Re: [OpenSIPS-Users] OpenSIPS CP 9.3.2 password mode ha1_sha256 for adding new user

[jacky z]

Hi Team,

Does ha1_sha256 work in general opensips config settings? I have the
following in the scripts:

modparam("auth_db", "calculate_ha1", 0)

modparam("auth_db", "password_column", "ha1_sha256")


but got the following error in the log:


/usr/sbin/opensips[28261]: ERROR:auth:auth_calc_HA1: Incorrect length of
pre-hashed credentials for the algorithm "MD5": 32 expected, 64 provided


It seems though the sha256 was specified, but the server still calculated
MD5 and compared with the database column ha1_sha256.

On Tue, Aug 9, 2022 at 5:39 PM Bogdan-Andrei Iancu 
wrote:

> Hi Bela,
>
> The OCP does not support ha1_sha256 AFAIK. Consider opening a feature
> request here https://github.com/OpenSIPS/opensips-cp/issues
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
>   https://www.opensips.org/events/Summit-2022Athens/
>
> On 6/29/22 9:10 AM, Bela H wrote:
>
> Hi all,
>
>
>
> Is there any way to add new subscriber from OpenSIPS CP 9.3.2 using
> password mode ha1_sha256?
>
> The ha1 (MD5(username:realm:password)) works fine but I had no luck with
> the value generation for the ha1_sha256 field in “subscriber” table.
>
>
>
> I have this setting:
>
> modparam("auth_db", "calculate_ha1", 0)
>
> modparam("auth_db", "password_column", "ha1_sha256")
>
>
>
> Thanks!
>
> Bela
>
>
>
>
>
>
>
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>
>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

I will, thanks.


On Wed, Sep 14, 2022, 4:55 PM Daniel Zanutti 
wrote:

> Hi
>
> Everytime opensips sends the BYE, it's generated inside local_route:
> https://www.opensips.org/Documentation/Script-Routes-3-1#toc6
>
> So put a xlog there to see why. Something like this:
> local_route
> {
>   if (is_method("BYE"))
>   {
>   xlog("L_ERR", "LOCAL_ROUTE - BYE - $DLG_end_reason - $ru - $ci");
>   }
> }
>
>
> On Wed, Sep 14, 2022 at 5:04 PM Johan De Clercq  wrote:
>
>> Xlog(….);
>>
>> Outlook voor iOS  downloaden
>> --
>> *Van:* Users  namens Saint Michael <
>> vene...@gmail.com>
>> *Verzonden:* Wednesday, September 14, 2022 9:56:41 PM
>> *Aan:* OpenSIPS users mailling list 
>> *Onderwerp:* Re: [OpenSIPS-Users] The update from yesterday makes all
>> calls fail after 20 seconds, how do I go back?
>>
>> how do I do this:
>> " put some log on local_route"
>> Sorry I am learning
>>
>>
>> On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
>> wrote:
>>
>> So your Opensips is hanging up the call.
>>
>> Do you see any log on it? Try put some log on local_route if you don't
>> see anything.
>>
>>
>>
>> On Wed, Sep 14, 2022 at 4:40 PM Saint Michael  wrote:
>>
>> This is a trace showing a BYE from Opensips, but none of the sides did
>> actually hangup.
>>
>>
>> On Wed, Sep 14, 2022 at 3:33 PM Saint Michael  wrote:
>>
>> I use opensips 3.1, and I did an update yesterday. in all the boxes that
>> I upgraded all calls fail after 20 seconds.
>>
>> cd /usr/src/opensips-3.1/
>> git pull
>> make clean;make proper;make all
>> make modules
>> make install
>> clearlog.sh
>> systemctl restart opensips
>> opensips -V
>>
>>
>>
>> How do I go back?
>>
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
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>>
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
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>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Daniel Zanutti]

Hi

Everytime opensips sends the BYE, it's generated inside local_route:
https://www.opensips.org/Documentation/Script-Routes-3-1#toc6

So put a xlog there to see why. Something like this:
local_route
{
  if (is_method("BYE"))
  {
  xlog("L_ERR", "LOCAL_ROUTE - BYE - $DLG_end_reason - $ru - $ci");
  }
}


On Wed, Sep 14, 2022 at 5:04 PM Johan De Clercq  wrote:

> Xlog(….);
>
> Outlook voor iOS  downloaden
> --
> *Van:* Users  namens Saint Michael <
> vene...@gmail.com>
> *Verzonden:* Wednesday, September 14, 2022 9:56:41 PM
> *Aan:* OpenSIPS users mailling list 
> *Onderwerp:* Re: [OpenSIPS-Users] The update from yesterday makes all
> calls fail after 20 seconds, how do I go back?
>
> how do I do this:
> " put some log on local_route"
> Sorry I am learning
>
>
> On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
> wrote:
>
> So your Opensips is hanging up the call.
>
> Do you see any log on it? Try put some log on local_route if you don't see
> anything.
>
>
>
> On Wed, Sep 14, 2022 at 4:40 PM Saint Michael  wrote:
>
> This is a trace showing a BYE from Opensips, but none of the sides did
> actually hangup.
>
>
> On Wed, Sep 14, 2022 at 3:33 PM Saint Michael  wrote:
>
> I use opensips 3.1, and I did an update yesterday. in all the boxes that I
> upgraded all calls fail after 20 seconds.
>
> cd /usr/src/opensips-3.1/
> git pull
> make clean;make proper;make all
> make modules
> make install
> clearlog.sh
> systemctl restart opensips
> opensips -V
>
>
>
> How do I go back?
>
>
>
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Johan De Clercq]

Xlog(….);

Outlook voor iOS downloaden

Van: Users  namens Saint Michael 

Verzonden: Wednesday, September 14, 2022 9:56:41 PM
Aan: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] The update from yesterday makes all calls fail 
after 20 seconds, how do I go back?

how do I do this:
" put some log on local_route"
Sorry I am learning


On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
mailto:daniel.zanu...@gmail.com>> wrote:
So your Opensips is hanging up the call.

Do you see any log on it? Try put some log on local_route if you don't see 
anything.



On Wed, Sep 14, 2022 at 4:40 PM Saint Michael 
mailto:vene...@gmail.com>> wrote:
This is a trace showing a BYE from Opensips, but none of the sides did actually 
hangup.


On Wed, Sep 14, 2022 at 3:33 PM Saint Michael 
mailto:vene...@gmail.com>> wrote:
I use opensips 3.1, and I did an update yesterday. in all the boxes that I 
upgraded all calls fail after 20 seconds.
cd /usr/src/opensips-3.1/
git pull
make clean;make proper;make all
make modules
make install
clearlog.sh
systemctl restart opensips
opensips -V



How do I go back?


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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

how do I do this:
" put some log on local_route"
Sorry I am learning


On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
wrote:

> So your Opensips is hanging up the call.
>
> Do you see any log on it? Try put some log on local_route if you don't see
> anything.
>
>
>
> On Wed, Sep 14, 2022 at 4:40 PM Saint Michael  wrote:
>
>> This is a trace showing a BYE from Opensips, but none of the sides did
>> actually hangup.
>>
>>
>> On Wed, Sep 14, 2022 at 3:33 PM Saint Michael  wrote:
>>
>>> I use opensips 3.1, and I did an update yesterday. in all the boxes that
 I upgraded all calls fail after 20 seconds.
>>>
>>> cd /usr/src/opensips-3.1/
>>> git pull
>>> make clean;make proper;make all
>>> make modules
>>> make install
>>> clearlog.sh
>>> systemctl restart opensips
>>> opensips -V
>>>
>>>
>>>
>>> How do I go back?
>>>
>>>
>>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
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>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Daniel Zanutti]

So your Opensips is hanging up the call.

Do you see any log on it? Try put some log on local_route if you don't see
anything.



On Wed, Sep 14, 2022 at 4:40 PM Saint Michael  wrote:

> This is a trace showing a BYE from Opensips, but none of the sides did
> actually hangup.
>
>
> On Wed, Sep 14, 2022 at 3:33 PM Saint Michael  wrote:
>
>> I use opensips 3.1, and I did an update yesterday. in all the boxes that
>>> I upgraded all calls fail after 20 seconds.
>>
>> cd /usr/src/opensips-3.1/
>> git pull
>> make clean;make proper;make all
>> make modules
>> make install
>> clearlog.sh
>> systemctl restart opensips
>> opensips -V
>>
>>
>>
>> How do I go back?
>>
>>
>>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

This is a trace showing a BYE from Opensips, but none of the sides did
actually hangup.


On Wed, Sep 14, 2022 at 3:33 PM Saint Michael  wrote:

> I use opensips 3.1, and I did an update yesterday. in all the boxes that I
>> upgraded all calls fail after 20 seconds.
>
> cd /usr/src/opensips-3.1/
> git pull
> make clean;make proper;make all
> make modules
> make install
> clearlog.sh
> systemctl restart opensips
> opensips -V
>
>
>
> How do I go back?
>
>
>
INVITE sip:1957408@208.78.161.135:5060 SIP/2.0
Via: SIP/2.0/UDP 
38.95.11.9:5060;branch=z9hG4bK-524287-1---8d06285b2fbb097f;rport
Max-Forwards: 70
Contact: 
To: 
From: "7869440397";tag=fa388212
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, 
INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 282

v=0
o=3cxPS 9251273505243136 8956523271159809 IN IP4 38.95.11.9
s=3cxPS Audio call
c=IN IP4 38.95.11.9
t=0 0
m=audio 9092 RTP/AVP 0 8 9 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
SIP/2.0 100 Giving it a try
Via: SIP/2.0/UDP 
38.95.11.9:5060;received=38.95.11.9;branch=z9hG4bK-524287-1---8d06285b2fbb097f;rport=5060
To: 
From: "7869440397";tag=fa388212
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Server: OpenSIPS (3.1.11 (x86_64/linux))
Content-Length: 0

INVITE sip:1957408@216.82.236.252:5060 SIP/2.0
v: SIP/2.0/UDP 208.78.161.135:5060;branch=z9hG4bK99a6.74acd986.0
t: 1957408 
f: 15617596254;tag=fa388212
CSeq: 1 INVITE
i: KNCLJbZ-sV2tfAQT5Qv7Ig..
m: 
Max-Forwards: 69
c: application/sdp
l: 195
k: replaces, timer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, 
INFO, MESSAGE, UPDATE
Identity: 
eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9hei50YXgvMSJ9.eyJhdHRlc3QiOiJCIiwiZGVzdCI6eyJ0biI6WyIxOTU0NDQ0NzQwOCJdfSwiaWF0IjoxNjYzMTgzMzU0LCJvcmlnIjp7InRuIjoiMTU2MTc1OTYyNTQifSwib3JpZ2lkIjoiMzU4ZDA0OGItZTAwNy0xMWVhLWFjMzctMzgtOTUtMTEtOSJ9.E5pg-DMkY06sGwFTK5JniLaEBsRhEMS192Zb1OIbdAzN0tWopRxWNRWvBrLxECb5juZWtPSNu3m_VgCDvUYi9A;info=;alg=ES256;ppt=shaken

v=0
o=3cxPS 9251273505243136 8956523271159809 IN IP4 38.95.11.9
s=3cxPS Audio call
c=IN IP4 38.95.11.9
t=0 0
m=audio 9092 RTP/AVP 0 8 9 3 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
SIP/2.0 100 Trying
v: SIP/2.0/UDP 208.78.161.135:5060;branch=z9hG4bK99a6.74acd986.0
From: 15617596254;tag=fa388212
To: 1957408 ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Content-Length: 0

SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 208.78.161.135:5060;branch=z9hG4bK99a6.74acd986.0
From: 15617596254;tag=fa388212
To: 1957408 ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Contact: 
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length:   238
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 572791 493214 IN IP4 216.82.236.252
s=SIP Media Capabilities
c=IN IP4 216.82.236.27
t=0 0
m=audio 30648 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
38.95.11.9:5060;received=38.95.11.9;rport=5060;branch=z9hG4bK-524287-1---8d06285b2fbb097f
From: "7869440397";tag=fa388212
To: ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Contact: 
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length: 257
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 572791 493214 IN IP4 208.78.161.135
s=SIP Media Capabilities
c=IN IP4 208.78.161.135
t=0 0
m=audio 39112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=nortpproxy:yes
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 208.78.161.135:5060;branch=z9hG4bK99a6.74acd986.0
From: 15617596254;tag=fa388212
To: 1957408 ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Contact: 
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length:   238
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 572791 493214 IN IP4 216.82.236.252
s=SIP Media Capabilities
c=IN IP4 216.82.236.27
t=0 0
m=audio 30648 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
38.95.11.9:5060;received=38.95.11.9;rport=5060;branch=z9hG4bK-524287-1---8d06285b2fbb097f
From: "7869440397";tag=fa388212
To: ;tag=gK04c793f2
Call-ID: KNCLJbZ-sV2tfAQT5Qv7Ig..
CSeq: 1 INVITE
Contact: 
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length: 257
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 572791 493214 IN IP4 208.78.161.135
s=SIP Media Capabilities
c=IN IP4 208.78.161.135
t=0 0
m=audio 39112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

[OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Saint Michael]

>
> I use opensips 3.1, and I did an update yesterday. in all the boxes that I
> upgraded all calls fail after 20 seconds.

cd /usr/src/opensips-3.1/
git pull
make clean;make proper;make all
make modules
make install
clearlog.sh
systemctl restart opensips
opensips -V



How do I go back?
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Re: [OpenSIPS-Users] Autoscaler in 3.2.x

[Yury Kirsanov]

Hi Bogdan,
Thanks a lot for your help and support! The only question I know have is
why OpenSIPS was going into a crash if all TCP processes were blocked
waiting for connection? It was starting to consume more and more memory and
then it was crashing with a segfault upon reaching then -m memory
parameter. I do understand that TCP listeners were in a blocking mode and
were not able to do any work until the session could be fully established,
not being able to forward any SIP packets, but isn't that a bug that
OpenSIPS was starting to eat memory and then crash? Do I need to open a bug
report on this? Thanks!

Best regards,
Yury.

On Wed, Sep 14, 2022 at 10:58 PM Bogdan-Andrei Iancu 
wrote:

> Hi Yury,
>
> You need to check the TCP setting and to be sure your OpenSIPS will (1)
> not try to perform TCP connect against destination known not to be able to
> accept (like TCP/WS end points behind NAT) - see the tcp_no_new_conn_bflag
> [1] - or (2) not block for long time while attempting a connect - see the
> tcp_connect_timeout [2] or consider enabling async [3].
>
> [1]
> https://www.opensips.org/Documentation/Script-CoreParameters-3-2#tcp_no_new_conn_bflag
> [2]
> https://www.opensips.org/Documentation/Script-CoreParameters-3-2#tcp_connect_timeout
> [3] https://opensips.org/html/docs/modules/3.2.x/proto_tcp.html#idp168992
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
>   https://www.opensips.org/events/Summit-2022Athens/
>
> On 9/13/22 12:01 PM, Yury Kirsanov wrote:
>
> Hi Bogdan,
> Thanks for this update, but it looks like I can't check autoscaler because
> of this first issue with blocking TCP connect. Is there a way to resolve
> it? Am I doing something wrong? Or is that something to do with OpenSIPS
> code? As yes, you're right, as soon as I restart OpenSIPS having a lot of
> SIP devices trying to connect to it - it goes crazy, starts to consume
> memory and stops to forward packets sitting there at 100% load until it
> runs out of memory and segfaults. Sometimes I can't even restart it to come
> to normal state to make it work, it just loops into same crash whatever I
> try to do.
>
> I've compiled OpenSIPS 3.3.1 with your patch and was able to start it but
> not sure, maybe I was just lucky this time.
>
> What should I do? Thanks!
>
> Best regards,
> Yury.
>
> On Tue, 13 Sept 2022, 18:56 Bogdan-Andrei Iancu, 
> wrote:
>
>> Hi Yury,
>>
>> it looks like you some multiple issues, overlapping here. The traps you
>> sent here have nothing to do with the auto-scaling, but with a blocking TCP
>> connect for SIP - most of the procs get blocked into a sync TCP connect.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>   https://www.opensips.org/events/Summit-2022Athens/
>>
>> On 9/12/22 4:39 PM, Yury Kirsanov wrote:
>>
>> Hi Bogdan,
>> I've applied the patch (had to find where to apply it manually for 3.2.8
>> downloaded from Web page, line 1568 instead of 1652) and restarted the
>> server with only about 300-350 SIP devices and immediately got into same
>> issue. I'm attaching two GDB dumps made within several minutes from each
>> other. Autoscale was now OFF, please see my previous message as currently
>> for some reason I'm experiencing lockups even when it's off :(
>>
>>
>> Best regards,
>> Yury.
>>
>> On Mon, Sep 12, 2022 at 7:48 PM Bogdan-Andrei Iancu 
>> wrote:
>>
>>> Hi Yuri,
>>>
>>> Could you give this patch a try? it should fix the blocking you
>>> experience (it should apply on 3.2 too).
>>>
>>> Best regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>>   https://www.opensips.org/events/Summit-2022Athens/
>>>
>>> On 9/7/22 2:54 PM, Bogdan-Andrei Iancu wrote:
>>>
>>> Hi Yury,
>>>
>>> Thanks for the details info here - let me do a review of some code and
>>> run some tests, as at this point I have a good idea on the direction to dig
>>> into.
>>>
>>> I will update here.
>>>
>>> Best regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>>   https://www.opensips.org/events/Summit-2022Athens/
>>>
>>> On 9/6/22 11:24 AM, Yury Kirsanov wrote:
>>>
>>> Hi Bogdan,
>>> Yes, I'm listening on all types of sockets including UDP, TCP and TLS on
>>> the outside public interface and then forward traffic into internal LAN via
>>> UDP only.
>>>
>>> Previously it was getting stuck quite easily, now I had to wait for a
>>> while before this actually happened. I've routed part of my customers to
>>> this server to obtain this result so I will have to do that again.
>>>
>>> As soon as I see one of the processes stuck I'll dot the trap command
>>> and send you all the details including 

Re: [OpenSIPS-Users] Autoscaler in 3.2.x

[Yury Kirsanov]

Hi Bogdan,
Looks like my problem was quite complex as I had following issues:

1. tcp_async was off
2. TCP timeouts were set to be very high

I've tried to just enable tcp_async and that didn't help - after restart
and TCP SYN storm OpenSIPS started to consume memory and processes got
locked up again. Then I started to tune other parameters. Here's like it
was before:

# Proto TCP
loadmodule "proto_tcp.so"
modparam("proto_tcp", "tcp_async", 1)
modparam("proto_tcp", "tcp_send_timeout", 5000)
modparam("proto_tcp", "tcp_async_local_connect_timeout", 5000)
modparam("proto_tcp", "tcp_async_local_write_timeout", 5000)
modparam("proto_tcp", "tcp_max_msg_chunks", 16)

I had a very high tcp_send_timout because some of our customers are
connecting from across the globe and have high latency times, of course
that's not 5 seconds but I set it that high just to make sure they will be
able to connect. Now I ended up with this config:

# Proto TCP
loadmodule "proto_tcp.so"
modparam("proto_tcp", "tcp_async", 1)
modparam("proto_tcp", "tcp_send_timeout", 1000)
modparam("proto_tcp", "tcp_async_local_connect_timeout", 500)
modparam("proto_tcp", "tcp_async_local_write_timeout", 500)
modparam("proto_tcp", "tcp_max_msg_chunks", 16)
modparam("proto_tcp", "tcp_parallel_handling", 1)

And looks like OpenSIPS is now able to survive restarts!

One more thing I tried before was to rate-limit TCP connections on iptables
- that also helped even in my incorrect configuration and with blocking TCP
mode. I rate-limited TCP SYN packets on my public interface on TCP ports
that go to OpenSIPS using iptables rate-limit module with 10 packets per
second and 50 packets burst - that also seemed to help. This can be
adjusted as required depending on new connections load. Hope this helps
someone who would run into the same troubles!

I will continue monitoring our OpenSIPS instances and if everything works
fine after restart I will enable auto-scaler to test it with the new patch.

Thanks a lot for your help, Bogdan, that's much appreciated!

Best regards,
Yury.

On Thu, Sep 15, 2022 at 1:22 AM Yury Kirsanov  wrote:

> Hi Bogdan,
> Thanks for your answer, I've checked my configs and yes, for some reason I
> had tcp_async off!!! I will definitely switch it on for now and then give
> it a try!!! Can't believe I missed that one!!!
>
> Best regards,
> Yury.
>
> On Wed, Sep 14, 2022 at 10:58 PM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Yury,
>>
>> You need to check the TCP setting and to be sure your OpenSIPS will (1)
>> not try to perform TCP connect against destination known not to be able to
>> accept (like TCP/WS end points behind NAT) - see the tcp_no_new_conn_bflag
>> [1] - or (2) not block for long time while attempting a connect - see the
>> tcp_connect_timeout [2] or consider enabling async [3].
>>
>> [1]
>> https://www.opensips.org/Documentation/Script-CoreParameters-3-2#tcp_no_new_conn_bflag
>> [2]
>> https://www.opensips.org/Documentation/Script-CoreParameters-3-2#tcp_connect_timeout
>> [3] https://opensips.org/html/docs/modules/3.2.x/proto_tcp.html#idp168992
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>   https://www.opensips.org/events/Summit-2022Athens/
>>
>> On 9/13/22 12:01 PM, Yury Kirsanov wrote:
>>
>> Hi Bogdan,
>> Thanks for this update, but it looks like I can't check autoscaler
>> because of this first issue with blocking TCP connect. Is there a way to
>> resolve it? Am I doing something wrong? Or is that something to do with
>> OpenSIPS code? As yes, you're right, as soon as I restart OpenSIPS having a
>> lot of SIP devices trying to connect to it - it goes crazy, starts to
>> consume memory and stops to forward packets sitting there at 100% load
>> until it runs out of memory and segfaults. Sometimes I can't even restart
>> it to come to normal state to make it work, it just loops into same crash
>> whatever I try to do.
>>
>> I've compiled OpenSIPS 3.3.1 with your patch and was able to start it but
>> not sure, maybe I was just lucky this time.
>>
>> What should I do? Thanks!
>>
>> Best regards,
>> Yury.
>>
>> On Tue, 13 Sept 2022, 18:56 Bogdan-Andrei Iancu, 
>> wrote:
>>
>>> Hi Yury,
>>>
>>> it looks like you some multiple issues, overlapping here. The traps you
>>> sent here have nothing to do with the auto-scaling, but with a blocking TCP
>>> connect for SIP - most of the procs get blocked into a sync TCP connect.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>>   https://www.opensips.org/events/Summit-2022Athens/
>>>
>>> On 9/12/22 4:39 PM, Yury Kirsanov wrote:
>>>
>>> Hi Bogdan,
>>> I've applied the patch (had to find where to apply it manually for 3.2.8
>>> downloaded from Web page, line 1568 instead of 1652) and restarted the
>>> server with only about 300-350 SIP devices and 

Re: [OpenSIPS-Users] Autoscaler in 3.2.x

[Yury Kirsanov]

Hi Bogdan,
Thanks for your answer, I've checked my configs and yes, for some reason I
had tcp_async off!!! I will definitely switch it on for now and then give
it a try!!! Can't believe I missed that one!!!

Best regards,
Yury.

On Wed, Sep 14, 2022 at 10:58 PM Bogdan-Andrei Iancu 
wrote:

> Hi Yury,
>
> You need to check the TCP setting and to be sure your OpenSIPS will (1)
> not try to perform TCP connect against destination known not to be able to
> accept (like TCP/WS end points behind NAT) - see the tcp_no_new_conn_bflag
> [1] - or (2) not block for long time while attempting a connect - see the
> tcp_connect_timeout [2] or consider enabling async [3].
>
> [1]
> https://www.opensips.org/Documentation/Script-CoreParameters-3-2#tcp_no_new_conn_bflag
> [2]
> https://www.opensips.org/Documentation/Script-CoreParameters-3-2#tcp_connect_timeout
> [3] https://opensips.org/html/docs/modules/3.2.x/proto_tcp.html#idp168992
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
>   https://www.opensips.org/events/Summit-2022Athens/
>
> On 9/13/22 12:01 PM, Yury Kirsanov wrote:
>
> Hi Bogdan,
> Thanks for this update, but it looks like I can't check autoscaler because
> of this first issue with blocking TCP connect. Is there a way to resolve
> it? Am I doing something wrong? Or is that something to do with OpenSIPS
> code? As yes, you're right, as soon as I restart OpenSIPS having a lot of
> SIP devices trying to connect to it - it goes crazy, starts to consume
> memory and stops to forward packets sitting there at 100% load until it
> runs out of memory and segfaults. Sometimes I can't even restart it to come
> to normal state to make it work, it just loops into same crash whatever I
> try to do.
>
> I've compiled OpenSIPS 3.3.1 with your patch and was able to start it but
> not sure, maybe I was just lucky this time.
>
> What should I do? Thanks!
>
> Best regards,
> Yury.
>
> On Tue, 13 Sept 2022, 18:56 Bogdan-Andrei Iancu, 
> wrote:
>
>> Hi Yury,
>>
>> it looks like you some multiple issues, overlapping here. The traps you
>> sent here have nothing to do with the auto-scaling, but with a blocking TCP
>> connect for SIP - most of the procs get blocked into a sync TCP connect.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>   https://www.opensips.org/events/Summit-2022Athens/
>>
>> On 9/12/22 4:39 PM, Yury Kirsanov wrote:
>>
>> Hi Bogdan,
>> I've applied the patch (had to find where to apply it manually for 3.2.8
>> downloaded from Web page, line 1568 instead of 1652) and restarted the
>> server with only about 300-350 SIP devices and immediately got into same
>> issue. I'm attaching two GDB dumps made within several minutes from each
>> other. Autoscale was now OFF, please see my previous message as currently
>> for some reason I'm experiencing lockups even when it's off :(
>>
>>
>> Best regards,
>> Yury.
>>
>> On Mon, Sep 12, 2022 at 7:48 PM Bogdan-Andrei Iancu 
>> wrote:
>>
>>> Hi Yuri,
>>>
>>> Could you give this patch a try? it should fix the blocking you
>>> experience (it should apply on 3.2 too).
>>>
>>> Best regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>>   https://www.opensips.org/events/Summit-2022Athens/
>>>
>>> On 9/7/22 2:54 PM, Bogdan-Andrei Iancu wrote:
>>>
>>> Hi Yury,
>>>
>>> Thanks for the details info here - let me do a review of some code and
>>> run some tests, as at this point I have a good idea on the direction to dig
>>> into.
>>>
>>> I will update here.
>>>
>>> Best regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS Summit 27-30 Sept 2022, Athens
>>>   https://www.opensips.org/events/Summit-2022Athens/
>>>
>>> On 9/6/22 11:24 AM, Yury Kirsanov wrote:
>>>
>>> Hi Bogdan,
>>> Yes, I'm listening on all types of sockets including UDP, TCP and TLS on
>>> the outside public interface and then forward traffic into internal LAN via
>>> UDP only.
>>>
>>> Previously it was getting stuck quite easily, now I had to wait for a
>>> while before this actually happened. I've routed part of my customers to
>>> this server to obtain this result so I will have to do that again.
>>>
>>> As soon as I see one of the processes stuck I'll dot the trap command
>>> and send you all the details including processes load, ps output and so on.
>>>
>>> For now I had to switch autoscaling off and just create many listeners.
>>> Do I understand correctly that I need to restart OpenSIPS in order to apply
>>> autoscaling profiles and reload-routes is not sufficient?
>>>
>>> Also, do I need separate UDP profiles for public and private interfaces?
>>> And do I need to apply autoscaling profile just to a socket or I need 

Re: [OpenSIPS-Users] How can I retrieve the callID and start time for a call inside event_route[E_RTPPROXY_DTMF] {

[Bogdan-Andrei Iancu]

There are several options here, unfortunately none really simple. Here 
is one:


a) after creating the dialog (at initial INVITE time), save the Call-id 
into a $dlg_val [1]

    $dlg_val(my_callid) = $ci;

b) in event route, use load_dialog_ctx() [2] - do NOT forget to pair it 
with the unload function

    load_dialog_ctx("$param(id)", "did");
    $var(callid) = $dlg_val(my_callid);
load_dialog_ctx();
    
    # you have the callid into $var(callid) for further processing

[1] https://opensips.org/html/docs/modules/3.2.x/dialog.html#pv_dlg_val
[2] 
https://opensips.org/html/docs/modules/3.2.x/dialog.html#func_load_dialog_ctx



Another option may be the usage of local cache to store the SIP callid 
under the dialog ID as key, at INVITE time:

https://opensips.org/html/docs/modules/3.2.x/dialog.html#pv_DLG_did
https://www.opensips.org/Documentation/Script-CoreFunctions-3-2#cache_store
And in event route simply use the cache_fetch() with the dialog ID as 
key also.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
  https://www.opensips.org/events/Summit-2022Athens/

On 9/14/22 5:31 PM, Saint Michael wrote:

Is there way to obtain the SIP Call ID from the "id" or when is_callid=0?
if I save the callID on a global variable, will it be visible 
around all the code?

Sorry I am still learning the ropes.




On Wed, Sep 14, 2022 at 9:38 AM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi,

As per docs [1], you have in event_route :

 id - represents the identifier of the call for which that event
was received.

 is_callid - is 0 if the id parameter represents the Dialog ID, or
1 if it is a callid.

[1]

https://opensips.org/html/docs/modules/3.2.x/rtpproxy.html#event_E_RTPPROXY_DTMF



Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
OpenSIPS Summit 27-30 Sept 2022, Athens
   https://www.opensips.org/events/Summit-2022Athens/  


On 9/14/22 10:46 AM, Saint Michael wrote:


My goal is to close the call as soon as the callee presses
any DTMF

when any DTMF is detected,  then I need to access the callID
variable, $ci
but it's nowhere to be found
Also I need $avp(start_time) and $avp(duration)
I am at a loss as to how to retrieve this information. The call
is connected
I need to fire a database query to close the call.
I imagine that when the call got connected, some variable stored
that unix-time .
But it does not seem to be available in this part of the code.
Philip







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Re: [OpenSIPS-Users] How can I retrieve the callID and start time for a call inside event_route[E_RTPPROXY_DTMF] {

[Saint Michael]

Is there way to obtain the SIP Call ID from the "id" or when is_callid=0?
if I save the callID on a global variable, will it be visible around all
the code?
Sorry I am still learning the ropes.




On Wed, Sep 14, 2022 at 9:38 AM Bogdan-Andrei Iancu 
wrote:

> Hi,
>
> As per docs [1], you have in event_route :
>
>  id - represents the identifier of the call for which that event was
> received.
>
>  is_callid - is 0 if the id parameter represents the Dialog ID, or 1 if it
> is a callid.
>
> [1]
> https://opensips.org/html/docs/modules/3.2.x/rtpproxy.html#event_E_RTPPROXY_DTMF
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
>   https://www.opensips.org/events/Summit-2022Athens/
>
> On 9/14/22 10:46 AM, Saint Michael wrote:
>
> My goal is to close the call as soon as the callee presses any DTMF
>
> when any DTMF is detected,  then I need to access the callID variable,
> $ci
> but it's nowhere to be found
> Also I need $avp(start_time) and $avp(duration)
> I am at a loss as to how to retrieve this information. The call is
> connected
> I need to fire a database query to close the call.
> I imagine that when the call got connected, some variable stored that
> unix-time .
> But it does not seem to be available in this part of the code.
> Philip
>
>
>
>
>
>
>>
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>
>
>
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Re: [OpenSIPS-Users] How can I retrieve the callID and start time for a call inside event_route[E_RTPPROXY_DTMF] {

[Bogdan-Andrei Iancu]

Hi,

As per docs [1], you have in event_route :

 id - represents the identifier of the call for which that event was 
received.


 is_callid - is 0 if the id parameter represents the Dialog ID, or 1 if 
it is a callid.


[1] 
https://opensips.org/html/docs/modules/3.2.x/rtpproxy.html#event_E_RTPPROXY_DTMF


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
  https://www.opensips.org/events/Summit-2022Athens/

On 9/14/22 10:46 AM, Saint Michael wrote:


My goal is to close the call as soon as the callee presses any DTMF

when any DTMF is detected,  then I need to access the callID variable, $ci
but it's nowhere to be found
Also I need $avp(start_time) and $avp(duration)
I am at a loss as to how to retrieve this information. The call is 
connected

I need to fire a database query to close the call.
I imagine that when the call got connected, some variable stored that 
unix-time .

But it does not seem to be available in this part of the code.
Philip







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Re: [OpenSIPS-Users] Connect to AWS RDS database with SSL enabled

[Vlad Patrascu]

Hi Jacky,

OpenSIPS will always require you to configure a client certificate for 
TLS client domains and will also present that certificate when 
connecting. But normally, a TLS server can simply choose not to verify 
the client certificate. I don't have any experience with AWS RDS though 
but it seems odd to not accept a connection only because the client did 
present a certificate.


Regards,

--
Vlad Patrascu
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 14.09.2022 05:42, jacky z wrote:

Hi Bogdan-Andrei,

I checked the mariadb documentation and found mariadb has two options 
to set ssl connection: two-way TSL and one-way TSL. It seems AWS RDS 
only supports one-way TSL, that is, TSL is used without a client cert. 
Does OPENSIPS support such one-way TSL to connect a database? Thanks!


On Wed, Sep 14, 2022 at 12:06 AM jacky z  wrote:

Hi Bogdan-Andrei,

I have set the "certificate" and "private_key" in my script, as I
explained in method 1. However, AWS RDS doesn't support a client
cert. Please refer to

https://stackoverflow.com/questions/53760104/how-to-configure-x509-client-certificate-based-authentication-to-connect-to-aws

Is there any workaround to use the public cert list provided by
AWS? Anyone has successfully used RDS with SSL connections? Thanks!

On Tue, Sep 13, 2022 at 9:54 PM Bogdan-Andrei Iancu
 wrote:

Set the certificate and key you have in the tls_mgm module,
for the "certificate" and "private_key" parameters.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
   https://www.opensips.org/events/Summit-2022Athens/

On 9/13/22 2:57 PM, jacky z wrote:

Hi Bogdan-Andrei,

I tried two methods.

Method 1:

#enabled TLS connection:
modparam("db_mysql", "use_tls", 1)

#setup a client domain:
modparam("tls_mgm", "client_domain", "dom1")
modparam("tls_mgm", "match_ip_address", "[dom1]*")
modparam("tls_mgm", "match_sip_domain", "[dom1]*")
modparam("tls_mgm","certificate",
"[dom1]/etc/ssl/certs/rootCACert.pem")
modparam("tls_mgm","private_key",
"[dom1]/etc/ssl/private/rootCAKey.pem")
modparam("tls_mgm","ca_list",
"[dom1]/etc/ssl/certs/rootCACert.pem")
modparam("tls_mgm","tls_method", "[dom1]SSLv23")
modparam("tls_mgm","verify_cert", "[dom1]0")
modparam("tls_mgm","require_cert", "[dom1]0")
# set db_url
modparam("usrloc", "db_url",
"mysql://root:1234@/opensips?tls_domain=dom1")
...

I couldn't figure out how to use global-bundle.pem AWS
provided with this method. No luck to get a connection with
RDS. If I don't use ssl, opensips can connect to RDS without
encryption.

Method 2:

I tried

modparam("usrloc", "db_url",

"mysql://root:1234@/opensips?ssl=true_ca_certs=/etc/ssl/certs/global-bundle.pem")

to include the AWS cert. Still no luck.

Thanks!

On Tue, Sep 13, 2022 at 4:52 PM Bogdan-Andrei Iancu
 wrote:

Hi,

sorry for my silly question, but how do you connect from
the OpenSIPS side ??

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
   https://www.opensips.org/events/Summit-2022Athens/

On 9/13/22 10:41 AM, jacky z wrote:

Hi Team,

We hope to connect to aws RDS database with ssl
encryption. We have setup a client domain according to
OPENSIPS documents. However, AWS RDS does not support
client cert as someone has confirmed with AWS

https://stackoverflow.com/questions/53760104/how-to-configure-x509-client-certificate-based-authentication-to-connect-to-aws

Is there any way to use the cert provided by AWS to
connect? AWS provides a global-bundle.pem

(https://docs.aws.amazon.com/AmazonRDS/latest/UserGuide/UsingWithRDS.SSL.html)
for such a connection, but we don't know how to include
it in the config file.

Thanks

Jacky z

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Re: [OpenSIPS-Users] Autoscaler in 3.2.x

[Bogdan-Andrei Iancu]

Hi Yury,

You need to check the TCP setting and to be sure your OpenSIPS will (1) 
not try to perform TCP connect against destination known not to be able 
to accept (like TCP/WS end points behind NAT) - see the 
tcp_no_new_conn_bflag [1] - or (2) not block for long time while 
attempting a connect - see the tcp_connect_timeout [2] or consider 
enabling async [3].


[1] 
https://www.opensips.org/Documentation/Script-CoreParameters-3-2#tcp_no_new_conn_bflag
[2] 
https://www.opensips.org/Documentation/Script-CoreParameters-3-2#tcp_connect_timeout

[3] https://opensips.org/html/docs/modules/3.2.x/proto_tcp.html#idp168992

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
  https://www.opensips.org/events/Summit-2022Athens/

On 9/13/22 12:01 PM, Yury Kirsanov wrote:

Hi Bogdan,
Thanks for this update, but it looks like I can't check autoscaler 
because of this first issue with blocking TCP connect. Is there a way 
to resolve it? Am I doing something wrong? Or is that something to do 
with OpenSIPS code? As yes, you're right, as soon as I restart 
OpenSIPS having a lot of SIP devices trying to connect to it - it goes 
crazy, starts to consume memory and stops to forward packets sitting 
there at 100% load until it runs out of memory and segfaults. 
Sometimes I can't even restart it to come to normal state to make it 
work, it just loops into same crash whatever I try to do.


I've compiled OpenSIPS 3.3.1 with your patch and was able to start it 
but not sure, maybe I was just lucky this time.


What should I do? Thanks!

Best regards,
Yury.

On Tue, 13 Sept 2022, 18:56 Bogdan-Andrei Iancu, > wrote:


Hi Yury,

it looks like you some multiple issues, overlapping here. The
traps you sent here have nothing to do with the auto-scaling, but
with a blocking TCP connect for SIP - most of the procs get
blocked into a sync TCP connect.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
OpenSIPS Summit 27-30 Sept 2022, Athens
   https://www.opensips.org/events/Summit-2022Athens/  


On 9/12/22 4:39 PM, Yury Kirsanov wrote:

Hi Bogdan,
I've applied the patch (had to find where to apply it manually
for 3.2.8 downloaded from Web page, line 1568 instead of 1652)
and restarted the server with only about 300-350 SIP devices and
immediately got into same issue. I'm attaching two GDB dumps made
within several minutes from each other. Autoscale was now OFF,
please see my previous message as currently for some reason I'm
experiencing lockups even when it's off :(



Best regards,
Yury.

On Mon, Sep 12, 2022 at 7:48 PM Bogdan-Andrei Iancu
mailto:bog...@opensips.org>> wrote:

Hi Yuri,

Could you give this patch a try? it should fix the blocking
you experience (it should apply on 3.2 too).

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  

OpenSIPS Summit 27-30 Sept 2022, Athens
   https://www.opensips.org/events/Summit-2022Athens/  


On 9/7/22 2:54 PM, Bogdan-Andrei Iancu wrote:

Hi Yury,

Thanks for the details info here - let me do a review of
some code and run some tests, as at this point I have a good
idea on the direction to dig into.

I will update here.

Best regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  

OpenSIPS Summit 27-30 Sept 2022, Athens
   https://www.opensips.org/events/Summit-2022Athens/  

On 9/6/22 11:24 AM, Yury Kirsanov wrote:

Hi Bogdan,
Yes, I'm listening on all types of sockets including UDP,
TCP and TLS on the outside public interface and then
forward traffic into internal LAN via UDP only.

Previously it was getting stuck quite easily, now I had to
wait for a while before this actually happened. I've routed
part of my customers to this server to obtain this result
so I will have to do that again.

As soon as I see one of the processes stuck I'll dot the
trap command and send you all the details including
processes load, ps output and so on.

For now I had to switch autoscaling off and just create
many listeners. Do I understand correctly that I need to
restart OpenSIPS in order to apply autoscaling profiles and
reload-routes is not sufficient?

Also, do I need separate UDP 

Re: [OpenSIPS-Users] Opensips CP Permissions "RELOAD on SERVER" produces error

[mtck01]

Thank you so much and my apologies.

 

 

Regards,

Martin

 

From: Bogdan-Andrei Iancu  
Sent: Tuesday, September 13, 2022 8:49 AM
To: mtck01 ; users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips CP Permissions "RELOAD on SERVER"
produces error

 

Be sure you have in the opensips.cfg file the 
loadmodule "permissions.so"
line

PS: and please post to the list, not privately ;)

Regards,



Bogdan-Andrei Iancu
 
OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 27-30 Sept 2022, Athens
  https://www.opensips.org/events/Summit-2022Athens/

On 9/13/22 3:31 PM, mtck01 wrote:

Hello, I am very new to OpenSIPS and have the same error.   I looked
everywhere but am still confused.  I installed all the modules but how do I
'load' the permissions module..?

 

Regards,

Martin

 

 

 

 

 
 Bogdan-Andrei Iancu
 Wed, 21 Jul 2021 23:59:41 -0700

Hi Jeff,
 

Are you sure the OpenSIPS (your CP is connected to ) has the permissions
module loaded ?

 
Regards,
 
Bogdan-Andrei Iancu
 
OpenSIPS Founder and Developer
    https://www.opensips-solutions.com
OpenSIPS Bootcamp 2021 online
   
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
 
On 7/21/21 4:17 PM, Jeff Wilkie wrote:
Opensips 3.1.2
CP 8.3.1
Debian 10
 

When adding permissions, I hit "Reload on Server" but I get the following
error:

 

Sending to*json:127.0.0.1:/mi < 
http://127.0.0.1:/mi>*:MI command failed with code -32601 (Method not
found)

 
I

I don't get this error on any other page but this one when attempting to
"Reload on Server".  Is there something specifically wrong with this page
and how it uses the MI command structure?  Again, All other pages that use
the "Reload on Server" give a 200ok and work as expected.

 
Thanks,
 
Jeff
 
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[OpenSIPS-Users] How can I retrieve the callID and start time for a call inside event_route[E_RTPPROXY_DTMF] {

[Saint Michael]

>
> My goal is to close the call as soon as the callee presses any DTMF

when any DTMF is detected,  then I need to access the callID variable,  $ci
but it's nowhere to be found
Also I need $avp(start_time) and $avp(duration)
I am at a loss as to how to retrieve this information. The call is connected
I need to fire a database query to close the call.
I imagine that when the call got connected, some variable stored that
unix-time .
But it does not seem to be available in this part of the code.
Philip






>
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