# Voice is not listening after call accept **URL:** \ **Category:** Asterisk Support\ **Created:** [November 12, 2013, 5:42am UTC](https://community.asterisk.org/t/voice-is-not-listening-after-call-accept/44429 "2013-11-12T05:42:42Z")\ **Posts on this page:** 3\ **Page:** 1
**Author:** ![ajaysaini](https://avatars.discourse-cdn.com/v4/letter/a/e47c2d/32.png) [@ajaysaini](https://community.asterisk.org/u/ajaysaini)\ **Post date:** [November 12, 2013, 5:42am UTC](https://community.asterisk.org/t/voice-is-not-listening-after-call-accept/44429/1 "2013-11-12T05:42:42Z")
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’ server2_CLI\> 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] server2_CLI\> Hope to hear from you soon!! Kind Regards, Ajay Saini ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [November 12, 2013, 8:08am UTC](https://community.asterisk.org/t/voice-is-not-listening-after-call-accept/44429/2 "2013-11-12T08:08:07Z")
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. ---
**Author:** ![ajaysaini](https://avatars.discourse-cdn.com/v4/letter/a/e47c2d/32.png) [@ajaysaini](https://community.asterisk.org/u/ajaysaini)\ **Post date:** [November 12, 2013, 8:26am UTC](https://community.asterisk.org/t/voice-is-not-listening-after-call-accept/44429/3 "2013-11-12T08:26:48Z")
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