[Spce-user] No sound with outbound call

Marco Teixeira admin at marcoteixeira.com
Sat Jan 2 10:55:12 EST 2016


Hi,

On a security side note, don't forget to replace your, and your peer's,
real IP address before posting on public mailling lists.

About your issue, Can you tell that INVITE packet size ?
Try to grep it out of a tcpdump:
"tcpdump port 5060 -nnn | grep YOUR_PEER_IP_HERE"

​(apt-get install tcpdump, if you need)​


---
Marco
---


On Sat, Jan 2, 2016 at 7:56 AM, <j.feugne at prisnets.ch> wrote:

> 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
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> https://lists.sipwise.com/listinfo/spce-user
>
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