[Freeswitch-users] Installing OPUS codec

Thu Aug 21 19:11:42 MSD 2014

I have it in my source directory. 
make mod_opus give me the following errors.[root at localhost mod_opus]# make mod_opus
cc     mod_opus.c   -o mod_opus
mod_opus.c:33:20: error: switch.h: No such file or directory
mod_opus.c:34:18: error: opus.h: No such file or directory
mod_opus.c:37: warning: data definition has no type or storage class
mod_opus.c:37: warning: parameter names (without types) in function declaration
mod_opus.c:38: warning: data definition has no type or storage class
mod_opus.c:38: warning: parameter names (without types) in function declaration
mod_opus.c:71: error: expected specifier-qualifier-list before âOpusEncoderâ
mod_opus.c:79: error: expected specifier-qualifier-list before âswitch_mutex_tâ
mod_opus.c:83: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âswitch_opus_fmtp_parseâ
mod_opus.c:200: error: expected declaration specifiers or â...â before âswitch_memory_pool_tâ
mod_opus.c: In function âgen_fmtpâ:
mod_opus.c:205: warning: incompatible implicit declaration of built-in function âsnprintfâ
mod_opus.c:205: warning: incompatible implicit declaration of built-in function âstrlenâ
mod_opus.c:209: warning: incompatible implicit declaration of built-in function âsnprintfâ
mod_opus.c:209: warning: incompatible implicit declaration of built-in function âstrlenâ
mod_opus.c:213: warning: incompatible implicit declaration of built-in function âsnprintfâ
mod_opus.c:213: warning: incompatible implicit declaration of built-in function âstrlenâ
mod_opus.c:218: warning: incompatible implicit declaration of built-in function âsnprintfâ
mod_opus.c:218: warning: incompatible implicit declaration of built-in function âstrlenâ
mod_opus.c:222: warning: incompatible implicit declaration of built-in function âsnprintfâ
mod_opus.c:222: warning: incompatible implicit declaration of built-in function âstrlenâ
mod_opus.c:226: warning: incompatible implicit declaration of built-in function âsnprintfâ
mod_opus.c:226: warning: incompatible implicit declaration of built-in function âstrlenâ
mod_opus.c:230: warning: incompatible implicit declaration of built-in function âsnprintfâ
mod_opus.c:230: warning: incompatible implicit declaration of built-in function âstrlenâ
mod_opus.c:234: error: invalid lvalue in assignment
mod_opus.c:237: error: âpoolâ undeclared (first use in this function)
mod_opus.c:237: error: (Each undeclared identifier is reported only once
mod_opus.c:237: error: for each function it appears in.)
mod_opus.c:237: warning: return makes pointer from integer without a cast
mod_opus.c: At top level:
mod_opus.c:241: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âswitch_opus_initâ
mod_opus.c:328: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âswitch_opus_destroyâ
mod_opus.c:347: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âswitch_opus_encodeâ
mod_opus.c:375: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âswitch_opus_decodeâ
mod_opus.c:400: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âopus_load_configâ
mod_opus.c: In function âSWITCH_MODULE_LOAD_FUNCTIONâ:
mod_opus.c:433: error: âswitch_codec_interface_tâ undeclared (first use in this function)
mod_opus.c:433: error: âcodec_interfaceâ undeclared (first use in this function)
mod_opus.c:440: error: âNULLâ undeclared (first use in this function)
mod_opus.c:442: error: âswitch_status_tâ undeclared (first use in this function)
mod_opus.c:442: error: expected â;â before âstatusâ
mod_opus.c:444: error: âstatusâ undeclared (first use in this function)
mod_opus.c:444: error: âSWITCH_FALSEâ undeclared (first use in this function)
mod_opus.c:444: error: âSWITCH_STATUS_SUCCESSâ undeclared (first use in this function)
mod_opus.c:449: error: âmodule_interfaceâ undeclared (first use in this function)
mod_opus.c:449: error: âpoolâ undeclared (first use in this function)
mod_opus.c:449: error: âmodnameâ undeclared (first use in this function)
mod_opus.c:453: error: âswitch_opus_fmtp_parseâ undeclared (first use in this function)
mod_opus.c:464: error: too many arguments to function âgen_fmtpâ
mod_opus.c:466: error: âSWITCH_CODEC_TYPE_AUDIOâ undeclared (first use in this function)
mod_opus.c:479: error: âswitch_opus_initâ undeclared (first use in this function)
mod_opus.c:480: error: âswitch_opus_encodeâ undeclared (first use in this function)
mod_opus.c:481: error: âswitch_opus_decodeâ undeclared (first use in this function)
mod_opus.c:482: error: âswitch_opus_destroyâ undeclared (first use in this function)
mod_opus.c:502: error: too many arguments to function âgen_fmtpâ
make: *** [mod_opus] Error 1

--Forwarded Message Attachment--
From: kamil.nigmatullin at gmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 21 Aug 2014 18:25:35 +0600
Subject: Re: [Freeswitch-users] Installing OPUS codec

Last time I compiled opus from git repo, I downloaded the source of the library itself from official opus's site(powered by Mozilla), put it to FS directory and then did 
make mod_opus

make mod_opus-install

2014-08-20 6:57 GMT+06:00 Kamrul Khan <dodul at live.com>:

Hi, Can anyone tell me how to install OPUS module in FREESWITCH ? I have the source in my source directory. 		 	   		  

_________________________________________________________________________

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consulting at freeswitch.org

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Tel: 77272323748
mob: 7 777 2517003
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Skype: Kamil Nigmatullin

--Forwarded Message Attachment--
From: bernhard at sessionbird.com
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 21 Aug 2014 15:03:26 +0200
Subject: Re: [Freeswitch-users] Fetching gateways via mod_curl_xml

    Hi Kamil,

    you are telling FreeSWITCH to add a gateway, but not registering it.
    I've highlighted the 2 occasions in the quote.

    Try it again, but this time set the register-param to true.

    Regards,

    Bernhard

    Am 21.08.2014 00:38, schrieb Kamil
      Nigmatullin:

        Could anyone please help. I want to fetch
          gateways(sip_profiles) with mod_curl_xml. This is what I
          respond via http

          <?xml version="1.0" encoding="UTF-8" standalone="no"?>

          <document type="freeswitch/xml">

          <section name="configuration">

            <configuration name="sofia.conf" description="Core
          Configuration"> 

              <profile name="external">

                <gateways>

                  <gateway name="supernode1">

                     <param name="username" value="supernode"/>

                     <param name="realm"
          value="188.138.xx.xx:5060"/>

                     <param name="password" value="xxxx"/>

                     <param name="register" value="false"/>

                     <param name="proxy"
          value="188.138.xx.xx:5060"/>

                     <param name="caller-id-in-from"
          value="true"/>

                     <param name="dtmf-type" value="inband"/>

                     <param name="rtp-timeout-sec" value="30"/>

                     <param name="from-domain"
          value="188.138.xx.xx:5060"/>

                     <param name="register" value="false"/>

                     <param name="NDLB-sendrecv-in-session"
          value="true"/>

                     <param name="vad" value="none"/>

                     <param name="rtp-ip" value="188.138.xx.xx"/>

                     <param name="sip-ip" value="188.138.xx.xx"/>

                     <param name="extension-in-contact"
          value="true"/>

                     <param name="rtp-autoflush-during-bridge"
          value="true"/>

                     <param name="user-agent-string" value="SBC"/>

                     <param name="NDLB-sendrecv-in-session"
          value="true"/>

                     <param name="inbound-late-negotiation"
          value="false"/>

                  </gateway>

               </gateways>

             </profile>

          </configuration>

          </section>

          </document>

        I see that FS requests and read this XML(each line ended by
        \r\n) without reporting any errors, but gateway does not appears
        at sofia status. Is there anything wrong in that XML Block?

          Thanks in advance.

            -- 

            Kamil Nigmatullin

      _________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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http://confluence.freeswitch.org
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--Forwarded Message Attachment--
From: emamirazavi at gmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 21 Aug 2014 18:33:52 +0430
Subject: [Freeswitch-users] No audio in several profiles

When i have two sofia profiles on two network interfaces and i register my phone on second profile, i have no audio. Everything is in local computer. Firewall does not exist. I have this warning: switch_core_session.c:1554 sofia/customer_access/1009 at 192.168.10.10:5060 using scheduler due to bypass media or media is not established.

Any idea? 

--Forwarded Message Attachment--
From: brian at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 21 Aug 2014 10:00:39 -0500
Subject: Re: [Freeswitch-users] WebRTC Calls via Kamailio rejecting with 488

We should support UDP/TLS/RTP/SAVPF without much issue, So i'm puzzled as to why you've had a problem, We have sipml5/jssip/sipjs all working without a problem direct to WSS/WS to FreeSWITCH.

On Wed, Aug 20, 2014 at 4:47 PM, Kamrul Khan <dodul at live.com> wrote:

No thats not what we want to do. But, we figured out the issue. Our sipml5 webclient is sending media profile thats not supported by slandered SIP. sipml5 sends something like, 

m=audio 56381 UDP/TLS/RTP/SAVPF 109 0 8 101
However, as far as we concern, according to standard SIP definition, the UDP/TLS/ part shouldn't be there. (Probably this is an addition for webRTP). Our Asterisk server took the call cause we had SRTP-DTLS configured there. 

So, now we stripped off the UDP/TLS/  part from kamailio (sending RTP/AVP only) and now calls are going through. 

Thanks for your help guys!

--Forwarded Message Attachment--
From: brian at freeswitch.org

To: freeswitch-users at lists.freeswitch.org
Date: Wed, 20 Aug 2014 15:59:41 -0500
Subject: Re: [Freeswitch-users] WebRTC Calls via Kamailio rejecting with 488

have you tried talking directly to FreeSWITCH via websockets?

On Wed, Aug 20, 2014 at 3:23 PM, Kamrul Khan <dodul at live.com> wrote:

No I didnt

--Forwarded Message Attachment--
From: brian at freeswitch.org
To: freeswitch-users at lists.freeswitch.org

Date: Wed, 20 Aug 2014 14:03:43 -0500
Subject: Re: [Freeswitch-users] WebRTC Calls via Kamailio rejecting with 488

Have you modified the default sofia profile codec preferences at all?

On Wed, Aug 20, 2014 at 1:47 PM, Kamrul Khan <dodul at live.com> wrote:

Hi,
This is the continuation of my previous mail, "Freeswitch srtp rejecting SRTP with 488‏". I have changed the subject line as according to the feedback from this mailing list Im convinced its not a SRTP issue. Let me describe my scenario a bit more detail. 

I am generating call from a webclient using simpl5. The call then hits a Kamailio server and Kamailio here acts like a webRTC proxy that forwards the call to Freeswitch. Freeswitch receives the call on udp port 5060. (not over websocket)

We tested the same thing with asterisk and thats working fine. but when we try it with Freeswitch it rejects the call with 488. And in console log it shows   [CS_NEW] [INCOMPATIBLE_DESTINATION]

Now, we assume (as per the feedback from this mailing list) the issue is not with SRTP but with codec negotiation. Therefore, for testing purpose we stripped off all the codec attributes from our INVITE leaving only rtpmap:0 PCMU/8000 and tested again. We were hoping to see something new in our console log this time but surprisingly it showed up the exact same log as before. The console log is given below. Please somebody give us some clue whats going wrong? Thanks in advanced.

2014-08-20 05:03:42.183856 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.2014-08-20 05:03:42.253550 [DEBUG] sofia.c:5847 IP 192.168.146.133 Rejected by acl "domains". Falling back to Digest auth.

2014-08-20 05:03:42.269373 [NOTICE] switch_channel.c:669 New Channel sofia/internal/alice at 192.168.146.133 [07526f12-2862-11e4-8afd-9de9b4eb98f8]2014-08-20 05:03:42.272360 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice at 192.168.146.133) Running State Change CS_NEW

2014-08-20 05:03:42.272360 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/alice at 192.168.146.133) State NEW2014-08-20 05:03:42.281094 [DEBUG] sofia.c:4153 Channel sofia/internal/alice at 192.168.146.133 entering state [received][100]

2014-08-20 05:03:42.281094 [DEBUG] sofia.c:4164 Remote SDP:v=0o=Mozilla-SIPUA-31.0 20849 1 IN IP4 0.0.0.0s=Doubango Telecom - firefoxt=0 0a=ice-ufrag:ca607972

a=ice-pwd:dd5932ce75801e50fae435678cd6755ba=fingerprint:sha-256 8B:B0:38:A0:DA:3D:7C:DE:7F:76:F0:35:AF:1A:DE:E9:25:8C:16:E7:D4:3F:75:B2:64:56:79:21:2D:16:7D:05m=audio 57096 UDP/RTP 0

c=IN IP4 184.69.59.132
a=ptime:20a=rtpmap:0 PCMU/8000a=fmtp:101 0-15a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-levela=setup:actpassa=candidate:0 1 UDP 2128609535 172.16.1.188 57096 typ host

a=candidate:1 1 UDP 1692467199 184.69.59.132 57096 typ srflx raddr 172.16.1.188 rport 57096a=candidate:5 1 UDP 2128543999 192.168.56.1 57097 typ host
a=candidate:10 1 UDP 2128478463 192.168.232.1 57098 typ hosta=candidate:15 1 UDP 2128412927 192.168.146.1 57099 typ host
a=candidate:0 2 UDP 2128609534 172.16.1.188 57100 typ hosta=candidate:1 2 UDP 1692467198 184.69.59.132 57100 typ srflx raddr 172.16.1.188 rport 57100
a=candidate:5 2 UDP 2128543998 192.168.56.1 57101 typ hosta=candidate:10 2 UDP 2128478462 192.168.232.1 57102 typ host
a=candidate:15 2 UDP 2128412926 192.168.146.1 57103 typ hosta=rtcp-mux
2014-08-20 05:03:42.281094 [NOTICE] sofia.c:4354 Hangup sofia/internal/alice at 192.168.146.133 [CS_NEW] [INCOMPATIBLE_DESTINATION]

2014-08-20 05:03:42.281094 [DEBUG] switch_channel.c:2102 Send signal sofia/internal/alice at 192.168.146.133 [KILL]2014-08-20 05:03:42.282146 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice at 192.168.146.133 [BREAK]

2014-08-20 05:03:42.282146 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice at 192.168.146.133) Running State Change CS_HANGUP2014-08-20 05:03:42.282146 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice at 192.168.146.133) State HANGUP

2014-08-20 05:03:42.282146 [DEBUG] mod_sofia.c:414 Channel sofia/internal/alice at 192.168.146.133 hanging up, cause: INCOMPATIBLE_DESTINATION2014-08-20 05:03:42.328550 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 488

2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:46 sofia/internal/alice at 192.168.146.133 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION

2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/alice at 192.168.146.133) State HANGUP going to sleep2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/alice at 192.168.146.133) State Change CS_HANGUP -> CS_REPORTING

2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice at 192.168.146.133 [BREAK]2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/alice at 192.168.146.133) Running State Change CS_REPORTING

2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice at 192.168.146.133) State REPORTING2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:53 sofia/internal/alice at 192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION

2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:590 (sofia/internal/alice at 192.168.146.133) State REPORTING going to sleep2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/alice at 192.168.146.133) State Change CS_REPORTING -> CS_DESTROY

2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/alice at 192.168.146.133 [BREAK]2014-08-20 05:03:42.328550 [DEBUG] switch_core_session.c:1164 Session 18 (sofia/internal/alice at 192.168.146.133) Locked, Waiting on external entities

2014-08-20 05:03:42.328550 [NOTICE] switch_core_session.c:1182 Session 18 (sofia/internal/alice at 192.168.146.133) Ended2014-08-20 05:03:42.328550 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/alice at 192.168.146.133 [CS_DESTROY]

2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/alice at 192.168.146.133) Running State Change CS_DESTROY2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice at 192.168.146.133) State DESTROY

2014-08-20 05:03:42.328550 [DEBUG] mod_sofia.c:341 sofia/internal/alice at 192.168.146.133 SOFIA DESTROY2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:60 sofia/internal/alice at 192.168.146.133 Standard DESTROY

2014-08-20 05:03:42.328550 [DEBUG] switch_core_state_machine.c:439 (sofia/internal/alice at 192.168.146.133) State DESTROY going to sleep

From: freeswitch-users-request at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 98, Issue 138
To: freeswitch-users at lists.freeswitch.org

Date: Wed, 20 Aug 2014 22:18:01 +0400

Send FreeSWITCH-users mailing list submissions to
	freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to

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You can reach the person managing the list at
	freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."

--Forwarded Message Attachment--
From: andretodd at verizon.net

To: freeswitch-users at lists.freeswitch.org
Date: Wed, 20 Aug 2014 13:35:01 -0400
Subject: Re: [Freeswitch-users] high pdd

Hi, I’m using the master. Should I try 1.4.7?

I’ve done a dump after things started to go bad/ backed up. 

I see 8193 threads in _thread_cond_timedwait. Looks like its on line 81 of thread_cond.c  res = WaitForSingleObject(cond->semaphore, timeout_ms);

The value of timeout_ms is 2016850392.

Not sure if this means anything.

Thanks for the support guys.

Andre

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West

Sent: Wednesday, August 20, 2014 11:57 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] high pdd

1.2.X is EOL, 

1.4.7 is the current release.

We have releases and tags in git that no longer carry the 'stable' moniker starting with 1.4.  

On Sun, Aug 17, 2014 at 1:39 PM, Andre Demattia <andretodd at verizon.net> wrote:

Thanks, I'm using the newest master and the last stable 1.2.23

We are willing to pay to get this fixed.
Andre 

From: Anthony Minessale

Sent: ‎8/‎17/‎2014 12:30 PM

To: Freeswitch-users

Subject: Re: [Freeswitch-users] high pdd

We dont have a lot of tools to troubleshoot windows.  You should mention windows in your subject to have a better chance to get the attention of our windows users. 
You should consider installing debian 7 on the same hardware and perform similar tests to help determine if your issue is o/s specific or not.

You also have not indicated the revision of FS.  If you have not worked with latest master, you should be also to rule out improvements. 
On Aug 17, 2014 9:54 AM, "Andre Demattia" <andretodd at verizon.net> wrote:

Yes that's off and I believe all the databases are off too. No authentication or registration is enabled.  The sizes of the database are 0 until I call the shutdown command then its about 90k.  I have the DBS in Ra drive too.

Windows seems pretty good until the 3000 sessions per min mark. Then I just get long pdd and the threads get backed up then no more sessions can be created. 

Funny thing is I has 10 profiles and the same results. I also tried two consoles with the same results.

FreeSWITCH is great.

I'd be grateful for help. I've worked on this for 2 years and hate to give up. 

If any one from consulting at freeswitch.org is willing to help I'd appreciate that.

Thanks again,
Andre 
From: Anthony Minessale

Sent: ‎8/‎17/‎2014 10:13 AM

To: Freeswitch-users

Subject: Re: [Freeswitch-users] high pdd

Did you comment out manage-presence ?  Also we have no real idea how well FS performs in windows under extreme load.
On Aug 17, 2014 5:42 AM, "Andre Demattia" <andretodd at verizon.net> wrote:

Thanks Ken, I'll try the consulting again.

Andre 
From: Ken Rice

Sent: ‎8/‎17/‎2014 1:19 AM

To: FreeSWITCH Users Help

Subject: Re: [Freeswitch-users] high pdd

if the last post does not work contact consulting at freeswitch.org for pro help from one of the developers... thats about as far as the list will take you

Ken
Sent from my iPad

On Aug 16, 2014, at 15:23, "Andre" <andretodd at verizon.net> wrote:

Going with the idea it’s IO related I did a db_cache status. I just disabled limit in my code by removing it but limit should be HASH not a database however I stills how a database.

Below is the results of the db cache status. I’m not real sure what its saying. I should have all registration and limit disabled and should not be using any sql database from my understanding.

Anyone know what the results mean? I’ve only tested 2 calls on this status.

[The entire original message is not included.]

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites

http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

[The entire original message is not included.]

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites

http://www.freeswitch.org
http://confluence.freeswitch.org
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-- 
Brian West

brian at freeswitch.org

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com

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--Forwarded Message Attachment--
From: stuart.mills3 at btopenworld.com

To: freeswitch-users at lists.freeswitch.org
Date: Wed, 20 Aug 2014 19:17:34 +0100
Subject: Re: [Freeswitch-users] Media after hang-up???

Hi Mike,

Thanks for the reply.

I know for sure this isn’t FreeSWITCH causing the issue, I get the CDR 
posted at the right time and durations are correct, it’s almost as if the far 
end PBX hasn’t noticed the hang-up, even though the BYE does get sent.

My carrier mentioned today (following a full trace from their end) that the 
PBX isn’t sending an ACK to the BYE, does that make any sense given the issue I 
am seeing? My main issue here is if it’s a PBX configuration issue, I am unable 
to look at and diagnose myself, so some sort of starting point would be really 
helpful .

Cheers,

Stuart

From: Michael Jerris 
Sent: Wednesday, August 20, 2014 5:25 PM
To: FreeSWITCH Users Help 

Subject: Re: [Freeswitch-users] Media after 
hang-up???

I 
have never seen an issue like this, a pcap might tell you more, but i seriously 
doubt we are still sending media after the call hangs up.  I would suggest 
taking a look at a pcap and the debug logs and seeing that the call is actually 
being hung up on the leg facing the pbx and we really stop sending media. 

Mike

On Aug 19, 2014, at 8:19 PM, Stuart Mills <stuart.mills3 at btopenworld.com> 
wrote:

  Hi,

  I’m experiencing a strange issue calling a destination within the UK, 
  whereby media seems to still be connected after hang up.

  I only know this as the destination is a PBX, supplied by Unify (formerly 
  Siemens), and that PBX on occasions records a voicemail, when a message is 
  taken and after the call is hung up, there is a continuous tone heard on the 
  end of the recording. The call logs show the right duration and the carrier 
  has also confirmed that the call ended as the logs suggest.

  So, I’m wondering if anyone else has come across the same sort of issue 
  using FreeSWITCH? The end user has told me they never used to experience this 
  issue using a different provider, and I can’t reproduce this issue at all 
  using other types of PBX, all calls end as they should and voicemail 
  recordings end when the call does.

  This issue doesn’t seem to be a FreeSWITCH one, just thought it worth an 
  ask to the group.

  Cheers,

  Stuart

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_________________________________________________________________________

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consulting at freeswitch.org

http://www.freeswitchsolutions.com

Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

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FreeSWITCH-users mailing list

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Brian West
brian at freeswitch.org

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com

http://www.freeswitchcookbook.com

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)

iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com

Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

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-- 

Brian West
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Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com

http://www.freeswitchcookbook.com

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

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