Hello Everyone,
I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API.
Call Flow:
Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s).
Does a reliable module for this already exist today?
Thanks in advance ! Stay safe.
Kamailio isn't really meant to initiate calls -- it's a SIP proxy.
It does have at least one gruesomely hacky way of doing so anyway:
https://kamailio.org/docs/modules/5.4.x/modules/dialog.html#dialog.f.dlg_bri...
And you can trigger it via the JSONRPC interface via HTTP using the `jsonrpcs` and `xhttp` modules:
https://kamailio.org/docs/modules/5.4.x/modules/dialog.html#dlg.r.bridge_dlg
https://kamailio.org/docs/modules/5.4.x/modules/jsonrpcs.html
https://kamailio.org/docs/modules/5.4.x/modules/xhttp.html
But if you ask me, this is a really, really bad idea, and the API call in question should be going out to the B2BUA, not to Kamailio. :-)
-- Alex
On 9/9/20 4:58 PM, Ameed Jamous wrote:
Hello Everyone,
I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API.
Call Flow:
Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s).
Does a reliable module for this already exist today?
Thanks in advance ! Stay safe.
-- photo *Ameed Jamous* Founder, TelecomsXChange
+1 954 246 3356 tel:+1+954+246+3356 | a.jamous@telecomsxchange.com mailto:a.jamous@telecomsxchange.com
www.telecomsxchange.com http://www.telecomsxchange.com | Skype: telecomsxchange <#SignatureSanitizer_SafeHtmlFilter_>
1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Ameed, you might be interested to look at Jambonz, which runs in node and is very HTTPS-friendly. Kind-of HTTP-to-SIP B2BUA. There are few presentations Dave Horton done on YouTube about it recently. He is very nice guy also has some mailings lists where you can ask questions like this. Good luck!
-Max
On Wed., Sep. 9, 2020, 1:59 p.m. Ameed Jamous, a.jamous@telecomsxchange.com wrote:
Hello Everyone,
I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API.
Call Flow:
Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s).
Does a reliable module for this already exist today?
Thanks in advance ! Stay safe.
-- [image: photo] *Ameed Jamous* Founder, TelecomsXChange
+1 954 246 3356 <+1+954+246+3356> | a.jamous@telecomsxchange.com
www.telecomsxchange.com | Skype: telecomsxchange <#m_-4549784323434264268_m_-4629235627212483015_SignatureSanitizer_SafeHtmlFilter_>
1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Maxim, Thanks for reminding me, I will look into Jambonz again. I think that might just work for the use case we're trying to achieve.
Thanks for the feedback Alex too.
-- [image: photo] *Ameed Jamous* Founder, TelecomsXChange
+1 954 246 3356 <+1+954+246+3356> | a.jamous@telecomsxchange.com
www.telecomsxchange.com | Skype: telecomsxchange <#SignatureSanitizer_SafeHtmlFilter_>
1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
On Thu, Sep 10, 2020 at 8:09 AM Maxim Sobolev sobomax@sippysoft.com wrote:
Ameed, you might be interested to look at Jambonz, which runs in node and is very HTTPS-friendly. Kind-of HTTP-to-SIP B2BUA. There are few presentations Dave Horton done on YouTube about it recently. He is very nice guy also has some mailings lists where you can ask questions like this. Good luck!
-Max
On Wed., Sep. 9, 2020, 1:59 p.m. Ameed Jamous, < a.jamous@telecomsxchange.com> wrote:
Hello Everyone,
I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API.
Call Flow:
Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s).
Does a reliable module for this already exist today?
Thanks in advance ! Stay safe.
-- [image: photo] *Ameed Jamous* Founder, TelecomsXChange
+1 954 246 3356 <+1+954+246+3356> | a.jamous@telecomsxchange.com
www.telecomsxchange.com | Skype: telecomsxchange <#m_7404560307509413927_m_-4549784323434264268_m_-4629235627212483015_SignatureSanitizer_SafeHtmlFilter_>
1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi Ameed,
also Yate should work for this, it can do js, php and python. https://docs.yate.ro/wiki/Programmer%27s_guide
Some code examples http://yate.null.ro/websvn/listing.php?repname=yate&path=%2Ftrunk%2Fshar...
Cheers
Gerry
On 10 Sep 2020, at 15:17, Ameed Jamous a.jamous@telecomsxchange.com wrote:
Maxim, Thanks for reminding me, I will look into Jambonz again. I think that might just work for the use case we're trying to achieve.
Thanks for the feedback Alex too.
-- Ameed Jamous Founder, TelecomsXChange +1 954 246 3356 tel:+1+954+246+3356 | a.jamous@telecomsxchange.com mailto:a.jamous@telecomsxchange.com www.telecomsxchange.com http://www.telecomsxchange.com/ | Skype: telecomsxchange x-msg://18/#SignatureSanitizer_SafeHtmlFilter_ 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
On Thu, Sep 10, 2020 at 8:09 AM Maxim Sobolev <sobomax@sippysoft.com mailto:sobomax@sippysoft.com> wrote: Ameed, you might be interested to look at Jambonz, which runs in node and is very HTTPS-friendly. Kind-of HTTP-to-SIP B2BUA. There are few presentations Dave Horton done on YouTube about it recently. He is very nice guy also has some mailings lists where you can ask questions like this. Good luck!
-Max
On Wed., Sep. 9, 2020, 1:59 p.m. Ameed Jamous, <a.jamous@telecomsxchange.com mailto:a.jamous@telecomsxchange.com> wrote: Hello Everyone,
I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API.
Call Flow:
Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s).
Does a reliable module for this already exist today?
Thanks in advance ! Stay safe.
-- Ameed Jamous Founder, TelecomsXChange +1 954 246 3356 tel:+1+954+246+3356 | a.jamous@telecomsxchange.com mailto:a.jamous@telecomsxchange.com www.telecomsxchange.com http://www.telecomsxchange.com/ | Skype: telecomsxchange x-msg://18/#m_7404560307509413927_m_-4549784323434264268_m_-4629235627212483015_SignatureSanitizer_SafeHtmlFilter_ 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org mailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org mailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello,
if looking around for a lightweight sip only b2bua, SEMS should be checked as well:
* https://github.com/sems-server/sems
Cheers, Daniel
On 10.09.20 15:42, Gerry | Rigatta wrote:
Hi Ameed,
also Yate should work for this, it can do js, php and python. https://docs.yate.ro/wiki/Programmer%27s_guide
Some code examples http://yate.null.ro/websvn/listing.php?repname=yate&path=%2Ftrunk%2Fshar...
Cheers
Gerry
On 10 Sep 2020, at 15:17, Ameed Jamous <a.jamous@telecomsxchange.com mailto:a.jamous@telecomsxchange.com> wrote:
Maxim, Thanks for reminding me, I will look into Jambonz again. I think that might just work for the use case we're trying to achieve.
Thanks for the feedback Alex too.
-- photo
*Ameed Jamous* Founder, TelecomsXChange +1 954 246 3356 tel:+1+954+246+3356 | a.jamous@telecomsxchange.com mailto:a.jamous@telecomsxchange.com www.telecomsxchange.com http://www.telecomsxchange.com/ | Skype: telecomsxchange x-msg://18/#SignatureSanitizer_SafeHtmlFilter_ 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
On Thu, Sep 10, 2020 at 8:09 AM Maxim Sobolev <sobomax@sippysoft.com mailto:sobomax@sippysoft.com> wrote:
Ameed, you might be interested to look at Jambonz, which runs in node and is very HTTPS-friendly. Kind-of HTTP-to-SIP B2BUA. There are few presentations Dave Horton done on YouTube about it recently. He is very nice guy also has some mailings lists where you can ask questions like this. Good luck! -Max On Wed., Sep. 9, 2020, 1:59 p.m. Ameed Jamous, <a.jamous@telecomsxchange.com <mailto:a.jamous@telecomsxchange.com>> wrote: Hello Everyone, I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API. Call Flow: Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s). Does a reliable module for this already exist today? Thanks in advance ! Stay safe. -- photo *Ameed Jamous* Founder, TelecomsXChange +1 954 246 3356 <tel:+1+954+246+3356> | a.jamous@telecomsxchange.com <mailto:a.jamous@telecomsxchange.com> www.telecomsxchange.com <http://www.telecomsxchange.com/> | Skype: telecomsxchange <x-msg://18/#m_7404560307509413927_m_-4549784323434264268_m_-4629235627212483015_SignatureSanitizer_SafeHtmlFilter_> 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org mailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Oh yes, SEMS is excellent.
— Sent from mobile, with due apologies for brevity and errors.
On Sep 10, 2020, at 10:05 AM, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
if looking around for a lightweight sip only b2bua, SEMS should be checked as well:
Cheers, Daniel
On 10.09.20 15:42, Gerry | Rigatta wrote:
Hi Ameed,
also Yate should work for this, it can do js, php and python. https://docs.yate.ro/wiki/Programmer%27s_guide
Some code examples http://yate.null.ro/websvn/listing.php?repname=yate&path=%2Ftrunk%2Fshar...
Cheers
Gerry
On 10 Sep 2020, at 15:17, Ameed Jamous a.jamous@telecomsxchange.com wrote:
Maxim, Thanks for reminding me, I will look into Jambonz again. I think that might just work for the use case we're trying to achieve.
Thanks for the feedback Alex too.
--
Ameed Jamous Founder, TelecomsXChange +1 954 246 3356 | a.jamous@telecomsxchange.com www.telecomsxchange.com | Skype: telecomsxchange 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
On Thu, Sep 10, 2020 at 8:09 AM Maxim Sobolev sobomax@sippysoft.com wrote:
Ameed, you might be interested to look at Jambonz, which runs in node and is very HTTPS-friendly. Kind-of HTTP-to-SIP B2BUA. There are few presentations Dave Horton done on YouTube about it recently. He is very nice guy also has some mailings lists where you can ask questions like this. Good luck!
-Max
On Wed., Sep. 9, 2020, 1:59 p.m. Ameed Jamous, a.jamous@telecomsxchange.com wrote:
Hello Everyone,
I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API.
Call Flow:
Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s).
Does a reliable module for this already exist today?
Thanks in advance ! Stay safe.
--
Ameed Jamous Founder, TelecomsXChange +1 954 246 3356 | a.jamous@telecomsxchange.com www.telecomsxchange.com | Skype: telecomsxchange 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Gerry/Daniel,
Thanks! I will review both projects as well.
On Thu, Sep 10, 2020 at 6:31 PM Alex Balashov abalashov@evaristesys.com wrote:
Oh yes, SEMS is excellent.
— Sent from mobile, with due apologies for brevity and errors.
On Sep 10, 2020, at 10:05 AM, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
if looking around for a lightweight sip only b2bua, SEMS should
be checked as well:
Cheers,
Daniel
On 10.09.20 15:42, Gerry | Rigatta
wrote:
Hi Ameed,
also Yate should work for this, it can do js, php
and python.
https://docs.yate.ro/wiki/Programmer%27s_guide
Some code examples
http://yate.null.ro/websvn/listing.php?repname=yate&path=%2Ftrunk%2Fshar...
Cheers
Gerry
On 10 Sep 2020, at 15:17, Ameed Jamous a.jamous@telecomsxchange.com
wrote:
Maxim, Thanks for reminding me, I will
look into Jambonz again. I
think that might just work for the use case
we're trying to achieve.
Thanks
for the feedback Alex too.
--
[image: photo]
*Ameed Jamous*
Founder,
TelecomsXChange
+1 954 246 3356 <+1+954+246+3356> | a.jamous@telecomsxchange.com
www.telecomsxchange.com
| Skype: telecomsxchange
1800 N Bayshore Dr https://www.google.com/maps/search/1800+N+Bayshore+Dr?entry=gmail&source=g, Suite CU1,
Miami, FL
33132
On Thu, Sep 10, 2020
at 8:09 AM Maxim Sobolev sobomax@sippysoft.com
wrote:
Ameed, you might be interested to look
at Jambonz, which runs in node and is very
HTTPS-friendly. Kind-of HTTP-to-SIP B2BUA. There
are few presentations Dave Horton done on YouTube
about it recently. He is very nice guy also has
some mailings lists where you can ask questions
like this. Good luck!
-Max
On Wed., Sep.
9, 2020, 1:59 p.m. Ameed Jamous, a.jamous@telecomsxchange.com
wrote:
Hello
Everyone,
I was wondering
what’s the most recommended module by the
community today to initiate calls via
Kamailio using REST API.
Call Flow:
Application —>
HTTP (Async) —> Kamailio —> B2BUA
—> Callee(s).
Does a reliable
module for this already exist today?
Thanks in advance !
Stay safe.
--
[image: photo]
*Ameed Jamous*
Founder,
TelecomsXChange
+1
954 246 3356 <+1+954+246+3356>
|
a.jamous@telecomsxchange.com
www.telecomsxchange.com
| Skype: telecomsxchange
1800 N Bayshore Dr https://www.google.com/maps/search/1800+N+Bayshore+Dr?entry=gmail&source=g, Suite CU1,
Miami, FL
33132
Kamailio (SER) - Users Mailing List
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
[image: photo] *Ameed Jamous* Founder, TelecomsXChange
+1 954 246 3356 <+1+954+246+3356> | a.jamous@telecomsxchange.com
www.telecomsxchange.com | Skype: telecomsxchange <#SignatureSanitizer_SafeHtmlFilter_>
1800 N Bayshore Dr, Suite CU1, Miami, FL 33132
I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump.
“ngrep -d any -qt -W byline port 5060”
Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured.
Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected. UA client = sip.js / wss Inbound works(but 1 way audio ill fix after outbound issue) Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump.
But I seet his when it happens knowing I ame getting something sent. Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041) Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
Are you using vlans? In the past I had issues with ngrep if the packets had vlan tags on them.
On Thu, Sep 10, 2020 at 6:10 PM Johnny Ritzer sudoritz@gmail.com wrote:
I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump.
“ngrep -d any -qt -W byline port 5060”
Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured.
Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected.
UA client = sip.js / wss Inbound works(but 1 way audio ill fix after outbound issue)
Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump.
But I seet his when it happens knowing I ame getting something sent.
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041) _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
No… or not that I know of
This is a temp GCP (as im building a POC to get this to a working concept.) trying to fine-tune config
From: sr-users sr-users-bounces@lists.kamailio.org Date: Thursday, September 10, 2020 at 3:33 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call) Are you using vlans? In the past I had issues with ngrep if the packets had vlan tags on them.
On Thu, Sep 10, 2020 at 6:10 PM Johnny Ritzer <sudoritz@gmail.commailto:sudoritz@gmail.com> wrote: I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump.
“ngrep -d any -qt -W byline port 5060”
Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured. Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected. UA client = sip.js / wss Inbound works(but 1 way audio ill fix after outbound issue) Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump.
But I seet his when it happens knowing I ame getting something sent. Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041) Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041) _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
You sure you’re using 5060 on kamailio?
On Fri, 11 Sep 2020 at 00:04, Johnny Ritzer sudoritz@gmail.com wrote:
No… or not that I know of
This is a temp GCP (as im building a POC to get this to a working concept.) trying to fine-tune config
*From:*sr-users sr-users-bounces@lists.kamailio.org
*Date: *Thursday, September 10, 2020 at 3:33 PM
*To: *Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org
*Subject: *Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
Are you using vlans? In the past I had issues with ngrep if the packets had vlan tags on them.
On Thu, Sep 10, 2020 at 6:10 PM Johnny Ritzer sudoritz@gmail.com wrote:
I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump.
“ngrep -d any -qt -W byline port 5060”
Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured.
Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected.
UA client = sip.js / wss
Inbound works(but 1 way audio ill fix after outbound issue)
Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump.
But I seet his when it happens knowing I ame getting something sent.
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
Ya I have other scenarios that show up like inbound call.
So I did this (made a test call) and saved output ngrep to pcap.
And open in wireshark and I see it as protocol UDP with SIP enclosed in data
ngrep -d any -qt -W byline port 5060 -O test.pcap
heres ngrep 2020/09/10 23:19:51.109093 10.111.0.4:5060 -> 127.0.0.1:9060 #1 .......[F... ...INVITE sip:8000@24.xx.xx.xx SIP/2.0. Via: SIP/2.0/WSS 4d8asfsdfag3f.invalid;branch=z9hG4bK2797686. Max-Forwards: 70. To: sip:8000@24.xx.xx.xx. From: "Ctx-WEBSIP-1138" sip:1138@24.xx.xx.xx;tag=jd85ifa0f2. Call-ID: 9pujji6vasdfasfrbn2. CSeq: 906 INVITE. Contact: sip:1138@24.xx.xx.xx;gr=urn:uuid:121211-a79d-4e76-a010-121211. Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER. Supported: gruu, outbound. User-Agent: SIP.js/0.7.8. Content-Type: application/sdp. Content-Length: 2047.
From: sr-users sr-users-bounces@lists.kamailio.org Date: Thursday, September 10, 2020 at 4:19 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call) You sure you’re using 5060 on kamailio?
On Fri, 11 Sep 2020 at 00:04, Johnny Ritzer <sudoritz@gmail.commailto:sudoritz@gmail.com> wrote:
No… or not that I know of
This is a temp GCP (as im building a POC to get this to a working concept.) trying to fine-tune config
From:
sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org>
Date: Thursday, September 10, 2020 at 3:33 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
Are you using vlans? In the past I had issues with ngrep if the packets had vlan tags on them.
On Thu, Sep 10, 2020 at 6:10 PM Johnny Ritzer <sudoritz@gmail.commailto:sudoritz@gmail.com> wrote:
I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump.
“ngrep -d any -qt -W byline port 5060”
Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured.
Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected.
UA client = sip.js / wss
Inbound works(but 1 way audio ill fix after outbound issue)
Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump.
But I seet his when it happens knowing I ame getting something sent.
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Regards,
David Villasmil email: david.villasmil.work@gmail.commailto:david.villasmil.work@gmail.com phone: +34669448337
Hmmmm seeing that port 127.0.0.1….9060 looks like to capture and forward to homer (I know I installed it at one time to try to some triage).
Maybe homer built for additional checks. I just disabled the module hosting that port and I no longer see it in ngrep.
my path issue is wss:// webrtc over to uac.reg and FW it off to PBX. So ill try and figure out where in kama.cfg to hopefully get it to fw to pbx and make call.
From: Johnny Ritzer sudoritz@gmail.com Date: Thursday, September 10, 2020 at 4:32 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call) Ya I have other scenarios that show up like inbound call.
So I did this (made a test call) and saved output ngrep to pcap.
And open in wireshark and I see it as protocol UDP with SIP enclosed in data
ngrep -d any -qt -W byline port 5060 -O test.pcap
heres ngrep 2020/09/10 23:19:51.109093 10.111.0.4:5060 -> 127.0.0.1:9060 #1 .......[F... ...INVITE sip:8000@24.xx.xx.xx SIP/2.0. Via: SIP/2.0/WSS 4d8asfsdfag3f.invalid;branch=z9hG4bK2797686. Max-Forwards: 70. To: sip:8000@24.xx.xx.xx. From: "Ctx-WEBSIP-1138" sip:1138@24.xx.xx.xx;tag=jd85ifa0f2. Call-ID: 9pujji6vasdfasfrbn2. CSeq: 906 INVITE. Contact: sip:1138@24.xx.xx.xx;gr=urn:uuid:121211-a79d-4e76-a010-121211. Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER. Supported: gruu, outbound. User-Agent: SIP.js/0.7.8. Content-Type: application/sdp. Content-Length: 2047.
From: sr-users sr-users-bounces@lists.kamailio.org Date: Thursday, September 10, 2020 at 4:19 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call) You sure you’re using 5060 on kamailio?
On Fri, 11 Sep 2020 at 00:04, Johnny Ritzer <sudoritz@gmail.commailto:sudoritz@gmail.com> wrote:
No… or not that I know of
This is a temp GCP (as im building a POC to get this to a working concept.) trying to fine-tune config
From:
sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org>
Date: Thursday, September 10, 2020 at 3:33 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
Are you using vlans? In the past I had issues with ngrep if the packets had vlan tags on them.
On Thu, Sep 10, 2020 at 6:10 PM Johnny Ritzer <sudoritz@gmail.commailto:sudoritz@gmail.com> wrote:
I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump.
“ngrep -d any -qt -W byline port 5060”
Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured.
Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected.
UA client = sip.js / wss
Inbound works(but 1 way audio ill fix after outbound issue)
Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump.
But I seet his when it happens knowing I ame getting something sent.
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Regards,
David Villasmil email: david.villasmil.work@gmail.commailto:david.villasmil.work@gmail.com phone: +34669448337
Dude, you need to learn how to write properly on a mailing list. This is not a chat room or an IRC channel.
On 9/10/20 7:41 PM, Johnny Ritzer wrote:
Hmmmm seeing that port 127.0.0.1….9060 looks like to capture and forward to homer (I know I installed it at one time to try to some triage).
Maybe homer built for additional checks. I just disabled the module hosting that port and I no longer see it in ngrep.
my path issue is wss:// webrtc over to uac.reg and FW it off to PBX. So ill try and figure out where in kama.cfg to hopefully get it to fw to pbx and make call.
*From: *Johnny Ritzer sudoritz@gmail.com *Date: *Thursday, September 10, 2020 at 4:32 PM *To: *Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org *Subject: *Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
Ya I have other scenarios that show up like inbound call.
So I did this (made a test call) and saved output ngrep to pcap.
And open in wireshark and I see it as protocol UDP with SIP enclosed in data
ngrep -d any -qt -W byline port 5060 -O test.pcap
heres ngrep
2020/09/10 23:19:51.109093 10.111.0.4:5060 -> 127.0.0.1:9060 #1
.......[F...
...INVITE sip:8000@24.xx.xx.xx SIP/2.0.
Via: SIP/2.0/WSS 4d8asfsdfag3f.invalid;branch=z9hG4bK2797686.
Max-Forwards: 70.
To: sip:8000@24.xx.xx.xx.
From: "Ctx-WEBSIP-1138" sip:1138@24.xx.xx.xx;tag=jd85ifa0f2.
Call-ID: 9pujji6vasdfasfrbn2.
CSeq: 906 INVITE.
Contact: sip:1138@24.xx.xx.xx;gr=urn:uuid:121211-a79d-4e76-a010-121211.
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER.
Supported: gruu, outbound.
User-Agent: SIP.js/0.7.8.
Content-Type: application/sdp.
Content-Length: 2047.
*From: *sr-users sr-users-bounces@lists.kamailio.org *Date: *Thursday, September 10, 2020 at 4:19 PM *To: *Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org *Subject: *Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
You sure you’re using 5060 on kamailio?
On Fri, 11 Sep 2020 at 00:04, Johnny Ritzer <sudoritz@gmail.com mailto:sudoritz@gmail.com> wrote:
No… or not that I know of This is a temp GCP (as im building a POC to get this to a working concept.) trying to fine-tune config *From: *sr-users <sr-users-bounces@lists.kamailio.org <mailto:sr-users-bounces@lists.kamailio.org>> *Date: *Thursday, September 10, 2020 at 3:33 PM *To: *Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org>> *Subject: *Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call) Are you using vlans? In the past I had issues with ngrep if the packets had vlan tags on them. On Thu, Sep 10, 2020 at 6:10 PM Johnny Ritzer <sudoritz@gmail.com <mailto:sudoritz@gmail.com>> wrote: I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump. “ngrep -d any -qt -W byline port 5060” Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured. Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected. UA client = sip.js / wss Inbound works(but 1 way audio ill fix after outbound issue) Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump. But I seet his when it happens knowing I ame getting something sent. Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041) Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041) _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Regards,
David Villasmil
email: david.villasmil.work@gmail.com mailto:david.villasmil.work@gmail.com
phone: +34669448337
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello,
wss is not usually on port 5060, it can be on port 5061 (if you haven't configured another tls socket specially for wss traffic). Anyhow, wss traffic is encrypted, so sngrep will not match it as sip traffic.
If you want to see locally the wss sip traffic, load sipdump module, it will write in text files (the version in master can save the traffic in pcap files as well, although it will appear as being udp traffic, because it was simpler to build the pcap headers, however you can sport from the headers/ports what was the trasport layer, or enable to add the extra meta-data sip header).
If you already have homer, then you should see the wss encrypted traffic there.
Cheers, Daniel
On 11.09.20 01:41, Johnny Ritzer wrote:
Hmmmm seeing that port 127.0.0.1….9060 looks like to capture and forward to homer (I know I installed it at one time to try to some triage).
Maybe homer built for additional checks. I just disabled the module hosting that port and I no longer see it in ngrep.
my path issue is wss:// webrtc over to uac.reg and FW it off to PBX. So ill try and figure out where in kama.cfg to hopefully get it to fw to pbx and make call.
*From: *Johnny Ritzer sudoritz@gmail.com *Date: *Thursday, September 10, 2020 at 4:32 PM *To: *Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org *Subject: *Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
Ya I have other scenarios that show up like inbound call.
So I did this (made a test call) and saved output ngrep to pcap.
And open in wireshark and I see it as protocol UDP with SIP enclosed in data
ngrep -d any -qt -W byline port 5060 -O test.pcap
heres ngrep
2020/09/10 23:19:51.109093 10.111.0.4:5060 -> 127.0.0.1:9060 #1
.......[F...
...INVITE sip:8000@24.xx.xx.xx SIP/2.0.
Via: SIP/2.0/WSS 4d8asfsdfag3f.invalid;branch=z9hG4bK2797686.
Max-Forwards: 70.
To: sip:8000@24.xx.xx.xx.
From: "Ctx-WEBSIP-1138" sip:1138@24.xx.xx.xx;tag=jd85ifa0f2.
Call-ID: 9pujji6vasdfasfrbn2.
CSeq: 906 INVITE.
Contact: sip:1138@24.xx.xx.xx;gr=urn:uuid:121211-a79d-4e76-a010-121211.
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER.
Supported: gruu, outbound.
User-Agent: SIP.js/0.7.8.
Content-Type: application/sdp.
Content-Length: 2047.
*From: *sr-users sr-users-bounces@lists.kamailio.org *Date: *Thursday, September 10, 2020 at 4:19 PM *To: *Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org *Subject: *Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
You sure you’re using 5060 on kamailio?
On Fri, 11 Sep 2020 at 00:04, Johnny Ritzer <sudoritz@gmail.com mailto:sudoritz@gmail.com> wrote:
No… or not that I know of This is a temp GCP (as im building a POC to get this to a working concept.) trying to fine-tune config *From: *sr-users <sr-users-bounces@lists.kamailio.org <mailto:sr-users-bounces@lists.kamailio.org>> *Date: *Thursday, September 10, 2020 at 3:33 PM *To: *Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org>> *Subject: *Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call) Are you using vlans? In the past I had issues with ngrep if the packets had vlan tags on them. On Thu, Sep 10, 2020 at 6:10 PM Johnny Ritzer <sudoritz@gmail.com <mailto:sudoritz@gmail.com>> wrote: I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump. “ngrep -d any -qt -W byline port 5060” Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured. Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected. UA client = sip.js / wss Inbound works(but 1 way audio ill fix after outbound issue) Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump. But I seet his when it happens knowing I ame getting something sent. Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041) Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041) _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Regards,
David Villasmil
email: david.villasmil.work@gmail.com mailto:david.villasmil.work@gmail.com
phone: +34669448337
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Thanks.
That helps a lot.
From: Daniel-Constantin Mierla miconda@gmail.com Date: Thursday, September 10, 2020 at 10:38 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org, Johnny Ritzer sudoritz@gmail.com Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
Hello,
wss is not usually on port 5060, it can be on port 5061 (if you haven't configured another tls socket specially for wss traffic). Anyhow, wss traffic is encrypted, so sngrep will not match it as sip traffic.
If you want to see locally the wss sip traffic, load sipdump module, it will write in text files (the version in master can save the traffic in pcap files as well, although it will appear as being udp traffic, because it was simpler to build the pcap headers, however you can sport from the headers/ports what was the trasport layer, or enable to add the extra meta-data sip header).
If you already have homer, then you should see the wss encrypted traffic there.
Cheers, Daniel On 11.09.20 01:41, Johnny Ritzer wrote: Hmmmm seeing that port 127.0.0.1….9060 looks like to capture and forward to homer (I know I installed it at one time to try to some triage).
Maybe homer built for additional checks. I just disabled the module hosting that port and I no longer see it in ngrep.
my path issue is wss:// webrtc over to uac.reg and FW it off to PBX. So ill try and figure out where in kama.cfg to hopefully get it to fw to pbx and make call.
From: Johnny Ritzer sudoritz@gmail.commailto:sudoritz@gmail.com Date: Thursday, September 10, 2020 at 4:32 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call) Ya I have other scenarios that show up like inbound call.
So I did this (made a test call) and saved output ngrep to pcap.
And open in wireshark and I see it as protocol UDP with SIP enclosed in data
ngrep -d any -qt -W byline port 5060 -O test.pcap
heres ngrep 2020/09/10 23:19:51.109093 10.111.0.4:5060 -> 127.0.0.1:9060 #1 .......[F... ...INVITE sip:8000@24.xx.xx.xx SIP/2.0. Via: SIP/2.0/WSS 4d8asfsdfag3f.invalid;branch=z9hG4bK2797686. Max-Forwards: 70. To: sip:8000@24.xx.xx.xxsip:8000@24.xx.xx.xx. From: "Ctx-WEBSIP-1138" sip:1138@24.xx.xx.xxsip:1138@24.xx.xx.xx;tag=jd85ifa0f2. Call-ID: 9pujji6vasdfasfrbn2. CSeq: 906 INVITE. Contact: sip:1138@24.xx.xx.xx;gr=urn:uuid:121211-a79d-4e76-a010-121211sip:1138@24.xx.xx.xx;gr=urn:uuid:121211-a79d-4e76-a010-121211. Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER. Supported: gruu, outbound. User-Agent: SIP.js/0.7.8. Content-Type: application/sdp. Content-Length: 2047.
From: sr-users sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org Date: Thursday, September 10, 2020 at 4:19 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call) You sure you’re using 5060 on kamailio?
On Fri, 11 Sep 2020 at 00:04, Johnny Ritzer <sudoritz@gmail.commailto:sudoritz@gmail.com> wrote:
No… or not that I know of
This is a temp GCP (as im building a POC to get this to a working concept.) trying to fine-tune config
From:
sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org>
Date: Thursday, September 10, 2020 at 3:33 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] sngrep nor tcpdump showing 5060 traffic but ngrep /kama logs show something ? (outbound call)
Are you using vlans? In the past I had issues with ngrep if the packets had vlan tags on them.
On Thu, Sep 10, 2020 at 6:10 PM Johnny Ritzer <sudoritz@gmail.commailto:sudoritz@gmail.com> wrote:
I can see kama logs./ngrep when I try to call out (WSS -> ) . But I cant see anything in sngrep nor tcpdump.
“ngrep -d any -qt -W byline port 5060”
Im not sure how/why or it could be something simple but what puzzles me is I know im sending data but its not captured.
Setup is pbx at 24.xx.xx.xx but ext is uac.reg to stay connected.
UA client = sip.js / wss
Inbound works(but 1 way audio ill fix after outbound issue)
Outbound doesn’t dial out (UA = webRTC client) using wss. Which led me to troubleshoot via sngrep/tcpdump.
But I seet his when it happens knowing I ame getting something sent.
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: INVITE from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: WARNING: <script>: ----RouteLogic--SIP request received on port 4443: from:1138 to:8040 srcip:70.xx.xx.xx
Sep 10 21:48:36 kamaProx /usr/sbin/kamailio[27480]: INFO: <script>: START: ACK from sip:1138@24.xx.xx.xx (IP: 70.xx.xx.xx:55041)
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