[OpenSIPS-Users] OpenSIPS2.4 Cluster Module Dialog Sync
Hi, I have to 2 openSIPS nodes setup in HA and one node is seed one. So whenever a backup node is rebooted it's unable to resync the dialog. It sends the dialog sync request tho. But it's discarding the receiving packets. Here are the logs. clusterer:send_sync_req: Sent sync request for capability 'dialog-dlg-repl' to node 1, cluster 1 Aug 23 17:45:10 openSIPS-1 /usr/local/sbin/opensips[20714]: INFO:clusterer:handle_sync_packet: discarding sync packet version 1, need version 2 Aug 23 17:45:10 openSIPS-1 /usr/local/sbin/opensips[20714]: INFO:clusterer:handle_sync_packet: discarding sync packet version 1, need version 2 Thanks for your help. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips/rtpporxy and early media
Hi Razvan
just one thing to be clear as I do not see anything in my debugging mode.
So after i get beck 180 ringing, must call:
rtpproxy_answer("rfo",,"2");
and
rtpproxy_stream2uas("/tmp/wav.wav", "-1"); (I did conversion and files
are like wav.wav.3, .0, .8)
Must I do something else? This is done on on_replay route. I guess I
must change to 183 session in progress?
thank you for help!
Miha
On 8/19/2019 8:57 PM, Miha via Users wrote:
Hi, Răzvan!
thank you, so i was thinking right :)
br
miha
On Mon, 19 Aug 2019 17:28:07 +0300
Răzvan Crainea wrote:
Hi, Miha!
You first need to convert the wav file you want to stream
to a RTP payload, one for each codec you support. To do
that, you can use the makeann tool that rtpproxy
provides[1].
Once you have those files (named file.3 for GSM, file.0
for PCMU. file.8 for PCMA), you need to call the
rtpproxy_stream2uac("file"). This will automatically do
the codec selection and choose the right file.
[1] https://github.com/sippy/rtpproxy/tree/master/makeann
Best regards,
Răzvan
On 8/19/19 4:07 PM, Miha via Users wrote:
Hello guys
first time doing this, normally I use freeswitch... Se
in combination with rtpproxy how to enable ringback tone.
I need to call rtpproxy_stream2() i add it as file? Or
there is some other option for this if I would like that
is played by UAS?
thank you for help!
miha
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http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] opensips-v2.4.6 - CRASH
Sorry, you mentioned 2.4 - the instructions are the same, but for 2.4
version (GIT and package) of course :)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
https://www.opensips.org/events/Summit-2019Amsterdam/
On 8/23/19 4:11 PM, Bogdan-Andrei Iancu wrote:
Hi Hamid,
This is an old known issue. Please update from latest 3.0 GIT branch
or use the nightly builds for packages.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
https://www.opensips.org/events/Summit-2019Amsterdam/
On 8/23/19 4:00 PM, Hamid Hashmi wrote:
If I create a dialog with in-dilaog pings like create_dialog("pPB") .
Service got crashed in local route.
(gdb) bt
#0 0x in ?? ()
#1 0x7ff20c965b3d in replicate_dialog_cseq_updated
(dlg=dlg@entry=0x7ff211d34af8, leg=leg@entry=0) at dlg_replication.c:858
#2 0x7ff20c90bd1b in send_leg_msg (dlg=dlg@entry=0x7ff211d34af8,
method=method@entry=0x7ff20cb7c650 , src_leg=out>, dst_leg=dst_leg@entry=0,
hdrs=hdrs@entry=0x0, body=body@entry=0x0,
func=func@entry=0x7ff20c901abc ,
param=param@entry=0x7ff211d34af8,
release=release@entry=0x7ff20c901674 ,
reply_marker=0x7ff211d7f35c "\001") at dlg_req_within.c:640
#3 0x7ff20c905579 in dlg_options_routine (ticks=,
attr=) at dlg_timer.c:958
#4 0x004ccdaf in handle_timer_job () at timer.c:730
#5 0x005ca36d in handle_io (idx=,
event_type=2, fm=0x7ff2139c6b10) at net/net_udp.c:265
#6 io_wait_loop_epoll (h=, t=,
repeat=) at net/../io_wait_loop.h:284
#7 udp_start_processes (chd_rank=chd_rank@entry=0x88b6d0
, startup_done=startup_done@entry=0x0) at
net/net_udp.c:389
#8 0x0041b543 in main_loop () at main.c:782
#9 main (argc=, argv=) at main.c:1439
Please find the debug logs in the attachment.
Regards
*Hamid R. Hashmi*
__
www.hrhashmi.blogspot.com
Mobile: +92 300 968 22 85 ; +92 322 636 32 66
Email: hamid.hashmi...@gmail.com
; _hamid2kviii@hotmail.com_
Sype: _hamidrhashmi_
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Re: [OpenSIPS-Users] opensips-v2.4.6 - CRASH
Hi Hamid,
This is an old known issue. Please update from latest 3.0 GIT branch or
use the nightly builds for packages.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
https://www.opensips.org/events/Summit-2019Amsterdam/
On 8/23/19 4:00 PM, Hamid Hashmi wrote:
If I create a dialog with in-dilaog pings like create_dialog("pPB") .
Service got crashed in local route.
(gdb) bt
#0 0x in ?? ()
#1 0x7ff20c965b3d in replicate_dialog_cseq_updated
(dlg=dlg@entry=0x7ff211d34af8, leg=leg@entry=0) at dlg_replication.c:858
#2 0x7ff20c90bd1b in send_leg_msg (dlg=dlg@entry=0x7ff211d34af8,
method=method@entry=0x7ff20cb7c650 , src_leg=out>, dst_leg=dst_leg@entry=0,
hdrs=hdrs@entry=0x0, body=body@entry=0x0,
func=func@entry=0x7ff20c901abc ,
param=param@entry=0x7ff211d34af8, release=release@entry=0x7ff20c901674
,
reply_marker=0x7ff211d7f35c "\001") at dlg_req_within.c:640
#3 0x7ff20c905579 in dlg_options_routine (ticks=,
attr=) at dlg_timer.c:958
#4 0x004ccdaf in handle_timer_job () at timer.c:730
#5 0x005ca36d in handle_io (idx=,
event_type=2, fm=0x7ff2139c6b10) at net/net_udp.c:265
#6 io_wait_loop_epoll (h=, t=,
repeat=) at net/../io_wait_loop.h:284
#7 udp_start_processes (chd_rank=chd_rank@entry=0x88b6d0
, startup_done=startup_done@entry=0x0) at
net/net_udp.c:389
#8 0x0041b543 in main_loop () at main.c:782
#9 main (argc=, argv=) at main.c:1439
Please find the debug logs in the attachment.
Regards
*Hamid R. Hashmi*
__
www.hrhashmi.blogspot.com
Mobile: +92 300 968 22 85 ; +92 322 636 32 66
Email: hamid.hashmi...@gmail.com
; _hamid2kviii@hotmail.com_
Sype: _hamidrhashmi_
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Re: [OpenSIPS-Users] OpenSIPS FreeSWITCH TCP support
Thanks for the update!
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
https://www.opensips.org/events/Summit-2019Amsterdam/
On 8/23/19 1:46 PM, Matthias Kneer via Users wrote:
Hi Bogdan,
thank you very much for your reply.
Do you do full TCP from UA to FS? or OpenSIPS converts from TCP (UA
side) to UDP (FS side) ?
Yes I do. In the meanwhile I was able to identify the issue.
I did a check in my openSIPS logic which checks if a request is
originating from one of the systems in my dispatcher list:
/ds_is_in_list("$si","$sp")/ I also did checked for the port at this
point, which is always 5060 for UDP. For TCP this is of course some
port in a high range - and also not listed in the dispatcher table.
After I removed the additional port check:
/ds_is_in_list("$si","")///everything started to work. FreeSWITCH
still lists the registration as UDP, with the TCP source port, but
since the traffic is properly flowing as TCP (also from the fs) that's
not something I care about.
Best regards,
Matthias
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[OpenSIPS-Users] opensips-v2.4.6 - CRASH
If I create a dialog with in-dilaog pings like create_dialog("pPB") . Service
got crashed in local route.
(gdb) bt
#0 0x in ?? ()
#1 0x7ff20c965b3d in replicate_dialog_cseq_updated
(dlg=dlg@entry=0x7ff211d34af8, leg=leg@entry=0) at dlg_replication.c:858
#2 0x7ff20c90bd1b in send_leg_msg (dlg=dlg@entry=0x7ff211d34af8,
method=method@entry=0x7ff20cb7c650 , src_leg=,
dst_leg=dst_leg@entry=0,
hdrs=hdrs@entry=0x0, body=body@entry=0x0, func=func@entry=0x7ff20c901abc
, param=param@entry=0x7ff211d34af8,
release=release@entry=0x7ff20c901674 ,
reply_marker=0x7ff211d7f35c "\001") at dlg_req_within.c:640
#3 0x7ff20c905579 in dlg_options_routine (ticks=,
attr=) at dlg_timer.c:958
#4 0x004ccdaf in handle_timer_job () at timer.c:730
#5 0x005ca36d in handle_io (idx=, event_type=2,
fm=0x7ff2139c6b10) at net/net_udp.c:265
#6 io_wait_loop_epoll (h=, t=, repeat=) at net/../io_wait_loop.h:284
#7 udp_start_processes (chd_rank=chd_rank@entry=0x88b6d0 ,
startup_done=startup_done@entry=0x0) at net/net_udp.c:389
#8 0x0041b543 in main_loop () at main.c:782
#9 main (argc=, argv=) at main.c:1439
Please find the debug logs in the attachment.
Regards
Hamid R. Hashmi
__
www.hrhashmi.blogspot.com
Mobile: +92 300 968 22 85 ; +92 322 636 32 66
Email: hamid.hashmi...@gmail.com;
hamid2kv...@hotmail.com
Sype: hamidrhashmi
Aug 23 12:45:18 sip-3 SafariOSIP-12[11703]: DBG:dialog:ref_dlg: ref dlg
0x7ff211d34af8 with 1 -> 4
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:dialog:send_leg_msg: sending
[OPTIONS] to caller (0)
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:dialog:ref_dlg: ref dlg
0x7ff211d34af8 with 1 -> 5
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:tm:t_uac:
next_hop=
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:mk_proxy: doing DNS
lookup...
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:tm:t_uac: sending socket is
10.71.3.12
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:tm:dlg2hash: 30869
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:tm:print_request_uri:
sip:923455908023@58.65.176.42:37166;transport=TLS;ob
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:tm:t_uac: building sip_msg from
buffer
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_msg: SIP Request:
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_msg: method:
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_msg: uri:
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_msg: version:
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_headers: flags=2
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_via_param: found
param type 232, = ; state=16
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_via: end of header
reached, state=5
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_headers: via found,
flags=2
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_headers: this is the
first via
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_headers:
flags=
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_to_param:
tag=869c10ef-2e19-4c8b-89de-b416d5546498
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:_parse_to: end of header
reached, state=29
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:_parse_to: display={},
ruri={sip:923455908...@ociqacall.waafi.com}
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:get_hdr_field: [81];
uri=[sip:923455908...@ociqacall.waafi.com]
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:get_hdr_field: to body
[]
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:get_hdr_field: cseq
: <1>
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:get_hdr_field:
content_length=0
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:get_hdr_field: found end
of header
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_headers: flags=78
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_to_param:
tag=3166b458-d24d-4e5b-8beb-dfb673170837
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:_parse_to: end of header
reached, state=29
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:_parse_to: display={},
ruri={sip:923345448...@ociqacall.waafi.com}
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: [923345448283 923455908023
982d846e-8f97-46db-941a-91dc43438d3f OPTIONS] LOCAL ROUTE OPTIONS
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_headers:
flags=
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:pv_get_contact_body: no
contact header!
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:destroy_index_avp: AVP
with the specified index not found
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:parse_headers:
flags=
Aug 23 12:45:19 sip-3 SafariOSIP-12[11703]: DBG:core:pv_get_contact_body: no
contact header!
Aug 23
Re: [OpenSIPS-Users] OpenSIPS FreeSWITCH TCP support
Hi Bogdan,
thank you very much for your reply.
Do you do full TCP from UA to FS? or OpenSIPS converts from TCP (UA
side) to UDP (FS side) ?
Yes I do. In the meanwhile I was able to identify the issue.
I did a check in my openSIPS logic which checks if a request is
originating from one of the systems in my dispatcher list:
/ds_is_in_list("$si","$sp")/ I also did checked for the port at this
point, which is always 5060 for UDP. For TCP this is of course some port
in a high range - and also not listed in the dispatcher table.
After I removed the additional port check:
/ds_is_in_list("$si","")///everything started to work. FreeSWITCH still
lists the registration as UDP, with the TCP source port, but since the
traffic is properly flowing as TCP (also from the fs) that's not
something I care about.
Best regards,
Matthias
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Re: [OpenSIPS-Users] OpenSIPS FreeSWITCH TCP support
Hi Matthias, Do you do full TCP from UA to FS? or OpenSIPS converts from TCP (UA side) to UDP (FS side) ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 7/22/19 7:26 PM, Matthias Kneer via Users wrote: Hello, I'm using OpenSIPS as a loadbalancer / failover proxy and mid registrar in AOR mode, also for doing the NAT keepalives, in front of 2 FreeSWITCHes. While using UDP everything works fine. Since I now want to implement TLS, I first started to test TCP to have the baseline working to go on with TLS afterwards. Unfortunately, while using TCP, none of TCP registered phones receives inbound calls - while outbound calls from these phones to other UDP registered phones work without an issue. The registration seems to work fine and also the information in the userlocation of OpenSIPS properly recognizing the registration over TCP: # opensipsctl ul show Domain:: location hash_size=512 AOR:: 2...@sip.example.org Contact:: sip:212@[EXTERNAL_PHONE_IP]:40009;transport=TCP;rinstance=ebc7149013d15489 Q= ContactID:: 1156589880790491998 Expires:: 18 Callid:: Y6nJ2r4Iwm6WNL_KHqcLOQ.. Cseq:: 2 User-agent:: Zoiper rv2.9.2 Received:: sip:[EXTERNAL_PHONE_IP]:40009;transport=tcp State:: CS_SYNC Flags:: 0 Cflags:: Socket:: tcp:[OPENSIPS_IP]:5060 Methods:: 5951 It looks like FreeSWITCH is not recognizing that the phones are registered through TCP even though the packets seem to contain the required information. Here's a dump of a the registration through TCP taken from the FreeSWITCH: https://pastebin.com/mbMffLK7 And here's the output of the active registrations on the FreeSWITCH for the above mentioned phone, which also contains the transport information in the contact header: Call-ID: Y6nJ2r4Iwm6WNL_KHqcLOQ.. User: 2...@sip.example.org Contact: "212 ZoIPer" Agent: Zoiper rv2.9.2 Status: Registered(UDP)(unknown) EXP(2019-07-22 18:02:32) EXPSECS(131) Ping-Status: Reachable Ping-Time: 0.00 Host: freeswitch IP: [OPENSIPS_IP] Port: 51540 Auth-User: 212 Auth-Realm: sip.example.org MWI-Account: 2...@sip.example.org Could someone guide me, at which point I do have to handle TCP different from UDP? I'm pretty new to this whole VoIP / SIP topic. Thanks in advance, Matthias ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] runtime changes of module parameters
Hi Mark, This feature was planned for 3.0, but it didn't make it - with limited resources, other features had higher priorities. See the end page of https://opensips.org/Development/Opensips-3-0-Planning Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 7/19/19 11:45 AM, Mark Farmer wrote: Good morning everyone I've been looking through the documentation pages for OpenSIPS 3.0 regarding module parameter changes during runtime but I don't seem to be able to find out how to actually do it. Have I missed something? Thanks Mark. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] is_contact_registered - not working as expected
Hi Peter,
Indeed, that's a good point - the `fix_nated_contact()` is changing the
contact (with the IP and port from the network level), messing up the
matching.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
https://www.opensips.org/events/Summit-2019Amsterdam/
On 8/22/19 10:16 PM, Peter Pulham wrote:
Hi,
A bit more testing and I realised that if I xlog the contact $ct it
printed the value of the contact in the INVITE:
Which looked fine and matched the usrloc table. However, if I xlog
$ct.field(uri) it printed a value with the public IP:
sip:5001@x.x.x.x:49710;ob
So it looks like I need to do my checking before I do any NAT contact
fixup.
This also explains why non NAT'd contacts were working.
Thanks for the help, set me on the right path to finally get to the issue.
Many thanks
On Thu, Aug 22, 2019 at 10:33 AM Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:
In this case, the function will perform a string matching between
the first valid contact URI from the message and the registered
contacts (the `Contact` field). Again, it is a string matching
over the entire SIP URI.
So far, based on your logs, I would say they should match. Could
you doublecheck again ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
https://www.opensips.org/events/Summit-2019Amsterdam/
On 8/22/19 12:07 PM, Peter Pulham wrote:
Hi,
I am calling is_contact_registered("location")
For the mi command, I used "mi ul_show_contact location
5...@domain.net " in opensips-cli
Many thanks
On Thu, Aug 22, 2019 at 10:00 AM Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:
Hi Peter,
How exactly do you invoke the is_contact_registered()
function (as params) for your script?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
https://www.opensips.org/events/Summit-2019Amsterdam/
On 8/21/19 10:34 PM, Peter Pulham wrote:
Hi,
Thanks for the input.
Adding the logging of $ct before call is_contact_registered
show this (also added a log for when it returned false):
Aug 21 19:24:52 ip-10-100-100-62
/usr/local/sbin/opensips[21114]: CONTACT OF REGISTER: "Peter
Pulham"
Aug 21 19:24:52 ip-10-100-100-62
/usr/local/sbin/opensips[21114]: IS CONTACT REGISTERED
RETURNED FALSE
The contact from the ul_show_contact command shows the same
contact:
{
"AOR": "5...@domain.net ",
"Contacts": [
{
"Contact": "sip:5001@y.y.y.y:49710;ob",
"ContactID": "1491535901589791905",
"Expires": 18,
"Q": "",
"Callid": "8..5AzWHdLmUQNXHdQhwDR3qMwOGPaWH",
"Cseq": 56286,
"User-agent": "Telephone 1.4",
"Received": "sip:x.x.x.x:49710",
"State": "CS_NEW",
"Flags": 0,
"Cflags": "NAT",
"Socket": "udp:z.z.z.z:5060",
"Methods": 8063
}
]
}
y.y.y.y is the same in the log as the mi output.
Through testing, I have realised that if the contact is not
NAT'd then the is_contact_registered returns true.
Any input appreciated.
On Wed, Aug 21, 2019 at 11:56 AM Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:
Hi Peter,
To doublecheck, do this:
* before calling the is_contact_registered(), use xlog()
to print the contact you are testing (use $ct).
* before running the test, use the ul_show_contact MI
function [1] to print the in-memory contacts for the
desired AOR.
[1]
https://opensips.org/html/docs/modules/2.4.x/usrloc.html#mi_ul_show_contact
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Summit 2019
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On 08/16/2019 06:38 PM, Peter Pulham wrote:
Hi,
I am using mid_registrar to save registrations to the
location table.
If I use is_registered - everything works as expected.
So AOR matching is working.
However, when I use is_contact_registered - it returns
false.
I have checked the contact header in the request
