OpenSIPS, Asterisk and LocalAgents for Queues

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OpenSIPS, Asterisk and LocalAgents for Queues

Postby michelepin » Wed Sep 16, 2015 4:34 am

Hi all,

i'm build and using a voip pbx system using OpenSIPS as a router (i need to serve thousand of users...) and an Asterisk server as media box, for IVR, queues and so on.

I've a PATTON PSTN GW (172.20.1.4), the VoIP OpenSIPS ROUTER (172.20.1.2) and the Asterisk BOX (172.20.1.5).

In queues, because i've some troubles telling Asterisk which users are online and available, i decide to use LocalAgent way to force calls to every agents. For example, in queue.conf i have:

[operator-phone-queue]
music = queue-default
strategy = linear
context = ivr-services ; Here we go when the caller presses a single digit, while in the queue
timeout = 15
wrapuptime = 10
announce-frequency = 30
announce-holdtime = yes
joinempty = yes
member => Local/SIP-5002@MemberConnector,1
member => Local/SIP-5023@MemberConnector,2

and in extensions.conf:

[MemberConnector]
exten => _[A-Za-z0-9].,1,Verbose(2,Connecting ${CALLERID(all)} to Agent at ${EXTEN})
same => n,Set(QueueMember=${FILTER(A-Za-z0-9\-,${EXTEN})})
same => n,Set(Technology=${CUT(QueueMember,-,1)})
same => n,Set(Device=${CUT(QueueMember,-,2)})
same => n,Noop("MemberConnector: calling queue member ${Technology}/voip-trunk/${Device}")
same => n,Dial(${Technology}/voip-trunk/${Device},30)
same => n,Hangup()

That way works well *BUT* i have a problem with RTP audio flow, because when, for example, i call from 4999 to the queue and 5002 or 5023 answers the call, i got no audio from 5002 to 4999 (but i hear sounds from 4999 to 5002). The SIP signalling was this:

Code: Select all
INVITE sip:5002@172.20.1.47:57907 SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKa165.92c040a1.0.
Via: SIP/2.0/UDP 172.20.1.5:5060;rport=5060;received=172.20.1.5;branch=z9hG4bK47310f8d.
Max-Forwards: 69.
From: <sip:4999@>;tag=as1e28f247.
To: <sip:5002@;tag=l3f2mwdv8j.
Contact: <sip:4999@172.20.1.5:5060>.
Call-ID: 252126f32e04b0364360b6d65c7dba1f@.
CSeq: 104 INVITE.
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length:240.
.
v=0.
o=root 862552143 862552145 IN IP4 172.20.1.5.
s=Asterisk PBX 11.13.1~dfsg-2+b1.
c=IN IP4 172.20.1.5.
t=0 0.
m=audio 16660 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

[...]

SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 172.20.1.5:5060;rport=5060;received=172.20.1.5;branch=z9hG4bK47310f8d.
From: <sip:4999@>;tag=as1e28f247.
To: "Michele" <sip:5002@>;tag=l3f2mwdv8j.
Call-ID: 252126f32e04b0364360b6d65c7dba1f@.
CSeq: 104 INVITE.
User-Agent: snom760/8.7.5.17.
Contact: <sip:5002@172.20.1.47:57907>;reg-id=1.
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE.
Allow-Events: talk, hold, refer, call-info.
Supported: timer, 100rel, replaces, from-change.
Content-Type: application/sdp.
Content-Length: 218.
.
v=0.
o=root 1421125882 1421125885 IN IP4 172.20.1.47.
s=call.
c=IN IP4 172.20.1.47.
t=0 0.
m=audio 60670 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


I think that the problem was the 172.20.1.5 (Asterisk box) as RTP endpoint and not 172.20.1.4 (Patton GW, where call 4999 was originated).

So, there's a solution ? Hints ?

Thanks, Michele
michelepin
Newsterisk
 
Posts: 1
Joined: Wed Sep 16, 2015 4:18 am
Location: Siena, ITALY

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