# Asterisk Audio Problems **URL:** \ **Category:** Asterisk SIP\ **Created:** [March 21, 2024, 11:42pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706 "2024-03-21T23:42:44Z")\ **Posts on this page:** 20\ **Page:** 1
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 21, 2024, 11:42pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/1 "2024-03-21T23:42:44Z")
asterisk-18.4.0-1.fc36.5.x86\_64 That’s Asterisk built on Fedora 36, installed via dnf Up until about 10 days ago things were going great and “everything” _seemed_ like it was working. Test 1 - my landline to my flowrouteDID - no audio at all Test 2 - my T-Mobile calling my flowraouteDID: - inbound audio works (I hear audio on my T-Mobile) - outbound audio fails - no audio on the far end For each test I 1. called my flowroute DID 2. Listened (to the message or silence) 3. Pressed an extension, and finally since there was no outbound audio 4. was dc by my PBX because no extension received I have tried reviewing tcpdump for RTP but I have not found anything (probably because of my lack of knowledge). Please find: . my flowroute PJsip config (immediatly below . full log files for each test . tcpdump files for each test I would sincerely appreciate any assistance. Thank you very much! [flowroute] type=registration transport=transport-udp-nat outbound\_auth=flowroute\_auth server\_uri=sip:us-east-va.sip.flowroute.com [client\_uri=sip:techprefix@us-east-va.sip.flowroute.com](mailto:client_uri=sip:techprefix@us-east-va.sip.flowroute.com) contact\_user=techprefix retry\_interval=60 [flowroute\_auth] type=auth auth\_type=userpass password=mypassword username=techprefix [realm=sip.flowroute.com](http://realm=sip.flowroute.com) [flowroute] type=aor contact=sip:us-east-va.sip.flowroute.com [flowroute] type=endpoint context=from-trunk disallow=all allow=ulaw outbound\_auth=flowroute\_auth aors=flowroute direct\_media=no [flowroute] type=identify endpoint=flowroute match=34.210.91.112/28 match=34.226.36.32/28 [full-landline-to-flowroute.txt](https://community.asterisk.org/uploads/short-url/wHC04UMM6QojRXg7ikk49dCOSqU.txt) (142.5 KB) [full-T-Mobile-line-to-flowroute.txt](https://community.asterisk.org/uploads/short-url/q9oLiqJPivorLfmlsp7reQay8ZW.txt) (152.8 KB) [watch-Landline-to-flowroute-tcpip240321-16-02-29.txt](https://community.asterisk.org/uploads/short-url/kH3u34kSmDYNvphn1unSDIQFA4s.txt) (10.4 KB) [watch-T-Mobile-to-flowroute-tcpip240321-16-44-12.txt](https://community.asterisk.org/uploads/short-url/w9ioqAoSO8COLi4HQ799oxXzWVi.txt) (12.2 KB) ---
**Author:** ![FrancoSmash](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/francosmash/32/6689_2.png) [@FrancoSmash](https://community.asterisk.org/u/FrancoSmash)\ **Post date:** [March 22, 2024, 12:04am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/2 "2024-03-22T00:04:02Z")
You need to configure your rtp.conf file Ports 10000 - 25000 udp ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 12:48am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/3 "2024-03-22T00:48:49Z")
OK, that’s embarrassing. I’m sorry. Please find attached: . my rtp.conf . updated tcpdumps [rtp.conf.txt](https://community.asterisk.org/uploads/short-url/mEzDBJCa1VbQLGQ73A1gO7TdQdu.txt) (6.4 KB) [watch-landline-to-flowroute-tcpip240321-20-22-09.txt](https://community.asterisk.org/uploads/short-url/fUzXqphkEk3qKPDIMeYtffh0PK2.txt) (352.9 KB) [watch-T-Mobile=phone-to-flowroute-tcpip240321-20-23-50.txt|attachment](https://community.asterisk.org/uploads/short-url/zuvVoDuicsVbEqeWHN7pPByVYTY.txt) (280.0 KB) ---
**Author:** ![FrancoSmash](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/francosmash/32/6689_2.png) [@FrancoSmash](https://community.asterisk.org/u/FrancoSmash)\ **Post date:** [March 22, 2024, 1:07am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/4 "2024-03-22T01:07:50Z")
[general] rtpstart=10000 rtpend=25000 icesupport=true [stunaddr:19302=stun.l.google.com](http://stunaddr:19302=stun.l.google.com) ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 4:37am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/5 "2024-03-22T04:37:21Z")
rtp.conf updated and attached. I restarted asterisk. Re-tried tests–same results. Test 1 - my landline to my flowrouteDID - no audio at all Test 2 - my T-Mobile calling my flowraouteDID: Please find attached full logs and tcpdumps. Thank you very much for your help [full-landline-to-flowroute.txt](https://community.asterisk.org/uploads/short-url/p04GyC09mGSxJx9piOj5wNRfcP1.txt) (157.4 KB) [full-T-Mobile-line-to-flowroute.txt](https://community.asterisk.org/uploads/short-url/ibxk3kMLDB6xxBSQE0yvZqUxyyU.txt) (149.2 KB) [rtp.conf.txt](https://community.asterisk.org/uploads/short-url/1YBa8TSwTlY6thAdhzm3GohX4SL.txt) (6.5 KB) [watch-landline-to-flowroute-tcpip240321-23-50-43.txt](https://community.asterisk.org/uploads/short-url/6lqQzyNJFSCs3uPeIIe7mtPDOva.txt) (267.0 KB) [watch-T-Mobile=phone-to-flowroute-tcpip240321-23-58-14.txt|attachment](https://community.asterisk.org/uploads/short-url/ykbU7dTAfOMwZLFfs6sJWfqF5nb.txt) (314.5 KB) ---
**Author:** ![jmokoena](https://avatars.discourse-cdn.com/v4/letter/j/b5ac83/32.png) [@jmokoena](https://community.asterisk.org/u/jmokoena)\ **Post date:** [March 22, 2024, 5:58am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/6 "2024-03-22T05:58:07Z")
Hi Check if g729 codec is installed. Regards Johannes ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 6:05am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/7 "2024-03-22T06:05:51Z")
Yes-and: Just to be thorough: ```auto *CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. ID TYPE NAME FORMAT DESCRIPTION ------------------------------------------------------------------------------------------------ 31 image png png (PNG Image) 6 audio g726 g726 (G.726 RFC3551) 4 audio alaw alaw (G.711 a-law) 2 audio g723 g723 (G.723.1) 20 audio speex speex (SpeeX) 21 audio speex speex16 (SpeeX 16khz) 22 audio speex speex32 (SpeeX 32khz) 24 audio g722 g722 (G722) 25 audio siren7 siren7 (ITU G.722.1 (Siren7, licensed from Polycom)) 32 video h261 h261 (H.261 video) 33 video h263 h263 (H.263 video) 8 audio adpcm adpcm (Dialogic ADPCM) 36 video h265 h265 (H.265 video) 44 audio silk silk8 (SILK Codec (8 KHz)) 45 audio silk silk12 (SILK Codec (12 KHz)) 46 audio silk silk16 (SILK Codec (16 KHz)) 47 audio silk silk24 (SILK Codec (24 KHz)) 28 audio g719 g719 (ITU G.719) 34 video h263p h263p (H.263+ video) 35 video h264 h264 (H.264 video) 19 audio g729 g729 (G.729A) 9 audio slin slin (16 bit Signed Linear PCM) 10 audio slin slin12 (16 bit Signed Linear PCM (12kHz)) 11 audio slin slin16 (16 bit Signed Linear PCM (16kHz)) 12 audio slin slin24 (16 bit Signed Linear PCM (24kHz)) 13 audio slin slin32 (16 bit Signed Linear PCM (32kHz)) 14 audio slin slin44 (16 bit Signed Linear PCM (44kHz)) 15 audio slin slin48 (16 bit Signed Linear PCM (48kHz)) 16 audio slin slin96 (16 bit Signed Linear PCM (96kHz)) 17 audio slin slin192 (16 bit Signed Linear PCM (192kHz)) 3 audio ulaw ulaw (G.711 u-law) 18 audio lpc10 lpc10 (LPC10) 27 audio testlaw testlaw (G.711 test-law) 43 audio none none ( codec) 42 image t38 t38 (T.38 UDPTL Fax) 39 video vp9 vp9 (VP9 video) 38 video vp8 vp8 (VP8 video) 5 audio gsm gsm (GSM) 37 video mpeg4 mpeg4 (MPEG4 video) 23 audio ilbc ilbc (iLBC) 40 text red red (T.140 Realtime Text with redundancy) 41 text t140 t140 (Passthrough T.140 Realtime Text) 29 audio opus opus (Opus Codec) 30 image jpeg jpeg (JPEG image) 7 audio g726aal2 g726aal2 (G.726 AAL2) 1 audio codec2 codec2 (Codec 2) 26 audio siren14 siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom)) *CLI> ``` ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 6:23am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/8 "2024-03-22T06:23:10Z")
> [@jmokoena](#): > > Hi > > Check if g729 codec is installed. > > Regards > Johannes Hi Johannes! This isn’t impatience–I jusr realized I may not have either understood or correctly answered your question. Please let me know if you need additional information. Thank you! ---
**Author:** ![EkFudrek](https://avatars.discourse-cdn.com/v4/letter/e/ce73a5/32.png) [@EkFudrek](https://community.asterisk.org/u/EkFudrek)\ **Post date:** [March 22, 2024, 7:18am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/9 "2024-03-22T07:18:19Z")
You need to describe your network configuration. Audio problems are often related to NAT problems. STUN and other NAT helpers are usually not necessary once you understand your network and have some control over your router. Also, since you mentioned Fedora, there’s always a good chance that SELinux will break things when it’s active. I think it makes only sense to look at the Asterisk configuration, if it is explicitly excluded that the root of the problem is outside Asterisk. This usually means a couple of network tests. ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 8:03am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/10 "2024-03-22T08:03:11Z")
> [@EkFudrek](#): > > You need to describe your network configuration. Asterisk PBX on 192.168.x.x network Edge server forwards: 5060udp to PBX 10000-25000 udp to PBX with regards to NAT, does this cover everything: ```auto [transport-udp-nat] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.1.0/24 external_media_address=mypublicIPaddr external_signaling_address=mypublicIPaddr ``` > [@EkFudrek](#): > > Also, since you mentioned Fedora, there’s always a good chance that SELinux will break things when it’s active. ```auto ]# getenforce Permissive ``` So SELinux is active, but permissive. I realize it may be a problem and beg any help that can be provided on this issue–I seem to remember it is possible to “set it to off” with a mandatory reboot. Should I try that? Also,I just turned the local firewall (on the PBX) off and tested to find both tests still failing in the same manner. I have restarted that firewall but can easily take it down, as needed. What network tests did you have in-mind? Is there anything else? And thank you very much! ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [March 22, 2024, 9:08am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/11 "2024-03-22T09:08:27Z")
You should know if the G.729 codec is installed, as, if using the supported one, you will have had to pay for it. However the command you need is module show, see, for example, [Your Asterisk Module Is Loaded, But Is It Running?](https://www.sangoma.com/your-asterisk-module-is-loaded-but-is-it-running/) ---
**Author:** ![EkFudrek](https://avatars.discourse-cdn.com/v4/letter/e/ce73a5/32.png) [@EkFudrek](https://community.asterisk.org/u/EkFudrek)\ **Post date:** [March 22, 2024, 11:51am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/12 "2024-03-22T11:51:55Z")
> [@ThePowerTool](#): > > with regards to NAT, does this cover everything You may be right, or not. You need to capture some VoIP traffic on the LAN and the WAN side of your router (or check the the firewall states, in case this is available). I know cases where the service provider reuses the ports from incoming traffic (to their servers) for signalling and then port forwarding is relatively pointless as it is assumed that the existing connection stays open all the time. In cases like that you need to configre something like “outbound NAT rules” (pfSense jargon) or so-called “1-1 NAT”, which is a bit more general. STUN may help in some scenarios, but I tend to forget which cases can be handled, so I’d rather look at what is actuall going on. ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 6:44pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/13 "2024-03-22T18:44:01Z")
> [@david551](#): > > You should know if the G.729 codec is installed, as, if using the supported one, you will have had to pay for it. I installed Asterisk using “dnf install asterisk”. I have only been using the default installation and no purchased modules to the best of my knowledge given I trust the Fedora repo to have a legitimate installation. Perhaps you can help me determine if something is wrong with regards to this. Here is what I have: ```auto *CLI> module show like 729 Module Description Use Count Status Support Level format_g729.so Raw G.729 data 0 Running core res_format_attr_g729.so G.729 Format Attribute Module 1 Running core ``` When I place a call to flowroute I see: ```auto *CLI> core show channels verbose Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgeID PJSIP/flowroute-0000 from-trunk myflowrouteDID 19 Up Read digit,custom/no-solicitor +myLandline 00:00:07 1 active channel 1 active call 3 calls processed *CLI> core show codec 19 19 G.729A (g729) ``` ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 7:44pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/14 "2024-03-22T19:44:26Z")
> [@EkFudrek](#): > > You need to capture some VoIP traffic on the LAN and the WAN side of your router (or check the the firewall states, Not counting my op (where I totally screwed the tcpdump captures), subsequent tcpdump-captures should contain all inbound and outbound packets from my PBX. Is it possible what you are discussing is in those “watch” files? Additionally, and I was hoping to avoid discussing this because I want to focus on one issue at a time, I have my PBX connected to another Asterisk PBX as follows: MyPBX (192.168.1.x) NAT MySPEdgerouter to Internet Internet to private router (192.168.10.x) private router to remotePBX (192.168.10.x) NAT I am able to call and receive calls over this connection with audio working normally. That-said, it uses a different codec: ```auto *CLI> core show channels verbose Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgeID PJSIP/6003-00000005 my-phone 2058 1 Up Dial PJSIP/xxx(remoteext)@RemotePBX myCIDdata 00:00:16 049e372d-4425-4a41-8 PJSIP/RemotePBX-0000 from-trunk 1 Up AppDial (Outgoing Line) xxxx 00:00:16 049e372d-4425-4a41-8 2 active channels 1 active call 5 calls processed *CLI> core show codec 1 1 Codec 2 (codec2) ``` Perhaps that means I am having a problem with my g729 codec??? ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [March 22, 2024, 8:15pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/15 "2024-03-22T20:15:20Z")
You only have the format modules for G.729, so you can only do pass through for it. ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 8:30pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/16 "2024-03-22T20:30:38Z")
I am not sure what to ask. I think what you are telling me is that (from my above osts) I am using g729 but don’t have the codec. I also thought my config (from op) showing: ```auto disallow=all allow=ulaw ``` means my Asterisk PBX shouldn’t use g729. But I am getting no-audio because it actually is g729. I would sincerely appreciate any information to help me figure this out. My thanks, in-advance!!! ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [March 22, 2024, 8:48pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/17 "2024-03-22T20:48:25Z")
I’m assuming you were asked about G.729 because you were trying to use it. I haven’t looked at all the details. ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 22, 2024, 11:30pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/18 "2024-03-22T23:30:36Z")
I _thought_ this indicated the g729 codec in-use: ```auto *CLI> core show channels verbose Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgeID PJSIP/flowroute-0000 from-trunk 17044453955 19 Up Read digit,custom/no-solicitor +myLandline 00:00:07 1 active channel 1 active call 3 calls processed *CLI> core show codec 19 19 G.729A (g729) ``` Am I correct that the 19 in the “19 Up” is the codec # so the subsequent command displays codec 19 as g729? And if I have: ```auto disallow=all allow=ulaw ``` Why would my PBX then use g729 rather than the allowed ulaw? Let me know what information to provide and I will do my best. I’d really like to get my flowroute connection audio working. Thanks!!! ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [March 23, 2024, 11:31am UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/19 "2024-03-23T11:31:52Z")
> [@ThePowerTool](#): > > Am I correct that the 19 in the “19 Up” is the codec # so the subsequent command displays codec 19 as g729? No. 19 is the step number in the dialplan and Up means the call has been answered. ---
**Author:** ![ThePowerTool](https://avatars.discourse-cdn.com/v4/letter/t/ea666f/32.png) [@ThePowerTool](https://community.asterisk.org/u/ThePowerTool)\ **Post date:** [March 23, 2024, 6:38pm UTC](https://community.asterisk.org/t/asterisk-audio-problems/101706/20 "2024-03-23T18:38:38Z")
Can you offer any suggestions? I re-read your posts but nothing jumped out at me that I missed (and you requested from me). Did I miss something? Thanks! [Next page](https://community.asterisk.org/t/asterisk-audio-problems/101706.md?page=2)