Re: [OpenSIPS-Users] Ms Teams to SBC call not working

2024-04-23 Thread Devang Dhandhalya via Users
Hello Rehan,

Thank you for the response

I have configured this dialed pattern : ^+91(\d{10})$ and select My SBC
which I have configured in Direct routing.
I assigned a phone number and Direct routing for my user using Microsoft
Teams admin center portal.
I configure voice routing policy as well and not assign PSTN usage Record
in it
Also configured dialplan with dial pattern: ^+91(\d{10})$ and voice routing
policy and dialplan is also assigned to my user

Please let me know if I miss any further configuration in MS Teams.

Regards,
Devang Dhandhalya



On Mon, Apr 22, 2024 at 10:20 PM Ahmed Rehan  wrote:

> Check your Voice Routing policy to the user you are calling from and also
> check the Route which is matching your dialed number has been assigned the
> SBC you have configured.
>
> On Mon, Apr 22, 2024 at 4:54 PM Devang Dhandhalya via Users <
> users@lists.opensips.org> wrote:
>
>> Hello All,
>>
>> I Configured Direct routing and all status related to SBC connectivity
>> are good and also Configured Voice routes with SBC for my user.
>> I am facing one issue for calls from MS Teams to SBC and getting below
>> error on MS teams Screen:
>>
>>
>> *We couldn't connect the call.With your calling license, you can only
>> call people within your organization. Talk to your IT admin to change your
>> license.*
>> What are the possible reasons for this error?
>>
>> Note: SBC to MS Teams Calls are working fine.
>>
>> I have configured below License in MS Teams account
>>
>> 1 ) Microsoft Teams Phone Standard
>> 2 ) Microsoft Teams
>> 3 ) Microsoft 365 Audio Conferencing
>>
>> Can Anyone tell me if I am missing any other license for Making outbound
>> calls from MS Teams to SBC using Direct routing?
>> Let me know if you require any further information related to MS Teams
>> Configuration.
>> Any suggestions would be appreciated.
>>
>> Many Thanks,
>> Devang Dhandhalya
>>
>>
>> *[image: https://www.ecosmob.com/opensips-summit/]
>> *
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>
> --
> Regards
> Ahmed Rehan
>


-- 
Regards,
*Devang Dhandhalya*

[image: Ecosmob Technologies Pvt. Ltd.] 

Ecosmob Technologies Pvt. Ltd.
https://www.ecosmob.com

VoIP | Web | Mobile | IoT | Big Data

-- 
* 
*
*Disclaimer*

In addition to 
generic Disclaimer which you have agreed on our website, any views or 
opinions presented in this email are solely those of the originator and do 
not necessarily represent those of the Company or its sister concerns. Any 
liability (in negligence, contract or otherwise) arising from any third 
party taking any action, or refraining from taking any action on the basis 
of any of the information contained in this email is hereby excluded.



*Confidentiality*
This communication (including any attachment/s) is 
intended only for the use of the addressee(s) and contains information that 
is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, 
distribution, or copying of this communication is prohibited. Please inform 
originator if you have received it in error.


*Caution for viruses, 
malware etc.*
This communication, including any attachments, may not be 
free of viruses, trojans, similar or new contaminants/malware, 
interceptions or interference, and may not be compatible with your systems. 
You shall carry out virus/malware 

Re: [OpenSIPS-Users] call not landing in agents

2024-04-23 Thread Prathibha B
Now the call is landing to an agents. But the music on hold continues
without stopping. How to resolve this?

On Mon, 22 Apr 2024 at 17:24, Prathibha B  wrote:

> I've setup a call center using opensips. When I call to a number, the call
> should land in one of the call center agents. I've browser phone as agent1
> and zoiper phone as agent2. The call is landing in agent2 after 45-50 mins
> of call establishment. But the call never lands in browser phone.
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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[OpenSIPS-Users] OpenSIPS 3.4 SIPREC Issue

2024-04-23 Thread Luis Leal via Users
Hi there,

We're encountering a curious issue with SIPREC in upgrading from 3.2 to 3.4.4 
and I was hoping someone would be able to shed some light on it.

There are two symptoms:

  1.  Errors in the opensips log
  2.  SIPREC invite with correct SDP details (as per rtpengine log) but stream 
metadata missing from the XML metadata

The errors in the log are as follows:

Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:src_start_recording: could not start recording!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:tm_start_recording: cannot start recording!

The curious part is that the above error happens before the 200 OK is received. 
The relevant SIP trace is:

12   21:33:50.20834810.1.17.15 5060 
10.249.224.954204   SIP/SDP   1125 Request: INVITE 
sip:1125@10.249.224.9:54204;ob |

...Snip...

21   21:33:53.26256010.1.17.15 5060 10.1.17.12  
   5060 SIP/SDP   1073 Status: 200 OK (INVITE) |

The SIPREC invite is still generated though but is missing stream details 
(participant details masked with +27X for privacy):

Session Initiation Protocol (SIP as raw text)
INVITE sip:10.1.17.24:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.17.15:5060;branch=z9hG4bK0235.aca30813.0
To: sip:10.1.17.24:5060
From: sip:10.1.17.24:5060;tag=c5d35275eae8a009626d3007dc8441a2-ce21
CSeq: 2 INVITE
Call-ID: B2B.364.22430.1713814432.535273629
Max-Forwards: 70
Content-Length: 1995
User-Agent: OpenSIPS (3.4.4 (x86_64/linux))
Require: siprec
Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42
Contact: sip:10.1.17.15:5060;+sip.src

--OSS-unique-boundary-42
Content-Type: application/sdp

v=0
o=- 7360776941148045834 7360776941148045834 IN IP4 10.1.17.8
s=rtpengine-12-3-1-2-0-mr12-3-1-2-1-el9
t=0 0
m=audio 31432 RTP/AVP 8 101
c=IN IP4 10.1.17.8
a=label:0
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1120210035 cname:060168be20ab122b
a=sendonly
a=rtcp:31433
m=audio 36760 RTP/AVP 8 101
c=IN IP4 10.1.17.8
a=label:1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendonly
a=rtcp:36761
a=ptime:20

--OSS-unique-boundary-42
Content-Type: application/rs-metadata+xml
Content-Disposition: recording-session



 complete
 
  2d409cb6-b066-4579-8c49-c6e6a7b9d600
 
 
  
   +27X
  
 
 
  
 
 
  2024-04-22T21:33:50+0200
 
 
  2024-04-22T21:33:50+0200
 
 
  2024-04-22T21:33:50+0200
 
 
 
 
 

--OSS-unique-boundary-42--

Is there a configuration item we're missing perhaps?

Kind regards

Luis Leal



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