Re: [Freeswitch-users] Understanding ACLs

[Brian West]

Open up the sip profile and make sure you apply the correct ACL and  
it'll work.

/b

On Oct 17, 2008, at 8:44 AM, Gavin Henry wrote:

> Hi,
>
> Can someone help me undertand domain ACLs?
>
> I'm forwarding a call to [EMAIL PROTECTED] and FS is giving:
>
> 2008-10-17 16:37:39 [DEBUG] sofia.c:3134 sofia_handle_sip_i_invite()
> IP 193.111.201.114 Rejected by acl domains. Falling back to Digest
> auth.


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Re: [Freeswitch-users] Understanding ACLs

[Brian West]

That would be it... ;) I need to add to that now.

/b

On Oct 17, 2008, at 9:59 AM, Michael Collins wrote:


Brian, is this the relevant wiki entry?
http://wiki.freeswitch.org/wiki/Acl#sip_profiles

Just confirming.

Thanks,
MC


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Re: [Freeswitch-users] Shared Line Appearance

[Brian West]

I haven't ever configured full SLA on the Polycom, I have emulated SLA  
on the snom but thats about it.  You can do BLA with the polycom with  
ease.

/b

On Oct 17, 2008, at 7:06 PM, Alexander J. Perovich wrote:

>
> Has anyone configured SLA for Polycom phones using freeswitch?  If  
> so, I
> have a project if you want it.


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Re: [Freeswitch-users] Setting the moh

[Brian West]

It looks like you have the default config for local_stream.conf.xml  
and you'll need to "make hd-moh-install" during the installation of  
FreeSWITCH to have the proper files in place for local_stream to work.

/b

On Oct 20, 2008, at 9:50 AM, Woody Dickson wrote:

> I am getting the following error when executing the meta_app (*4).   
> The error is complaining about moh not found.  Does anyone know  
> where I can define the moh?  If I want to define different moh for  
> each user, is it doable?


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Re: [Freeswitch-users] Passwords in clear text

[Brian West]

Honestly is that much of an issue?  Your machine should be secure  
enough to not allow anyone but the user FreeSWITCH is running as read  
the configs in the first place.  I'm not even that paranoid :P

/b

On Oct 20, 2008, at 9:09 AM, Peter P GMX wrote:

> Thanks,
>
> I got it for the directory password (a1-hash).
>
> But what about the voicemail-password and the passwords stored for
> external gateways?
>
> Best regards
> Peter


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Re: [Freeswitch-users] sdp header rewrite

[Brian West]

Which version of FreeSWITCH are you using?

/b

On Oct 20, 2008, at 2:25 AM, Gabriel Kuri wrote:

> I'm having an issue with the linksys spa devices when enabling inbound
> proxy media mode (inbound-proxy-media=true) and late negotiation
> (inbound-late-negotiation=true) in the sofia profile. The spa
> immediately sends a BYE when the call is answered by the called party.
> For whatever reason, it works fine between two linksys devices  
> directly
> connected to FS, but when the call goes out to the PSTN via the SIP
> provider, the spa isn't happy and sends a BYE.
>
> After comparing the raw SIP packets on the wire (tcpdump) and between
> enabling/disabling proxy-media mode and late negotiation, the only
> difference I notice is the port in the m= line of the SDP header.
>
> According to the freeswitch log, the rtp port would be rewritten to
> 28044 in the sdp header of the SIP packet sent to the spa device.  
> But on
> the wire, the port is rewritten to 0, which I'm guessing is why the  
> spa
> isn't happy and sending a BYE.
>
> Here's the excerpt from the freeswitch log showing FS rewriting the  
> port
> to 28044 for the packet going to the spa device.
>
>
> [DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp()
> sofia/internal/@mydomain.net Patched SDP
> ---
> v=0
> o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX
> s=sip call
> c=IN IP4 XX.XX.XX.XX
> t=0 0
> m=audio 24174 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> +++
> v=0
> o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX
> s=sip call
> c=IN IP4 YY.YY.YY.YY
> t=0 0
> m=audio 28044 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
>
> However, the packet on the wire reveals FS rewriting the port to  
> 0  ...
>
> v=0.
> o=01Nextone 7852943629956191733 8120394851828294756 IN IP4  
> YY.YY.YY.YY.
> s=sip call.
> c=IN IP4 YY.YY.YY.YY.
> t=0 0.
> m=audio 0 RTP/AVP 96 101.
> a=rtpmap:96 G729/8000.
> a=fmtp:96 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
>
> Is this a bug or is there some other problem?
>
> Thanks for the help,
> Gabe
>
>
>
>
>
>
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Re: [Freeswitch-users] Problem with PlayAndGetDigits

[Brian West]

Keith,
I tried something like this and it worked fine.

digits = session:playAndGetDigits(1, 1, 3, 3000, "#*", "/tmp/ 
sr8k.wav", "/tmp/test.wav", "1|2|3|5")


Can you verify this?

/b

On Oct 18, 2008, at 4:14 AM, Keith Wood wrote:


Hi Brian,

Here is the script:

digits = session:playAndGetDigits(1, 1, 3, 3000, #*, /audio/ 
admin_menu.wav , /audio/invalid_input.wav ,1|2|3|5 )


I basically copied from the wiki.

Thanks,
Keith


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Re: [Freeswitch-users] overloaded 'CoreSession_execute'

[Brian West]

It would be session:execute as outlined here 
http://wiki.freeswitch.org/wiki/Mod_lua#session 
:execute

/b


On Oct 20, 2008, at 4:36 PM, Damon Brown wrote:

> digits = session:playAndGetDigits(1, 10, 10, 5000, "#",  
> "dialno.wav", "ivr-imsorry.wav", "\\d+")
> freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n");
> cmd = "sofia/gateway/broadvox/1" .. digits .. "@broadvox";
> freeswitch.consoleLog("info", "Command: ".. cmd .."\n");
> session.execute("bridge", cmd)


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Re: [Freeswitch-users] Setting the moh

[Brian West]

in the defaults the rate is selected by the channel rate.  You have no  
need to run a 16k hold music on an 8k channel.

/b

On Oct 20, 2008, at 9:15 PM, Woody Dickson wrote:

>  There are three rates specified ( 8000, 16000, and 32000), so how  
> do I select which one to use?


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Re: [Freeswitch-users] Setting the moh

[Brian West]

Turn off silences suppression ?

/b

On Oct 20, 2008, at 9:51 PM, Woody Dickson wrote:

> Hi Brian,
>
> What do you think could be the reason for the bad MOH sound quality?
>
> I am just thinking that it could be some setting issue.
>
> Thanks,
> Woody


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Re: [Freeswitch-users] sdp header rewrite

[Brian West]

It's an spa issue but we work around it so that shouldn't matter!

/b

Sent from my iPhone

On Oct 20, 2008, at 9:59 PM, Gabriel Kuri <[EMAIL PROTECTED]> wrote:

> I ran into this posting which is similar, although not exactly the  
> same,
> as the failure mode I'm experiencing.
>
>http://bugs.digium.com/view.php?id=11483
>
> following what this other person tried to temporarily fix the issue, I
> changed the name of the rtpmap on the linksys spa from G729a to G729  
> and
> it works - FS no longer transmits an audio port of 0 in the sdp  
> headers
> when inbound-proxy-media and late-negotiation are enabled.
>
> correct sdp header excerpt on a call ...
>
> v=0.
> o=01Nextone 2341985734634606731 5798373005113647141 IN IP4  
> YY.YY.YY.YY.
> s=sip call.
> c=IN IP4 YY.YY.YY.YY.
> t=0 0.
> m=audio 25454 RTP/AVP 18 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
>
> So is this an underlying issue with the linksys spa units or FS?
>
> Gabe
>
>
> Gabriel Kuri wrote:
>> I'm having an issue with the linksys spa devices when enabling  
>> inbound
>> proxy media mode (inbound-proxy-media=true) and late negotiation
>> (inbound-late-negotiation=true) in the sofia profile. The spa
>> immediately sends a BYE when the call is answered by the called  
>> party.
>> For whatever reason, it works fine between two linksys devices  
>> directly
>> connected to FS, but when the call goes out to the PSTN via the SIP
>> provider, the spa isn't happy and sends a BYE.
>>
>> After comparing the raw SIP packets on the wire (tcpdump) and between
>> enabling/disabling proxy-media mode and late negotiation, the only
>> difference I notice is the port in the m= line of the SDP header.
>>
>> According to the freeswitch log, the rtp port would be rewritten to
>> 28044 in the sdp header of the SIP packet sent to the spa device.  
>> But on
>> the wire, the port is rewritten to 0, which I'm guessing is why the  
>> spa
>> isn't happy and sending a BYE.
>>
>> Here's the excerpt from the freeswitch log showing FS rewriting the  
>> port
>> to 28044 for the packet going to the spa device.
>>
>>
>> [DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp()
>> sofia/internal/@mydomain.net Patched SDP
>> ---
>> v=0
>> o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX
>> s=sip call
>> c=IN IP4 XX.XX.XX.XX
>> t=0 0
>> m=audio 24174 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>>
>> +++
>> v=0
>> o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX
>> s=sip call
>> c=IN IP4 YY.YY.YY.YY
>> t=0 0
>> m=audio 28044 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>>
>>
>> However, the packet on the wire reveals FS rewriting the port to  
>> 0  ...
>>
>> v=0.
>> o=01Nextone 7852943629956191733 8120394851828294756 IN IP4  
>> YY.YY.YY.YY.
>> s=sip call.
>> c=IN IP4 YY.YY.YY.YY.
>> t=0 0.
>> m=audio 0 RTP/AVP 96 101.
>> a=rtpmap:96 G729/8000.
>> a=fmtp:96 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>>
>> Is this a bug or is there some other problem?
>>
>> Thanks for the help,
>> Gabe
>>
>>
>>
>>
>>
>>
>>
>
>
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Re: [Freeswitch-users] playback: Easy ways to handle different situations?

[Brian West]

On Oct 21, 2008, at 3:29 AM, Dennis wrote:

> Hi,
>
> I am fiddling arround with fs and different situations to handle
> playbacks and wonder, if there are some tricks or options, to tell
> "playback", to handle playbacks in different ways?
>
> What  I found out till now:
>
> 1.) If i want to stop a playback, I have to send a "break"...
>
> 2.) If I playback a soundfile (#1) and tell fs to play another
> soundfile (#2), while #1 is still playing back, fs behaves as follow:
> #1 will pause, #2 will be played back and after #2 was played back, #1
> will continue to play back.
> Why is this the default behavior?
>

If you add "event-lock:true" to your message you send then it will do  
what you want.  It will play the files in order as you expect.

/b


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Re: [Freeswitch-users] USER_NOT_REGISTERED (?)

[Brian West]

double check your firewall.

/b

On Oct 21, 2008, at 8:13 AM, henkoegema wrote:

> I'm sure somebody knows the anwser to my problem.
>
> I really can't find it myself.   :confused:


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Re: [Freeswitch-users] USER_NOT_REGISTERED (?)

[Brian West]

Its best you find us on IRC.  #freeswitch @ irc.freenode.net to help  
with this.. others can ask questions and help you get to a solution.

/b

On Oct 21, 2008, at 12:58 PM, henkoegema wrote:

>
>
> Brian West-3 wrote:
>>
>> double check your firewall.
>>
>
> I'm using an Edimax Wireless router with the Firewall module function
> Disabled.
>
> -- 
> View this message in context: 
> http://www.nabble.com/USER_NOT_REGISTERED-%28-%29-tp20050486p20095923.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Passwords in clear text

[Brian West]

Its called TLS...

/b

On Oct 21, 2008, at 4:30 PM, Peter P GMX wrote:

> In our environment DTMF is of course transported via SRTP so this is
> more secure (although the key exchange by SDES is known to have  
> security
> issues, as rtp streams may be replayed by a 3rd party, there is no
> replay prevention mechanism in SDES and therefore also not in
> freeswitch, hein?).


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Re: [Freeswitch-users] DTMF Star Event Inconsistent

[Brian West]

Are you using inband dtmf anywhere in this mix?

/b

On Oct 27, 2008, at 3:13 PM, Klaus Teller wrote:

> As far as i can tell, there is one single channel.  Call is  
> initiated via the socket interface to the extension 1003 and parked.  
> Or does parking generate a second channel?
>
> I'm using Xlite to listen on 1003 and for sending DTMF digits on the  
> parked channel. The wireshark trace also shows one single call going  
> from my computer to the Freeswitch box located on another computer.  
> And in this trace, events are not duplicated.
>
> I have updated and here is the new log information for 5** (DTMF-5,  
> DTMF-*, DTMF-*)
>
> Klaus.
>


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Re: [Freeswitch-users] freeswitch as a stateful SIP proxy or back2back user agent?

[Brian West]

Follow this thread 
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-October/007516.html

/b

On Oct 28, 2008, at 3:26 PM, Ryan McDougall wrote:

> Apologies if this has been answered somewhere already, but does
> freeswitch expose an API that would make it appropriate as a SIP
> proxy?
>
> Any advice you could share would be greatly appreciated.
>
> Cheers,
>
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Re: [Freeswitch-users] Clustering FreeSWITCH

[Brian West]

I'll have to 100% disagree with this statement.

NAPTR and SRV are how it should always be done.  Toss in some GEO dns  
and you have many of the problems solved.  SRV records should never be  
optional they should be required to function properly.  The NATPR  
records order preference of records which works in many hard and soft  
phones.


Example which this works:

<92>:host -t NAPTR bkw.org
bkw.org has NAPTR record 10 10 "s" "SIPS+D2T" "" _sips._tcp.bkw.org.
bkw.org has NAPTR record 20 20 "s" "SIP+D2S" "" _sip._sctp.bkw.org.
bkw.org has NAPTR record 30 30 "s" "SIP+D2T" "" _sip._tcp.bkw.org.
bkw.org has NAPTR record 40 40 "s" "SIP+D2U" "" _sip._udp.bkw.org.

<93>:host -t SRV _sips._tcp.bkw.org
_sips._tcp.bkw.org has SRV record 10 0 5061 sip.bkw.org.

<94>:host -t SRV _sip._sctp.bkw.org.
_sip._sctp.bkw.org has SRV record 10 0 5060 sip.bkw.org.

<95>:host -t SRV _sip._tcp.bkw.org.
_sip._tcp.bkw.org has SRV record 10 0 5060 sip.bkw.org.

<96>:host -t SRV _sip._udp.bkw.org.
_sip._udp.bkw.org has SRV record 10 0 5060 sip.bkw.org.

With these records in place my Eyebeam will register to my FreeSWITCH  
instance via TLS since it was listed as the highest preference.  The  
same goes for my Snom phones on my desk.  They see the NAPTR's, SRV's  
and do exactly what I told them to do via DNS.


The internet wouldn't exist today without DNS and if your DNS is that  
fragile you need to figure out why because without it we would be in  
for a world of hurt Not sure about you but I don't wanna remember  
what 4 billion IP's go to.


Bottom line is NO SRV NO NAPTR you're doing it wrong in my opinion  
because as a SIP UA you have to look them all up anyway since its NOT  
optional as per the spec.


/b


On Oct 29, 2008, at 6:54 PM, Yuval Hertzog wrote:

4. I would strongly recommend reconsidering the use of any DNS  
feature (such as SRV records) when deploying a telephony  
infrastructure. Of course, it all depends what this deployment is  
for. DNS is commonly used in the ITSP space due to the vast audience  
but enterprises (all sized) are recommended to refrain adding DNS to  
the list of point-of-failures in their telephony architectures.


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Re: [Freeswitch-users] Limit number of outbound channels

[Brian West]

http://wiki.freeswitch.org/wiki/Mod_limit

That should get you on the right track.

/b

On Oct 30, 2008, at 3:56 PM, Chav Paskov wrote:

> Hi, Everybody,
> i was wondering if there is an option or command that allows to limit
> the number of outbound channels per gateway under external profile.
> Regards
> Chav


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Re: [Freeswitch-users] Call Forwarding from phone

[Brian West]

Turn off the aggressive nat detection on the profile... if its on that  
would explain what is going on.

/b

On Oct 30, 2008, at 4:24 PM, Noah Silverman wrote:

> Hello,
>
> I have a configuration (dialplan?) question.
>
> One of my users used the "call forward" button on his sip phone.  The
> phone is directly registered to FS.
>
> The forwarded calls fail.  It looks like the phone is sending a
> "redirect" message to FS, but then the call is then not getting routed
> correctly through FS.  So, the call is dropped.
>
> Is there something specific I need to configure to account for
> possible call forwarding from phones??
>
> Thanks,
>
> -Noah


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Re: [Freeswitch-users] Call Forwarding from phone

[Brian West]

Well not if they are configured properly and use stun... It shouldn't  
ever be the responsibility of the registrar to overcome broken  
clients... thats how we got into this SIP mess in the first place.

/b

On Oct 30, 2008, at 5:03 PM, Noah Silverman wrote:

> Ok,
>
> But don't I need the aggressive nat detection since most of my clients
> will be behind nat??
>
> -N


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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

[Brian West]

Turn on the TPORT_LOG=1 ./freeswitch and let me see the challenge  
packet.

/b

On Oct 30, 2008, at 4:45 PM, Wellie Chao wrote:

> Here is what I have:
>
> 
>  
>
>
>
>
>
>  
> 
>
> Whether register is true or false doesn't seem to make a difference  
> (except that Freeswitch then comes up with broadview in NOREG  
> state). On calls from Metaswitch to Freeswitch, it's the same  
> problem, and I get the same message in the Freeswitch logs:
>
> 2008-10-30 17:39:04 [ERR] sofia_reg.c:1089  
> sofia_reg_handle_sip_r_challenge() No Matching gateway found
>
> I presume this is the same thing with the 401 Unauthorized packet  
> being sent by Metaswitch in response to Freeswitch's BYE message.  
> Note that the call itself goes just fine. I pick up, both sides can  
> hear each other. Just the hangup gets messed up and for some reason  
> Metaswitch expects an authenticated BYE message even though the  
> connection was not authenticated in the beginning when Metaswitch  
> initiated it. The packet trace shows this and it's very odd.
>
> Is that what you meant when you said set up a gateway in Freeswitch  
> that has reg=false and the proper credentials?


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Re: [Freeswitch-users] Call Forwarding from phone

[Brian West]

Its not so much for FreeSWITCH its for your phones to figure out their  
public IP so they don't lie when they register.

http://sourceforge.net/project/showfiles.php?group_id=47735

/b

On Oct 30, 2008, at 8:28 PM, Noah Silverman wrote:

> Can anyone recommend a "best" open source STUN server to work with  
> FS??
>
> Thanks,
>
> - N


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Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

[Brian West]

They really shouldn't be challenging the bye like that because not  
every device can answer the challenge properly.  Either case it  
doesn't matter if you have a proper domain 206.57.23.143 in your user  
directory with the proper user and realm set to meet this challenge.
We have a setting in FreeSWITCH to auth all packets and when we did  
that we found out that some devices just do not work properly when  
doing that.  I need to see your original gateway XML.  You can email  
it off list if you like and let me try to solve this.


/b

On Oct 30, 2008, at 7:52 PM, Wellie Chao wrote:


Here is the BYE from Freeswitch to Metaswitch:

send 683 bytes to udp/[64.115.128.6]:5060 at 00:44:46.607025:


   BYE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 216.57.23.143;rport;branch=z9hG4bKNa693jZ8SD54D
   Max-Forwards: 70
   From: ;tag=r4yBmtX3U0Hrr
   To:
[EMAIL PROTECTED]:5060;transport=udp>;tag=Broadview1+1+25f76f 
+cc3ba534

   Call-ID: [EMAIL PROTECTED]
   CSeq: 106588607 BYE
   Contact: 
   User-Agent: FreeSWITCH-mod_sofia/1.0.1-9171
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
SUBSCRIBE,

NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0




Metaswitch is not so happy with the BYE message:

recv 491 bytes from udp/[64.115.128.6]:5060 at 00:44:46.630445:


   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP
216.57.23.143 
;received=216.57.23.143;branch=z9hG4bKNa693jZ8SD54D;rport=5060

   From: ;tag=r4yBmtX3U0Hrr
   To:
[EMAIL PROTECTED]:5060;transport=udp>;tag=Broadview1+1+25f76f 
+cc3ba534

   Call-ID: [EMAIL PROTECTED]
   CSeq: 106588607 BYE
   WWW-Authenticate: Digest
realm 
="SipLocal",nonce="3e952db60fb8",stale=false,algorithm=MD5,qop="auth"

   Server: DC-SIP/2.0
   Organization:
   Supported: 100rel
   Content-Length: 0




Right after receiving the 401 Unauthorized message from Metaswitch,
Freeswitch emits the following error on the console:

2008-10-30 20:44:46 [ERR] sofia_reg.c:1089
sofia_reg_handle_sip_r_challenge() No Matching gateway found

At this point, the caller (the endpoint connected to Metaswitch) just
remains on the line, never having received the BYE.

Did you take a look at the packet traces I captured? The carrier
gave me a packet trace for an Aastra PBX/softswitch, and it had the  
same

interaction, but immediately upon receiving the 401 Unauthorized from
Metaswitch, the Aastra machine then sent a second BYE, this time with
authentication. Is there some way I can tell Freeswitch to do the  
same?


Regards,
Wellie


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Re: [Freeswitch-users] VxML Parser?

[Brian West]

But that doesn't keep us from doing such a thing as an application  
interface.  ;)

/b



On Nov 2, 2008, at 2:51 PM, Andrew Gilbert wrote:

> Quick answer, it's a switch/b2bua and not a vxml parser.
>
>

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Re: [Freeswitch-users] Domain Resolution Problem

[Brian West]

Can you show me the exact originate line you're using?

/b

On Nov 3, 2008, at 3:18 PM, Klaus Teller wrote:

> Hi Michael,
>
> Here is what i got on the consolo (Log level 7):


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[Freeswitch-users] Transcoding to GSM using SOX

[Brian Wood]

I'm trying to transcode a .wav to a .gsm using SOX v14.0.1. What I get 
is a completely garbled when using playback in FreeSWITCH. I have 
verified that at least one other application (Quicktime Player) can in 
fact play the resulting .gsm file.

I have tried the following:

sox william.wav -t gsm william.gsm
sox william.wav -t gsm -r 8000 william.gsm resample -ql
sox william.wav -t gsm -r 8000 -c1 william.gsm resample -ql

And I am playing the file using:

originate sofia/internal/[EMAIL PROTECTED] 
&playback(/home/user/william.gsm)

Other formats such as uLaw and aLaw (-t ul and -t al, respectively) seem 
to work fine.

Does anyone know of a set of parameters for SOX that will make this work?

At the end of the day, I just want it to work, so if anyone can 
recommend any other known/good command line tools to encode GSM that 
would be fine too.

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Re: [Freeswitch-users] Transcoding to GSM using SOX

[Brian West]

How are you trying to play these files?

/b


On Nov 3, 2008, at 9:01 PM, Brian Wood wrote:

> I'm trying to transcode a .wav to a .gsm using SOX v14.0.1. What I get
> is a completely garbled when using playback in FreeSWITCH. I have
> verified that at least one other application (Quicktime Player) can in
> fact play the resulting .gsm file.
>
> I have tried the following:
>
>sox william.wav -t gsm william.gsm
>sox william.wav -t gsm -r 8000 william.gsm resample -ql
>sox william.wav -t gsm -r 8000 -c1 william.gsm resample -ql
>
> And I am playing the file using:
>
>originate sofia/internal/[EMAIL PROTECTED]
> &playback(/home/user/william.gsm)
>
> Other formats such as uLaw and aLaw (-t ul and -t al, respectively)  
> seem
> to work fine.
>
> Does anyone know of a set of parameters for SOX that will make this  
> work?
>
> At the end of the day, I just want it to work, so if anyone can
> recommend any other known/good command line tools to encode GSM that
> would be fine too.
>
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Re: [Freeswitch-users] stun.freeswitch.org: No route to host

[Brian West]

Yes the IP did change but you shouldn't be relying on our STUN server  
for production.  I can take out your network with one issuance of  
pkill -9 stund  :P

You're better off running your own stun server if you depend on it for  
production.

/b

On Nov 4, 2008, at 8:02 AM, Birgit Arkesteijn wrote:

> Hi Brian,
>
> Thanks again for your reply.
>
> Looking at our firewall rules (STUN_FREESWITCH="208.64.203.130"), for
> some reason the IP address has changed from 208.64.203.130 to  
> 216.27.14.70.
> Does that ring a bell?
>
> I'll ask the person(s) in charge to change the rules.
>
> Cheers, Birgit


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Re: [Freeswitch-users] stun.freeswitch.org: No route to host

[Brian West]

This is what we run on stun.freeswitch.org http://sourceforge.net/projects/stun/

I highly recommend checking it out  ;)

/b

On Nov 4, 2008, at 8:02 AM, Birgit Arkesteijn wrote:

> Hi Brian,
>
> Thanks again for your reply.
>
> Looking at our firewall rules (STUN_FREESWITCH="208.64.203.130"), for
> some reason the IP address has changed from 208.64.203.130 to  
> 216.27.14.70.
> Does that ring a bell?
>
> I'll ask the person(s) in charge to change the rules.
>
> Cheers, Birgit
>
>
> On 04/11/08 13:32, Brian West wrote:
>> They are the same IP.
>>
>> /b
>>
>> On Nov 4, 2008, at 7:24 AM, Birgit Arkesteijn wrote:
>>
>>> It's odd that I can reach "freeswitch01.bandwidth.com", but not
>>> "stun.freeswitch.org"
>
> -- 
> -- Birgit Arkesteijn, [EMAIL PROTECTED],
> -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK
> -- Company no: 1769350
> -- Registered Office:
> -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK.
> -- tel.: +44 (0)161 237 0660
> -- http://www.westhawk.co.uk>
>
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Re: [Freeswitch-users] stun.freeswitch.org: No route to host

[Brian West]

I have two ip's bound on the same machine and start the stun server to  
use both.

/b

On Nov 4, 2008, at 8:19 AM, Birgit Arkesteijn wrote:

> 3. You must have two IPs bound to the machine that you want to setup  
> the
> STUN server on or, use two different machines. Preferrably, you would
> setup the STUN server to have the IPs on the local machine through
> ifconfig.
>
> I'd have to check with the more 'network savvy' if we have that  
> available.


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Re: [Freeswitch-users] Transcoding to GSM using SOX

[Brian Wood]

 From the CLI, I'm doing:

originate sofia/internal/[EMAIL PROTECTED] &playback(/home/user/william.gsm)


... just as a quick test. I've been using a combination of softphones 
and a Polycom 501 as the destination. The .gsm file plays fine in an 
application such as quicktime.  .wav, .al, and .ul seem to work fine 
with the playback command.


Brian West wrote:
> How are you trying to play these files?
>
> /b
>   


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Re: [Freeswitch-users] stun.freeswitch.org: No route to host

[Brian West]

They are the same IP.

/b

On Nov 4, 2008, at 7:24 AM, Birgit Arkesteijn wrote:

> It's odd that I can reach "freeswitch01.bandwidth.com", but not
> "stun.freeswitch.org"


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Re: [Freeswitch-users] Transcoding to GSM using SOX

[Brian West]

I use that every now and then it works fine... I'll have to retest  
this shortly and see... what sox version are you using?


/b

On Nov 4, 2008, at 8:28 AM, Brian Wood wrote:


From the CLI, I'm doing:

originate sofia/internal/[EMAIL PROTECTED] &playback(/home/user/ 
william.gsm)



... just as a quick test. I've been using a combination of softphones
and a Polycom 501 as the destination. The .gsm file plays fine in an
application such as quicktime.  .wav, .al, and .ul seem to work fine
with the playback command.


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Re: [Freeswitch-users] Transcoding to GSM using SOX

[Brian Wood]

SOX v14.0.1. I'm running FreeSWITCH trunk 10208.


Brian West wrote:
> I use that every now and then it works fine... I'll have to retest 
> this shortly and see... what sox version are you using?
>
> /b
>
> On Nov 4, 2008, at 8:28 AM, Brian Wood wrote:
>
>> From the CLI, I'm doing:
>>
>> originate sofia/internal/[EMAIL PROTECTED] 
>> <mailto:sofia/internal/[EMAIL PROTECTED]> &playback(/home/user/william.gsm)
>>
>>
>> ... just as a quick test. I've been using a combination of softphones 
>> and a Polycom 501 as the destination. The .gsm file plays fine in an 
>> application such as quicktime.  .wav, .al, and .ul seem to work fine 
>> with the playback command.
>
> 
>
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Re: [Freeswitch-users] Transcoding to GSM using SOX

[Brian Wood]

Confirmed. Fixed in 10238. Thanks!

Note to anyone else out there, you need to set the sample rate 
(otherwise playback is slow), so the command line you want is:

sox william.wav -t gsm -r 8000 william.gsm



Anthony Minessale wrote:
> nevermind, try latest trunk we found the issue.
>
>
> On Tue, Nov 4, 2008 at 9:20 AM, Anthony Minessale 
> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
>
> Try reversing the order you load mod_sndfile and mod_voipcodecs
> and let me know if it makes any diff.
>
>
>
> On Mon, Nov 3, 2008 at 9:01 PM, Brian Wood
> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
>
> I'm trying to transcode a .wav to a .gsm using SOX v14.0.1.
> What I get
> is a completely garbled when using playback in FreeSWITCH. I have
> verified that at least one other application (Quicktime
> Player) can in
> fact play the resulting .gsm file.
>
> I have tried the following:
>
>sox william.wav -t gsm william.gsm
>sox william.wav -t gsm -r 8000 william.gsm resample -ql
>sox william.wav -t gsm -r 8000 -c1 william.gsm resample -ql
>
> And I am playing the file using:
>
>originate sofia/internal/[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> &playback(/home/user/william.gsm)
>
> Other formats such as uLaw and aLaw (-t ul and -t al,
> respectively) seem
> to work fine.
>
> Does anyone know of a set of parameters for SOX that will make
> this work?
>
> At the end of the day, I just want it to work, so if anyone can
> recommend any other known/good command line tools to encode
> GSM that
> would be fine too.
>
> ___
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>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> iax:[EMAIL PROTECTED]/888
> <http://iax:[EMAIL PROTECTED]/888>
> googletalk:[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> pstn:213-799-1400
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>
> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>
> iax:[EMAIL PROTECTED]/888 
> <http://iax:[EMAIL PROTECTED]/888>
> googletalk:[EMAIL PROTECTED] 
> <mailto:[EMAIL PROTECTED]>
> pstn:213-799-1400
> 
>
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Re: [Freeswitch-users] Inbound calls question

[Brian West]

You need to first make sure the profile is on its own IP and port.   
Then I would recommend you turn auth-calls=false or use ACL's to  
authenticate the callers.

http://wiki.freeswitch.org/wiki/ACL


The external profile is a good one to look at for a base.. it listens  
on port 5080 and doesn't auth calls... and sends them to the public  
context.

/b


On Nov 4, 2008, at 10:24 AM, [EMAIL PROTECTED] wrote:

> Hi,
>
>How can my FS accept inbound SIP calls from other gateways
> without the need of a registration from their part? I only need to  
> be able
> to accept inbound calls from specific gateway IPs. I tried creating my
> own profile
> and gateway but it fails : "Error Creating SIP UA for profile:  
> myprofile"
>
> Can someone give me some first-step directions?


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Re: [Freeswitch-users] Wrong IP on ACK?

[Brian West]

You need to set localnet and externip or externhost on Asterisk so it  
doesn't lie about its IP.

/b

On Nov 4, 2008, at 11:10 AM, David Aldworth wrote:

> For some reason when trunking with Asterisk PBX's (yes, I know) FS
> wants to send the ACK to the internal ip found in the Contact field of
> the 200 OK. We have the force rport setting on but it's still not
> responding to that IP. Register's work. Most of the sip signalling
> works, just when the customer specifies the Contact filed with an
> internal ip. Below is a packet capture and our external.xml conf file.
>
> U 2008/11/04 09:17:08.259672 64.74.188.23:5060 -> 68.188.189.202:5060
> INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0.
> Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bKc6y6pr80HyeaN.
> Max-Forwards: 68.
> From: "user" ;tag=Dt6v81cDZXa3B.
> To: .
> Call-ID: de7c471c-252e-122c-3cba-5f1bac93b621.
> CSeq: 106789378 INVITE.
> Contact: .
> User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
> Supported: 100rel, timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, call-info, sla, include-session-
> description, presence.winfo, message-summary.
> Min-SE: 120.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 370.
>
> U 2008/11/04 09:21:32.861720 68.188.189.202:5060 -> 64.74.188.23:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> 64.74.188.23
> ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060.
> From: "user" ;tag=4UF788r8ct8aD.
> To: .
> Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
> CSeq: 106789510 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: .
> Content-Length: 0.
> .
>
> U 2008/11/04 09:21:32.861845 68.188.189.202:5060 -> 64.74.188.23:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 64.74.188.23
> ;branch=z9hG4bKZFem0FmX4g19H;received=64.74.188.23;rport=5060.
> From: "user" ;tag=4UF788r8ct8aD.
> To: ;tag=as1da4b7aa.
> Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
> CSeq: 106789510 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: .
> Content-Type: application/sdp.
> Content-Length: 285.
> .
> v=0.
> o=root 10970 10970 IN IP4 192.168.0.5.
> s=session.
> c=IN IP4 192.168.0.5.
> t=0 0.
> m=audio 15876 RTP/AVP 18 0 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
>
> U 2008/11/04 09:21:32.862573 64.74.188.23:5060 -> 192.168.0.5:5060
> ACK sip:[EMAIL PROTECTED] SIP/2.0.
> Via: SIP/2.0/UDP 64.74.188.23;rport;branch=z9hG4bK0r7c2a501SQvD.
> Max-Forwards: 70.
> From: "user" ;tag=4UF788r8ct8aD.
> To: ;tag=as1da4b7aa.
> Call-ID: 7c2d8578-252f-122c-3cba-5f1bac93b621.
> CSeq: 106789510 ACK.
> Contact: .
> Content-Length: 0.
> .
>
>
>
>
>
> external.xml
>
>
>   
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>   
>
> ___
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Re: [Freeswitch-users] Wrong IP on ACK?

[Brian West]

You can use the param "NDLB-force-rport" to force it to use rport no  
matter what.

/b

On Nov 4, 2008, at 11:43 AM, David Aldworth wrote:

> Hi bkw -
>
> We did that and it does indeed fix the issue. However, in the case
> that you have multiple SIP UA's behind a router there tend to be many
> dynamically generated ports in use. The obvious solution would be to
> statically map a port to an internal IP and then set the externip and
> localhost settings. I agree, this would work. Except if you are using
> a dsl or cable modem provider that also like to update your WAN ip on
> a regular basis. But what confuses me more is that all the sip
> messaging works fine right up until the ACK we send back to the 200
> OK. Obviously FS is sending the ACK to the Contact field IP but is
> there a way in FS to tell it to just respond on the IP and port that
> the 200 OK came from? I thought that is what the force rport setting
> did but i guess it does not.
>
> David


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Re: [Freeswitch-users] Wrong IP on ACK?

[Brian West]

Make sure you restart the profile for it to take effect.

/b

On Nov 4, 2008, at 11:49 AM, David Aldworth wrote:

> That is actually already on.
>
> Any idea?


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Re: [Freeswitch-users] RTMP Support (Flash)

[Brian West]

Do you have any links about this project?

/b

On Nov 4, 2008, at 6:47 AM, jocke eriksson wrote:

> Or one could wait for project pacifica to get up to speed in the  
> Adobe labs, witch i think would be a better solution.


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Re: [Freeswitch-users] Error Creating SIP UA for profile: internal-ipv6

[Brian West]

Chances are you don't have IPv6 enabled.  You can ignore it.

/b

On Nov 4, 2008, at 12:49 PM, [EMAIL PROTECTED] wrote:

I'm getting this error when ever I start FS and wonder what the  
cause is and if I need to fix it.


[ERR] sofia.c:554 sofia_profile_thread_run() Error Creating SIP UA  
for profile: internal-ipv6


Here is the console output:


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Re: [Freeswitch-users] stun.freeswitch.org: No route to host

[Brian West]

Nope its not down at all.  Works perfectly fine here.

/b

On Nov 4, 2008, at 7:07 AM, Birgit Arkesteijn wrote:

> Hi,
>
> Is stun.freeswitch.org down?
> Our FreeSWITCH installation broke overnight (no configuration changes
> made) because stun can no longer be reached.
>
> # telnet stun.freeswitch.org 3478
>  Trying 216.27.14.70...
>  telnet: connect to address 216.27.14.70: No route to host
>
> # traceroute stun.freeswitch.org
> ends with ..
> 18  * dr1-g2-2.cry.hostedsolutions.com (216.27.30.86)  76.936 ms   
> 77.497 ms
>  19  freeswitch01.bandwidth.com (216.27.14.70)  81.201 ms !X  81.730  
> ms
> !X  81.900 ms !X
>
> Cheers, Birgit


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Re: [Freeswitch-users] Error Creating SIP UA for profile: internal-ipv6

[Brian West]

I think he was talking to the guys that did the project integration on  
windows about that.  I have personally only used it on linux and mac.


/b

On Nov 4, 2008, at 2:02 PM, [EMAIL PROTECTED] wrote:


Thanks Brian,

Also, would you be kind enough to ask Mike Jerris if he has gotten  
my emails about the problem I'm having with the "pizza" demo  
crashing on Windows.
I haven't heard back from him in a few days and was wondering if  
he'd seen it or he's just to busy right now.


Best regards.

Mark.


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Re: [Freeswitch-users] Question on call transfer

[Brian West]

We already support this in the latest svn trunk:

Here are the options from the internal.xml sip_profile that would  
allow you to do this.







On Nov 5, 2008, at 5:39 AM, Rajagopal, Sridhar (Sridhar) wrote:

> Hi all,
>
> I am planning to implement changes in freeswitch code to support call
> transfer even during media bypass mode.
> Please provide me some insight on how this can be done.
>
> Thanks in advance and appreciate ur help.
>
> Regards
> Sridhar


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Re: [Freeswitch-users] Anybody tried with Trunk between asterisk and freeswitch...?

[Brian West]

http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

/b

On Nov 5, 2008, at 12:31 AM, sambasivarao Vemula wrote:



Hi,
Any body tried with Trunkig between freeswitch and asterisk .
If any body tried and its working fine .
Please share the details.
Regards
Samba


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Re: [Freeswitch-users] Voicemail remote dialing in to retrieve messages

[Brian West]

Something like this would do it:

 
 
   
 

 
   
 

This is in the default config as of a few weeks ago.

/b



On Nov 5, 2008, at 12:52 AM, David Walker wrote:

> I have been using freeswitch for over 18 months and watching it  
> being developed since its inception. This is one great software!!! I  
> have been able to do many things that I gathered from the list along  
> with the wiki and hope to contribute back. Which leads me to ask has  
> anyone created a remote dial in retrieval dialplan to check  
> voicemail if your out of the office? I cannot find any info on this  
> kind of procedure and I do not want to recreate something if it has  
> already been developed, but again I might not be looking in the  
> right place.


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Re: [Freeswitch-users] Problem in Routing G729A Calls

[Brian West]

I would need to see a sip trace of this taking place.  If you're using  
the passthru codec we do pass the fmtp options thru when we receive  
them.

/b

On Nov 5, 2008, at 8:26 AM, shehzad p wrote:

>
>
> I have to route the inbound calls of G729A codec.
> Calls comes to my freeswitch with codec G729A (As "annexb=no" is set)
>
> But when i route calls to termination gateway, calls are dropped  
> (because
> of "annexb=no " is not set)
>
> Why "annexb=no" is removed while i route the calls?
> How can I set "annexb=no'? (I am using javascript for routing the  
> calls)
>
> Does following SDP variables can help me in solving above problem?  
> How to
> use those variables?
> http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation
>
> Warm thanks in advance...
> MSP
> -- 
> View this message in context: 
> http://www.nabble.com/Problem-in-Routing-G729A-Calls-tp20342694p20342694.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>

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Re: [Freeswitch-users] Inband DTMF Problem

[Brian West]

Inband DTMF on voip isn't the most reliable thing in the first  
place... anyway to use RFC2833 or SIP Info?

/b

On Nov 5, 2008, at 9:08 AM, Klaus Teller wrote:

> Hi Folks,
>
> It's me again with a DTMF issue. Here is what's going on. I have a  
> remote IVR and i'm writing some code to communicate with it,  
> exercise it, and test it. The remote IVR is playing DTMF as inband.
>
> Right now, Freeswitch can detect all DTMF digits from the IVR. The  
> interaction between my code and the IVR is essentially  a  
> combination of  playing files and sending and reading DTMFs.
>
> The issue: whenever my code plays a file or send a DTMF digit, the  
> following attempt to read DTMF fails. By failing i mean that there  
> will be a spurious DTMF detected by Freeswitch that was not sent by  
> the IVR.
>
> That is, if i have something like:
>
> sendDTMF("3"); getDigits("38373", "#");  // # is the termination  
> character
>
> After executing the sendDTMF, getDigits fails because it immediately  
> detects a pound. Yet this pound is not sent by the IVR. I can check  
> this by having wireshark running both on the IVR's computer and on  
> Freeswitch's computer.
>
> Any idea what may be causing this problem? Is there some sort of  
> Echo going on or something like that?
>
> Thanks,
> Klaus.


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Re: [Freeswitch-users] FS and MySQL

[Brian West]

You can use JS and ODBC to access mysql... pick lua, perl, java,  
python, mono/c#

/b
PS: Your milage will vary depending on what is available per language.

On Nov 5, 2008, at 9:40 AM, Helmut Kuper wrote:

> Hello,
>
> is there a way in freeswitch to do a mysql query from the dialplan?
>
> regards
> helmut


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Re: [Freeswitch-users] Inband DTMF Problem

[Brian West]

Extract the RTP audio into wav/au files in wireshark and lets look at  
them.

/b

On Nov 5, 2008, at 2:30 PM, Klaus Teller wrote:

> Hi Anthony,
>
> Let me add some few facts.
>
> 1) I am sending DTMF using RFC2833 and the IVR is using Inband
> 2) The problem is not limited to when i send DTMF. It also happens  
> when i try to play a file.
> 3) The functions are from my java code. They send commands to  
> Freeswitch via socket interface.
> 4) The same functions are working perfectly when the remote IVR  
> sends DTMF via RFC2833.
> 5) I have a pause of 2000 ms after sending the DTMF.
>
>
> Let me reformulate my problem. I have a spurious dtmf-# being  
> detected by freeswitch where nothing was sent by the other party.
>
> The strange thing to me is that Wireschark (running on the same  
> machine as Freeswitch) detects the proper digit sequence.
>
> Klaus.


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Re: [Freeswitch-users] Wrong IP on ACK?

[Brian West]

0.0.0.0/0 should match all IP space.

/b

On Nov 5, 2008, at 5:16 PM, David Aldworth wrote:

> Anthony, In hopes of matching all IP's we added a very simple:
>
> 
> 
>
> To the acl.conf.xml and we added:
>
> 
>
> To the sip profile. Unfortunately there was no affect. What would be  
> the correct acl to match all IP's?
>
> David


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Re: [Freeswitch-users] Different problems with "pizza" demo on newer build.

[Brian West]

Looks like the score is messed up.  Are you using ps_pizza.js?

/b

On Nov 5, 2008, at 6:44 PM, [EMAIL PROTECTED] wrote:

OK, I did my first FS "build" today from download  
"freeswitch-1.0.1.tar.gz" on Windows XP Pro with Visual C++ 2008  
Express instead of using my previous MSI installed build which  
apparently uses the older FS version 1.0.9570.


I'm not getting a crash as I did before with the MSI build on the  
"pizza" demo. Instead, in the newer build, the demo gets stuck in a  
loop and can't break out no matter what I say. See below:


Any advice?

Thank you in advance.
Mark.


2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Heard [ε■ε■]
2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Hit score 0/1/78
2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:382  
console_log() We don't understand this
2008-11-05 16:22:21 [DEBUG] mod_pocketsphinx.c:342  
pocketsphinx_asr_resume() Manually Resuming
2008-11-05 16:22:21 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:25 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:25 [DEBUG] switch_core_session.c:430  
switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 
 [BREAK]
2008-11-05 16:22:26 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:26 [DEBUG] mod_pocketsphinx.c:389  
pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0
2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() XML:


  TAKEOUT
  TAKEOUT

2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Heard [TAKEOUT]
2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Hit score 0/1/78
2008-11-05 16:22:26 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:382  
console_log() We don't understand this
2008-11-05 16:22:26 [DEBUG] switch_core_session.c:430  
switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 
 [BREAK]
2008-11-05 16:22:27 [DEBUG] mod_pocketsphinx.c:342  
pocketsphinx_asr_resume() Manually Resuming
2008-11-05 16:22:27 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:33 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:33 [DEBUG] switch_core_session.c:430  
switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 
 [BREAK]
2008-11-05 16:22:34 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:34 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:34 [DEBUG] switch_core_session.c:430  
switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 
 [BREAK]
2008-11-05 16:22:35 [DEBUG] mod_pocketsphinx.c:389  
pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0
2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() XML:


  TAKEOUT
  TAKEOUT

2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Heard [TAKEOUT]
2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Hit score 0/1/78
2008-11-05 16:22:35 [DEBUG] js_modules/SpeechTools.jm:382  
console_log() We don't understand this
2008-11-05 16:22:37 [DEBUG] mod_pocketsphinx.c:342  
pocketsphinx_asr_resume() Manually Resuming
2008-11-05 16:22:37 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:42 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:42 [DEBUG] switch_core_session.c:430  
switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 
 [BREAK]
2008-11-05 16:22:43 [DEBUG] mod_pocketsphinx.c:389  
pocketsphinx_asr_get_results() Recognized: DELIVERY, Score: 0
2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() XML:


  DELIVERY
  DELIVERY

2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Heard [DELIVERY]
2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Hit score 0/1/78
2008-11-05 16:22:43 [DEBUG] js_modules/SpeechTools.jm:382  
console_log() We don't understand this
2008-11-05 16:22:44 [DEBUG] mod_pocketsphinx.c:342  
pocketsphinx_asr_resume() Manually Resuming
2008-11-05 16:22:44 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:50 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:50 [DEBUG] switch_core_session.c:430  
switch_core_session_receive_message() Kill so

[Freeswitch-users] test

[Brian West]

test

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[Freeswitch-users] test

[Brian West]

test

ignore please.

/b


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[Freeswitch-users] test again

[Brian West]

test

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[Freeswitch-users] Mailing list Maint.

[Brian West]

I have done a little bit of maintenance on our list server tonight.   
Things will level out as DNS starts to settle down.


Please report any issues to me directly off list please.

/b

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Re: [Freeswitch-users] avoiding moh for meta_app

[Brian West]

Try export instead of set here.

/b

On Nov 6, 2008, at 7:30 AM, Woody Dickson wrote:

>  


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Re: [Freeswitch-users] Anybody tried with Trunk between asterisk and freeswitch...?

[Brian West]

Was this not helpful? 
http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

/b

On Nov 6, 2008, at 8:23 AM, Peter P GMX wrote:

> It's rather simple
> - Setup a sip user on asterisk with username/password
> - Setup a gateway in freeswitch with the asterisk user credentials  
> (ip,
> username, password of asterisk)
> - Define a route in the dialplan (e.g. default.xml) to route certain
> numbers to the asterisk gateway
> e.g.
> 
> 
> 
> 
>  data="sofia/gateway/asterisk/[EMAIL PROTECTED]"/>
> 
> 
>
> You should already be able to make outgoing calls via asterisk.
>
> Best regards
> Peter
>
>
> sambasivarao Vemula schrieb:
>>
>>
>>
>> Hi,
>>
>> Any body tried with Trunkig between freeswitch and asterisk .
>>
>> If any body tried and its working fine .
>>
>> Please share the details.
>>
>> Regards
>>
>> Samba
>>
>>
>>
>>
>>
>> DISCLAIMER == This e-mail may contain privileged and
>> confidential information which is the property of Persistent Systems
>> Ltd. It is intended only for the use of the individual or entity to
>> which it is addressed. If you are not the intended recipient, you are
>> not authorized to read, retain, copy, print, distribute or use this
>> message. If you have received this communication in error, please
>> notify the sender and delete all copies of this message. Persistent
>> Systems Ltd. does not accept any liability for virus infected mails.
>>
>> 
>>
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Re: [Freeswitch-users] att_xfer+loopback

[Brian West]

Viktor,
	For the user channel its user/[EMAIL PROTECTED]  ... as for the rest  
can you show me your entire config?


http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer

/b

On Nov 6, 2008, at 9:45 AM, x y wrote:


Hi!

I'm new to freeswitch, and I'm trying to make an att_xfer in a  
dialplan, but instead of giving a sofia/${domain}/${called_number}  
as , i would like to use a loopback/${called_number},  
because i would like to transfer the call not just to different  
extensions. Is there any way to achive this? When i'm trying to do  
this like that:




in a A-call->B-att_xfer->C situation, A gets hanged up, as soon as  
the bridge has been estabilished between B and C.


Btw, i have tryed out att_xfer by giving user/${legal_user} as  
. I've found att_xfer this way a kind of instable,  
sometimes it worked perfectly, sometimes not: A and C did not hang  
up, but there weren't succesfully connected (1 time from 10). The  
log printed the same at both cases.


Cheers:
Viktor


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[Freeswitch-users] More codec playback issues

[Brian Wood]

I'm doing some more testing with various codecs. I'm seeing some more 
weirdness with the selection of a codec by file extension. I'm testing 
this on FreeSWITCH trunk 10270.

The following script works with the .wav and .gsm extension, but 
everything else seems to fail.

Using lowercase names (such as .g722) fail when recording with a "[ERR] 
switch_core_file.c:66 switch_core_perform_file_open() Invalid file 
format [g722] for [/tmp/recording.g722]!". Which is fine, I can use 
uppercase, but then that plays back garbled audio.

This script should beep, record until you stop speaking, beep again, 
pause, beep, playback, beep, hang up.


   var allDigits = "";
   function on_dtmf(session, type, obj, arg)
   {
  try {
 if (type == "dtmf") {
allDigits += obj.digit;
console_log("info", "DTMF digit: " + session.name + " [" + 
obj.digit + "] len [" + obj.duration + "]\n\n");
 }
  } catch (e) {
 console_log("err", e + "\n");
  }
  return true;
   }


// .gsm works, .wav works
//filename = "/tmp/recording.gsm"
//filename = "/tmp/recording.wav"

// These all result in garbled playback
//filename = "/tmp/recording.GSM"
filename = "/tmp/recording.G722"
/filename = "/tmp/recording.SPEEX"
//filename = "/tmp/recording.LPC"
//filename = "/tmp/recording.PCMU"
//filename = "/tmp/recording.PCMA"
//filename = "/tmp/recording.AMR"



session.waitForAnswer(1);
session.execute("sleep", "2000");

console_log("info", "*BEGIN*");
session.execute("gentones", "%(500,0,800)");
rtn = session.recordFile(filename, on_dtmf, "", 240, 500, 3);
session.execute("gentones", "%(500,0,800)");
console_log("info", "*END*");


session.execute("sleep", "2000");
session.execute("gentones", "%(500,0,800)");
session.streamFile(filename);
session.execute("gentones", "%(500,0,800)");

session.execute("sleep", "1000");
session.hangup();


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Re: [Freeswitch-users] More codec playback issues

[Brian West]

When you're doing a playback with the native codecs you shouldn't put  
any extension on it.  It will automagically select the file based on  
the current codec of the channel.

filename = "/tmp/recording"

Then have a file with all the codecs you support then FreeSWITCH will  
pick the right one.


On Nov 6, 2008, at 1:06 PM, Brian Wood wrote:

> I'm doing some more testing with various codecs. I'm seeing some more
> weirdness with the selection of a codec by file extension. I'm testing
> this on FreeSWITCH trunk 10270.
>
> The following script works with the .wav and .gsm extension, but
> everything else seems to fail.
>
> Using lowercase names (such as .g722) fail when recording with a  
> "[ERR]
> switch_core_file.c:66 switch_core_perform_file_open() Invalid file
> format [g722] for [/tmp/recording.g722]!". Which is fine, I can use
> uppercase, but then that plays back garbled audio.

These are native file formats and should only play a G722 to a G722  
channel.  They are also case sensitive.

> This script should beep, record until you stop speaking, beep again,
> pause, beep, playback, beep, hang up.

What exactly are you trying to accomplish?

/b

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Re: [Freeswitch-users] More codec playback issues

[Brian Wood]

Brian West wrote:
> When you're doing a playback with the native codecs you shouldn't put  
> any extension on it.  It will automagically select the file based on  
> the current codec of the channel.
>
> filename = "/tmp/recording"
>
> Then have a file with all the codecs you support then FreeSWITCH will  
> pick the right one.
>   

But what if I want to force a specific codec?

>> This script should beep, record until you stop speaking, beep again,
>> pause, beep, playback, beep, hang up.
>> 
>
> What exactly are you trying to accomplish?
>
>   
Ultimately, what I'm trying to do is prevent FreeSWITCH from transcoding 
on the fly. I would like to have prompts and menus pre-encoded in all 
the codecs I expect to be using with my various endpoints and devices.

That script is just a really stupid test that I'll never use for 
anything. I was trying to make SOX create a .G722 file, but it was 
playing back all garbled, much like my GSM experience earlier in the 
week. This script is an attempt to force FreeSWITCH to record in a 
specific codec to determine if the problem was with SOX and the other 
G722 encoder I tried.

I may be trying to do something that really isn't necessary, but it 
seems like you should be able to do it.



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Re: [Freeswitch-users] More codec playback issues

[Brian West]

On Nov 6, 2008, at 1:46 PM, Brian Wood wrote:

> Brian West wrote:
>>
> But what if I want to force a specific codec?

You turn on inbound late negotiation and make the decision in the  
dialplan.

>>> This script should beep, record until you stop speaking, beep again,
>>> pause, beep, playback, beep, hang up.
>>>
>>
>> What exactly are you trying to accomplish?
>>
>>
> Ultimately, what I'm trying to do is prevent FreeSWITCH from  
> transcoding
> on the fly. I would like to have prompts and menus pre-encoded in all
> the codecs I expect to be using with my various endpoints and devices.

Have the files in all the codecs and use playback without a file  
extension it'll pick the one that matches the current codec.

> That script is just a really stupid test that I'll never use for
> anything. I was trying to make SOX create a .G722 file, but it was
> playing back all garbled, much like my GSM experience earlier in the
> week. This script is an attempt to force FreeSWITCH to record in a
> specific codec to determine if the problem was with SOX and the other
> G722 encoder I tried.

We fixed the error in the GSM codec earlier this week.  Not sure that  
sox will create the right file format for G722.  I might be wrong!

>
>
> I may be trying to do something that really isn't necessary, but it
> seems like you should be able to do it.
>
>
>
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Re: [Freeswitch-users] More codec playback issues

[Brian Wood]

Brian West wrote:
> You turn on inbound late negotiation and make the decision in the  
> dialplan.
>   
But I'm trying to do this on outbound/originate sessions. So nothing's 
going through the dialplan. :) I'm initiating the script with originate 
commands, from the CLI or XML-RPC.

> We fixed the error in the GSM codec earlier this week.  Not sure that  
> sox will create the right file format for G722.  I might be wrong!
>
>   

I'm also not sure if SOX is producing a valid file. I also tried using 
the ITU's ansi-c encoder with the same result, but I'm not completely 
convinced it works either. I was trying to see if I could make 
FreeSWITCH record and play back in various codecs, so I would have a 
known/good file to try to decode with SOX and the ITU's code.


I also tried changing the filename in my script to /tmp/recording. I've 
seen the auto extension select work for playback. If I don't specify an 
extension when recording, it throws an error about the file format being 
invalid. If I record as /tmp/recoding.G722 and playback as 
/tmp/recording, it auto-selects the .G722 codec because that's what my 
phone is using, but the playback is garbled.


So, I guess the better question for me to ask... is specifying a 
extension to session.recordFile() the correct way to force recording in 
a specifc codec (if, for some reason you _really_ want to do this)? OR, 
is that just not supported and why I get garbled playback? .gsm and .wav 
seem to work, and a hexdump of the files looks sane, so why not the others?

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Re: [Freeswitch-users] Different problems with "pizza" demo on newer build.

[Brian West]

gmail got caught in the cross fire of the mail list server move  
lastnight... they dont' like it when you change IP's apparently.  It  
should be flowing fine as of this morning.


Can you forward me the warning?

/b

On Nov 6, 2008, at 4:22 PM, [EMAIL PROTECTED] wrote:

Anthony, look in your gmail account since I got a warning about my  
email being to long to be placed here.


Mark.


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Re: [Freeswitch-users] att_xfer+loopback

[Brian West]

Why aren't you prefixing the $1 with 668 in the first example?

/b

On Nov 7, 2008, at 2:03 AM, x y wrote:


Btw, the xfer is a transfer with loopback channel, and it works fine.


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Re: [Freeswitch-users] Recording through Default.xml not working

[Brian West]

I have just tested this again and it works fine.  The called extension  
pressing *2 and it record fine.  Are you talking about your user of  
record_session?  Because if you are already recording it.. why even  
have the bind_meta option to record it?


/b

On Nov 6, 2008, at 11:59 PM, Baskar wrote:


Hi,

 Recoding is done through default.xml.
 For past 1 month recoding is working fine.
 Suddenly for past 3day recording is not working.
 I did not modify any thing in deafult .xml .

 My deafult.xml file



  
   
   

   
   
   
   
   
   
   

   
   
   

   
   
   
   
   
   
   















  


   
  

  
  
  
  
  data="effective_caller_id_number=true"/>

  
  
  
  
  

 
   
   
  




  


  


correct me were is wrong

--
Warm Regards,
N.Baskar

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Re: [Freeswitch-users] Socket outbound: How to bridge two calls?

[Brian West]

On Nov 7, 2008, at 4:30 AM, Dennis wrote:

> ok, now i understand to way intercept should work (i was always
> wonderung, why there only was one uuid).
>
> BUT, it still does not work. Could you please be so kind and have a
> look at http://pastebin.freeswitch.org/6033. there you can see the
> results of our latest tests.
>
>
> 1.) intercept with inbound-uuid and (originated) outbound-uuid (now
> the right way) => does not work for us.

Can you clarify this?  It makes no sense.  You should already have two  
calls established before you run intercept.

> 2.) uuid_bridge with inbound-uuid and (originated) outbound-u
> does not work for us.
> this does not work either. we do not get any events with
> channel_bridge or something. after we make a bridge, nothing happens
> till we hang up.
>
> might it be possible, that there is a reason that i does not work and
> that the reason is the same because intercept does not work?

I made two calls that where either parked or in a conference and I  
execute uuid_bridge and they bridge as expected.


> 3.) uuid_bridge with two originated calls => WORKS
> we have an inbound call and then we make two originates. then we
> uuid_bridge both originated calls. we can talk to each other and get
> all channel events like bridge and unbridge.
>
> it seems, that there is a problem, when we want to do the whole thing
> with the inbound.
>
> the only problem is, that we originate both calls with
> hangup_after_bridge=false, but as soon as one of the bridged
> originated calls hangs up, the other call is also beeing hangup.
>
>
> we just can't see, what we are doing wrong or what we could try or
> change to make it work.
>
>
> thanks,
> dennis
>
>
>
> 2008/11/6 Anthony Minessale <[EMAIL PROTECTED]>:
>> you are missing execute-app-arg
>>
>> sendmsg 
>> call-command: execute
>> execute-app-name: intercept
>> execute-app-arg: > (outbound leg)
>
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Re: [Freeswitch-users] att_xfer+loopback

[Brian West]

Didn't you say in your previous email using loopback worked?

/b

On Nov 7, 2008, at 8:32 AM, x y wrote:



Because I just wantet to try that the att_xfer is working with user  
channel. The 668 and 669 perfixes are only made to be assured about  
that my extensions will be processed instead of the default  
config's, and because of that, I could transfer the call to any  
dialable number in the config later on. The usage of the user  
channel wont help me in this matter, cos its very similar to the  
original sofia method in the att_xfer example on freeswitch wiki.  
Basicly, i just want to know that there is any option to use  
loopback channel in att_xfer.


Cheers,
Viktor


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Re: [Freeswitch-users] Different problems with "pizza" demo on newer build.

[Brian West]

-rw-r--r--  2 brian  staff  1325 Jul  8 20:35 pizza_arso/pizza_arso.lm
-rw-r--r--  2 brian  staff  1797 Jul  8 19:51 pizza_crust/pizza_crust.lm
-rw-r--r--  2 brian  staff   667 Jul  7 20:08 pizza_order/pizza_order.lm
-rw-r--r--  2 brian  staff  1122 Jul  8 19:46 pizza_size/pizza_size.lm
-rw-r--r--  2 brian  staff  2529 Jul  8 20:05 pizza_specialty/ 
pizza_specialty.lm
-rw-r--r--  2 brian  staff  3367 Jul  8 20:30 pizza_toppings/ 
pizza_toppings.lm

-rw-r--r--  2 brian  staff   885 Jul  8 19:55 pizza_type/pizza_type.lm
-rw-r--r--  2 brian  staff   724 Jul  7 20:18 pizza_yesno/pizza_yesno.lm

They all look fine to me.  Where did you download yours from?

/b


On Nov 7, 2008, at 1:22 PM, [EMAIL PROTECTED] wrote:


Hi Anthony and Brian.

I should have done this yesterday but I'm learning debugging in this  
IDE environment as I go.

Anyway, have you looked at the .lm grammar files in the "pizza demo"?

For the grammar file download in the wiki, I finding 9 .lm files:

6859, pizza_arso, pizza_crust, pizza_order, pizza_size,  
pizza_specialty, pizza_toppings, pizza_type, pizza_yesno.


Of these, 4 files are empty:

pizza_crust, pizza_size, pizza_specialty, pizza_toppings.

I commented out the related sections in ps_pizza.js and the "pizza  
demo" seemed to work in this shortened form.
It took the type of order (nothing else), order confirmation, when  
to come pick-up and then hung-up.


Could I get the rest of the .lm files? Updating the grammar files  
download for "pizza" would nice.


Thanks.

Mark.



-Original Message-
From: Anthony Minessale <[EMAIL PROTECTED]>
To: freeswitch-users@lists.freeswitch.org
Sent: Thu, 6 Nov 2008 6:22 am
Subject: Re: [Freeswitch-users] Different problems with "pizza" demo  
on newer build.


can you please download and build the development snapshot instead  
of the 1.0.1 since we are almost to 1.0.2 now.


http://files.freeswitch.org/freeswitch-snapshot.tar.gz


On Wed, Nov 5, 2008 at 10:05 PM, <[EMAIL PROTECTED]> wrote:
Yup, I'm using ps_pizza.js.

Mark

-Original Message-
From: Brian West <[EMAIL PROTECTED]>
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 5 Nov 2008 4:50 pm
Subject: Re: [Freeswitch-users] Different problems with "pizza" demo  
on newer build.


Looks like the score is messed up.  Are you using ps_pizza.js?

/b

On Nov 5, 2008, at 6:44 PM, [EMAIL PROTECTED] wrote:

OK, I did my first FS "build" today from download  
"freeswitch-1.0.1.tar.gz" on Windows XP Pro with Visual C++ 2008  
Express instead of using my previous MSI installed build which  
apparently uses the older FS version 1.0.9570.


I'm not getting a crash as I did before with the MSI build on the  
"pizza" demo. Instead, in the newer build, the demo gets stuck in a  
loop and can't break out no matter what I say. See below:


Any advice?

Thank you in advance.
Mark.


2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Heard [ε■ε■]
2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Hit score 0/1/78
2008-11-05 16:22:19 [DEBUG] js_modules/SpeechTools.jm:382  
console_log() We don't understand this
2008-11-05 16:22:21 [DEBUG] mod_pocketsphinx.c:342  
pocketsphinx_asr_resume() Manually Resuming
2008-11-05 16:22:21 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:25 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:25 [DEBUG] switch_core_session.c:430  
switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 
 [BREAK]
2008-11-05 16:22:26 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:26 [DEBUG] mod_pocketsphinx.c:389  
pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 0
2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() XML:


  TAKEOUT
  TAKEOUT

2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Heard [TAKEOUT]
2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:150  
console_log() Hit score 0/1/78
2008-11-05 16:22:26 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:26 [DEBUG] js_modules/SpeechTools.jm:382  
console_log() We don't understand this
2008-11-05 16:22:26 [DEBUG] switch_core_session.c:430  
switch_core_session_receive_message() Kill sofia/internal/[EMAIL PROTECTED] 
 [BREAK]
2008-11-05 16:22:27 [DEBUG] mod_pocketsphinx.c:342  
pocketsphinx_asr_resume() Manually Resuming
2008-11-05 16:22:27 [DEBUG] switch_ivr_play_say.c:912  
switch_ivr_play_file() Codec Activated [EMAIL PROTECTED] 1 channels 20ms
2008-11-05 16:22:33 [DEBUG] switch_ivr_play_say.c:1203  
switch_ivr_play_file() done playing file
2008-11-05 16:22:33 [DEBUG] switch_core_

Re: [Freeswitch-users] Increase max-sessions and sessions-per-second

[Brian West]

Double check to make sure you're editing the right switch.conf.xml in / 
usr/local/freeswitch/conf/autoload_configs/

/b

On Nov 8, 2008, at 9:12 AM, shehzad p wrote:

>
>
> I need to increase Max Sessions, and sessions-per-second
> In switch.conf.xml, after increasing "max-session" (originally 1000)  
> to
> 2000,
> and then start freeswitch, there it still shows 1000 Max session  
> only?,
>
> I can increase it by CLI command "fsctl max_session 2000",
> but only from configuration file it doesn't work.
>
> Same happens for session-per-session
>
> I suspect that switch.conf.xml file is not being parsed,  
> whilefreeswitch
> starts
> but again all xml files are included in freeswitch.xml file as below:
>  
>   
>  
>
> So what would be the cause.
> Thanks...
> MSP
>
> -- 
> View this message in context: 
> http://www.nabble.com/Increase-max-sessions-and-sessions-per-second-tp20396980p20396980.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Problems with ./configure on OSX

[Brian West]

try ./bootstrap.sh again

/b

On Nov 9, 2008, at 2:14 PM, martin joseph wrote:

> Hey FreeSwitchers,


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Re: [Freeswitch-users] Strange Sofia error

[Brian West]

It should be harmless.

/b

On Nov 9, 2008, at 4:18 PM, Noah Silverman wrote:

>
> sofia.c:197 sofia_event_callback() event [nua_r_options] status [408]
> [Request Timeout] session: n/a


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Re: [Freeswitch-users] Strange Sofia error

[Brian West]

That tells me you're running older code... can you tell me what rev  
you're on?

/b

On Nov 9, 2008, at 5:06 PM, Noah Silverman wrote:

> It seems to be happening when calls get dropped.  When speaking to one
> of my users, calls get dropped after 2-4 minutes.
>
> It seems like I get a 408 error followed by a "bad frame" error.
>
> It only happens with this particular user.  Works fine for everyone
> else.  Could this be a function of a lousy DSL line or maybe his weird
> firewall?
>
> Here is what I see in the debug as the drop happens:


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Re: [Freeswitch-users] Strange Sofia error

[Brian West]

Oh after looking closer you have NAT problems?  Since  
recovery_on_timer_expire is being set as the hangup cause.


/b

On Nov 9, 2008, at 5:06 PM, Noah Silverman wrote:

> RECOVERY_ON_TIMER_EXPIRE


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Re: [Freeswitch-users] Different problems with "pizza" demo on newer build.

[Brian West]

On Nov 9, 2008, at 12:25 PM, [EMAIL PROTECTED] wrote:


OK Brian.

Do you know off hand if I would have to use them when making up a  
new grammar?


http://www.speech.cs.cmu.edu/tools/lmtool.html




Also, if you didn't catch this in my previous post, I have a couple  
requests/suggestion/questions for your teams consideration:


1) This page (http://wiki.freeswitch.org/wiki/Mod_pocketsphinx) on  
Mod Pocketsphinx needs updating to indicate this easy to use  
resource for building grammar files but if everyone is busy I could  
do it if there are no objections.


Please feel free to update the wiki.  Thats why its there.. its a  
community project.





2) I believe the latest "snapshot" release of FS that Anthony wanted  
me to try was using Pocketsphinx version 0.499 but the latest stable  
version of pocketsphinx is 0.5 (http://sourceforge.net/forum/forum.php?forum_id=843940 
). It'sabout 50% smaller and up to 18% faster than the previous one,  
and introduces a new, re-entrant and "modern" API. Are you planning  
to use this updated version of pocketsphinx soon?


Sphinx base is 0.499 and PocketSphinx is 0.599...  We can't use  
anything in the PocketSphinx 0.4x range as its NOT threadsafe.







Thanks again.

Mark.


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Re: [Freeswitch-users] Problems with ./configure on OSX

[Brian West]

install the latest dev tools CD/DVD image from connect.apple.com

/b

On Nov 9, 2008, at 2:14 PM, martin joseph wrote:

> Hey FreeSwitchers,
>
> I have been running 1.01 (which I built from a tarball) without any
> problems.


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Re: [Freeswitch-users] How to find an extension in public.xml ?

[Brian West]

You would use the transfer app.

/b

On Nov 10, 2008, at 9:30 AM, henkoegema wrote:

>> How do I do that ?


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Re: [Freeswitch-users] Different problems with "pizza" demo on newer build.

[Brian West]

Well on linux I install a makefile in /usr/local/freeswitch/grammar  
that will take all .sent files and build both the .lm and .dic files  
based on the .sent files.  On windows I haven't tested the perl  
scripts I wrote for this on linux... chances are they'll just work.  I  
highly recommend you download the files from here that are zero  
length.  http://www.bkw.org/pizza/



/b

On Nov 9, 2008, at 12:25 PM, [EMAIL PROTECTED] wrote:


OK Brian.

Do you know off hand if I would have to use them when making up a  
new grammar?


Also, if you didn't catch this in my previous post, I have a couple  
requests/suggestion/questions for your teams consideration:


1) This page (http://wiki.freeswitch.org/wiki/Mod_pocketsphinx) on  
Mod Pocketsphinx needs updating to indicate this easy to use  
resource for building grammar files but if everyone is busy I could  
do it if there are no objections.


2) I believe the latest "snapshot" release of FS that Anthony wanted  
me to try was using Pocketsphinx version 0.499 but the latest stable  
version of pocketsphinx is 0.5 (http://sourceforge.net/forum/forum.php?forum_id=843940 
). It'sabout 50% smaller and up to 18% faster than the previous one,  
and introduces a new, re-entrant and "modern" API. Are you planning  
to use this updated version of pocketsphinx soon?



Thanks again.

Mark.



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Re: [Freeswitch-users] How to find an extension in public.xml ?

[Brian West]

As per the wiki page

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer


/b

On Nov 10, 2008, at 10:10 AM, henkoegema wrote:

> Just something like:
>   (??)
> at the end of the file default.xml ?
>


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Re: [Freeswitch-users] Problems with ./configure on OSX

[Brian West]

If you have the latest dev tools and you do a fresh check out and  
then ./bootstrap.sh it should work fine. It does on my 10.5 box.

/usr/bin/libtool != gnu libtool aka glibtool

/b


On Nov 10, 2008, at 12:09 PM, martin joseph wrote:

> After a build.sh I get the following.
>
> I have NOT updated my dev tools as I am still running 10.4.11, but I
> will go over to apple and check if there is an update (I think I am
> current).
>
> Thanks for the help.
> Marty
>
>
> /usr/bin/libtool: can't map file: /usr/src/freeswitch/trunk/libs/
> apr/.libs ((os/kern) invalid argument)
> make[2]: *** [libfreeswitch.la] Error 1
> Making all in src
> Making all in mod
>
> making all mod_amr
> make[5]: *** No rule to make target `/usr/src/freeswitch/trunk/
> libfreeswitch.la', needed by `mod_amr.so'.  Stop.
> make[4]: *** [all] Error 1
> make[3]: *** [mod_amr-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
>  + FreeSWITCH Build Complete ---+
>  + FreeSWITCH has been successfully built.  +
>  + Install by running:  +
>  +  +
>  +  make install+
>  +--+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2


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Re: [Freeswitch-users] How to find an extension in public.xml ?

[Brian West]

You could transfer to public at the end of default.xml

/b

On Nov 10, 2008, at 6:21 AM, henkoegema wrote:

> Q: How can I force FS to look in ../dialplan/public.xml if it can't  
> find an
> entry in default.xml ?


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Re: [Freeswitch-users] Different problems with "pizza" demo on newer build.

[Brian West]

You don't use the .set.arpabo files at all.. just the .dic and .lm  
files.


/b

On Nov 9, 2008, at 11:57 AM, [EMAIL PROTECTED] wrote:

A further follow up on the grammar download from the FS Mod  
pocketsphinx wiki page.


The contents of these 4 folders .lm files, pizza_crust, pizza_size,  
pizza_specialty, pizza_toppings, is empty but what lmtool generates  
is found in these folders .sent.arpabo files.


On the other hand, the remaining folders, pizza_arso, pizza_order,  
pizza_type, pizza_yesno, have content in their .lm files but nothing  
inside their .sent.arpabo files.


It seems that .lm files are needed, but What are these .sent.arpabo  
files and are they needed?


Mark


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Re: [Freeswitch-users] SIP packets sending rport instead of 5060

[Brian West]

Can you send me the sip invite and dialog?

/b

On Nov 10, 2008, at 5:02 PM, Josh Forman wrote:

> I've been trying to send out SIP calls from a softphone to an outside
> line.  the call makes it to the freeswitch box and is sent to our
> cisco router, but the source port is coming up as "rport" instead of
> 5060.
> In what may be a related problem when the router sends messages the
> freeswitch box, freeswitch does not return an ACK message.
> The gateway and sip profile settings are pasted below.


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Re: [Freeswitch-users] Question about FIFO event

[Brian West]

You can put your patch up on jira.freeswitch.org

/b

On Nov 10, 2008, at 9:01 PM, Juan Backson wrote:

> Mike,
>
> If Cesar already have the patch, I don't need to re-invent the wheel
> anymore.  Otherwise, I will try to write one myself.
>
> Cesar,
> Would you please send me a copy of your patch?
>
> Thanks,
> Woody


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Re: [Freeswitch-users] FACILITY_NOT_SUBSCRIBED

[Brian West]

This isn't strange at all its part of the default configs.

/b

On Nov 11, 2008, at 9:28 AM, [EMAIL PROTECTED] wrote:


What does your dialplan look like?  I see the error, but I can't quite
tell what is wrong.  It looks like there is some sort of strange
variable assignment going on "variable string 0 =
[EMAIL PROTECTED]".


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Re: [Freeswitch-users] FACILITY_NOT_SUBSCRIBED

[Brian West]

That isn't the problem because I run with this config every day in my  
testing.

/b

On Nov 10, 2008, at 11:50 AM, Peter P GMX wrote:

> Addendum:
>
> I fixed the "sip_secure_media=[undef]"  problem. However no change.
>
> Best regards
> peter


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Re: [Freeswitch-users] FACILITY_NOT_SUBSCRIBED

[Brian West]

If you look very close in the dialog box it says what they are.  Its  
"pastebin" and "freeswitch"


You failed the test :P

/b

On Nov 11, 2008, at 4:14 PM, Peter P GMX wrote:

http://pastebin.freeswitch.org asked  
for
login credentials. Any idea where to get them from? I googled  
around, no

solution fund. Wiki credentials don't work.
Best regards

Peter


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Re: [Freeswitch-users] FACILITY_NOT_SUBSCRIBED

[Brian West]

I have tried to raise this... so I have a request here for someone in  
the community to step up and either write us a new pastebin that  
doesn't suck or find me one that doesn't take 100 depends to install.   
So if you wish to help with the pastebin please email me off list.

/b

On Nov 11, 2008, at 5:07 PM, Michael Collins wrote:

> FYI,
>
> There is like a 1400 line limit in pastebin, or something like that,  
> so
> you have to be careful when you have a large dump to put there.
> -MC


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Re: [Freeswitch-users] Using Meta-app with curl together

[Brian West]

It should be requesting the record extension in the context features.   
Is that not what you get?

/b

On Nov 12, 2008, at 4:02 AM, Doug Blacksone wrote:

> 


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Re: [Freeswitch-users] mod_python sqlalchemy core dumps

[Brian West]

Well this backtrace doesn't help at all.  See the line starting with  
#0... at that point you need to type "bt" and email that to the list..  
and just the part after you type "bt"

/b

On Nov 12, 2008, at 8:31 PM, . wrote:

> Program terminated with signal 11, Segmentation fault.
> #0  0x00e140df in PyObject_Malloc () from /usr/lib/libpython2.4.so.1.0
>
> unfortunately I'm still not entirely sure what this adds up to, other
> than the obvious memory allocation problem.  Is this a direct issue
> with sqlalchemy/the python mysql backend? It would seem that the
> hangup hook has some issue with one of these, as sqlalchemy works fine
> without the hangup hook, and the hangup hook seems generally ok
> without sqlalchemy, but put the two together and you've got problems.
>
> i wonder if switching up to python2.5 would be a good idea?
>
> Cheers


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Re: [Freeswitch-users] Strange (?) port number.

[Brian West]

You might have done that but it registered to FS and reported that as  
the port number in the contact... verify this by doing a sip trace.

/b

On Nov 13, 2008, at 5:53 AM, henkoegema wrote:

> But I programmed port 5090.


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Re: [Freeswitch-users] Strange (?) port number.

[Brian West]

Because thats the contact's port number.

/b

On Nov 13, 2008, at 5:09 AM, henkoegema wrote:



All my remote extensions are showing port number 5090, except the  
extension

2013.
Why does it show port 56588 ?

Call-ID:OTJlMGM3NWY4OTAyMDIwODEwODMwYjM5N2UyYjBiZGU.
User:   [EMAIL PROTECTED]
Contact: 	"Simon" >

Agent:  Zoiper for Windows rev.599
Status: Registered(UDP)(unknown) EXP(2008-11-13 11:59:51)
Host:   freeswitch


Henk.


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Re: [Freeswitch-users] E-mail as user id and bridging

[Brian West]

You can't have two @'s in a uri.  You do know its already domain  
based.. you have domains that have users inside them?  so you can dial user/[EMAIL PROTECTED] 
 already without any extra thought?  You can have multiple domains in  
the directory... with users in each domain.


/b

On Nov 14, 2008, at 2:20 AM, Jan Kubr wrote:


Hi guys,
I'd like to have e-mail addresses as user ids in the directory,  
something like:


 

Now is there any way I can bridge a call to this user?
This:




doesn't work because of the two @ signs. I tried putting the id in a
variable, replace @ with @, but neither helped.

Thanks,
Jan Kubr


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Re: [Freeswitch-users] Question on CDR CSV

[Brian West]

Yes its correct.  we log to the account code specific file and the  
master csv file.

/b

On Nov 15, 2008, at 1:26 PM, Shelby Ramsey wrote:

> Is this correct?  Just want to make sure I don't miss or duplicate  
> calls in the CDR records ...


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Re: [Freeswitch-users] How many calls can FreeSwitch bridge simultaeously

[Brian West]

I think thats what he means!

/b

On Nov 16, 2008, at 12:35 PM, David Knell wrote:

> Bzzt.  That's 1200 call *legs*, or 600 bridged calls.
>
> Cheers --
>
> Dave


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Re: [Freeswitch-users] E-mail as user id and bridging

[Brian West]

Its better to use a dash... a dot is valid in the username part of an  
email address... so what dot do you split on?

Example

[EMAIL PROTECTED]
[EMAIL PROTECTED]

Even worse

[EMAIL PROTECTED]

Lets go even worse... [EMAIL PROTECTED]

So which do you split on :)  Not very clear is it?

/b

On Nov 16, 2008, at 4:13 PM, Jan Kubr wrote:

> I'll stick to what I'm doing now which is that I replace the at-sign
> with a dot and take that as a username.
> Sorry for the newbie question and thanks all for the answers.
>
> Jan


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Re: [Freeswitch-users] DTMF

[Brian West]

 == The digits you wish to pass.

Tip... try then ask ;)

/b

On Nov 17, 2008, at 5:33 AM, Baskar wrote:


Hi,

I want to pass the DTMF digits through api command

i find the api command
api uuid_send_dtmf  

I just want to know what is  what is the value to pass in  
that parameter



Thanks in advance

--
Warm Regards,
N.Baskar


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Re: [Freeswitch-users] MOH problems

[Brian West]

Have you updated to the latest SVN trunk?

/b

On Nov 17, 2008, at 2:05 AM, Helmut Kuper wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> just to get an answer ..
>
> regards
> Helmut


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