I’m new to the asterisk. I configured the sip.conf and extension.conf but after accept the call, voice is not there. I mean voice is not reaching to each other.
sip.conf
[general]
context=internal
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
canreinvite=no
nat=yes
session-timers=refuse
[7001]
type=friend
host=dynamic
secret=7001
context=internal
[7002]
type=friend
host=dynamic
secret=7002
context=internal
extension.conf
[internal]
exten => 7001,1,Answer()
exten => 7001,2,Dial(SIP/7001,60)
exten => 7001,3,Playback(vm-nobodyavail)
exten => 7001,4,VoiceMail(7001@main)
exten => 7001,5,Hangup()
exten => 7002,1,Answer()
exten => 7002,2,Dial(SIP/7002,60)
exten => 7002,3,Playback(vm-nobodyavail)
exten => 7002,4,VoiceMail(7002@main)
exten => 7002,5,Hangup()
iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
Logs of the asterisk
== Parsing ‘/etc/asterisk/logger.conf’: Found
Asterisk Queue Logger restarted
[Nov 12 10:50:06] NOTICE[3258]: chan_sip.c:27732 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002
– Registered SIP ‘7002’ at 192.168.1.121:5060
> Saved useragent “Zoiper rev.14091” for peer 7002
[Nov 12 10:50:06] NOTICE[3258]: chan_sip.c:27732 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002
– Registered SIP ‘7001’ at 192.168.1.170:5060
> Saved useragent “Zoiper for Windows 2.41 r19887” for peer 7001
[Nov 12 10:51:10] NOTICE[3258]: chan_sip.c:27732 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7001
[Nov 12 10:51:11] NOTICE[3258]: chan_sip.c:27732 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7001
== Using SIP RTP CoS mark 5
– Executing [7002@internal:1] Answer(“SIP/7001-00000000”, “”) in new stack
– Executing [7002@internal:2] Dial(“SIP/7001-00000000”, “SIP/7002,60”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/7002
– SIP/7002-00000001 is ringing
> 0x7fdb140216a0 – Probation passed - setting RTP source address to 192.168.1.170:8000
– SIP/7002-00000001 answered SIP/7001-00000000
– Remotely bridging SIP/7001-00000000 and SIP/7002-00000001
> 0x7fdb1403b930 – Probation passed - setting RTP source address to 192.168.1.121:8000
== Spawn extension (internal, 7002, 2) exited non-zero on ‘SIP/7001-00000000’
[Nov 12 10:54:15] WARNING[3258][C-00000000]: chan_sip.c:22960 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog ‘1c67b74c1367fb787f09f29f639e505f@192.168.1.160:5060’. Giving up.
== Using SIP RTP CoS mark 5
– Executing [7001@internal:1] Answer(“SIP/7002-00000002”, “”) in new stack
– Executing [7001@internal:2] Dial(“SIP/7002-00000002”, “SIP/7001,60”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/7001
– SIP/7001-00000003 is ringing
> 0x7fdb1403df50 – Probation passed - setting RTP source address to 192.168.1.121:8000
– SIP/7001-00000003 answered SIP/7002-00000002
– Remotely bridging SIP/7002-00000002 and SIP/7001-00000003
> 0x7fdb140597d0 – Probation passed - setting RTP source address to 192.168.1.170:8000
== Spawn extension (internal, 7001, 2) exited non-zero on 'SIP/7002-00000002’
server2CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status Description
7001/7001 192.168.1.170 D a 5060 Unmonitored
7002/7002 192.168.1.121 D a 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
server2CLI>
Hope to hear from you soon!!
Kind Regards,
Ajay Saini