i need help, i created a sip trunk with other sip server, but the cme give this error:
070846: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:900@200.111.156.150 SIP/2.0
Via: SIP/2.0/UDP 54.207.30.33:5080;rport;branch=z9hG4bKD7U28atZZ6r7B
Max-Forwards: 69
From: "Extension 1000" <sip:FreeSWITCH@200.111.156.150>;tag=Br453j6KcmKFN
To: <sip:900@200.111.156.150>
Call-ID: ef0174b1-5735-1233-6ebd-02585e7b1bc5
CSeq: 73838188 INVITE
Contact: <sip:gw+aerosan@54.207.30.33:5080;transport=udp;gw=aerosan>
User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 223
X-FS-Support: update_display,send_info
Remote-Party-ID: "Extension 1000" <sip:1000@200.111.156.150>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1428321972 1428321973 IN IP4 54.207.30.33
s=FreeSWITCH
c=IN IP4 54.207.30.33
t=0 0
m=audio 26532 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
070847: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 54.207.30.33:5080;rport;branch=z9hG4bKD7U28atZZ6r7B
From: "Extension 1000" <sip:FreeSWITCH@200.111.156.150>;tag=Br453j6KcmKFN
To: <sip:900@200.111.156.150>;tag=3888E968-22ED
Date: Mon, 06 Apr 2015 19:23:05 GMT
Call-ID: ef0174b1-5735-1233-6ebd-02585e7b1bc5
CSeq: 73838188 INVITE
Allow-Events: telephone-event
Warning: 304 200.111.156.150 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
the sip trunk configuration:
dial-peer voice 3611 voip
permission orig
description aws
huntstop
preference 1
destination-pattern 100.
session protocol sipv2
session target ipv4:54.207.30.33:5080
incoming called-number 1000
voice-class codec 2
no vad
the voice configuration:
voice service voip
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
supplementary-service h450.12
sip
!
voice class codec 2
codec preference 1 g711ulaw
!
voice class codec 1
codec preference 1 g711ulaw
please help me
