# Voicemail not played after one succesful call **URL:** \ **Category:** Asterisk SIP\ **Created:** [February 6, 2017, 8:43am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642 "2017-02-06T08:43:06Z")\ **Posts on this page:** 20\ **Page:** 1
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 6, 2017, 8:43am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/1 "2017-02-06T08:43:06Z")
Good morning, when routing a call to voicemail the very first call is played correctly. All following calls don’t play the audio. In the CLI ii is shown that the voicemal file is played but no audio and Asterisk hangs up regardless of talking into the headset or not. > ``` > -- Executing [022528301470@Telekom_in:1] Dial("PJSIP/Telekom_in-00000006", "SCCP/29,30") in new stack > > SCCP: sccp_requestChannel returned Line 29 not currently registered -> Try again later > > ``` > > [Feb 6 09:25:42] WARNING[25311][C-00000007]: app\_dial.c:2530 dial\_exec\_full: Unable to create channel of type ‘SCCP’ (cause 44 - Requested channel not available) > == Everyone is busy/congested at this time (1:0/0/1) > – Executing [022528301470@Telekom\_in:2] VoiceMail(“PJSIP/Telekom\_in-00000006”, “29@voicemail”) in new stack > – \ Playing ‘vm-intro.g722’ (language ‘en’) > – \ Playing ‘beep.g722’ (language ‘en’) > – Recording the message > – x=0, open writing: /var/spool/asterisk/voicemail/voicemail/29/tmp/sUva5X format: wav, 0x75b0e0ec > [Feb 6 09:25:52] WARNING[25311][C-00000007]: app.c:1638 \_\_ast\_play\_and\_record: No audio available on PJSIP/Telekom\_in-00000006?? > – User hung up > == Parsing ‘/var/spool/asterisk/voicemail/voicemail/29/INBOX/msg0004.txt’: Found > == Spawn extension (Telekom\_in, 022528301470, 2) exited non-zero on ‘PJSIP/Telekom\_in-00000006’ A retry in debug mode showed: > Feb 6 10:24:32] WARNING[25996][C-00000002]: app.c:1638 \_\_ast\_play\_and\_record: No audio available on PJSIP/Telekom\_in-00000001?? > – User hung up > == Parsing ‘/var/spool/asterisk/voicemail/voicemail/11/INBOX/msg0013.txt’: Found > == Spawn extension (Telekom\_in, 022528301471, 2) exited non-zero on ‘PJSIP/Telekom\_in-00000001’ No audio available? Please advise what to do. Regards, Joerg ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [February 6, 2017, 11:05am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/2 "2017-02-06T11:05:12Z")
You will need to provide the output of “pjsip set logger on” and “rtp set debug on” for a call that does not work. It may be that they are not sending media or if you are behind NAT that it is getting blocked. ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 6, 2017, 11:10am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/3 "2017-02-06T11:10:08Z")
Hello Joshua and thanks again for your answer. First of all I have to apologize for bothering you again and again with Newbie questions. By testing and torturing my spouse with my testing I found out that this behaviour only happens when no telephone is registered (as I tend to switch off my VoIP phones if no one’s in the house). Do you want me to attach the logfile in text format? Best regards to Canada, Joerg ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [February 6, 2017, 11:10am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/4 "2017-02-06T11:10:59Z")
If you’d like some insight into what may be the cause then the log info needs to be available somehow. ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 6, 2017, 11:26am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/5 "2017-02-06T11:26:46Z")
Is there any way I can upload such a big file? I plain text it’s not possible and as attachment only pictures are accepted. ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [February 6, 2017, 11:27am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/6 "2017-02-06T11:27:15Z")
You can use [pastebin.com](http://pastebin.com) and provide a link. ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 6, 2017, 11:48am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/7 "2017-02-06T11:48:53Z")
[http://pastebin.com/xh44Hnc7](http://pastebin.com/xh44Hnc7) Thanks! ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [February 6, 2017, 12:03pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/8 "2017-02-06T12:03:25Z")
We are sending both an answer and media to them, but they are not acknowledging the answer and not sending us media. I’d look at the network level to make sure their response is not getting dropped by a firewall. ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 6, 2017, 12:24pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/9 "2017-02-06T12:24:29Z")
If it would be a firewall it would drop all the communication events. But if there is an active endpoint there’ll be no loss auf audio. If the endpoint is not active and there is no ringing (because the endpoint could not be found) only then there’s a loss auf audio… This doesn’t sound like a firewall drop to me… ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [February 6, 2017, 12:25pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/10 "2017-02-06T12:25:32Z")
Well, the other side is not sending media and it is not responding to our 200 OK. Eliminating the firewall eliminates another part of the equation if one is present. Otherwise you’ll need to talk to your provider as everything looks correct from our side. ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 6, 2017, 1:20pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/11 "2017-02-06T13:20:35Z")
Eliminated firewall. Without success… Still no audio ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [February 6, 2017, 1:20pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/12 "2017-02-06T13:20:36Z")
Firewalls can create dynamic rules, so it is possible that one is being created and then expiring. ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 6, 2017, 1:41pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/13 "2017-02-06T13:41:44Z")
I deleted all fw rules, rebooted the router and still no success ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 7, 2017, 8:00pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/14 "2017-02-07T20:00:54Z")
Good evening everyone, do I need to forward all RTP ports to the Asterisk server? I disabled my firewall and used port forwarding in one try and no port forwarding on another test (I rebooted the router in between). - I have on the first call from outside no voice output on the inside phone but output on the outside - After one more or less successful call on the next calls I can accpept the call on the inside phone but the caller still has the calling signal. > [Feb 7 20:54:39] WARNING[3836][C-00000001]: app\_dial.c:2530 dial\_exec\_full: Unable to create channel of type ‘SCCP’ (cause 44 - Requested channel not available) > – Called PJSIP/22 > – PJSIP/22-00000001 is ringing > – PJSIP/22-00000001 answered PJSIP/Telekom\_in-00000000 > – Channel PJSIP/22-00000001 joined ‘simple\_bridge’ basic-bridge \<63d9fea4-4156-4152-9798-6aa880390ffb\> > – Channel PJSIP/Telekom\_in-00000000 joined ‘simple\_bridge’ basic-bridge \<63d9fea4-4156-4152-9798-6aa880390ffb\> > \> Bridge 63d9fea4-4156-4152-9798-6aa880390ffb: switching from simple\_bridge technology to native\_rtp > \> Remotely bridged ‘PJSIP/Telekom\_in-00000000’ and ‘PJSIP/22-00000001’ - media will flow directly between them > \> Remotely bridged ‘PJSIP/Telekom\_in-00000000’ and ‘PJSIP/22-00000001’ - media will flow directly between them > \> 0x75d21378 – Probation passed - setting RTP source address to 192.168.178.57:5008 > \> 0x75d19600 – Probation passed - setting RTP source address to 217.0.7.7:41110 > – Channel PJSIP/Telekom\_in-00000000 left ‘native\_rtp’ basic-bridge \<63d9fea4-4156-4152-9798-6aa880390ffb\> > == Spawn extension (Telekom\_in, 022528301471, 1) exited non-zero on ‘PJSIP/Telekom\_in-00000000’ > – Channel PJSIP/22-00000001 left ‘native\_rtp’ basic-bridge \<63d9fea4-4156-4152-9798-6aa880390ffb\> > – Executing [022528301471@Telekom\_in:1] Dial(“PJSIP/Telekom\_in-00000002”, “PJSIP/22&SCCP/11,30”) in new stack > \> SCCP: sccp\_requestChannel returned Line 11 not currently registered → Try again later > > [Feb 7 20:55:40] WARNING[3844][C-00000002]: app\_dial.c:2530 dial\_exec\_full: Unable to create channel of type ‘SCCP’ (cause 44 - Requested channel not available) > – Called PJSIP/22 > – PJSIP/22-00000003 is ringing > – PJSIP/22-00000003 answered PJSIP/Telekom\_in-00000002 > – Channel PJSIP/22-00000003 joined ‘simple\_bridge’ basic-bridge \<5d53b3c0-a7b8-4405-9977-ea1adfcb504d\> > – Channel PJSIP/Telekom\_in-00000002 joined ‘simple\_bridge’ basic-bridge \<5d53b3c0-a7b8-4405-9977-ea1adfcb504d\> > \> Bridge 5d53b3c0-a7b8-4405-9977-ea1adfcb504d: switching from simple\_bridge technology to native\_rtp > \> Remotely bridged ‘PJSIP/Telekom\_in-00000002’ and ‘PJSIP/22-00000003’ - media will flow directly between them > \> Remotely bridged ‘PJSIP/Telekom\_in-00000002’ and ‘PJSIP/22-00000003’ - media will flow directly between them > \> 0x75d296a8 – Probation passed - setting RTP source address to 192.168.178.57:5014 > – Contact 20/sip:20@192.168.178.57:5070 is now Unknown. RTT: 0.000 msec > – Contact 21/sip:21@192.168.178.57:5070 is now Unknown. RTT: 0.000 msec > – Channel PJSIP/22-00000003 left ‘native\_rtp’ basic-bridge \<5d53b3c0-a7b8-4405-9977-ea1adfcb504d\> > – Channel PJSIP/Telekom\_in-00000002 left ‘native\_rtp’ basic-bridge \<5d53b3c0-a7b8-4405-9977-ea1adfcb504d\> > == Spawn extension (Telekom\_in, 022528301471, 1) exited non-zero on ‘PJSIP/Telekom\_in-00000002’ > – Contact 22/sip:22@192.168.178.57:5070 is now Unknown. RTT: 0.000 msec > – Contact 23/sip:23@192.168.178.57:5070 is now Unknown. RTT: 0.000 msec Please advise how to solve the problem. Do you need another debug log? ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [February 7, 2017, 8:04pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/15 "2017-02-07T20:04:45Z")
What is the Asterisk configuration for pjsip.conf? ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 7, 2017, 8:20pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/16 "2017-02-07T20:20:04Z")
Sorry, please explain… I’m not sure what you would like to know… ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [February 7, 2017, 8:24pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/17 "2017-02-07T20:24:04Z")
What is the configuration present in the pjsip.conf file, without passwords? ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 7, 2017, 8:46pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/18 "2017-02-07T20:46:05Z")
[global] type=global user\_agent=LasRamblas endpoint\_identifier\_order=ip,username ;default\_from\_user=012345662008 [transport-udp] type=transport protocol=udp bind=0.0.0.0 local\_net=192.168.178.0/24 local\_net=10.10.10.0/27 external\_media\_address=[vpn.test.de](http://vpn.test.de) external\_signaling\_address=[vpn.test.de](http://vpn.test.de) [transport-tcp] type=transport protocol=tcp bind=0.0.0.0 local\_net=192.168.178.0/24 local\_net=10.10.10.0/27 [acl] type=acl deny=0.0.0.0/0.0.0.0 ; Telekom und sipgate permit=217.0.0.0/13 permit=217.10.64.0/20 permit=217.116.112.0/20 permit=212.9.32.0/19 ; eigenes LAN permit=192.168.178.0/24 permit=10.10.10.0/27 ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* ; externe SIP-Accounts ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [01\_Telekom\_012345662008] type=registration transport=transport-udp outbound\_auth=01\_Telekom\_012345662008\_auth server\_uri=sip:tel.t-online.de [client\_uri=sip:+4912345662008@tel.t-online.de](mailto:client_uri=sip:+4912345662008@tel.t-online.de) contact\_user=012345662008 retry\_interval=60 forbidden\_retry\_interval=300 expiration=480 auth\_rejection\_permanent=false [01\_Telekom\_012345662008\_auth] type=auth auth\_type=userpass [password=12340001@t-online.de](mailto:password=12340001@t-online.de) username=012345662008 [realm=tel.t-online.de](http://realm=tel.t-online.de) [01\_Telekom\_012345662008\_out] type=endpoint transport=transport-udp context=unspecified disallow=all allow=g722 allow=alaw outbound\_auth=01\_Telekom\_012345662008\_auth aors=01\_Telekom\_0123456620080\_out callerid=012345662008 from\_user=012345662008 from\_domain=[tel.t-online.de](http://tel.t-online.de) timers=no rtp\_symmetric=yes [01\_Telekom\_\_012345662008\_out] type=aor [contact=sip:+4912345662008@tel.t-online.de](mailto:contact=sip:+4912345662008@tel.t-online.de) [Telekom\_in] type=endpoint transport=transport-udp context=Telekom\_in disallow=all allow=g722 allow=alaw outbound\_auth=01\_Telekom\_012345662008\_auth [Telekom\_in] type=identify endpoint=Telekom\_in match=217.0.0.0/13 ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [02\_Telekom\_012345601470] type=registration transport=transport-udp outbound\_auth=02\_Telekom\_012345601470\_auth server\_uri=sip:tel.t-online.de [client\_uri=sip:+4912345601470@tel.t-online.de](mailto:client_uri=sip:+4912345601470@tel.t-online.de) contact\_user=012345601470 retry\_interval=60 forbidden\_retry\_interval=300 expiration=480 auth\_rejection\_permanent=false [02\_Telekom\_012345601470\_auth] type=auth auth\_type=userpass [password=12340001@t-online.de](mailto:password=12340001@t-online.de) username=012345601470 [realm=tel.t-online.de](http://realm=tel.t-online.de) [02\_Telekom\_012345601470\_out] type=endpoint transport=transport-udp context=unspecified disallow=all allow=g722 allow=alaw outbound\_auth=02\_Telekom\_012345601470\_auth aors=02\_Telekom\_012345601470\_out callerid=012345601470 from\_user=012345601470 from\_domain=[tel.t-online.de](http://tel.t-online.de) timers=no rtp\_symmetric=yes [02\_Telekom\_012345601470\_out] type=aor [contact=sip:+4912345601470@tel.t-online.de](mailto:contact=sip:+4912345601470@tel.t-online.de) ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [03\_Telekom\_012345601471] type=registration transport=transport-udp outbound\_auth=03\_Telekom\_012345601471\_auth server\_uri=sip:tel.t-online.de [client\_uri=sip:+4912345601471@tel.t-online.de](mailto:client_uri=sip:+4912345601471@tel.t-online.de) contact\_user=012345601471 retry\_interval=60 forbidden\_retry\_interval=300 expiration=480 auth\_rejection\_permanent=false [03\_Telekom\_012345601471\_auth] type=auth auth\_type=userpass [password=12340001@t-online.de](mailto:password=12340001@t-online.de) username=012345601471 [realm=tel.t-online.de](http://realm=tel.t-online.de) [03\_Telekom\_012345601471\_out] type=endpoint transport=transport-udp context=unspecified disallow=all allow=g722 allow=alaw outbound\_auth=03\_Telekom\_012345601471\_auth aors=03\_Telekom\_012345601471\_out callerid=012345601471 from\_user=012345601471 from\_domain=[tel.t-online.de](http://tel.t-online.de) timers=no rtp\_symmetric=yes [03\_Telekom\_012345601471\_out] type=aor [contact=sip:+4912345601471@tel.t-online.de](mailto:contact=sip:+4912345601471@tel.t-online.de) ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [04\_sipgate\_Joerg\_012345669796] type=registration retry\_interval=20 max\_retries=10 contact\_user=1234567 expiration=120 transport=transport-udp outbound\_auth=04\_sipgate\_Joerg\_auth client\_uri=sip:1234567@sipgate.de:5060 server\_uri=sip:sipgate.de:5060 [04\_sipgate\_Joerg\_auth] type=auth username=1234567 password=ABCDEFG [04\_sipgate\_Joerg\_aor] type=aor [contact=sip:1234567@sipgate.de](mailto:contact=sip:1234567@sipgate.de) [04\_sipgate\_Joerg\_identity] type=identify endpoint=04\_sipgate\_Joerg [match=sipgate.de](http://match=sipgate.de) [04\_sipgate\_Joerg] type=endpoint context=04\_sipgate\_Joerg-in dtmf\_mode=rfc4733 disallow=all allow=alaw rtp\_symmetric=yes force\_rport=yes rewrite\_contact=yes timers=yes from\_user=1234567 from\_domain=[sipgate.de](http://sipgate.de) language=en outbound\_auth=04\_sipgate\_Joerg\_auth aors=04\_sipgate\_Joerg\_aor ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* ; interne SIP-Accounts ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [20] type=endpoint transport=transport-udp context=internalsip disallow=all allow=g722 allow=alaw auth=20-auth aors=20 mailboxes=20 [20-auth] type=auth auth\_type=userpass password=1234 username=20 realm=domain.local [20] type=aor max\_contacts=1 remove\_existing=true [20] type=identify endpoint=20 ;match=192.168.178.57 ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [21] type=endpoint transport=transport-udp context=internalsip disallow=all allow=g722 allow=alaw auth=21-auth aors=21 mailboxes=21 [21-auth] type=auth auth\_type=userpass password=1234 username=21 realm=domain.local [21] type=aor max\_contacts=1 remove\_existing=true ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [22] type=endpoint transport=transport-udp context=internalsip disallow=all allow=g722 allow=alaw auth=22-auth aors=22 mailboxes=22 [22-auth] type=auth auth\_type=userpass password=1234 username=22 realm=domain.local [22] type=aor max\_contacts=1 remove\_existing=true ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [23] type=endpoint transport=transport-udp context=internalsip disallow=all allow=g722 allow=alaw auth=23-auth aors=23 mailboxes=23 [23-auth] type=auth auth\_type=userpass password=1234 username=23 realm=domain.local [23] type=aor max\_contacts=1 remove\_existing=true ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [30] type=endpoint transport=transport-udp context=internalsip disallow=all allow=g722 allow=alaw auth=30-auth aors=30 mailboxes=30 [30-auth] type=auth auth\_type=userpass password=1234 username=30 realm=domain.local [30] type=aor max\_contacts=1 remove\_existing=true ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [31] type=endpoint transport=transport-udp context=internalsip disallow=all allow=g722 allow=alaw auth=31-auth aors=31 mailboxes=31 [31-auth] type=auth auth\_type=userpass password=5678 username=31 realm=domain.local [31] type=aor max\_contacts=1 remove\_existing=true ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [32] type=endpoint transport=transport-udp context=internalsip disallow=all allow=g722 allow=alaw auth=32-auth aors=32 mailboxes=32 [32-auth] type=auth auth\_type=userpass password=1234 username=32 realm=domain.local [32] type=aor max\_contacts=1 remove\_existing=true ;\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\* [98] type=endpoint transport=transport-udp context=internalsip disallow=all allow=g722 allow=alaw auth=98-auth aors=98 mailboxes=98 [98-auth] type=auth auth\_type=userpass password=123456 username=98 realm=domain.local [98] type=aor max\_contacts=1 remove\_existing=true [98] type=identify endpoint=98 match=192.168.178.0/27 match=10.10.10.0/27 ---
**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\ **Post date:** [February 7, 2017, 8:51pm UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/19 "2017-02-07T20:51:58Z")
I would suggest using an explicit IP address for the external\_media\_address and external\_signaling\_address options. You will also need to ensure the external RTP ports (10000-20000 by default) are forwarded to Asterisk. You will also want to set the direct\_media option to no on any endpoints which are behind NAT. ---
**Author:** ![Tenere\_750](https://avatars.discourse-cdn.com/v4/letter/t/65b543/32.png) [@Tenere\_750](https://community.asterisk.org/u/Tenere_750)\ **Post date:** [February 8, 2017, 5:10am UTC](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642/20 "2017-02-08T05:10:21Z")
Good morning and thanks. As I do have a dynamic external IP I can’t use a stitc IP for external\_media\_address and external\_signaling\_address. OR is there a function/variable within Asterisk to determine and use the external IP? Just to make sure I forwarded ports 10000 to 50000 to Asterisk. I’m using a Cisco 881 router an this is the forwarding part of my config > ip nat pool PortFwd-VoIP 192.168.178.207 192.168.178.207 netmask 255.255.255.0 type rotary > ip nat inside source list 101 interface Dialer1 overload > ip nat inside source static tcp 192.168.178.207 5060 interface Dialer1 5060 > ip nat inside source static udp 192.168.178.207 5060 interface Dialer1 5060 > ip nat inside destination list 110 pool PortFwd-VoIP > ! > access-list 101 permit ip 192.168.178.0 0.0.0.255 any > access-list 110 permit udp any any range 10000 51000 > access-list 111 permit udp any eq 5060 any > access-list 111 permit udp any range 10000 51000 any > access-list 111 deny ip any any log > ! > interface Dialer1 > ip access-group 111 in Setting direct\_media option to no I’ll try as soon as I’m able to test. Regards, Joerg [Next page](https://community.asterisk.org/t/voicemail-not-played-after-one-succesful-call/69642.md?page=2)