://kamailio.org/docs/modules/stable/modules/tm.html#tm.p.wt_timer 
> <https://kamailio.org/docs/modules/stable/modules/tm.html#tm.p.wt_timer>
> 
> On 11/24/20 3:24 AM, me.proje...@yahoo.com 
> <mailto:me.proje...@yahoo.com> wrote:
>  > Hi When ACK is received on Kamailio from PBX t_check_trans() fails.
>  > 200 OK sent PBX from Teams and ACK from PBX is provided here.
>  >
>  > What could be the issue.
>  >
>  > SIP/2.0 200 OK
>  > FROM: "Ext1" <mailto:1102@PBX_IP>>;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f
>  > TO:  <mailto:1101@KAM_IP>>;tag=e667397cb47c457a860222931e320b41
>  > CSEQ: 25772 INVITE
>  > CALL-ID: a6685754-b402-4a15-82b2-5fb2551e5b58
>  > VIA: SIP/2.0/UDP
>  > 
> PBX_IP:7790;received=PBX_IP;branch=z9hG4bKPj170912fc-0908-40bb-8248-9f6cb560332d;rport=7790
>  
> 
>  >
>  > RECORD-ROUTE:
>  > 
> ,,
>  
> 
>  >
>  > CONTACT:
>  > 
> 
>  
> 
>  >
>  > CONTENT-LENGTH: 353
>  > SUPPORTED: timer
>  > CONTENT-TYPE: application/sdp
>  > ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
>  > REQUIRE: timer
>  > SESSION-EXPIRES: 1800;refresher=uac
>  > SERVER: Microsoft.PSTNHub.SIPProxy v.2020.11.16.5 i.USEA.5
>  > X-Siptrace-Fromip: tls:KAM_IP:7790
>  > X-Siptrace-Toip: udp:PBX_IP:7790
>  > X-Siptrace-Time: 1606205334 607029
>  > X-Siptrace-Method: INVITE
>  > X-Siptrace-Dir: out
>  >
>  > ACK
>  > 
> sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;alias=.114.132.46~5061~3
>  
> 
>  > SIP/2.0
>  > Via: SIP/2.0/UDP
>  > PBX_IP:7790;rport;branch=z9hG4bKPjaa30c14c-c965-473d

Re: [SR-Users] ACK Relay Issue Transaction Not Found

2020-11-24 Thread me.proje...@yahoo.com
 Thanks for your reply. I can't see the ACK being relayed to MSTeams in network 
capture through in Relay Route
 if(is_method("ACK|BYE")) { sip_trace(); setflag(22); }   but call has been 
established successfully. is ACK is being relayed hiddenly by Kamailio ?
On Tuesday, November 24, 2020, 09:35:32 PM GMT+5, Alex Balashov 
 wrote:  
 
 Hi,

If you take a look at the documentation for t_check_trans(), you'll see 
that it behaves in very different ways depending on the type of message 
it is handling, and is supposed to return a negative/false value in some 
scenarios or abort script execution -- context is very important:

https://kamailio.org/docs/modules/stable/modules/tm.html#tm.f.t_check_trans

There are two types of ACKs - hop-by-hop ACKs, which acknowledge 
negative final replies, and end-to-end ACKs, which are for successful 
final 2xx replies and meant to be relayed by Kamailio end-to-end, in the 
manner of a normal in-dialog request.

An ACK of this nature is a separate request and a separate transaction, 
though its CSeq is identical to that of the INVITE transaction it is 
meant to acknowledge. However, a 2xx final reply will close an INVITE 
transaction, subject to `max_inv_lifetime` and `wt_timer`[1] and all 
that stuff:

The documentation speaks to this point:

    "For end-to-end ACKs (ACKs to 2xx responses for forwarded
    INVITE transactions) it will return true if the corresponding
    INVITE transaction is found and still active and false if not.

    Note that the e2e ACK matching is more of a hint then a
    certainty. A delayed e2e ACK might arrive after the
    transaction wait time elapses, when the INVITE transaction
    no longer exists and thus would not match anything. There
    are also cases when tm would not keep all the information
    needed for e2e ACK matching (since this is not needed for
    a stateful proxy and it requires additional memory,
    tm will not keep this information unless needed by
    some other module or callbacks)."

So, in other words, you should not expect t_check_trans() to return true 
for an end-to-end ACK. It is reasonable to expect that in the vast 
majority of cases, it will return false. The ACK should be relayed in 
the same manner as any other loose-routed in-dialog request, e.g. a 
reinvite or a BYE. If you follow the logic of the stock configuration 
that Kamailio ships with, this will happen correctly by default.

-- Alex

[1] https://kamailio.org/docs/modules/stable/modules/tm.html#tm.p.wt_timer

On 11/24/20 3:24 AM, me.proje...@yahoo.com wrote:
> Hi When ACK is received on Kamailio from PBX t_check_trans() fails.
> 200 OK sent PBX from Teams and ACK from PBX is provided here.
> 
> What could be the issue.
> 
> SIP/2.0 200 OK
> FROM: "Ext1";tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f
> TO: ;tag=e667397cb47c457a860222931e320b41
> CSEQ: 25772 INVITE
> CALL-ID: a6685754-b402-4a15-82b2-5fb2551e5b58
> VIA: SIP/2.0/UDP 
> PBX_IP:7790;received=PBX_IP;branch=z9hG4bKPj170912fc-0908-40bb-8248-9f6cb560332d;rport=7790
>  
> 
> RECORD-ROUTE: 
> ,,
>  
> 
> CONTACT: 
> 
>  
> 
> CONTENT-LENGTH: 353
> SUPPORTED: timer
> CONTENT-TYPE: application/sdp
> ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
> REQUIRE: timer
> SESSION-EXPIRES: 1800;refresher=uac
> SERVER: Microsoft.PSTNHub.SIPProxy v.2020.11.16.5 i.USEA.5
> X-Siptrace-Fromip: tls:KAM_IP:7790
> X-Siptrace-Toip: udp:PBX_IP:7790
> X-Siptrace-Time: 1606205334 607029
> X-Siptrace-Method: INVITE
> X-Siptrace-Dir: out
> 
> ACK 
> sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;alias=.114.132.46~5061~3
>  
> SIP/2.0
> Via: SIP/2.0/UDP 
> PBX_IP:7790;rport;branch=z9hG4bKPjaa30c14c-c965-473d-bc49-9f223f9f634c
> From: "Ext1" ;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f
> To: ;tag=e667397cb47c457a860222931e320b41
> Call-ID: a6685754-b402-4a15-82b2-5fb2551e5b58
> CSeq: 25772 ACK
> Route: 
> Route: 
> 
>  
> 
> Route: 
> Max-Forwards: 70
> User-Agent: Asterisk
> Content-Length:  0
> X-Siptrace-Fromip: udp:PBX_IP:7790
> X-Siptrace-Toip: udp:KAM_IP:7790
> X-Siptrace-Time: 1606205334 615495
> X-Siptrace-Method: ACK
> X-Siptrace-Dir: in
> 
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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[SR-Users] ACK Relay Issue Transaction Not Found

2020-11-24 Thread me.proje...@yahoo.com
Hi When ACK is received on Kamailio from PBX t_check_trans() fails. 200 OK sent 
PBX from Teams and ACK from PBX is provided here.
What could be the issue.
                                                                                
                          SIP/2.0 200 OK                                        
                                                                                
                                   FROM: 
"Ext1";tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f                
                                                                          TO: 
;tag=e667397cb47c457a860222931e320b41                          
                                                                              
CSEQ: 25772 INVITE                                                              
                                                                                
         CALL-ID: a6685754-b402-4a15-82b2-5fb2551e5b58                          
                                                                                
                  VIA: SIP/2.0/UDP 
PBX_IP:7790;received=PBX_IP;branch=z9hG4bKPj170912fc-0908-40bb-8248-9f6cb560332d;rport=7790
                                             RECORD-ROUTE: 
,,
                                                                                
                 CONTACT: 

                                                                                
                                                                CONTENT-LENGTH: 
353                                                                             
                                                                         
SUPPORTED: timer                                                                
                                                                                
         CONTENT-TYPE: application/sdp                                          
                                                                                
                  ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY                   
                                                                                
                           REQUIRE: timer                                       
                                                                                
                                    SESSION-EXPIRES: 1800;refresher=uac         
                                                                                
                                             SERVER: Microsoft.PSTNHub.SIPProxy 
v.2020.11.16.5 i.USEA.5                                                         
                                                      X-Siptrace-Fromip: 
tls:KAM_IP:7790                                                                 
                                                               X-Siptrace-Toip: 
udp:PBX_IP:7790                                                                 
                                                                
X-Siptrace-Time: 1606205334 607029                                              
                                                                                
         X-Siptrace-Method: INVITE                                              
                                                                                
                  X-Siptrace-Dir: out
                                                                                
                         ACK 
sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=c825abba-7203-442c-8461-994522d14365;x-c=25f3c2ce3a635971b809dc3be872e51b/s/1/63c9eea743284db99366cabf5f7dcbf5;alias=.114.132.46~5061~3
 SIP/2.0                                                                        
                                                                       Via: 
SIP/2.0/UDP 
PBX_IP:7790;rport;branch=z9hG4bKPjaa30c14c-c965-473d-bc49-9f223f9f634c          
                                                                From: "Ext1" 
;tag=e08f0a03-1120-4e13-b0e2-90d1a027c05f                      
                                                                   To: 
;tag=e667397cb47c457a860222931e320b41                          
                                                                              
Call-ID: a6685754-b402-4a15-82b2-5fb2551e5b58                                   
                                                                                
         CSeq: 25772 ACK                                                        
                                                                                
                  Route: 
                  
                                                                           
Route: 

                                                                               
Route:             
                                                                                
         Max-Forwards: 70                                                   

Re: [SR-Users] ACK Not Relayed to MS Teams

2020-11-23 Thread me.proje...@yahoo.com
 Just one more update 
in ACK  t_check_trans() fail
What shall be reason
On Monday, November 23, 2020, 08:12:10 PM GMT+5, me.proje...@yahoo.com 
 wrote:  
 
  Hi George Gogdlize, config file is mess at this stage. Will be able to get it 
working then can share it, at this moment stuck in this scenario if you have 
something working do let me know so I can implement that fix.
Thanks,
On Monday, November 23, 2020, 06:24:37 PM GMT+5, George Goglidze 
 wrote:  
 
 Hi, 
Any chance you could share the relevant config? I’m doing exact same thing 
integrating teams with on-premise pbx via Kamaili.
Get Outlook for iOSFrom: sr-users  on 
behalf of me.proje...@yahoo.com 
Sent: Monday, November 23, 2020 1:07:15 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] ACK Not Relayed to MS Teams Hi,
I am integrating PBX with Teams via Kamailio
Call is working but ACK delivery has an issue 

This is the ACK Kamailio receives from PBX
I process the ACK with 
handle_ruri_alias();

and 
record_route_preset(AS per GUIDE LINES)

But this change is not reflected in RELAY 

ACK from PBX 
ACK 
sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=0c344c6a-cd75-4095-9891-bf9445c28b82;x-c=5691aa770226562db7d3e16ab090e196/s/1/e857255ef2344e03bd1d12e270ef5e6c;ias=52.114.132.46~5061~3
 SIP/2.0                                                                        
                                                           Via: SIP/2.0/UDP 
PBX_IP:7790;rport;branch=z9hG4bKPjc50440d0-77db-4467-94cb-2f93c4f88538          
                                                          From: "USER" 
;tag=ab87b9b5-eee3-420c-8472-2eec82a769ac                     
                                                              To: 
;tag=a2ce378b106949cf8bac8f978bb573fb                     
                                                                             
Call-ID: 3dbc6680-fec3-4ba5-8232-6dda8c669bca                                   
                                                                                
   CSeq: 23811 ACK                                                              
                                                                                
      Route: 
             
                                                                          
Route: 

                                                                         Route: 
                   
                                                                            
Max-Forwards: 70                                                                
                                                                                
   User-Agent: Asterisk                                                         
                                                                                
      Content-Length:  0                                                        
                                                                                
         X-Siptrace-Fromip: udp:PBX_IP:7790                                     
                                                                                
    X-Siptrace-Toip: udp:KAMAILIO_IP:7790                                       
                                                                                
     X-Siptrace-Time: 1606135546 516717                                         
                                                                                
        X-Siptrace-Method: ACK                                                  
                                                                                
           X-Siptrace-Dir: in

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Re: [SR-Users] ACK Not Relayed to MS Teams

2020-11-23 Thread me.proje...@yahoo.com
 Hi George Gogdlize, config file is mess at this stage. Will be able to get it 
working then can share it, at this moment stuck in this scenario if you have 
something working do let me know so I can implement that fix.
Thanks,
On Monday, November 23, 2020, 06:24:37 PM GMT+5, George Goglidze 
 wrote:  
 
 Hi, 
Any chance you could share the relevant config? I’m doing exact same thing 
integrating teams with on-premise pbx via Kamaili.
Get Outlook for iOSFrom: sr-users  on 
behalf of me.proje...@yahoo.com 
Sent: Monday, November 23, 2020 1:07:15 PM
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] ACK Not Relayed to MS Teams Hi,
I am integrating PBX with Teams via Kamailio
Call is working but ACK delivery has an issue 

This is the ACK Kamailio receives from PBX
I process the ACK with 
handle_ruri_alias();

and 
record_route_preset(AS per GUIDE LINES)

But this change is not reflected in RELAY 

ACK from PBX 
ACK 
sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=0c344c6a-cd75-4095-9891-bf9445c28b82;x-c=5691aa770226562db7d3e16ab090e196/s/1/e857255ef2344e03bd1d12e270ef5e6c;ias=52.114.132.46~5061~3
 SIP/2.0                                                                        
                                                           Via: SIP/2.0/UDP 
PBX_IP:7790;rport;branch=z9hG4bKPjc50440d0-77db-4467-94cb-2f93c4f88538          
                                                          From: "USER" 
;tag=ab87b9b5-eee3-420c-8472-2eec82a769ac                     
                                                              To: 
;tag=a2ce378b106949cf8bac8f978bb573fb                     
                                                                             
Call-ID: 3dbc6680-fec3-4ba5-8232-6dda8c669bca                                   
                                                                                
   CSeq: 23811 ACK                                                              
                                                                                
      Route: 
             
                                                                          
Route: 

                                                                         Route: 
                   
                                                                            
Max-Forwards: 70                                                                
                                                                                
   User-Agent: Asterisk                                                         
                                                                                
      Content-Length:  0                                                        
                                                                                
         X-Siptrace-Fromip: udp:PBX_IP:7790                                     
                                                                                
    X-Siptrace-Toip: udp:KAMAILIO_IP:7790                                       
                                                                                
     X-Siptrace-Time: 1606135546 516717                                         
                                                                                
        X-Siptrace-Method: ACK                                                  
                                                                                
           X-Siptrace-Dir: in

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[SR-Users] ACK Not Relayed to MS Teams

2020-11-23 Thread me.proje...@yahoo.com
Hi,
I am integrating PBX with Teams via Kamailio
Call is working but ACK delivery has an issue 

This is the ACK Kamailio receives from PBX
I process the ACK with 
handle_ruri_alias();

and 
record_route_preset(AS per GUIDE LINES)

But this change is not reflected in RELAY 

ACK from PBX 
ACK 
sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=0c344c6a-cd75-4095-9891-bf9445c28b82;x-c=5691aa770226562db7d3e16ab090e196/s/1/e857255ef2344e03bd1d12e270ef5e6c;ias=52.114.132.46~5061~3
 SIP/2.0                                                                        
                                                           Via: SIP/2.0/UDP 
PBX_IP:7790;rport;branch=z9hG4bKPjc50440d0-77db-4467-94cb-2f93c4f88538          
                                                          From: "USER" 
;tag=ab87b9b5-eee3-420c-8472-2eec82a769ac                     
                                                              To: 
;tag=a2ce378b106949cf8bac8f978bb573fb                     
                                                                             
Call-ID: 3dbc6680-fec3-4ba5-8232-6dda8c669bca                                   
                                                                                
   CSeq: 23811 ACK                                                              
                                                                                
      Route: 
             
                                                                          
Route: 

                                                                         Route: 
                   
                                                                            
Max-Forwards: 70                                                                
                                                                                
   User-Agent: Asterisk                                                         
                                                                                
      Content-Length:  0                                                        
                                                                                
         X-Siptrace-Fromip: udp:PBX_IP:7790                                     
                                                                                
    X-Siptrace-Toip: udp:KAMAILIO_IP:7790                                       
                                                                                
     X-Siptrace-Time: 1606135546 516717                                         
                                                                                
        X-Siptrace-Method: ACK                                                  
                                                                                
           X-Siptrace-Dir: in

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[SR-Users] Re-Invite from callee With RTPEngine between RTP--SRTP

2020-11-19 Thread me.proje...@yahoo.com
Hi, I have have Kamailio as SBC one-end is SRTP and other end is RTP
So I m using 
rtpengine_manage("replace-origin replace-session-connection DTLS=passive 
OSRTP-offer ICE=force SRTP")rtpengine_manage("replace-origin 
replace-session-connection DTLS=off SDES-off ICE=remove RTP")

1-Asterisk(RTP) -2-Kamailio/RTPEngine -- 3- Carrier with SRTP
I detect source IPs and to manage this all. 
When call is made from 1 to 3 all works and during call 3 send re-Invite to 
Hold Call .. We need to call rtpengine_manage and also when on 200OK reply of 
re-invite from 1.

How shall I detect that 1- if this is Re-Invite (has_totag() )and 2- its from 
Callee.
Please advise.

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Re: [SR-Users] How to Add Customer Table During Installation

2020-11-05 Thread me.proje...@yahoo.com
 Hi, 
I added two scripts 1- to create tables 2- Insert some basic data in address 
table
Statement 1 executed and tables created but second .sql file not executed 
although it showed in logs. What shall I do to insert some basic data?
Thanks.
On Thursday, November 5, 2020, 07:03:04 PM GMT+5, me.proje...@yahoo.com 
 wrote:  
 
  Hi,
Thanks for your reply. Some times we need to provide Initial Data in some 
tables so I think we can use something you suggested. 
On Thursday, November 5, 2020, 06:00:23 PM GMT+5, Daniel-Constantin Mierla 
 wrote:  
 
  Hello, 
  if you put the sql to create the table in 
share/kamailio/mysql/customtable-create.sql 
  then you can run: 
  kamdbctl add-tables customtable 
  This has to be done after the kamdbctl create command. 
  If you plan for long term, also taking in consideration the option to add 
management of customtable records in the cli, I can recommend to look at 
kamcli: 
    * https://github.com/kamailio/kamcli 
  It can also create the customtable using: 
  kamcli db create-tables-group customtable 
  The 'kamcli db' offers what old 'kamdbctl' and 'kamctl db ...' do. 
  But then you can just add your own python file in 
'kamcli/commands/cmd_customtable.py' to manage records in that table (for 
example, you can use cmd_aliasdb.py as a starting point). 
  Cheers,
 Daniel
  
  On 05.11.20 13:40, me.proje...@yahoo.com wrote:
  
 
 Hi, 
  I want add some custom Table Creation during installation of Kamailio.  
  I want to add my Custom Table during the execution of  
  kamdbctl create
  
  command. Currently I am adding my script in  
  share/kamailio/mysql/auth_db-create.sql 
  Thanks,  
  ___
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 -- 
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www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla ___
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Re: [SR-Users] How to Add Customer Table During Installation

2020-11-05 Thread me.proje...@yahoo.com
 Hi,
Thanks for your reply. Some times we need to provide Initial Data in some 
tables so I think we can use something you suggested. 
On Thursday, November 5, 2020, 06:00:23 PM GMT+5, Daniel-Constantin Mierla 
 wrote:  
 
  Hello, 
  if you put the sql to create the table in 
share/kamailio/mysql/customtable-create.sql 
  then you can run: 
  kamdbctl add-tables customtable 
  This has to be done after the kamdbctl create command. 
  If you plan for long term, also taking in consideration the option to add 
management of customtable records in the cli, I can recommend to look at 
kamcli: 
    * https://github.com/kamailio/kamcli 
  It can also create the customtable using: 
  kamcli db create-tables-group customtable 
  The 'kamcli db' offers what old 'kamdbctl' and 'kamctl db ...' do. 
  But then you can just add your own python file in 
'kamcli/commands/cmd_customtable.py' to manage records in that table (for 
example, you can use cmd_aliasdb.py as a starting point). 
  Cheers,
 Daniel
  
  On 05.11.20 13:40, me.proje...@yahoo.com wrote:
  
 
 Hi, 
  I want add some custom Table Creation during installation of Kamailio.  
  I want to add my Custom Table during the execution of  
  kamdbctl create
  
  command. Currently I am adding my script in  
  share/kamailio/mysql/auth_db-create.sql 
  Thanks,  
  ___
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 -- 
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www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla   ___
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[SR-Users] How to Add Customer Table During Installation

2020-11-05 Thread me.proje...@yahoo.com
Hi,
I want add some custom Table Creation during installation of Kamailio. 
I want to add my Custom Table during the execution of 
kamdbctl create

command. Currently I am adding my script in 
share/kamailio/mysql/auth_db-create.sql
Thanks,___
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[SR-Users] Kamailio Relay Replies Issue

2020-10-15 Thread me.proje...@yahoo.com
I have DID provider which is sending INVITES to Kamailio and kamailio has UAC 
module with asterisk to forward Calls to Asterisk with uac_auth()

1-DID Provider > (TLS)2-Kamailio (UDP) -> (UDP) 3-Asterisk 

Call from Kamailio to Asterisk is forwarded and answered but NON of the reply 
received in "MANAGE_REPLY" route is not relayed back to DID provider call get 
disconnected.

=SIP DUMP 
INVITE FROM DID
INVITE sip:+1...@kamailioserer.com:5061;user=phone;transport=tls SIP/2.0FROM: 
Alga 
Tech;tag=28989e96e28d47d891b4d8d21d85fa92TO:
 CSEQ: 1 INVITECALL-ID: 
936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 70VIA: SIP/2.0/TLS 
DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: 
CONTACT: 
CONTENT-LENGTH:
 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy 
v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: 
INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih
---Invite to Asterisk with AUTH
INVITE sip:+1...@kamailioserer.com:5061;user=phone;transport=tls SIP/2.0FROM: 
Alga 
Tech;tag=28989e96e28d47d891b4d8d21d85fa92TO:
 CSEQ: 1 INVITECALL-ID: 
936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 69VIA: SIP/2.0/TLS 
DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: 
CONTACT: 
CONTENT-LENGTH:
 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy 
v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: 
INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih
--
>From Asterisk to Kamailio
SIP/2.0 100 TryingVia: SIP/2.0/UDP 
KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via:
 SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: 
Record-Route:
 
Record-Route:
 Call-ID: 
936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" 
;tag=28989e96e28d47d891b4d8d21d85fa92To: 
CSeq: 1 INVITEServer: 
AsteriskContent-Length:  0TH: uch
--
>From Asterisk to Kamailio

SIP/2.0 180 RingingVia: SIP/2.0/UDP 
KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via:
 SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: 
Record-Route:
 
Record-Route:
 Call-ID: 
936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" 
;tag=28989e96e28d47d891b4d8d21d85fa92To: 
;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq:
 1 INVITEServer: AsteriskContact: Allow: OPTIONS, 
SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, 
MESSAGE, REFERContent-Length:  0
--
>From Asterisk to Kamailio

SIP/2.0 180 RingingVia: SIP/2.0/UDP 
KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via:
 SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: 
Record-Route:
 
Record-Route:
 Call-ID: 
936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" 
;tag=28989e96e28d47d891b4d8d21d85fa92To: 
;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq:
 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, 
ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: 
Content-Length:  0TH: uch
--
>From Asterisk to Kamailio
SIP/2.0 200 OKVia: SIP/2.0/UDP 
KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via:
 SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: 
Record-Route:
 
Record-Route:
 Call-ID: 
936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" 
;tag=28989e96e28d47d891b4d8d21d85fa92To: 
;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq:
 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, 
ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: 
Supported: 100rel, timer, replaces, 
norefersubSession-Expires: 3600;refresher=uacRequire: timerContent-Type: 
application/sdpContent-Length:   299TH: uch___
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