Hi,
Other than using rtpproxy/NAThelper modules, is there any way to
bypass/workaround SIP ALG enabled on many WiFi routers? Although SIP ALG
was designed to help with NAT, in most cases it does the opposite and
breaks SIP.
Nabeel
___
Users mailing
, Patrick Wakano wrote:
> Using TLS!
> Also configuring your systems/devices to use other port than 5060 may do
> the trick...
>
> On Mon, May 2, 2016 at 9:14 AM, Nabeel wrote:
>
>> Hi,
>>
>> Other than using rtpproxy/NAThelper modules, is there any way to
>
A possible solution to this seems to be a 'SIP tunnel' server. The server
would tunnel the SIP and UDP packets over a common TCP port such as 80 or
443, which are more likely to be open and unblocked on Wi-Fi routers for
browsing, Email, etc. The tunnel server would then send this data to
OpenSIPS
Please check the following SIP trace taken within a WiFi network. The call
fails to connect despite the INVITE request and using a non-standard port.
Could this be caused by SIP ALG, or some unopened RTP port on the router?
http://pastebin.com/raw/C4iymTbh
_
Please read why UDP is better for VoIP:
https://www.onsip.com/blog/udp-versus-tcp-for-voip
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s point I am successfully using sip tls on port 443
>> without any issues as of yet. It's bypassing some isp enforced algs as
>> well as those enforced by local routers. :-).
>>
>>
>>
>>
>> On Thu, May 5, 2016 at 3:35 PM, Nabeel wrote:
>>
>>
see anything in the log.
On 5 May 2016 at 21:14, Tito Cumpen wrote:
> Nabeel,
>
>
> Did you verify that your opensips server is listening on this non Standard
> port ?
>
> run
>
> netstat -lnp | grep opensips
>
> On Thu, May 5, 2016 at 4:12 PM, Nabeel wrote:
>
ime. I
am using OpenSIPS version 2.1.
Nabeel
On 6 May 2016 9:16 am, "Bogdan-Andrei Iancu" wrote:
> Hi,
>
> Hard to analyze a call based on the INVITE packet only :). Still the SIP
> signaling does not show any ALG interference (also not sure if the capture
> was done befor
Founder and Developerhttp://www.opensips-solutions.com
>
> On 06.05.2016 12:56, Nabeel wrote:
>
> Hi,
>
> Thanks for the idea about packet compression. By 'call fails to connect',
> I meant the call does not connect to the callee, ie. the callee's phone
> does no
Can packet fragmentation be verified (to be sure that it is packet
fragmentation)?
On 6 May 2016 5:28 pm, "Nabeel" wrote:
> The trace I posted earlier is what I see with tcpdump when attempting a
> call. There is no other INVITE shown in the trace:
> http://pastebin.com/r
Hi,
I tried to use mc_compact but got this error:
CRITICAL:core:yyerror: parse error in config file
/usr/local//etc/opensips/opensips.cfg, line 201, column 12-13: unknown
command , missing loadmodule?
Then trying to load the module I got this error:
CRITICAL:core:yyerror: parse error in config
Do I have to rebuild my OpenSIPS installation completely just for this
module?
On 13 May 2016 12:16 am, "Ionut Ionita" wrote:
> Hi,
>
> Did you build the module? Is the module loaded into your script?
>
> Regards,
>
> Ionut Ionita
> OpenSIPS Developer
>
&
>
> The next question - is this INVITE reaching your opensips script ? to be
> sure that the OS delivers the UDP packet to the opensips application.
I don't have any firewall on my server. Why would the UDP packet get
blocked between entering the server and reaching opensips script? The
opensips
ey are never delivered at application level.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 17.05.2016 16:05, Nabeel wrote:
>
> The next question - is this INVITE reaching your opensips script ? to be
>>
OpenSIPS and
Asterisk instead? Have there been changes to database structure which can
cause problems?
Nabeel
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ARNING[17112] res_odbc.c: Connection is down attempting
> to reconnect...
>
Also I had to change 'nat=yes' to 'nat=force_rport,comedia' as it is
deprecated.
Nabeel
On 14 June 2016 at 11:08, Bogdan-Andrei Iancu wrote:
> Hi Nabeel,
>
> We will update
es_config_odbc.c: SQL Prepare
> failed![SELECT * FROM sipusers WHERE name = ? AND host = ?]
> [Jun 30 01:07:53] WARNING[17112] res_odbc.c: Connection is down attempting
> to reconnect.
I am using OpenSIPS 2.2, not 2.3 as stated earlier.
Nabeel
On 30 June 2016 at 10:18, Bogdan-Andrei Iancu wr
The tutorial contains a mistake where the priority ordering in
extensions.conf should start with 1, not n:
; Voicemail
> exten => _VMR_.,1,Ringing
> exten => _VMR_.,n,Wait(1)
> exten => _VMR_.,n,Answer
> exten => _VMR_.,n,Wait(1)
> exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
> exten => _VMR_.,n,Hang
D port = ?]
> [Jul 2 02:29:22] WARNING[18269][C-]: app.c:1633
> __ast_play_and_record: No audio available on SIP/domain.com-??
On 2 July 2016 at 02:23, Nabeel wrote:
> The tutorial contains a mistake where the priority ordering in
> extensions.conf should start with 1, not
app.c:1633
> __ast_play_and_record: No audio available on SIP/domain.com-0005??
On 2 July 2016 at 02:36, Nabeel wrote:
> The issue in my last Email has solved the error about missing extension.
> Now the following errors remain:
>
> [Jul 2 02:29:18] WARNING[18226][C-000
In the latest version of Asterisk, there is a new file voicemail.conf which
must be configured correctly for voicemail, but the tutorial does not
mention this file at all. Please let me know how to configure this file for
integration with OpenSIPS.
Nabeel
tabase error. Is
this the correct way to fix the error for voicemail integration?
2. How can voicemail.conf file be configured to use variable substitution
for all users in the OpenSIPS subscriber table?
Nabeel
On 2 July 2016 at 13:41, Nabeel wrote:
> In the latest version of Asterisk, ther
k
Realtime:
"This tutorial presents the concept and implementation of a realtime
integration of OpenSIPS SIP server and Asterisk media server."
I am a bit puzzled by your suggestion, but I will try asking in the
Asterisk mailing list.
Nabeel
__
ng correctly in my current setup.
Nabeel
On 4 Jul 2016 8:48 a.m., "Bogdan-Andrei Iancu" wrote:
> Hi,
>
> What is the definition you used for this new column ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.
uffix in addition to the phone
number, instead of just the phone number which is in the subscriber table.
Nabeel
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Now all phone numbers in the subscriber table are correctly linked to their
mailboxes and I can successfully retrieve voice messages.
I have set 'odbcstorage=asterisk' and 'odbctable=voicemessages' in
voicemail.conf, but the 'voicemessages&#x
Hi Bogdan,
I have been able to solve that problem.
The issue was that I had asterisk compiled with file storage enabled
instead of ODBC storage. I recompiled asterisk with ODBC storage enabled
and now database storage is working.
Thanks.
Nabeel
On 14 Jul 2016 11:15 a.m., "Bogdan-Andrei
I also found the correct way to deal with the LIMIT problem. Asterisk has a
built-in way to deal with this. In file* /etc/asterisk/res_odbc.conf*, the
following should be added under [asterisk] :
limit => 5
share_connections => no
Now everything is working well without problems.
arts from default opensips.cfg file because they
clash with the voicemail reply:
if (!db_does_uri_exist()) {
>send_reply("420","Bad Extension");
>exit;
> }
>
>t_newtran();
able to play a busy tone
on the caller's side. Please let me know how to fix this.
Nabeel
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x27;m not sure who exactly should generate the reply but I think it depends
on how OpenSIPS is interacting with the Voicemail server during a call to
voicemail. Perhaps it's not treating the voicemail service as another
'user' so the caller didn't receive the busy signal.
Nabee
Is there any way to make OpenSIPS handle a call with voicemail in exactly
the same way as a call with another SIP user?
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On 10 August 2016 at 19:38, Bogdan-Andrei Iancu wrote:
> Hi Nabeel,
>
> OpenSIPS does not assume anything by default. If you want to have any new
> calls to user A rejected (if A already in a call, with other users or any
> service), you need to script this.
>
In the case of
Merry Christmas to all.
Happy new year.
On 23 Dec 2016 4:11 p.m., "Bogdan-Andrei Iancu" wrote:
> Hi all,
>
> As we all move one by one into the holiday mode, we - the entire OpenSIPS
> team - want to wish you Merry Christmas and A Happy New 2017. Thanks to all
> of you 2016 was a good year with
Hi,
I run an ejabberd server for xmpp as well as OpenSIPS for SIP and was
wondering if these two could be integrated in any way in the future. These
servers run concurrently but they are completely independent of each other,
although for the same user base.
Nabeel
On 12 Jan 2017 4:48 p.m
Hi,
You can set client to use random port instead of standard 5060.
But a better way is to set the client to only allow your required domain,
if possible.
On 20 Apr 2017 9:51 p.m., "Uzair Hassan" wrote:
Hello all,
I have setup a opensips 2.3 on a new server and I'm getting ghost calls
into my
My understanding of ghost calls is that they go directly via the client
through a loophole in the IP range rather than through the SIP server
itself. In this case, server-based solutions don't seem likely to work?
On 20 Apr 2017 10:08 p.m., "Mundkowsky, Robert" wrote:
> User authentication at SI
Robert" wrote:
> Do you mean a client is using SIP/RTP to make a call direct to backend
> servers and bypassing the opensips proxy? Or somehow just using RTP without
> SIP to bypass the opensips proxy?
>
>
>
>
>
> Robert
>
>
>
> *From:* Users [mailto:users-b
In case the call is attempted via your server, you can add the following to
opensips.cfg to block sip scanners:
if($ua=~"friendly-scanner") {
xlog("L_ERROR", "Auth error for $fU@$fd from $si method $rm
user-agent (friendly-scanner)\n");
drop();
exit;
}
if($ua=~"sipv
Hi,
I am recently getting the following error, '437 Allocation Mismatch', in
multiple Turn servers I have tried. The same error occurs in Coturn and
reTurnServer on multiple physical server instances and when using all
versions of the client I am using, which never had this issue. The common
facto
cal/sbin/opensips -c -f
> /usr/local/etc/opensips/opensips.cfg
> Restart=always
> TimeoutStopSec=30s
> LimitNOFILE=1048576
> LimitNPROC=1048576
>
[Install]
> WantedBy=multi-user.target
Please advise how to fix this urgently.
Nabeel
___
Anyone? I am using OpenSIPS version 2.2.
On 10 Aug 2017 8:02 am, "Nabeel" wrote:
> I found the cause of this. My OpenSIPS server has gone ballistic and is
> running hundreds of processes/instances concurrently; please see this
> screenshot: https://www.dropbox.co
; it's a TURN CLIENT <--> TURN Server problem. It could be related to the
> TURN client implementation.
>
>
> BR
>
> Max M.
> --
> *Von:* Users im Auftrag von Nabeel <
> nabeelshik...@gmail.com>
> *Gesendet:* Donnerstag, 10. A
Congrats to OpenSIPS.
On 9 October 2017 at 12:05, Bogdan-Andrei Iancu wrote:
>
> We are all proud to announce that the OpenSIPS project is a winner of the
> Google Open Source Peer Bonus - this is an official recognition from Google
> in terms of the OSS they use.
>
> "We’re excited to announce
Hello,
Please help me understand this error below. Is it of concern?
The server is not bound to localhost, it is bound to public IPs.
> Oct 20 05:10:12 server2 /usr/local/sbin/opensips[923]:
> ERROR:core:proto_udp_send:
> sendto(sock,0x7f1f89c2bcf0,4,0,0x7ffc5f146db0,16): Invalid argument(22)
>
ersion and OS you are using?
>
> --
> regards,
>
> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445
>
> On 20 October 2017 at 09:22, Nabeel wrote:
>
>> Hello,
>>
>> Please help me understand this error below. Is it of concern?
>> The server is no
Hi,
Getting a lot of these errors lately which has caused OpenSIPS to crash:
*ERROR:registrar:update_contacts: invalid cseq for aor XXX*
I have added t_newtran() before save(location) in the config but the error
still seems to occur.
Nabeel
<https://www.avast.com/sig-email?utm_medium=em
Hi,
I'm trying to set up OpenSIPS with TLS support and connecting to my server
with an SIP client (Lumicall - http://lumicall.org/).
The settings in my opensips.cfg file are as follows:
listen=tls:87.xx.xxx.42:5061 as server0.domain.com:5061
>
> loadmodule "proto_tls.so"
> modparam("proto_tls"
del: [TCP_worker] io_watch_del op on index 0 16
> (0x8874c0, 16, 0, 0x10,0x3) fd_no=3 called
Jun 22 11:28:03 server0 /usr/local/sbin/opensips[1963]:
> DBG:core:tcpconn_release: releasing con 0x7f5cc27ce1a0, state -2, fd=-1,
> id=3
Jun 22 11:28:03 server0 /usr/local/sbin/opensips[196
accept: New TLS connection from 87.81.230.42:45098 failed
> to accept: rejected by client
Jun 22 11:28:03 server0 /usr/local/sbin/opensips[1963]:
> ERROR:proto_tls:tls_read_req: failed to do pre-tls reading
Jun 22 11:28:03 server0 /usr/local/sbin/opensips[1963]:
> DBG:core:io_watch_del: [T
In the OpenSIPS documentation, I see the following:
*SSLv23* - means OpenSIPS will accept any of the above methods, but the
> initial SSL hello must be v2 (in the initial hello all the supported
> protocols are advertised enabling switching to a higher and more secure
> version). The initial v2 he
I removed the following line from the config file and the handshake error
disappeared:
modparam("proto_tls", "ciphers_list", "NULL")
However, now I get the following error when connecting to the server with
the SIP client:
06-23 21:31:56.418 14512-18595/com.domain E/NativeCrypto﹕ ssl=0x5af0aca8
This is the full log is it using SSL version 2 which is disabled in
OpenSIPs?
In particular, this part:
"SIP/2.0 500 Server error occurred (7/TM)
Via: SIP/2.0/TLS"
06-23 21:45:39.790 14512-21632/com.domain I/org.zoolu.net.TcpSocket﹕
Initializing SSLContext for first use
06-23 21:45:39.8
How can I enable SSL version 2 on OpenSIPS?
On 23 Jun 2015 21:59, "Nabeel" wrote:
> This is the full log is it using SSL version 2 which is disabled in
> OpenSIPs?
> In particular, this part:
>
> "SIP/2.0 500 Server error occurred (7/TM)
> Via: SIP/2.0/T
.wikipedia.org/wiki/Transport_Layer_Security#SSL_1.0.2C_2.0_and_3.0
>
>
> On 23/06/2015 6:45 PM, Nabeel wrote:
> > How can I enable SSL version 2 on OpenSIPS?
> >
> > On 23 Jun 2015 21:59, "Nabeel" > <mailto:nabeelshik...@gmail.com>> wrote:
> >
> >
# opensipsctl tls rootCA
ERROR: root CA config file (/usr/local//etc/opensips//tls/ca.conf) does not
exist
In fact, that whole tls directory is empty, even though my OpenSIPS
instance has been compiled with tls support. Where can I download the CA
files?
___
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 24.06.2015 17:14, Nabeel wrote:
>
> # opensipsctl tls rootCA
> ERROR: root CA config file (/usr/local//etc/opensips//tls/ca.conf) does
> not exist
>
> In fact, that whole tls director
drei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 24.06.2015 17:14, Nabeel wrote:
>
> # opensipsctl tls rootCA
> ERROR: root CA config file (/usr/local//etc/opensips//tls/ca.conf) does
> not exist
>
> In fact, that whole tls director
Where are the 'example' openssl certificates as mentioned in the link
above? In the source files folder, there is no /etc/tls folder, and there
are no example certificates in the [source]/examples folder either.
On 25 June 2015 at 00:26, Nabeel wrote:
> I tried installing Open
I just installed version 1.11.5 of OpenSIPS and this version does have all
the TLS files included. I should have downloaded this version all along
because version 2.1 clearly needs to be fixed.
On 25 June 2015 at 00:36, Nabeel wrote:
> Where are the 'example' openssl certificates
All the TLS files seems to be in place. For 2.1 there is no specific
> switch for TLS, it is by default present, there is not need for extra
> options or env variables. Just to "make install"
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp
sips-solutions.com
>
> On 25.06.2015 14:05, Nabeel wrote:
>
> I did not download the sources from git; I downloaded directly from the
> OpenSIPS website from this link:
> http://opensips.org/pub/opensips/latest/src/
>
> If git is more reliable, the download links should peeh
Have you tried removing libperl5.18, then install libperl5.14, then try
installing opensips-perl-modules again:
http://packages.ubuntu.com/precise/libperl5.14
On 26 June 2015 at 21:11, Jeff Pyle wrote:
> Hello,
>
> I have been tasked with configuring a lightweight installation of OpenSIPS
> on
rsion mismatches:
>
> dpkg: dependency problems prevent configuration of libperl5.14:
> libperl5.14 depends on perl-base (= 5.14.2-6ubuntu2.4); however:
> Version of perl-base on system is 5.18.2-2ubuntu1.
>
>
> - Jeff
>
> On Fri, Jun 26, 2015 at 4:45 PM, Jeff Pyle
I am trying to connect to OpenSIPS via TLS, but getting this error "500
Server Error occurred (7/T/M)".
I suspect this is because OpenSIPS has SSL version 2 disabled by default
because I see the following in the log:
DBG:core:tcp_send: buf=#012SIP/2.0 500 Server error occurred
(7/TM)#015#012Via:
could be causing this?
On 1 Jul 2015 12:55, "Saúl Ibarra Corretgé" wrote:
>
> On 01 Jul 2015, at 13:36, Nabeel wrote:
>
> > I am trying to connect to OpenSIPS via TLS, but getting this error "500
> Server Error occurred (7/T/M)".
> >
> > I
Hi,
I get the following error when attempting to connect my SIP client to
OpenSIPS. I understand that OpenSIPS has accepted the connection
but then the client rejects the certificate sent by OpenSIPS. However, the
CA root certificate (from CAcert.org) is included in the client's trust
store, so
I'm running OpenSIPS with TLS support compiled and enabled. The first SRV
record OpenSIPS searches for when connecting is _sip._udp. (as shown
below).
It should be _sips._tcp. How can this be fixed?
DBG:core:get_record: lookup(_sip._udp.domain.com, 33) failed
connection
was "rejected
by client" in this case; it is more true that the connection was rejected
by OpenSIPS because the client did not provide a client certificate when
OpenSIPS was expecting one.
On 4 July 2015 at 05:51, Nabeel wrote:
> Hi,
>
> I get the following error
I believe the correct word would be 'refused' in that case, not 'rejected'
:)
On 5 July 2015 at 08:59, Podrigal, Aron wrote:
> Just a teaser. The client has rejected to provide a certificate as
> requested by opensips :)
> On Jul 5, 2015 3:37 AM, "Nabeel&quo
:
> Hi,
>
> What is the complete SIP URI that OpenSIPS tries to resolve ? What kind of
> query (NATPR , SRV, AAA) and the query string depends on the RURI you have.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
Hi,
How can I include the proto/port indications in the URI?
On 6 Jul 2015 18:14, "Bogdan-Andrei Iancu" wrote:
> Hi Nabeel,
>
> According to RFC3263 (Locating SIP Servers), if no proto/port indications
> are found in the URI, a NAPTR lookup should be tried (to determine
Hi,
I added a NAPTR record to my domain for TLS. OpenSIPS resolves this
correctly and does the SRV lookup for the correct port, but then fails to
register with the following error. Please advise how to fix.
DBG:core:tcp_read_req: last char=0x0A, parsed msg=#012REGISTER sip:
server.mydomain.com
.xx.xxx.42:42997 failed to
accept
ERROR:core:tls_accept: TLS error: (ret=-1, err=5, errno=32/Broken pipe):
On 7 July 2015 at 04:31, Nabeel wrote:
> Hi,
>
> I added a NAPTR record to my domain for TLS. OpenSIPS resolves this
> correctly and does the SRV lookup for the correct port, but t
Just tested by changing the listening port from TLS to TCP (5061 to 5060)
and registration occurs without any errors. This means something in the
TLS connection is failing: maybe the record route URI? How do I set this
anyway?
On 7 Jul 2015 05:16, "Nabeel" wrote:
> Also getting th
Hi,
>
> add a port:
> $rp = 5070
>
> add a transport param:
> add_uri_param("transport=tls");
> or set a outgoing/send socket with TLS
> force_send_socket("tls:ip:port");
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Develope
Hi,
I am able to register my SIP clients to OpenSIPS, but when I attempt to
make a call, the call does not connect to the other client. I see the call
screen open and close briefly on the dialling client (approx. 1 second) and
the following is seen in the log:
DBG:rr:find_first_route: No Route h
I have attached to this Email the SIP capture taken with TCPdump. Please
check it with Wireshark and let me know.
On 10 Jul 2015 08:28, "Podrigal, Aron" wrote:
> Can you provide a sip capture?
> On Jul 10, 2015 3:04 AM, "Nabeel" wrote:
>
>> Hi,
>>
&
I checked the SIP capture in Wireshark and it seems the client is using the
wrong port:
"Src port 3921 (3921) [Client IP]
Dst port 5061 (5061) [OpenSIPS IP]"
The client is set to use port 5061 in its settings and I have the following
in OpenSIPS config:
$rp = 5061
add_uri_param("transport=tls
In the log, I see that 'uri' does not have ';transport=tls', but the
'ruri' does have ';transport=tls' :
DBG:core:parse_msg: uri:
DBG:core:parse_to: display={}, ruri={sip:usern...@mydomain.com
;transport=tls}
On 11 July 2015 at 07:33, Nabeel
I have attached the output of "# ngrep -tqd any -W byline port 5061" as you
suggested.
Please let me know how I can add transport=tls to contact, because in my
config file I see nothing about contact header.
On 12 July 2015 at 05:10, Podrigal, Aron wrote:
> Hello Nabeel,
>
>
Well, that was the output of that command. How do I decrypt it?
On 12 Jul 2015 13:03, "Podrigal, Aron" wrote:
> Encrypted :)
> On Jul 12, 2015 3:17 AM, "Nabeel" wrote:
>
>> I have attached the output of "# ngrep -tqd any -W byline port 5061" as
>
I am currently running repro SIP server as backend for my mobile VoIP
application (https://www.resiprocate.org/About_Repro). Due to some bugs in
the TLS code of repro, I decided to switch to OpenSIPS, but unfortunately
having difficulty configuring it to work in the same way. If anyone has
experi
Hi,
I have a STUN/TURN server set up in my SIP clients which also support ICE.
However, OpenSIPS does not make use of this STUN/TURN server when
attempting to make a call.
How do I configure OpenSIPS to use the STUN/TURN server at a given port?
Which is better to use: RTPproxy or STUN/TURN serve
t; Tito
>
> On Thu, Jul 23, 2015 at 2:27 PM, Nabeel wrote:
>
>> Hi,
>>
>> I have a STUN/TURN server set up in my SIP clients which also support
>> ICE. However, OpenSIPS does not make use of this STUN/TURN server when
>> attempting to make a call.
&
Hi,
How do I fix this error without recompiling OpenSIPS from source? It
occurs when making some calls:
ERROR:core:append_branch: max nr of branches exceeded
ERROR:registrar:lookup: failed to append a branch
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I understand that Asterisk is a PBX but it also has core SIP
functionality. What are the disadvantages of using Asterisk over OpenSIPS
for a basic SIP service?
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Perhaps a better way to word my question: what are advantages of using
OpenSIPS over Asterisk for a basic SIP service?
On 25 Jul 2015 21:33, "Nabeel" wrote:
> I understand that Asterisk is a PBX but it also has core SIP
> functionality. What are the disadvantages of using Asteri
I am using rtpproxy with OpenSIPS, I can register to the server but I get
the following error when making a call:
ERROR:core:parse_via: bad port
ERROR:core:parse_via: #015#012From:
;tag=z9hG4bK07391365#015#012Call-ID:
638873549209@10.31.188.230#015#012CSeq: 1 CANCEL#015#012Contact:
#015#012Expire
Hi,
You are getting the message "self signed certificate in certificate chain"
because you haven't included your server's root certificate in the command,
with either -CApath or -CAfile option, for example add the following to the
command: -CApath /etc/ssl/certs
Then the response you receive shou
I just found that this error only occurs on mobile network connections, not
on Wi-Fi. So something in the mobile network IP
On 27 July 2015 at 18:42, Nabeel wrote:
> I am using rtpproxy with OpenSIPS, I can register to the server but I get
> the following error when making
I just found that this error only occurs on mobile network connections, not
on Wi-Fi. So something in the mobile network IP may be malforming the via
header?
ERROR:core:parse_via: bad port
ERROR:core:parse_via: #015#012From:
;tag=z9hG4bK88229229#015#012Call-ID:
598445986695@10.31.188.230#015#012
On 12 July 2015 at 13:38, Podrigal, Aron wrote:
>
> https://wiki.freeswitch.org/wiki/Packet_Capture#Analyze_a_packet_capture_with_SIP_TLS_on_port_5061
> On Jul 12, 2015 8:05 AM, "Nabeel" wrote:
>
>> Well, that was the output of that command. How do I decrypt it?
Hi,
I used 'make menuconfig' to create a script with NAT support and a few of
the other options like 'use_dbusrloc'.
When I check with 'opensips -c' I get a bunch of errors saying 'failed to
load module' (see examples below). I used the script as provided from the
script generator, with only IP/d
That is indeed correct, the script was pointing to the incorrect path with
'lib' instead of 'lib64'.
Fixing this removed the errors. Thanks.
On 5 Aug 2015 08:59, "Bogdan-Andrei Iancu" wrote:
> Hi Nabeel,
>
> You mentioned getting some "failed to
Hi,
I am using the residential script generated by 'make menuconfig', with UDP
and NAT support enabled. I added "alias=domain.com" to the config because
otherwise the UA did not register with my domain (usern...@domain.com). When
I attempt to make a call, I see '408 Request Timeout' in the sip tr
xcept
with the addition of "alias=domain.com". I have attached my config file at
this link:
http://pastebin.com/0QRyC938
On 6 August 2015 at 05:00, SamyGo wrote:
> Hi Nabeel,
> Quick question; what is this destination ip? 192.168.0.19:60912 ? -
> Destination
> User Age
g call to private IP and hence timeout
> occurs.
>
> Do print some log lines while doing uac_nat_test while registering. Also
> use opensipsctl command line tool to see details of online users..AFAIR its
> "opensipsctl ul show"
> See what flags are set for the callee.
t_timeout' options for TCP, but
please let me know if there are similar options for UDP.
On 6 Aug 2015 12:08, "Bogdan-Andrei Iancu" wrote:
> Nabeel,
>
> I suppose you OpenSIPS seats on a public IP, right ? The callee looks to
> have a private IP. And, at IP level, it is i
elper", "nortpproxy_str", "a=nortpproxy:yes\r\n")*
I'm not sure in exactly what combination works best, but perhaps these
should be included in the default residential script?
Thanks for the help... I'll be back with more questions.
On 6 August 2015 at 15:41, Bogdan-And
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