Hi Magnus,
attaching cfg files is useless, as no one will debug the script, but we
will help you to debug your script.
So, for the non-working case (PSTN to SIP) does your script force
RTPproxy in INVITE and 200 OK ?
Regards,
Bogdan
On 03/29/2012 01:52 AM, [email protected] wrote:
I have phones (some behind NAT) connecting to Opensips server an
Asterisk and an rtpproxy as seen below:
rtpproxy started with
ps -aux | grep rtpproxy
root 15666 0.0 0.0 14472 920 ? Ssl Mar23 0:05
./rtpproxy -F -l 189.254.2.19 -s udp:* 7890 -d DBUG LOG_LOCAL3
UAC1 username =
100------------Firewall/router--------------------Opensips
1.7---------- RTP PROXY------------Asterisk 1.6
192.168.1.10 192.168.1.1
65.254.63.212 189.254.2.19 190.61.201.89
external ip dinamic 169.254.2.2
- Calls between UAC are OK (both SIP and RTP).
- Calls UAC for PSTN is OK.
- Did numbers is received in Asterisk, and destination for UAC
registered in opensips, but no work audio .
(EX User call cellphone for DID 54115368566, call is received in
asterisk, and destination for user 100, registered in opensips)
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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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