TB 2.8 Problem - No audio from incoming calls using Broadvoice SIP trunk

Your subject says it is happening on the broadvoice trunk, your comment says it is happening on your rhino card. Could you please clarify if it is doing it on one or the other or both? Could you please post a log of the call? The simplest way to do this is open up ssh, type asterisk -vvvvvvr make a call, copy and paste what shows up.

Ok. I played around some more and found that the same problem is true for extension on the same local network as well.

However, I found one odd thing. The system does NOT allow me to have externip set to my external IP address any longer. I am guessing that this is the issue. Basically, I can't register my trunk once I have specified the external IP in sip_general_custom.conf.

Here is my network setup:

1. Smoothwall forwarding 5 external IPs to internal machines (configuration didn't changes for 8 month plus now).
2. Trixbox uses valid hostname (pbx.domain.com) that is linked in local DNS to 192.168.x.x.
3. pbx.domain.com is also valid on the external network and linked to a valid IP 99.11.x.x.
4. Trixbox shows external and internal IP on status page correctly (with or without any configuration in sip_general_custom.conf.
5. sip_general_custom.conf contains

nat=yes
#externhost=pbx.domain.com
externip=99.11.x.x
localnet=192.168.x.x/255.255.255.0

6. I also tried nat=no and externhost instead of externip, but once I have externip or externhost specified I can no longer register my Broadvoice trunk.
7. My BroadVoice trunk worked fine on 2.6 with no incoming settings and the outgoing settings listed below.

username=5551235555
type=peer
secret=<mysecret>
qualify=yes
insecure=very
host=sip.broadvoice.com
fromuser=5551235555
fromdomain=sip.broadvoice.com
dtmfmode=rfc2833
disallow=all
context=from-pstn
canreinvite=no
allow=ulaw

8. The Asterisk log doesn't show any errors and so I didn't add that here...

I really hope that someone can help me with that. This is our only home phone and it is really bugging us that we can't hear incoming callers...

Thanks,
Jens

Hi Scott,

Thank you for getting back to me.

Yes, that makes sense. What's puzzling to me is that I didn't really change anything except upgrading to 2.8.

I believe the key of the problem is that I can no longer specify externip in my configuration file. I mean even the trixbox status page shows me the same external IP address anyhow, but once I explicitly specify it in the configuration file I can no longer register any SIP trunks...

I guess I might have to go back to 2.6.x once again, but then I will have the DTMF problem again... Either way I am left with problems...

Here is the asterisk log people were looking for:

Connected to Asterisk 1.6.0.9-samy-r27 currently running on trixbox (pid = 2503)
Verbosity was 3 and is now 6
    -- Remote UNIX connection
    -- Starting simple switch on 'DAHDI/1-1'
    -- Executing [5002@from-internal:1] Macro("DAHDI/1-1", "exten-vm,novm,5002") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("DAHDI/1-1", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("DAHDI/1-1", "AMPUSER=5001") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("DAHDI/1-1", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("DAHDI/1-1", "1?Set(REALCALLERIDNUM=5001)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("DAHDI/1-1", "AMPUSER=5001") in new stack
    -- Executing [s@macro-user-callerid:5] Set("DAHDI/1-1", "AMPUSERCIDNAME=Line 1") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/1-1", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("DAHDI/1-1", "AMPUSERCID=5001") in new stack
    -- Executing [s@macro-user-callerid:8] Set("DAHDI/1-1", "CALLERID(all)="Line 1" <5001>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("DAHDI/1-1", "REALCALLERIDNUM=5001") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("DAHDI/1-1", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("DAHDI/1-1", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:12] Set("DAHDI/1-1", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("DAHDI/1-1", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("DAHDI/1-1", "Using CallerID "Line 1" <5001>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("DAHDI/1-1", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("DAHDI/1-1", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("DAHDI/1-1", "EXTTOCALL=5002") in new stack
    -- Executing [s@macro-exten-vm:5] Set("DAHDI/1-1", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("DAHDI/1-1", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("DAHDI/1-1", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("DAHDI/1-1", "record-enable,5002,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("DAHDI/1-1", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("DAHDI/1-1", "recordingcheck,20090711-121440,1247339674.0") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck,20090711-121440,1247339674.0: Inbound recording not enabled
    -- <DAHDI/1-1>AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("DAHDI/1-1", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("DAHDI/1-1", "dial,"",trw,5002") in new stack
    -- Executing [s@macro-dial:1] GotoIf("DAHDI/1-1", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("DAHDI/1-1", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
  == Manager 'admin' logged on from 127.0.0.1
 dialparties.agi: Caller ID name is 'Line 1' number is '5001'
       > dialparties.agi: USE_CONFIRMATION:  'FALSE'
       > dialparties.agi: RINGGROUP_INDEX:   ''
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 5002 to extension map
    -- dialparties.agi: Extension 5002 cf is disabled
    -- dialparties.agi: Extension 5002 do not disturb is disabled
       > dialparties.agi: extnum 5002 has:  cw: 1; hascfb: 0 [] hascfu: 0 []
       > dialparties.agi: ExtensionState: 0
    -- dialparties.agi: dbset CALLTRACE/5002 to 5001
    -- dialparties.agi: Filtered ARG3: 5002
  == Manager 'admin' logged off from 127.0.0.1
    -- <DAHDI/1-1>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("DAHDI/1-1", "DAHDI/2,"",trw") in new stack
    -- Called 2
    -- DAHDI/2-1 is ringing
    -- DAHDI/2-1 is ringing
    -- DAHDI/2-1 answered DAHDI/1-1
    -- Executing [h@macro-dial:1] Macro("DAHDI/1-1", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("DAHDI/1-1", "vw") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("DAHDI/1-1", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("DAHDI/1-1", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("DAHDI/1-1", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("DAHDI/1-1", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("DAHDI/1-1", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'DAHDI/1-1' in macro 'hangupcall'
  == Spawn extension (macro-dial, h, 1) exited non-zero on 'DAHDI/1-1'
    -- Hungup 'DAHDI/2-1'
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'DAHDI/1-1' in macro 'dial'
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'DAHDI/1-1' in macro 'exten-vm'
  == Spawn extension (from-internal, 5002, 1) exited non-zero on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'

Thanks,
Jens

Hi Scott,

I realize that it is working for many people, but it doesn't work for me...

I activated sip debug in asterisk and captured the following log (cellphone to trixbox):
http://pastebin.ca/1492230

Here is the result of running sip show peer:

  * Name       : BroadVoice
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-pstn
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  FromUser     : 5551235555
  FromDomain   : sip.broadvoice.com
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : *97
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Text Support : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : sip.broadvoice.com
  Addr->IP     : 147.135.8.128 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Transport    : UDP
  Def. Username: 5551235555
  SIP Options  : 100rel
  Codecs       : 0x4 (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing :  No
  100 on REG   : No
  Status       : OK (108 ms)
  Useragent    :
  Reg. Contact :
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs

I still can't see anything being wrong... Since it is a fresh install I only activated ulaw so far to keep things simple.

I also noticed that I can externip in my configuration, but once I add localnet my trunk no longer registers... Really weird!

localnet=192.168.x.x/255.255.255.0

Thank you for looking into this in advance!

Best regards,
Jens

P.S.: Please let me know if I didn't provide what you were looking for an I will try again.

Scott,

The full IP is:

localnet=192.168.1.0/255.255.255.0

I believe that's correct, right?

Thanks,
Jens

Quick question, have you tried a sip phone? Does it work with 2 way audio? If you haven't, download xlite and give it a whirl. It may not be a trunk issue if a sip phone works. Then the rhino card could be concentrated on.

I'm sure you're just one of the early adopters of 2.8. Not many people have even tried it yet. If this is indeed a driver issue, we'll see more people on the forums with this issue in the coming months.
Anyways, just a shot in the dark.

yum remove rhino*
yum install -y rhino-$(uname -r)

I also noticed combing the forums you have had this issue before, what did you do to get it working on 2.6 10 months ago?

Seriously though, I have nothing. I read Rhino's support is very nice if you find that it is a driver issue and you haven't already opened a ticket. https://support.rhinoequipment.com/
Worst case, back to trixbox 2.6.2.3.

Just a quick update:

I started working with Rhino Support and we found that the problem also exists if I make a call from Rhino extension to another.

Unfortunately, I believe that still allows the issue to be a driver issue or a trixbox network issue.

Since I also see more and more people posting about problems with their network setup I am wondering what's going on.

Even though I tried tons of things on my end I have been unable to get trixbox to accept a localnet entry in my configuration file. I mean I can add it, but then I can no longer register any trunk...

I will post new updates once they become available.

Best regards,
Jens

Jon,

I am using DSL with 5 static IPs.

The modem has only one network connector and so I need to connect it to a router or, in my case, smoothwall to get all my addresses linked to the local network. I can investigate if there is a method that would allow me to do otherwise, but I am currently not aware of it. I would also think that most people here are using some kind of router or firewall to protect their trixbox environment.

That said I think I understand where you are going with your suggestion and I guess it would actually work. However, this would just confirm that something has changed in the way networks are being handled by trixbox and I would assume this to be a bug on the trixbox end and not in smoothwall.

Anyhow, if someone can make a suggestion how I can easily test that without pulling my entire environment apart I am game to test it.

Best regards,
Jens

An update for the curious we are working on this issue with Jens. So far testing seems to exclude our driver and Hardware. We have fully qualified our hardware on 2.8 and have seen no issues with basic operation. That said we are always happy to look in to things. A test call was recorded and there was bi-directional audio which indicates the bottom 3 layers are doing their thing:

Hardware
rcbfx
DAHDI -> application layer

All these components worked.

So either chan_dahdi is not getting the audio or something in the way jens box is handling audio streams is failing.

my guess would be the latter as this happens on chan_sip as well.

Ok. I did some more testing and I think I made some progress.

It seems that it call comes down to not having localnet in my config file. I still have to do a test between my Rhino extensions when I get back home, but I noticed that all audio issues between SIP extensions go away once I have localnet in my config file. The only issue now is that I can't register my trunk with BroadVoice once I have localnet in my config file.

Since it worked just fine with older Trixbox versions I am also pretty sure that something changed on the trixbox end as well.

Going through my notes I noticed that I did simplify my DNS setup this time. I used to run a secondary DNS linked to my windows domain server on trixbox, but I didn't see much use for that and so I didn't install it when I upgraded. Not sure, but this is pretty much the only difference on my end...

If somebody has any ideas on why I can't register my trunk with localnet specified please let me know.

Thanks,
Jens

Hi Jon,

I tried opendns as suggested above. Results are unchanged in comparison to using ATT DNS (smoothwall uses ATT DNS forwarders).

With localnet:

Name/username              Host            Dyn Nat ACL Port     Status     
BroadVoice/5551234567      147.135.8.128        N      5060     UNREACHABLE 

Without localnet:

Name/username              Host            Dyn Nat ACL Port     Status     
BroadVoice/5551234567      147.135.8.128        N      5060     OK (110 ms) 

I am lost. My only explanation is that something has changed in CentOS... I should mention that I can also connect to BroadVoice using xten as a client without problems. Same subnet and any DNS...

I will also test Rhino to Rhino with localnet on Friday, but I am almost certain that it will work as well. I guess I am stuck with either way having audio between extensions or not having audio on incoming calls, but being able to make calls...

Any linux gurus here?

Thanks,
Jens

I am guessing that the audio problem is caused by this:

Peer             User/ANR    Call ID          Format           Hold     Last Message   
147.135.32.221   5551234567  149a42dc2c4dbe8  0x0 (nothing)    No                                 
147.135.32.221   5551234567  1a80025-a8@147.  0x4 (ulaw)       No       Rx: ACK                   
2 active SIP dialogs

I should probably see ulaw in the first line as well... :(

Thanks,
Jens

I tried a few more things, but I couldn't get it to work. 3 weeks of time wasted. Time to inform management (my wife) of failure... ;)

Well, not exactly. I went back to 2.6.2.3 and everything worked (2 ZAPTEL extensions, 3 sip extensions and 1 sip trunk) right away. Ok. Since I noticed that both releases are now running on the same kernel I didn't try a different host name on the box yet... Didn't want to ruin my success right away...

Looking at the facts it seems to be safe assumption that the problem is caused somewhere in the DAHDI layer. The fact that I seem to be one of the only people running into this problem so far is probably mostly based on the fact of people not having tried it very much yet. However, I do see quite a lot of people having audio problems lately...

Since I am not a VOIP guru and since I don't feel knowledgeable enough to open an asterisk bug ticket I feel that rolling back to the current stable solution was my best option.

In general I am a bit disappointed that the overall support in the forum seem to have dropped quite a lot. With a few exceptions there aren't too many people left that are willing to help others. I mean I am not an expert, but I am trying to help people with trixbox whenever I can. I can see that help isn't always appreciated and some people just do a hit and run approach to get their questions answered (at the minimum people could report back when something worked for them) resulting in a lot of knowledgeable folks stopping to answer questions all together. Needless to say that this is a bad development for everyone here...

Anyhow, once again I would like to say thank you to the people that were trying to help me. I am hoping that my test efforts will go somewhere and, maybe, I will try upgrading again in a few months when the product has become more mature.

Best regards,
Jens

I do not have Rhino cards, but I am experiencing a similar issue
trixbox 2.8 w Asterisk 1.6
Dnynamic DNS
Server NATed; remote client NATed
NO customnization of config files directly - all configs done via trixbox UI changes
Local and External IP shows correctly on System Status window

When I connect from outside into my Server, with either x-lite or an Aastra phone (53i), I register, I can dial, I can make the connection, but no audio passes so any voicemails are ) seconds long.

Any suggestions?

Richard