sip_nat.conf help for audio dropout ---UPDATE---

General Help
1

FreePBX 2.8.1.0
asterisk 1.8.2
dell poweredge 1850
centOS 5.5
asus wl-500g premium v2 dd-wrt mega
behind NAT
linksys spa922 & spa942

recent change:
upgrade fail2ban to this: http://pbxinaflash.com/forum/showthread.php?t=9410

Issue:
Anywhere from 1 minute 30 seconds to 3 minutes into an incoming or outgoing call, the audio drops and becomes one way. It seems that if the call is inbound, I can hear them and if I place the call when the audio drops, they will hear me. Keeping the connection open for (up to ten minutes) does not resolve the issue, so the only way the audio is restored is when you hang up and dial back. This started about Wednesday night and I can’t seem to figure out the issue. I have not closed any ports on the router or on the server. Ping and latency results from various VOIP related sites are excellent; the log file does not point any failure out like I’ve seen in previous logs. It does not matter what phone or extension I use to make/receive the call and rebooting the server doesn’t resolve the issue.

VoIP test statistics

Jitter: you --> server: 0.7 ms

Jitter: server --> you: 0.8 ms

Packet loss: you --> server: 0.0 %

Packet loss: server --> you: 0.0 %

Packet discards: 0.0 %

Packets out of order: 0.0 %

Estimated MOS score: 4.2

It looks like the download consistency is a little sucky

Speed test statistics

Download speed: 2422120 bps

Upload speed: 1945968 bps

Download consistency of service: 65 %

Upload consistency of service: 99 %

Download test type: socket

Upload test type: socket

Maximum TCP delay: 109 ms

Average download pause: 9 ms

Minimum round trip time to server: 59 ms

Average round trip time to server: 59 ms

Estimated download bandwidth: 24800000bps

Route concurrency: 10.238964

Download TCP forced idle: 64 %

Maximum route speed: 8886096bps

My voip provider is VOIPO.com and they don’t really support BYOD so they just suggested things to check like the ports and ping tests.

Are there any freepbx settings I could edit?

Thanks in advance,
Sean

2

So I got tired of trying things that were not working so I re-imaged and re-installed everything. Right now things seem to be going well, but something kinda weird is now happening. The server has two NIC cards but traffic is only being used on one of the ports, 192.168.10.22.
I only only used this IP for everything but previously 192.168.10.24 would also carry SIP traffic and talk to the VOIP provider.

I think the original issue was something I’ve been overlooking this whole time.
the contact information at the voip provider was [email protected] instead of the servers public IP address.

3

Can you do a test real quick, go to tools - Asterisk sip settings, set your external ip as 0.0.0.0 . I’m not saying this will fix anything but I would like to know your result.

4

Typically I use a dynamic host name since my ip assigned by the internet provider is not static.

Setting this to 0.0.0.0 did not connect me to my VOIP provider and I monitored the activity on the router and nothing really happened. When I dialled out the first two times, ringing occurred and then my own music on hold came on.
The time after that, ringing occurred and never connected me to the number I dialled.

5

phone conservations are now lasting 7+ minutes so at least I can talk to people for a little bit and not think its going to silence my audio right away; however, this is not always the case but it is becoming more common.

The recent changes I made:

-limited the number of upd ports open to 20001
-disabled cron job that rotated the asterisk logs monthly. No idea what was causing an issue but it was not rotating correctly.

I don’t think it is my internet provider because the callers audio still remains and as far as I can hear during the call, it never breaks up or misses any packets either way.

6

new code

http://pastebin.com/srjJh4fs

7

monitoring the traffic from router to server and receive this message as soon as the audio drops

tcpdump dst 192.168.10.24
21:02:24.577385 IP sip-.com > asterisk2: ICMP sip-.com udp port 50240 unreachable, length 208

Why would it be at port 50240? I have 10000-20000 UDP open I have not seen documentation to have ports higher than that open.

8

life is much better with pastebin. thanks for the suggestion.

9

Hi

Sorry - I didn’t type that quite correctly earlier.

ngrep -d any -P “” -W byline -T -i -t “805xxxxxx3” port 5060

Brian

10

Thanks, I’ll try that code out when the phone is in my possession.

Did another tcpdump on the router and this time the audio lasted only a few seconds. the code is at http://pastebin.com/vmLQ3W0h

line 698 is interesting…

11:22:18.143387 IP human > asterisk: ICMP human udp port 5062 unreachable, length 556

All of the sip-byod.voipwelcome.com.61594 > asterisk.17688: UDP, length 172 lines occurred until I hung the phone up.

11

pastebin for code ngrep -d any -P “” -W byline -T -i -t “805xxxxxx3” port 5060
http://pastebin.com/y0FyLBgL
the line BYE sip:18xxxx
is when I hang up the asterisk phone since the audio drops

on the router the commands
tcpdum dst 192.168.10.22
and
tcpdum dst 192.168.10.24

Don’t seem to show much of anything other than

21:30:04.879636 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:04.899914 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:04.919469 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:04.939414 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:04.959302 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:04.979809 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:04.999956 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:05.020224 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:05.029439 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:05.049235 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:05.069815 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:05.089330 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:05.110340 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172
21:30:05.129667 IP sip-voipwelcome.com.56xxx > asterisk.14704: UDP, length 172

and

21:30:25.212834 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.232942 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.253140 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.273510 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.292795 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.313054 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.332619 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.353154 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.372653 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.392641 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.412836 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.432629 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
21:30:25.453597 IP sip-voipwelcome.com.56xxx > asterisk2.14704: UDP, length 172
12

I receive this message while just monitoring
-bash-3.2# ngrep -d any -P “” -W byline port 5060

Warning: 392 67.xxx.xx.18:5060 “Noisy feedback tells: pid=11950 req_src_ip=70.xxx.xxx.x4 req_src_port=3349 in_uri=sip:sip.com out_uri=sip:sip.voipwelcome.com via_cnt==1”

13

Do I need to post more information to maybe help someone point me to the right direction to resolve this?

14

thanks for looking at that. How could I do a SIP trace?

the original code you provided gives this error

-bash-3.2# ngrep -d any -P -W byline -T -i -t 805xxxxxx3 port 5060
interface: any
pcap compile: syntax error in filter expression

ssh logs from my ddwrt router using tcpdump dst 192.168.10.22

human is the router name
Please see the code at: http://pastebin.com/dRYpuHDQ

I called my asterisk box on google voice and as soon as the audio dropped (1m 30 sec in), the below started to appear very fast

10:52:11.213381 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.232871 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.252692 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.272655 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.292611 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.312685 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.332695 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.352676 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.373017 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
10:52:11.392550 IP yu-in-f126.1e100.net.19295 > asterisk.40060: UDP, length 172
15

Please use pastebin for long log files. Thank you.

16

Hello,

Looks like the linksys hangs up…

U 192.168.10.28:5061 -> 192.168.10.22:5060
BYE sip:[email protected]:5060 SIP/2.0…Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-64476ee9;rport…From: sip:[email protected];tag=208e9640
33b553b7o0…To: sip:[email protected];tag=as776ff54f…Call-ID: [email protected]…CSeq: 103 BYE…Max-Forwards: 70…Authorization: Diges
t username=“2600”,realm=“asterisk”,nonce=“6c4109ed”,uri=“sip:[email protected]:5060”,algorithm=MD5,response=“72fb53380ee3899dbc186ee7a08b3211”…User-A
gent: Linksys/SPA922-6.1.5(a)…Content-Length: 0…

though there is an options just before that U 192.168.10.22:5060 -> 192.168.10.28:5061
OPTIONS sip:[email protected]:5061 SIP/2.0…Via: SIP/2.0/UDP 192.168.10.22:5060;branch=z9hG4bK55f6a2e7;rport…Max-Forwards: 70…From: “Unknown” <sip:Unkn
[email protected]>;tag=as23306105…To: sip:[email protected]:5061…Contact: sip:[email protected]:5060…Call-ID: 2301e1b5473f82dd276da2fa33e2f7de
@192.168.10.22:5060…CSeq: 102 OPTIONS…User-Agent: FPBX-2.8.1(1.8.2)…Date: Fri, 25 Feb 2011 01:56:04 GMT…Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFE
R, SUBSCRIBE, NOTIFY, INFO, PUBLISH…Supported: replaces, timer…Content-Length: 0…

would really need to see the SIP provider traces also.

17

Hello,

Can I point out a couple of things that this is most likely to be.

Have you got rtptimeout parameter set in sip.conf ? If so turn it off.

I have not used bandwidth but it’s possible that they are sending a reinvite at some point during the media flow that either asterisk or an ALG firewall in the path is not handling correctly.

to debug this, dial a number say 6171234567 and then use ngrep:

ngrep -d any -P -W byline -T -i -t 6171234567 port 5060

this will capture all SIP packets related to the call. If you post this trace here it should point to the issue straight away.

hope this helps.
Brian

18

So I made a 30 second phone call to a land line and left a message. Pretty sure the audio dropped since the answering machine hung up on me.

192.168.10.22 is the asterisk server and 192.168.10.28 is the spa922 with extensions 2600

-bash-3.2# ngrep 192.168.10.28 port 5060
interface: eth0 (192.168.10.0/255.255.255.0)
filter: (ip) and ( port 5060 )
match: 192.168.10.28

U 192.168.10.28:5061 -> 192.168.10.22:5060
  INVITE sip:[email protected] SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-7d740b67;rport..From: <sip:[email protected]>;tag=208e964033
  b553b7o0..To: <sip:[email protected]>..Call-ID: [email protected]: 101 INVITE..Max-Forwards: 70..Contact: <sip:[email protected]:5
  061>..Expires: 240..User-Agent: Linksys/SPA922-6.1.5(a)..Content-Length: 401..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: re
  places..Content-Type: application/sdp....v=0..o=- 24625784 24625784 IN IP4 192.168.10.28..s=-..c=IN IP4 192.168.10.28..t=0 0..m=audio 16440 RTP/AVP 0 2 4 
  8 18 96 97 98 101..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729a/8000..a=rtpmap:96 G726-40/
  8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv..                    
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-7d740b67;received=192.168.10.28;rport=5061..From: <sip:[email protected]>;ta
  g=208e964033b553b7o0..To: <sip:[email protected]>;tag=as159b9130..Call-ID: [email protected]: 101 INVITE..Server: FPBX-2.8.1(1.8.2)
  ..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..WWW-Authenticate: Digest algorithm=MD5, r
  ealm="asterisk", nonce="6c4109ed"..Content-Length: 0....                                                                                                  
#
U 192.168.10.28:5061 -> 192.168.10.22:5060
  ACK sip:[email protected] SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-7d740b67;rport..From: <sip:[email protected]>;tag=208e964033b55
  3b7o0..To: <sip:[email protected]>;tag=as159b9130..Call-ID: [email protected]: 101 ACK..Max-Forwards: 70..Contact: <sip:[email protected]
  8.10.28:5061>..User-Agent: Linksys/SPA922-6.1.5(a)..Content-Length: 0....                                                                                 
#
U 192.168.10.28:5061 -> 192.168.10.22:5060
  INVITE sip:[email protected] SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-3f1a6b47;rport..From: <sip:[email protected]>;tag=208e964033
  b553b7o0..To: <sip:[email protected]>..Call-ID: [email protected]: 102 INVITE..Max-Forwards: 70..Authorization: Digest username="26
  00",realm="asterisk",nonce="6c4109ed",uri="sip:[email protected]",algorithm=MD5,response="c08cb5bfffd058dc8517c854937432b2"..Contact: <sip:[email protected]
  8.10.28:5061>..Expires: 240..User-Agent: Linksys/SPA922-6.1.5(a)..Content-Length: 401..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supp
  orted: replaces..Content-Type: application/sdp....v=0..o=- 24625784 24625784 IN IP4 192.168.10.28..s=-..c=IN IP4 192.168.10.28..t=0 0..m=audio 16440 RTP/A
  VP 0 2 4 8 18 96 97 98 101..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729a/8000..a=rtpmap:96
   G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv..           
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-3f1a6b47;received=192.168.10.28;rport=5061..From: <sip:[email protected]>;tag=208e
  964033b553b7o0..To: <sip:[email protected]>..Call-ID: [email protected]: 102 INVITE..Server: FPBX-2.8.1(1.8.2)..Allow: INVITE, ACK,
   CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:[email protected]:5060>..Content-Length: 0..
  ..                                                                                                                                                        
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-3f1a6b47;received=192.168.10.28;rport=5061..From: <sip:[email protected]
  >;tag=208e964033b553b7o0..To: <sip:[email protected]>;tag=as776ff54f..Call-ID: [email protected]: 102 INVITE..Server: FPBX-2.8.1(1.
  8.2)..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:[email protected]:5
  060>..Content-Type: application/sdp..Content-Length: 305....v=0..o=root 266103694 266103694 IN IP4 192.168.10.22..s=Asterisk PBX 1.8.2..c=IN IP4 192.168.1
  0.22..t=0 0..m=audio 15202 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=ptime:20..a=sendrecv..                                                                                                 
#############
U 192.168.10.28:5061 -> 192.168.10.22:5060
  NOTIFY sip:192.168.10.22 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-8dc4ad0b;rport..From: <sip:[email protected]>;tag=84ea475db01dcf56o0
  ..To: <sip:192.168.10.22>..Call-ID: [email protected]: 16406 NOTIFY..Max-Forwards: 70..Contact: <sip:[email protected]:5061>..Event: 
  keep-alive..User-Agent: Linksys/SPA922-6.1.5(a)..Content-Length: 0....                                                                                    
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-8dc4ad0b;received=192.168.10.28;rport=5061..From: <sip:[email protected]>;tag=84ea475d
  b01dcf56o0..To: <sip:192.168.10.22>;tag=as0971749e..Call-ID: [email protected]: 16406 NOTIFY..Server: FPBX-2.8.1(1.8.2)..Allow: INVITE
  , ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Content-Length: 0....                                   
######################
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-3f1a6b47;received=192.168.10.28;rport=5061..From: <sip:[email protected]>;tag=208e9640
  33b553b7o0..To: <sip:[email protected]>;tag=as776ff54f..Call-ID: [email protected]: 102 INVITE..Server: FPBX-2.8.1(1.8.2)..Allow: I
  NVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Contact: <sip:[email protected]:5060>..Content-
  Type: application/sdp..Content-Length: 305....v=0..o=root 266103694 266103695 IN IP4 192.168.10.22..s=Asterisk PBX 1.8.2..c=IN IP4 192.168.10.22..t=0 0..m
  =audio 15202 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000
  ..a=fmtp:101 0-16..a=ptime:20..a=sendrecv..                                                                                                               
#
U 192.168.10.28:5061 -> 192.168.10.22:5060
  ACK sip:[email protected]:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-6f6dbc15;rport..From: <sip:[email protected]>;tag=208e9640
  33b553b7o0..To: <sip:[email protected]>;tag=as776ff54f..Call-ID: [email protected]: 102 ACK..Max-Forwards: 70..Authorization: Diges
  t username="2600",realm="asterisk",nonce="6c4109ed",uri="sip:[email protected]",algorithm=MD5,response="c08cb5bfffd058dc8517c854937432b2"..Contact: <s
  ip:[email protected]:5061>..User-Agent: Linksys/SPA922-6.1.5(a)..Content-Length: 0....                                                                   
#
U 192.168.10.28:5061 -> 192.168.10.22:5060
  NOTIFY sip:192.168.10.22 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-f51de872;rport..From: <sip:[email protected]>;tag=84ea475db01dcf56o0
  ..To: <sip:192.168.10.22>..Call-ID: [email protected]: 16407 NOTIFY..Max-Forwards: 70..Contact: <sip:[email protected]:5061>..Event: 
  keep-alive..User-Agent: Linksys/SPA922-6.1.5(a)..Content-Length: 0....                                                                                    
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-f51de872;received=192.168.10.28;rport=5061..From: <sip:[email protected]>;tag=84ea475d
  b01dcf56o0..To: <sip:192.168.10.22>;tag=as0971749e..Call-ID: [email protected]: 16407 NOTIFY..Server: FPBX-2.8.1(1.8.2)..Allow: INVITE
  , ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Content-Length: 0....                                   
#################
U 192.168.10.28:5061 -> 192.168.10.22:5060
  NOTIFY sip:192.168.10.22 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-fea1e596;rport..From: <sip:[email protected]>;tag=84ea475db01dcf56o0
  ..To: <sip:192.168.10.22>..Call-ID: [email protected]: 16408 NOTIFY..Max-Forwards: 70..Contact: <sip:[email protected]:5061>..Event: 
  keep-alive..User-Agent: Linksys/SPA922-6.1.5(a)..Content-Length: 0....                                                                                    
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-fea1e596;received=192.168.10.28;rport=5061..From: <sip:[email protected]>;tag=84ea475d
  b01dcf56o0..To: <sip:192.168.10.22>;tag=as0971749e..Call-ID: [email protected]: 16408 NOTIFY..Server: FPBX-2.8.1(1.8.2)..Allow: INVITE
  , ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Content-Length: 0....                                   
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  OPTIONS sip:[email protected]:5061 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.22:5060;branch=z9hG4bK55f6a2e7;rport..Max-Forwards: 70..From: "Unknown" <sip:Unkn
  [email protected]>;tag=as23306105..To: <sip:[email protected]:5061>..Contact: <sip:[email protected]:5060>..Call-ID: 2301e1b5473f82dd276da2fa33e2f7de
  @192.168.10.22:5060..CSeq: 102 OPTIONS..User-Agent: FPBX-2.8.1(1.8.2)..Date: Fri, 25 Feb 2011 01:56:04 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFE
  R, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Content-Length: 0....                                                                    
#
U 192.168.10.28:5061 -> 192.168.10.22:5060
  SIP/2.0 200 OK..To: <sip:[email protected]:5061>;tag=b9f42129b732dffai0..From: "Unknown" <sip:[email protected]>;tag=as23306105..Call-ID: 2301e1b5473
  [email protected]:5060..CSeq: 102 OPTIONS..Via: SIP/2.0/UDP 192.168.10.22:5060;branch=z9hG4bK55f6a2e7;rport=5060..Server: Linksys/SPA922
  -6.1.5(a)..Content-Length: 0..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: replaces....                                      
###########################
U 192.168.10.28:5061 -> 192.168.10.22:5060
  BYE sip:[email protected]:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-64476ee9;rport..From: <sip:[email protected]>;tag=208e9640
  33b553b7o0..To: <sip:[email protected]>;tag=as776ff54f..Call-ID: [email protected]: 103 BYE..Max-Forwards: 70..Authorization: Diges
  t username="2600",realm="asterisk",nonce="6c4109ed",uri="sip:[email protected]:5060",algorithm=MD5,response="72fb53380ee3899dbc186ee7a08b3211"..User-A
  gent: Linksys/SPA922-6.1.5(a)..Content-Length: 0....                                                                                                      
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-64476ee9;received=192.168.10.28;rport=5061..From: <sip:[email protected]>;tag=208e9640
  33b553b7o0..To: <sip:[email protected]>;tag=as776ff54f..Call-ID: [email protected]: 103 BYE..Server: FPBX-2.8.1(1.8.2)..Allow: INVI
  TE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Content-Length: 0....                                 
##########
U 192.168.10.28:5061 -> 192.168.10.22:5060
  NOTIFY sip:192.168.10.22 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-b7d9bf68;rport..From: <sip:[email protected]>;tag=84ea475db01dcf56o0
  ..To: <sip:192.168.10.22>..Call-ID: [email protected]: 16409 NOTIFY..Max-Forwards: 70..Contact: <sip:[email protected]:5061>..Event: 
  keep-alive..User-Agent: Linksys/SPA922-6.1.5(a)..Content-Length: 0....                                                                                    
#
U 192.168.10.22:5060 -> 192.168.10.28:5061
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.28:5061;branch=z9hG4bK-b7d9bf68;received=192.168.10.28;rport=5061..From: <sip:[email protected]>;tag=84ea475d
  b01dcf56o0..To: <sip:192.168.10.22>;tag=as0971749e..Call-ID: [email protected]: 16409 NOTIFY..Server: FPBX-2.8.1(1.8.2)..Allow: INVITE
  , ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Content-Length: 0....                                   
####exit
115 received, 0 dropped

Does this help?

19

I’m having the same issue. See http://www.freepbx.org/forum/freepbx/users/one-way-audio-on-a-small-percentage-of-incoming-calls

I’ve been working with bandwidth.com for MONTHS with no resolve. The Asterisk server has a static ip with 5060 opened to bandwidth.com, rtp open from anywhere.

Running asterisk 1.6 with freepbx 2.8

Have you resolve your issue with VOIPO.com? Can you provide the ip address the rtp is coming from maybe VOIPO.com is using level3 also.

20

That’s very interesting!

I am not yet ready to throw voipo under the bus but they do business with bandwidth.com and level3. Voipo is not officially BYOD supported so they only advise using STUN on the phones.

There are many things I have done in which could have affected this: upgrading fail2ban, changing the domain name of the server and hostname, installed a couple more small scripts, using voipo which is not officially byod. All of these things could result in the audio dropping off.

I MAY re-image my entire system and try it all over again, but this is not something I am looking forward to as I’ve been customizing the system for awhile now.