Voice is not listening after call accept

Asterisk Asterisk Support
1

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

2

You have a remote bridge, so the two end points have been requested to send the media (voice) directly between themselves.

I guess canreinvite was finally removed from Asterisk. It has been deprecated for the best part of half a decade. You should be using directmedia.

nat=yes is also depecated,. Note if you are inside NAT calling out, this is not necessarily needed, but you are missing any option that would allow Asterisk to work out its external address.

More generally, you seem to posting a number of questions about hte same system and possibly the same problem, without linking the questions together, and in none of these questions have you provided the debugging information that is needed to quickly solve problems.

3

Now I successfully got one side sound but sound echo. What cold be problem that echo the sound.

and My sound listen to me back not to other user to which I made call.

Kind Regards