[SR-Users] No Audio For Outbound Calls

Tue May 11 10:17:36 CEST 2021

Hello,

Could you share your Asterisk's sip context where you set up connectivity
with telco? Just delete all usernames and passwords, if any....

Your setup is very similar to the one I have with a very large telco in my
country, and it works fine as long as you always keep Asterisk on the RTP
path (no re-invites allowed, NAT always active)

*Sérgio Charrua*

*www.voip.pt <http://www.voip.pt/>*
Tel.: +351  <callto:+351+91+104+12+66>21 130 71 77

Email : *sergio.charrua at voip.pt <sergio.charrua at voip.pt>*

This message and any files or documents attached are strictly confidential
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It is intended only for the individual or entity named. If you are not the
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On Tue, May 11, 2021 at 8:02 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> So let me get this right:
>
> asterisk (10.0.x.x)--->(192.168.0.192) proxy (10.0.x.x)--->(10.0.x.x)telco
> op
>
> There's something i'm not seeing.
> Can you explain further like i did above?
>
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Mon, May 10, 2021 at 8:42 PM Kashish Raheja <
> kashishraheja1809 at gmail.com> wrote:
>
>> Yes, the telecom operator is on the private network. 10.0.X.X is the SBC
>> IP of the telecom operator to which we register. 10.0.X.X is reachable only
>> through the second network interface. The complete flow is given below:
>>
>> [image: image.png]
>>
>> Here 3.236.72.101 is Asterisk Server, 192.168.0.192 is local first
>> network interface IP, 10.0.87.230 is second network interface IP
>> and 10.0.76.9 is telecom operator SBC IP to which we do SIP register.
>>
>> We are running the rtpproxy on local IP (192.168.0.192) in the following
>> way:
>>
>> *rtpproxy -F -p /var/run/rtpproxy.pid -u asterisk -l 192.168.0.192 -s
>> udp:localhost:7722*
>>
>>
>> Thanks.
>> Regards
>> Kashish
>>
>>
>> On Mon, May 10, 2021 at 4:04 PM Kashish Raheja <
>> kashishraheja1809 at gmail.com> wrote:
>>
>>> Here are the SIP Traces:
>>>
>>> *Asterisk Server to Kamailio Server (SDP Packet):*
>>>
>>> 2021/05/10 15:54:52.835255 10.0.X.X:5060 -> 10.0.X.X:5060
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP
>>> 192.168.0.192;branch=z9hG4bK2599.de1bcd2ba5f8bfc86afb083b0a9e3f65.0;received=10.0.X.X;rport=5060,SIP/2.0/UDP
>>> 3.236.X.X:5060;branch=z9hG4bK5a69547a;received=3.236.X.X;rport=5060
>>> Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
>>> Call-ID: 58eb00885daef7ff3a67ad0e235e817a at 14.98.22.110
>>> From: <sip:68XXXXX at 10.0.X.X>;tag=as2b21d944
>>> To: <sip:09413745250 at 192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
>>> CSeq: 102 INVITE
>>> Allow:
>>> INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
>>> Contact: <sip:09413745250 at 10.0.X.X
>>> :5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
>>> User-Agent: ZTE Softswitch/1.0.0
>>> Require: timer
>>> Session-Expires: 7200;refresher=uac
>>> Content-Length: 182
>>> Content-Type: application/sdp
>>>
>>> v=0
>>> o=- 1936 20890 IN IP4 10.0.X.X
>>> s=SBC call
>>> c=IN IP4 10.0.X.X
>>> t=0 0
>>> m=audio 37874 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=rtpmap:8 PCMA/8000/1
>>>
>>>
>>> *Kamailio Server to Telecom Operator Carrier (SDP Packet):*
>>>
>>> 2021/05/10 15:54:52.835419 192.168.0.192:5060 -> 3.X.X.X:5060
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 3.236.72.101:5060
>>> ;branch=z9hG4bK5a69547a;received=3.236.72.101;rport=5060
>>> Record-Route: <sip:192.168.0.192;lr;ftag=as2b21d944>
>>> Call-ID: 58eb00885daef7ff3a67ad0e235e817a at 14.98.22.110
>>> From: <sip:68XXXXX at 10.0.X.X>;tag=as2b21d944
>>> To: <sip:09413745250 at 192.168.0.192:5060>;tag=aa2c806-Huku2c07186a1
>>> CSeq: 102 INVITE
>>> Allow:
>>> INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
>>> Contact: <sip:09413745250 at 10.0.X.X
>>> :5060;Hpt=8e72_16;CxtId=3;TRC=ffffffff-ffffffff>
>>> User-Agent: ZTE Softswitch/1.0.0
>>> Require: timer
>>> Session-Expires: 7200;refresher=uac
>>> Content-Length: 182
>>> Content-Type: application/sdp
>>>
>>> v=0
>>> o=- 1936 20890 IN IP4 10.0.X.X
>>> s=SBC call
>>> c=IN IP4 10.0.X.X
>>> t=0 0
>>> m=audio 37874 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=rtpmap:8 PCMA/8000/1
>>>
>>> Regards
>>> Kashish
>>>
>>>
>>> On Mon, May 10, 2021 at 2:37 PM Kashish Raheja <
>>> kashishraheja1809 at gmail.com> wrote:
>>>
>>>> Hi All,
>>>>
>>>> I have set up Kamailio in the following manner:
>>>>
>>>> Kamailio (Physical Server: Register to Telecom Operator Carrier SIP
>>>> trunk) ---> Asterisk Server (on Cloud having public IP)
>>>>
>>>> I am successfully able to route the call to Asterisk server on Cloud
>>>> when I make a call to the number provided by the carrier and there is audio
>>>> also on both sides.
>>>>
>>>> However, when I am making an outbound call from Asterisk server to the
>>>> number through Kamailio, there is no audio when I pick up the call. I have
>>>> tried to capture the traces but not able to understand the exact problem
>>>> here.
>>>>
>>>> Note: I am running the RTP proxy on Kamailio server.
>>>>
>>>> Any help on why this might be happening?
>>>>
>>>> Thanks.
>>>> Regards
>>>> Kashish
>>>> +919413745250
>>>>
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