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) andn

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:

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: ;tag=as1e28f247. To: . 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: ;tag=as1e28f247. To: “Michele” ;tag=l3f2mwdv8j. Call-ID: 252126f32e04b0364360b6d65c7dba1f@. CSeq: 104 INVITE. User-Agent: snom760/8.7.5.17. Contact: ;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).

Just to be more clear, the flow is:

[PSTN Net 4999]—->[PATTON GW | 172.20.1.4]—->[OpenSIPS
172.20.1.2]—>[Asterisk BOX (Queues) | 172.20.1.5]—->[OpenSIPS
172.20.1.2]—->(ring 5002)—->(answer 5002)—>(Call established but no audio)

So, there’s a solution ? Hints ?

Thanks, Michele