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: <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).
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
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi
di Siena
tel: 0577.(23)5000 - fax: 0577.(23)2053
Per trovare una soluzione rapida ai tuoi problemi tecnici
consulta le FAQ di Ateneo, http://www.faq.unisi.it
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