Hai All,
I want to make a remote extensions in My asterisk …I created and made a call from extension to extension…Call Established But Am not able to hear the voice…
Please help me…to solve this issue
My configuration like this
In My router I forwarde ports like this
TCP-80 to 83 for ssh
TCP/5060 to 6000
UDP/1025 to 65534
in My Asterisk I used 3Cx Softphone in my PC and Zoiper in my mobile
rtp.conf
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=1025
rtpend=65534
====sip.conf===
[general]
externip=xxx.xxx.xxx.xx
port=5060
NAT=yes
bindaddr=0.0.0.0
localnet=192.168.1.197/255.255.255.0
srvlookup=no
tos=lowdelay
qualify=yes
[100]
username=100
secret=100
type=Friend
callerid=100
host=Dynamic
directmedia=no
directrtpsetup=no
NAT=yes
auth=Md5
qualify=yes
directmedia=no
insecure=port,invite
dtmfmode=auto
disallow=all
allow=ulaw,gsm,ilbc,alaw
Callgroup=1
pickupgroup=1-9,13
context=from-internal
mailbox=ay@gmail.com
[202]
username=202
secret=202
type=Friend
directmedia=no
directrtpsetup=no
qualify=yes
callerid=202
host=Dynamic
NAT=yes
auth=Md5
insecure=port,invite
dtmfmode=auto
disallow=all
allow=ulaw,gsm,ilbc,alaw
Callgroup=1
pickupgroup=1-9,13
context=from-internal
mailbox=ay@gmail.com
=================SIP DEBUG LOGS=================================
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: ** Our prefcodec: (nothing)
[Dec 4 12:37:48] VERBOSE[2436][C-0000002c] chan_sip.c: Audio is at 24734
[Dec 4 12:37:48] VERBOSE[2436][C-0000002c] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Dec 4 12:37:48] VERBOSE[2436][C-0000002c] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 4 12:37:48] VERBOSE[2436][C-0000002c] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Dec 4 12:37:48] VERBOSE[2436][C-0000002c] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: – Done with adding codecs to SDP
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Done building SDP. Settling with this capability: (gsm|ulaw|alaw)
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Initializing already initialized SIP dialog ODk2ZWQ0NWY3OWI3NDdjNzg1ZGExNTkwOWQyMTcwOTc. (presumably reinvite)
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 0 [ 55]: INVITE sip:100@123.237.130.2:5010;transport=UDP SIP/2.0
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 117.253.164.200:5060;branch=z9hG4bK1e1db8ba;rport
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 3 [ 49]: From: sip:101@192.168.1.197:5060;tag=as1fee546d
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 4 [ 50]: To: "100"sip:100@192.168.1.197:5060;tag=1d1c4c00
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 5 [ 39]: Contact: sip:101@117.253.164.200:5060
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 6 [ 53]: Call-ID: ODk2ZWQ0NWY3OWI3NDdjNzg1ZGExNTkwOWQyMTcwOTc.
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 8 [ 31]: User-Agent: Asterisk PBX 11.5.1
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 11 [ 81]: Remote-Party-ID: “101” sip:101@192.168.1.197;party=called;privacy=off;screen=no
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
[Dec 4 12:37:48] VERBOSE[2436][C-0000002c] chan_sip.c: Reliably Transmitting (no NAT) to 123.237.130.2:5010:
INVITE sip:100@123.237.130.2:5010;transport=UDP SIP/2.0^M
Via: SIP/2.0/UDP 117.253.164.200:5060;branch=z9hG4bK1e1db8ba;rport^M
Max-Forwards: 70^M
From: sip:101@192.168.1.197:5060;tag=as1fee546d^M
To: "100"sip:100@192.168.1.197:5060;tag=1d1c4c00^M
Contact: sip:101@117.253.164.200:5060^M
Call-ID: ODk2ZWQ0NWY3OWI3NDdjNzg1ZGExNTkwOWQyMTcwOTc.^M
o=root 1357123801 1357123802 IN IP4 117.253.164.200^M
s=Asterisk PBX 11.5.1^M
c=IN IP4 117.253.164.200^M
t=0 0^M
m=audio 24734 RTP/AVP 0 8 3 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:3 GSM/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=sendrecv^M
m=video 0 RTP/AVP 34^M
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1699
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] chan_sip.c: Trying to put ‘INVITE sip:’ onto UDP socket destined for 123.237.130.2:5010
[Dec 4 12:37:48] VERBOSE[2436][C-0000002c] frame.c: << [ TYPE: Control (4) SUBCLASS: Unknown control ‘22’ (22) ] [SIP/101-0000000a]
[Dec 4 12:37:48] DEBUG[2436][C-0000002c] res_rtp_asterisk.c: Difference is 896, ms is 132
[Dec 4 12:37:49] DEBUG[1901] chan_sip.c: SIP TIMER: Rescheduling retransmission #1699 (1) INVITE - 5
[Dec 4 12:37:49] DEBUG[1901] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 746 ms (t1 373 ms (Retrans id #1699))
[Dec 4 12:37:49] VERBOSE[1901] chan_sip.c: Retransmitting #1 (no NAT) to 123.237.130.2:5010:
INVITE sip:100@123.237.130.2:5010;transport=UDP SIP/2.0^M
Via: SIP/2.0/UDP 117.253.164.200:5060;branch=z9hG4bK1e1db8ba;rport^M
Max-Forwards: 70^M
From: sip:101@192.168.1.197:5060;tag=as1fee546d^M
To: "100"sip:100@192.168.1.197:5060;tag=1d1c4c00^M
Contact: sip:101@117.253.164.200:5060^M
Call-ID: ODk2ZWQ0NWY3OWI3NDdjNzg1ZGExNTkwOWQyMTcwOTc.^M
CSeq: 102 INVITE^M
User-Agent: Asterisk PBX 11.5.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Remote-Party-ID: “101” sip:101@192.168.1.197;party=called;privacy=off;screen=no^M
Content-Type: application/sdp^M