Re: [OpenSIPS-Users] Reject unsupported codec?
Hi Bogdan, That pointed me in the right direction and I'm able to reject calls. I ended up using codec_exists_re() Thank you! On Thu, 7 Jan 2021 at 16:37, Bogdan-Andrei Iancu wrote: > Hi, > > See the codec checking related functions here -> > https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#func_codec_exists > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 2020 online > https://opensips.org/training/OpenSIPS_eBootcamp_2020/ > > On 1/7/21 6:33 PM, solarmon wrote: > > Hi, > > On opensips 2.4.x how would I best check what codec is being offered, and > reject the call if it ONLY offers a codec that is not supported by us. For > example, if we only want to support G.711 PCMA/PCMU. > > Thank you. > > ___ > Users mailing > listUsers@lists.opensips.orghttp://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] Reject unsupported codec?
Hi, See the codec checking related functions here -> https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#func_codec_exists Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 2020 online https://opensips.org/training/OpenSIPS_eBootcamp_2020/ On 1/7/21 6:33 PM, solarmon wrote: Hi, On opensips 2.4.x how would I best check what codec is being offered, and reject the call if it ONLY offers a codec that is not supported by us. For example, if we only want to support G.711 PCMA/PCMU. Thank you. ___ 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
[OpenSIPS-Users] Reject unsupported codec?
Hi, On opensips 2.4.x how would I best check what codec is being offered, and reject the call if it ONLY offers a codec that is not supported by us. For example, if we only want to support G.711 PCMA/PCMU. Thank you. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Digest Auth with LDAP/RADIUS
Hi Michael,
What you can do is to grab some online digest auth calculator and to
doublecheck the auth responses on each side (opensips and radius)
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS Bootcamp 2020 online
https://opensips.org/training/OpenSIPS_eBootcamp_2020/
On 1/6/21 6:56 PM, bobsy via Users wrote:
Hello everyone,
I’m attempting to use digest auth on Freeradius with LDAP and plaintext
userPassword’s.
When the radius server goes to auth the digest hashes don’t match up.
authenticate {
(17) digest: A1 = bobsy:opensips.vale.ski:password
(17) digest: A2 = REGISTER:sip:opensips.vale.ski
H(A1) = 0342aafbaea975d9fde3c46f3f093993
H(A2) = b0605d01a41aac18c7f1a84c8ca1c4f5
(17) digest: KD =
0342aafbaea975d9fde3c46f3f093993:5ff5eaca15917970591b0edf7c7c6bbd13698c0dd5e6:b0605d01a41aac18c7f1a84c8ca1c4f5
EXPECTED a8d6639edfd61ac7b1bb247f7832b8e5
RECEIVED a817470a4e1612532d167bed0354a88b
(17) digest: FAILED authentication
(17) [digest] = reject
(17) } # authenticate = reject
(17) Failed to authenticate the user
I have calculate_ha1 set to 1.
Any insight would be great.
And after this is resolved maybe someone can help me find out why the Kerberos
module looks for “User-Password”. I believe it should be looking for
“Cleartext-Password” and that’s why Kerberos won’t work for me.
Regards,
Michael Vale.
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Re: [OpenSIPS-Users] Mediaproxy configuration
Sorry... should have added that OpenSIPS box is acting as mid-registrar
On Thu, 7 Jan 2021, 12:12 Mark Allen, wrote:
> I wonder if anyone can help me with this? I am trying to configure
> Mediaproxy to handle RTP traffic coming from outside our local network.
> Here's the setup:
>
> UAC ---> IPA ---> IPB ---> Mediaproxy / OpenSIPS ---> Asterisk
>
> IPA (a public IP address 4x.xxx.xxx.xxx) maps ports ports 5060 and 1
> to 65535 to IPB (local IP address 192.168.xxx.xxx). IPB is actually a
> Virtual IP managed by keepalived.
> UAC is MizuDroid app running on my Android phone connected to my home
> network (NATed) with a public IP of 5.xxx.xxx.xxx. Everything else relates
> to our office network.
> Mediaproxy Dispatcher and Relay are both running on the same (OpenSIPS)
> system
>
> SIP conversation between UAC and Asterisk via OpenSIPS looks to be working
> fine. Endpoints connect, exchange data, and hangup. The problem is with SDP
> addressing (NAT problem) causing no audio either way, which is what I want
> Mediaproxy to handle.
>
> In opensips.cfg I'm passing control for calls arriving at IPA to
> Mediaproxy...
>
> if (is_method("INVITE")) {
> if (!has_totag()) {
> if ($fd == "4x.xxx.xxx.xxx") {
> xlog("Passing control to Mediaproxy...");
> engage_media_proxy();
> }
> }
> }
>
> In /etc/mediaproxy/config.ini all settings are defaults except for setting
> dispatcher as IPB...
>
> dispatchers = 192.168.xxx.xxx
>
> ...and I've tried it with and without advertised_ip set to IPA...
>
> advertised_ip = 4x.xxx.xxx.xxx
>
>
> I can see that Mediaproxy is taking control of calls as instructed and
> making changes to SDP but it's not solving my audio problems. What am I
> doing wrong
>
>
>
>
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[OpenSIPS-Users] Mediaproxy configuration
I wonder if anyone can help me with this? I am trying to configure
Mediaproxy to handle RTP traffic coming from outside our local network.
Here's the setup:
UAC ---> IPA ---> IPB ---> Mediaproxy / OpenSIPS ---> Asterisk
IPA (a public IP address 4x.xxx.xxx.xxx) maps ports ports 5060 and 1 to
65535 to IPB (local IP address 192.168.xxx.xxx). IPB is actually a Virtual
IP managed by keepalived.
UAC is MizuDroid app running on my Android phone connected to my home
network (NATed) with a public IP of 5.xxx.xxx.xxx. Everything else relates
to our office network.
Mediaproxy Dispatcher and Relay are both running on the same (OpenSIPS)
system
SIP conversation between UAC and Asterisk via OpenSIPS looks to be working
fine. Endpoints connect, exchange data, and hangup. The problem is with SDP
addressing (NAT problem) causing no audio either way, which is what I want
Mediaproxy to handle.
In opensips.cfg I'm passing control for calls arriving at IPA to
Mediaproxy...
if (is_method("INVITE")) {
if (!has_totag()) {
if ($fd == "4x.xxx.xxx.xxx") {
xlog("Passing control to Mediaproxy...");
engage_media_proxy();
}
}
}
In /etc/mediaproxy/config.ini all settings are defaults except for setting
dispatcher as IPB...
dispatchers = 192.168.xxx.xxx
...and I've tried it with and without advertised_ip set to IPA...
advertised_ip = 4x.xxx.xxx.xxx
I can see that Mediaproxy is taking control of calls as instructed and
making changes to SDP but it's not solving my audio problems. What am I
doing wrong
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Re: [OpenSIPS-Users] Quality Routing Module in Opensips_3.1
Hi Opensips Team,
Could you please provide an update on the above query on Qrouting.
Best Regards
Saurabh Chopra
+918861979979
On Mon, Jan 4, 2021 at 5:23 PM Saurabh Chopra wrote:
> Hi Tony/Opensips Team,
>
> Happy New Year,
>
> I have tried to test with default values in my configuration file but no
> luck.The call is still going to the first gateway i.e. 104.XX.XX.XX. If
> possible could you please help us at configuration side, what parameters
> should be allowed to test this Qrouting module. Below is the output
> for opensips-cli -x mi qr_status:-
>
> "Carrier": {
> "CRID": "cr1",
> "Gateways": [
> {
> "GWID": "gw1",
> "ASR": "-1.00/9",
> "CCR": "-1.00/9",
> "PDD": "-1.00/7",
> "AST": "-1.00/7",
> "ACD": "-1.00/7"
> },
> {
> "GWID": "gw2",
> "ASR": "-1.00/0",
> "CCR": "-1.00/0",
> "PDD": "-1.00/0",
> "AST": "-1.00/0",
> "ACD": "-1.00/0"
> }
> ]
>
>
> Best Regards
> Saurabh Chopra
> +918861979979
>
>
> On Mon, Dec 21, 2020 at 4:18 PM Saurabh Chopra
> wrote:
>
>> Hi Tony/Opensips Team,
>>
>> Will test it with default values as per your suggestion and will post the
>> result of statistics for each of the gateways.
>>
>>
>> Best Regards
>> Saurabh Chopra
>> +918861979979
>>
>>
>> On Sun, Dec 20, 2020 at 3:09 PM Tomi Hakkarainen
>> wrote:
>>
>>> Hi,
>>>
>>> never used myself but as reading the doc and your config, here some of
>>> my thoughts.
>>>
>>> I see you are setting min_samples to zero and My guess is that that way
>>> they will stay healthy forever?
>>> Maybe adjust the config of min_samples to something like default or 15
>>> and look how it behaves...
>>> also have you viewed what the statistics show while testing? ( opensips-cli
>>> -x mi qr_status )
>>> Would like to hear how it goes :)
>>>
>>> Tomi
>>>
>>> On 18. Dec 2020, at 15.03, Saurabh Chopra wrote:
>>>
>>>
>>> Hi All,
>>>
>>> Kindly update me on the query raised on Qrouting.
>>>
>>> Best Regards
>>> Saurabh Chopra
>>> +918861979979
>>>
>>>
>>> On Thu, Dec 17, 2020 at 3:43 PM Saurabh Chopra
>>> wrote:
>>>
Hi All,
I want to test the new quality routing module, previously i have tested
the dynamic routing and it works for me. But somehow, qrouting module is
not running as per my expectation. My understanding is qrouting module
helps us to choose a better gateway at run time as per statistics like
ASR,PDD,AST etc. I took two asterisk gateways
1:- 162.243.XX.XXX
2:- 104.131.XXX.XXX
I have deliberately given 15sec wait on 104.131.XXX.XXX asterisk after
this it will send 200 OK response for the call. So as per qrouting module,
AST statistics for 104.131.XXX.XXX gateway would somewhat be lower than
this 162.243.XX.XXX.
So,I am expecting the call should mostly be reached to 162.243.XX.XXX
gateway instead of 104.131.XXX.XXX, but this is not happening as calls are
reaching to 104.131.XXX.XXX gateway which has poor statistics i.e AST.
*Configuration done at mysql is given below:-*
mysql> select * from dr_rules;
++-++-+--+-+---+--+--+---++
| ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
sort_alg | sort_profile | attrs | description|
++-++-+--+-+---+--+--+---++
| 1 | 1 || |0 | |
gw2=50,gw1=50 | Q|1 | | XXX_gateway |
++-++-+--+-+---+--+--+---++
1 row in set (0.00 sec)
mysql> select * from dr_gateways;
++--+--+--+---++---++---++--+
| id | gwid | type | address | strip | pri_prefix | attrs
| probe_mode | state | socket | description |
++--+--+--+---++---++---++--+
| 1 | gw1 |3 | 162.243.XX.XXX:5080 |
[OpenSIPS-Users] [Feature] Script driven B2B with OpenSIPS 3.2
Hi all, The work on 3.2 is already ongoing, and one by one, the new features will emerge. One of the first (actually a continuation from 3.1) is the script driven B2B - shortly said, instead of using the nasty and intricate XML scripts to drive the B2B logic, you can now directly use the OpenSIPS routing script for that. Simpler and more powerful. For more, see https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/ Enjoy it, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy Relay start error - 'MediaRelayBase' is not defined
Yep - the software loaded successfully, integrated with OpenSIPS and I can see in the log file that it is trying to handle RTP traffic. Just need to work out the correct configuration now! :) Thanks for the help Adrian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy Relay start error - 'MediaRelayBase' is not defined
Hi Adrian - thanks for getting back to me
I'm getting an unexpected error on apt-get update using the following in
sources.list...
# AG Projects software
deb http://ag-projects.com/debian unstable main
deb-src http://ag-projects.com/debian unstable main
...I get...
Err:10 https://ag-projects.com/debian unstable Release
404 Not Found [IP: 85.17.186.10 443]
E: The repository 'http://ag-projects.com/debian unstable Release' no
longer has a Release file.
...I've checked with nmap and I can see port 443 on 85.17.186.10
I've set it to use 'buster' instead of 'unstable' ('buster' was previously
giving me a similar error)...
# AG Projects software
deb http://ag-projects.com/debian buster main
deb-src http://ag-projects.com/debian buster main
...and that seems to now be working as I get...
Hit:5 https://ag-projects.com/debian buster InRelease
...and when I install packages I'm seeing...
Get:1 https://ag-projects.com/debian buster/main amd64
mediaproxy-common amd64 4.0.5buster [63.1 kB]
Get:2 https://ag-projects.com/debian buster/main amd64
mediaproxy-dispatcher all 4.0.5buster [18.1 kB]
Get:3 https://ag-projects.com/debian buster/main amd64 mediaproxy-relay
all 4.0.5buster [18.6 kB]
Fetched 99.8 kB in 0s (535 kB/s)
...so hopefully I'm good to go now.
Thanks for your help. I'll post a follow-up once I've got it up and running
On Wed, 6 Jan 2021 at 21:59, Adrian Georgescu wrote:
> This was a bug.
>
> You must update to the latest mediaproxy version:
>
> sudo apt update
> sudo apt install mediaproxy-relay mediaproxy-common mediaproxy-dispatcher
>
> Regards,
> Adrian
>
> On 6 Jan 2021, at 12:59, Mark Allen wrote:
>
> Hi all - not sure what I'm missing here...
>
> I'm installing Mediaproxy onto our Debian Buster box which is also running
> OpenSIPS 3.1 but when I start the relay I'm getting an error in syslog...
>
> 15:40:07 opensipsx media-relay[4983]: INFO Starting MediaProxy Relay
> 4.0.4
> 15:40:07 opensipsx media-relay[4983]: INFO Set resource limit for
> maximum open file descriptors to 11000
> 15:40:07 opensipsx media-relay[4983]: CRITICAL Failed to create MediaProxy
> Relay: name 'MediaRelayBase' is not defined
> 15:40:07 opensipsx media-relay[4983]: ERRORTraceback (most recent call
> last):#012ERROR File "/usr/bin/media-relay", line 100, in
> #012ERRORfrom mediaproxy.relay import MediaRelay#012ERROR
>File "/usr/lib/python3/dist-packages/mediaproxy/relay.py", line 290, in
> #012ERRORclass MediaRelay(MediaRelayBase):#012ERROR
> NameError: name 'MediaRelayBase' is not defined
> 15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Main process
> exited, code=exited, status=1/FAILURE
> 15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Failed with
> result 'exit-code'.
>
> I'm starting the relay with the command...
>
> systemctl start mediaproxy-relay
>
> ...I installed Mediaproxy using the Debian package using the instructions
> at http://mediaproxy.ag-projects.com/installation-guide/ and
> https://github.com/AGProjects/mediaproxy. mediaproxy-dispatcher is
> starting successfully.
>
> In the /etc/mediaproxy/config.ini file - everything is left at the default
> setting except for...
>
> dispatchers = xxx.xxx.xxx.xxx
> advertised_ip = xxx.xxx.xxx.xxx
>
> Certificates are in place in /etc/mediaproxy/tls
>
>
> Anybody got any ideas about where I've gone wrong
>
>
>
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>
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