[Spce-user] No sound with outbound call
j.feugne at prisnets.ch
j.feugne at prisnets.ch
Sat Jan 2 02:56:17 EST 2016
hello,
I'm just starting with Sipwise and I've setup a VM running version
mr4.1.1 CE.
I have followed the instructions of the manual and all subscribers can
call each others.
I also setup a sip peer to voicetrading, and here is waht I have withthe
command line ngrep sip on my vm while placing and outband call:
NB, voicetrading is expecting for premium rate all number to be leading
with 00000 (5 zero)
U 84.253.60.150:5060 -> 77.72.169.129:5060
INVITE sip:0000041227312221 at 77.72.169.129:5060;transport=udp
SIP/2.0..Record-Route: <sip:84.253.60.150;r2=on;
lr=on;ftag=2AF8D981-568781960002DB55-51003700;ngcplb=yes>..Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=2AF8
D981-568781960002DB55-51003700;ngcplb=yes>..Via: SIP/2.0/UDP
84.253.60.150;branch=z9hG4bK603f.7793d5b01aaccef
36467f073399c00c2.0..Via: SIP/2.0/UDP
127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKVg2PkaeG;rport=5080..Fr
om:
<sip:41225017070 at prisnets.adoovoice.com>;tag=2AF8D981-568781960002DB55-51003700..To:
<sip:000004122731222
1 at 77.72.169.129>..CSeq: 11 INVITE..Call-ID:
scRGTP71RF5cdeju0O0qmw.._b2b-1..Max-Forwards: 69..Allow: INVITE,
ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE..Supported:
replaces, norefersub, extended-refer, outbo
und, path, X-cisco-serviceuri..P-Asserted-Identity:
<sip:41225017070 at prisnets.adoovoice.com>..Authorization:
Digest username="", realm="prisnets.adoovoice.com",
nonce="4087963395", uri="sip:0000041227312221 at 77.72.169.1
29:5060;transport=udp", response="c2fb4cd3a1235efd35e99190aedc45c2",
algorithm=MD5..Content-Type: application
/sdp..Content-Length: 322..Contact:
<sip:ngcp-lb at 84.253.60.150:5060;ngcpct=7369703a3132372e302e302e313a353038
30>....v=0..o=Zoiper 0 0 IN IP4 84.253.60.150..s=Zoiper..c=IN IP4
84.253.60.150..t=0 0..m=audio 32468 RTP/AVP
3 0 8 18 101..a=rtpmap:3 GSM/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp
:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-16..a=sendrecv..a=rtcp:32469..a=direction:both
..
After almost 1 minute with no ringing sound nor answer the call hang up.
Can someone please help me point out where the problem is comming and
how I can solve it?
Thank you
Jephthé Feugné
More information about the Spce-user
mailing list