From jason at jasonjgw.net  Thu Apr  1 00:00:52 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 18:00:52 +1100
Subject: [Freeswitch-users] ZRTP on the FreeSWITCH PBX - questions on
 the log report
In-Reply-To: <COL108-W1015593859EC5279008F8CD91D0@phx.gbl>
References: <COL108-W1015593859EC5279008F8CD91D0@phx.gbl>
Message-ID: <20100401070052.GA13609@jdc.jasonjgw.net>

SERGE TUMBA <serget68 at msn.com> wrote:
> 
> I recently deployed ZRTP on the FreeSWITCH PBX on a Linux machine and I have
> two X-Lite softphones (on Windows) properly connected and secure with zfone
> and I am wondering why the FreeSWITCH log is returning the following errors.

These have been discussed previously on the list, and as I remember, they are
normal errors resulting from the fact that the first several packets aren't
encrypted.
> Also, the zfone on the machine receiving calls won't stay secure for a
> longtime after the two softphones are connected. I need help to understand
> this.

Can you be more specific? How do you know it doesn't stay secure? Someone who
knows Zfone will have to answer this.

I regularly use ZRTP successfully with calls between FreeSWITCH systems, and
they are always set up and maintained as secure ZRTP-encrypted sessions.

I do however get the erros that you quote as a normal part of the process.



From jason at jasonjgw.net  Thu Apr  1 00:09:15 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 18:09:15 +1100
Subject: [Freeswitch-users] ZRTP protocol measurement
In-Reply-To: <COL108-W11B3E7D9EEF7FA21F093D0D91D0@phx.gbl>
References: <COL108-W11B3E7D9EEF7FA21F093D0D91D0@phx.gbl>
Message-ID: <20100401070915.GB13609@jdc.jasonjgw.net>

SERGE TUMBA <serget68 at msn.com> wrote:
> 
> I would like to know how to measure the performence of the zrtp using
> FreeSWITCH that connect two X-Lite softphones which use the zfone for
> encrypting voice packets on both end phones.

What do you want to know? There are almost as many performance measurements as
there are people doing the measuring. I would suggest measuring in whatever
way you normally would, comparing ZRTP sessions with sessions that do not
involve ZRTP, and seeing if there are performance differences that affect your
usage scenario.
> 
> Also, can someone contrast and compare ZRTP to SRTP focusing on these two
> protocol behaviors.

Have a look at http://www.zfone.com/ for a description of ZRTP. From an
operational perspective, the main difference is that in configuring SRTP, you
need to use TLS to secure the SIP signaling; otherwise, the cryptographic keys
are transmitted in the clear, which completely eliminates the security.
Setting up TLS securely requires a public-key infrastructure whereby each side
verifies the identity of the other.

In ZRTP, the negotiation takes place entirely in the RTP stream; there are
several protection mechanisms provided to prevent third-parties from
masquerading as one of the end-points (namely, key finger-prints, displayed to
the user as words that can be verified in the conversation, and
the use of mathematically related keys in subsequent sessions between the same
parties, but without diminishing security). No public-key infrastructure is
needed, hence no X.509 certificates or TLS are required.



From jason at jasonjgw.net  Thu Apr  1 00:13:48 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 18:13:48 +1100
Subject: [Freeswitch-users] Trying to test mod_silk
In-Reply-To: <20100329040146.GA19949@jdc.jasonjgw.net>
References: <20100327035611.GA3552@jdc.jasonjgw.net>
	<FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>
	<20100327232215.GA8287@jdc.jasonjgw.net>
	<4BAF7871.3020909@aktzero.com>
	<20100329040146.GA19949@jdc.jasonjgw.net>
Message-ID: <20100401071348.GA13720@jdc.jasonjgw.net>

Jason White <jason at jasonjgw.net> wrote:
 
> http://pastebin.freeswitch.org/12565
> with SILK at 24000h specified in my configuration as the only codec, and after
> having verified that mod_silk was loaded. All of the SILK codecs (8000, 12000,
> etc., including 24000) were logged as having been registered.
> 
> I didn't notice anything interesting in the logs that I've posted other than
> the 48 khz codec being offered, presumably due to the fact that mod_portaudio
> was configured for 48 khz. Note that this always works perfectly well with a
> variety of codecs; FreeSWITCH resamples the input, as necessary, prior to
> encoding it.

Following a suggestion by Frank Carmickle, I confirmed that the profile status
shows SILK as the first codec in both inbound and outbound lists. It still
isn't showing up in the SIP request, though, so the problem remains exactly as
described earlier in this thread.



From lists at infosecurity.ch  Thu Apr  1 02:32:20 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 01 Apr 2010 11:32:20 +0200
Subject: [Freeswitch-users] FS to answer a call with ZRTP and do
	text-to-speech of SaS
Message-ID: <4BB46824.6060905@infosecurity.ch>

Hi all,

i would like to call from the product my company is going to release 
(www.privatewave.com, mobile voice encryption with ZRTP for Nokia, 
iPhone, Blackberry, Android) a FS extension and, without the ZRTP 
enrollment, having the extension to be answered negotiating ZRTP and 
playing with Text to Speech the SaS .

Is this possible with current FS/ZRTP integration?

Fabio



From a.afzali2003 at gmail.com  Thu Apr  1 02:33:10 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 1 Apr 2010 14:03:10 +0430
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
Message-ID: <g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>

same issue for me ( also with error in updating openzap directory), I just
used :

make clean
make

-- afshin

On Thu, Apr 1, 2010 at 12:06 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hello,
>
> Trying to update the svn source.
>
> svn up
>
> make current
>
> I got the following Errors.
>
>
> cd libs/openzap && autoconf
> configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
>       If this token and others are legitimate, please use m4_pattern_allow.
>       See the Autoconf documentation.
> configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
> configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
> configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
> make: *** [libs/openzap/Makefile] Error 1
>
>
> How to fix the make error.
>
> Thanks
> Lloyd
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Thu Apr  1 02:47:45 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 20:47:45 +1100
Subject: [Freeswitch-users] FS to answer a call with ZRTP and do
 text-to-speech of SaS
In-Reply-To: <4BB46824.6060905@infosecurity.ch>
References: <4BB46824.6060905@infosecurity.ch>
Message-ID: <20100401094745.GA15156@jdc.jasonjgw.net>

Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote:
> Hi all,
> 
> i would like to call from the product my company is going to release 
> (www.privatewave.com, mobile voice encryption with ZRTP for Nokia, 
> iPhone, Blackberry, Android) a FS extension and, without the ZRTP 
> enrollment, having the extension to be answered negotiating ZRTP and 
> playing with Text to Speech the SaS .
> 
> Is this possible with current FS/ZRTP integration?

Possibly, but you'll have to deal with licencing issues first associated with
the ZRTP library. It's under the GNU AGPLv3 licence, as I recall.

ZRTP is negotiated whenever the RTP stream starts; there's no point delaying
it. If there's no ZRTP support on the other end, it just continues as an
unencrypted call.



From lists at infosecurity.ch  Thu Apr  1 02:58:48 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 01 Apr 2010 11:58:48 +0200
Subject: [Freeswitch-users] FS to answer a call with ZRTP and do
 text-to-speech of SaS
In-Reply-To: <20100401094745.GA15156@jdc.jasonjgw.net>
References: <4BB46824.6060905@infosecurity.ch>
	<20100401094745.GA15156@jdc.jasonjgw.net>
Message-ID: <4BB46E58.8060308@infosecurity.ch>

On 01/04/10 11.47, Jason White wrote:
> ZRTP is negotiated whenever the RTP stream starts; there's no point delaying
> it. If there's no ZRTP support on the other end, it just continues as an
> unencrypted call.
>    
So if a call is ZRTP enabled FS automatically negotiate it.
Is there already some prototype and/or support to extract the negotiated 
SaS and do a text to speech?

I would like to arrange a "test echo service" that's ZRTP enabled.

Fabio



From jason at jasonjgw.net  Thu Apr  1 03:20:36 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 1 Apr 2010 21:20:36 +1100
Subject: [Freeswitch-users] FS to answer a call with ZRTP and do
 text-to-speech of SaS
In-Reply-To: <4BB46E58.8060308@infosecurity.ch>
References: <4BB46824.6060905@infosecurity.ch>
	<20100401094745.GA15156@jdc.jasonjgw.net>
	<4BB46E58.8060308@infosecurity.ch>
Message-ID: <20100401102036.GA15452@jdc.jasonjgw.net>

Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote:
> On 01/04/10 11.47, Jason White wrote:
> > ZRTP is negotiated whenever the RTP stream starts; there's no point delaying
> > it. If there's no ZRTP support on the other end, it just continues as an
> > unencrypted call.
> >    
> So if a call is ZRTP enabled FS automatically negotiate it.

That's what ZRTP is designed to do.
> Is there already some prototype and/or support to extract the negotiated 
> SaS and do a text to speech?

The SAS is available in a variable - I can't remember the details now, and the
text to speech should be easy at that point.




From nagalenoj at gmail.com  Thu Apr  1 03:37:02 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 1 Apr 2010 16:07:02 +0530
Subject: [Freeswitch-users] Grouping channels in span
Message-ID: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>

Dear friends,
    I'm using Sangoma A102 and I've configured it with wanpipe. I've
question with regard to grouping the spans. There are 2 spans in the card
and I would want to group the spans separately, so when I want to make call
I could specify as either openzap/1/ or openzap/2/...
    But, Now I'm unable to make calls like this. It is asking me to give the
dial string as openzap/1/a/123 at g1, openzap/1/a/123 at g2..

    I don't want this way., I would want to group the spans differently.
Kindly help me.

-- 
Regards,
Nagalenoj H.
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From moises.silva at gmail.com  Thu Apr  1 06:04:05 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 1 Apr 2010 09:04:05 -0400
Subject: [Freeswitch-users] Grouping channels in span
In-Reply-To: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>
References: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>
Message-ID: <i2mc4d05cbe1004010604n3119b859rc6afdda32f643fff@mail.gmail.com>

There can be only 1 boost span currently. All b-channels must be added to
that span. Then, in /etc/wanpipe/smg_pri.conf you configure which channels
belong to which group. You can see how to configure spans, channels and
groups here:

http://wiki.sangoma.com/wanpipe-pri-advanced-options

<http://wiki.sangoma.com/wanpipe-pri-advanced-options>Then when dialing from
FreeSWITCH use the syntax requested:

openzap/1/1234 at g1 to dial using group 1 defined in smg_pri.conf or
openzap/1/1234 at g2 to dial using group 2 and so on.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Thu, Apr 1, 2010 at 6:37 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>     I'm using Sangoma A102 and I've configured it with wanpipe. I've
> question with regard to grouping the spans. There are 2 spans in the card
> and I would want to group the spans separately, so when I want to make call
> I could specify as either openzap/1/ or openzap/2/...
>     But, Now I'm unable to make calls like this. It is asking me to give
> the dial string as openzap/1/a/123 at g1, openzap/1/a/123 at g2..
>
>     I don't want this way., I would want to group the spans differently.
> Kindly help me.
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From cucku.cucku at yahoo.com.vn  Thu Apr  1 07:17:48 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Thu, 1 Apr 2010 22:17:48 +0800 (SGT)
Subject: [Freeswitch-users]  :  need help on call to gateway
In-Reply-To: <c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
	<c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
Message-ID: <384781.44441.qm@web76212.mail.sg1.yahoo.com>

Hi Francios 

Thank you for you recommend

from your recommendation, i add the host name : sip.yeah.com into the 
/etc/hosts

but i am still getting error on DNS 
 Failed with status DNS Error [503]

it seems that FS needs resolve the sip.yeah.com from the DNS server.
the FS support register with domain - domain is not real and the parameter = register-proxy. Is there the same paramater for make call out??

i do ping sip.yeah.com successfull but when i do nslookup and the DNS server response cannot find the domain name

[root at localhost external]# nslookup sip.yeah.com
Server:         8.8.8.8
Address:        8.8.8.8#53

** server can't find sip.yeah.com: NXDOMAIN

[root at localhost external]# ping sip.yeah.com
PING sip.yeah.com (118.69.239.250) 56(84) bytes of data.
64 bytes from sip.yeah.com (118.69.239.250): icmp_seq=2 ttl=250 time=6.84 ms

Thank you
Ha`


________________________________
T?: Fran?ois Legal <devel at thom.fr.eu.org>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 20:51:17, Th? T?, 31 th?ng 3 2010 
Ch? ??: Re: [Freeswitch-users] V?:  V?:  V?:  need help on call to gateway


Then either provide DNS to that system so that it can resolve the address, either create a static binding in /etc/hosts file on that host for sip.yeah.com
 
On Wed, 31 Mar 2010 20:36:37 +0800 (SGT), false <cucku.cucku at yahoo.com.vn> wrote:
Hi David
>
>when change the 
>the Register message send out with wrong format : from sip:071234 at x.x.x.x
>the right format : sip:071234 at sip.yeah.com
>
>is there any way to fix it
>
>Thank you
>
>
>
>
>
________________________________
 T?: David Ponzone 
>??n: freeswitch-users at lists.freeswitch.org
>G?i ng?y: 15:50:00, Th? T?, 31 th?ng 3 2010 
>Ch? ??: Re: [Freeswitch-users] V?:  V?:  need help on call to gateway
>
>I guess you need to put the IP of the proxy also in:
>
>   
>David Ponzone  Direction Technique
>email: david.ponzone at ipeva.fr
>tel:      01 74 03 18 97
>gsm:   06 66 98 76 34
>
>
>Service Client IPeva
>tel:      0811 46 26 26
>www.ipeva.fr  -   www.ipeva-studio.com
>
>
>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>Le 31/03/2010 ? 10:22, false a ?crit :
>
>Hi David
>>
>>the domain sip.yeah.com cannot resolve in the FS host
>>the thing here is FW support  for Registration to remote SIP server
>>so the domain no need to resolve
>>how to do the same thing when call out, no need to resolve the domain name: sip.yeah.com,
>>
>>Thank you
>>
>>
>>
>>
________________________________
 T?: David Ponzone <david.ponzone at gmail.com>
>>??n: freeswitch-users at lists.freeswitch.org
>>G?i ng?y: 15:12:50, Th? T?, 31 th?ng 3 2010 
>>Ch? ??: Re: [Freeswitch-users] V?: need help on call to gateway
>>
>>This looks like a DNS error.
>>
>>Are you sure you can resolve sip.yeah.com from the FS host ?
>>
>>
>>David Ponzone  Direction Technique
>>email: david.ponzone at ipeva.fr
>>tel:      01 74 03 18 97
>>gsm:   06 66 98 76 34
>>
>>
>>Service Client IPeva
>>tel:      0811 46 26 26
>>www.ipeva.fr  -   www.ipeva-studio.com
>>
>>
>>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>>Le 31/03/2010 ? 09:51, false a ?crit :
>>
>>Hi David and Brian
>>>
>>>sorry to bother you again
>>>the domain name : sip.yeah.com is not real, How to fix the issue
>>>
>>>FreeSwitch register to sip server with user 071234 successull
>>>FreeSwitch receive incomming from sip server and then transfer to 1000 extension successfull
>>>the extension make call out using the sip server is fail
>>>and the error :
>>>sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
>>>nta: for  "sip.yeah.com" query "sip.yeah.com" A (cached)
>>>nua(0x9473108): call state changed: init -> calling, sent offer
>>>soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c], [(nil)]) called
>>>nua(0x9473108): event i_state INVITE sent
>>>nua(0x9473108): event r_invite 503 DNS Error
>>>nua(0x9473108): call state changed: calling -> init
>>>nua(0x9473108): event i_state 503 DNS Error
>>>nua(0x9473108): event i_terminated 503 DNS Error
>>>
>>>
>>>i create 3 file:
>>>    1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/external for register to sip server
>>>    1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/default for outbound call
>>>    1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/dialplan/public for incomming call
>>>
>>>the content of 071234.xml  :
>>>
>>> 
>>>  
>>>  
>>>  
>>>  
>>>  
>>>  
>>>  
>>>  
>>> 
>>>
>>>
>>>the content of 00_inbound_did_fpt.xml:
>>>
>>>  
>>>    
>>>      
>>>      
>>>    
>>>  
>>>
>>>
>>>the content of  01_fpt.net.xml:
>>>
>>>  
>>>    
>>>      
>>>    
>>>  
>>>
>>>
>>>
>>>
________________________________
 T?: David Ponzone <david.ponzone at gmail.com>
>>>??n: freeswitch-users at lists.freeswitch.org
>>>G?i ng?y: 14:26:13, Th? Ba, 30 th?ng 3 2010 
>>>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>>
>>>False,
>>>
>>>I think Brian is sleeping so I give you a quick answer.
>>>For OUTBOUND:
>>>Following my previous mail, you have to check if your user 1000 has the default context  configured.
>>>If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/.
>>>(You may also remove from this path the remaining .xml files from the default install)
>>>In 01_whatever.xml, put:
>>>    
>>>      
>>>        
>>>      
>>>    
>>>
>>>So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.
>>>For INBOUND:
>>>In conf/dialplan/public/, add a file named whatever.xml with the following content:
>>>   
>>>      
>>>        
>>>      
>>>    
>>>
>>>Here, YOUR_DID is the DID number your provider allocated to you.
>>>That is an exact match, so if you're not sure about the exact format, remove the leading ^, and match that on the longest part of the number you know about.
>>>Of course, after doing all this, reloadxml.
>>>It will perhaps not fit in your  config exactly, but you should get the idea to get started.
>>>
>>>
>>>David Ponzone  Direction Technique
>>>email: david.ponzone at ipeva.fr
>>>tel:      01 74 03 18 97
>>>gsm:   06 66 98 76 34
>>>
>>>
>>>Service Client IPeva
>>>tel:      0811 46 26  26
>>>www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>>
>>>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>>
>>>
>>>Le 30/03/2010 ? 09:01, false a ?crit :
>>>
>>>Hi Brian
>>>>
>>>>could you guide me more to config on freeswitch
>>>>
>>>>i want xilte with user 1000 can make call out to sip server 
>>>>and the prefix when make call out to sip server : 0
>>>>and receive incomming call from SIP server
>>>>
>>>>what file name i should create 
>>>>
>>>>Thank you
>>>>
>>>>
>>>>--- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ?? vi?t:
>>>>
>>>>
>>>>>T?: Brian West <brian at freeswitch.org>
>>>>>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>>>>??n: freeswitch-users at lists.freeswitch.org
>>>>>Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37
>>>>>
>>>>>
>>>>>its looking in context default... not public.
>>>>>
>>>>>/b
>>>>>
>>>>>
>>>>>On Mar 29, 2010, at 11:33 PM, false wrote:
>>>>>
>>>>>Hi all
>>>>>>
>>>>>>i use freeswitch to register to sip server
>>>>>>the domain is not real: sip.yeah.com
>>>>>>the outbound proxy  : 118.69.145.5
>>>>>>
>>>>>>i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
>>>>>>
>>>>>>    sip.yeah.com">
>>>>>>        
>>>>>>            
>>>>>>        sip.yeah.com"/>
>>>>>>        
>>>>>>        
>>>>>>        
>>>>>>        
>>>>>>        
>>>>>>    
>>>>>>
>>>>>>
>>>>>>so i use xlite register to freeswitch with user 1000
>>>>>>i start freeswitch , freeswitch start ok,
>>>>>>freeswitch register to sip server successfull with username 071234
>>>>>>xlite register to freeswitch successfull
>>>>>>
>>>>>>so i edit the public.xml
>>>>>>for incomming call from sip server, i will forward to extension 1000
>>>>>>    sip.yeah.com">
>>>>>>      
>>>>>>        
>>>>>>      
>>>>>>    
>>>>>>
>>>>>>but when i use  xlite make call out from freeswitch to sip server and get no routing
>>>>>> 
>>>>> 
>>>>________________________________
 >>>>T?t h?n, tho?ng g?n h?n, nhanh h?n - Tr?i nghi?m Yahoo! Mail m?i h?m nay!_______________________________________________
>>>>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
>>>>http://www.freeswitch.org
>>>>
>>>________________________________
 Thi?t k? ngay m?t Pingbox ??c ??o cho ri?ng b?n!
>>>T?o m?t n?i ?? chat tr?n blog l? chuy?n nh?._______________________________________________
>>>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
>>>http://www.freeswitch.org
>>________________________________
 >>Yahoo! Mail nay NHANH H?N - Th? ngay!_______________________________________________
>>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
>>http://www.freeswitch.org
>________________________________
 >Yahoo! Mail nay nhanh v? nhi?u kh?ng gian tho?ng h?n.H?y tr?i nghi?m ngay h?m nay! 


      Xem h?nh c? ?m m?i v?i ?ng B?t v? c? T?m?! T?i phi?n b?n Yahoo! Messenger m?i nh?t b?ng ti?ng Vi?t t?i ??y.
http://vn.messenger.yahoo.com
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From brian at freeswitch.org  Thu Apr  1 07:37:15 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 1 Apr 2010 09:37:15 -0500
Subject: [Freeswitch-users] :  need help on call to gateway
In-Reply-To: <384781.44441.qm@web76212.mail.sg1.yahoo.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
	<c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
	<384781.44441.qm@web76212.mail.sg1.yahoo.com>
Message-ID: <EE7D0215-5EE3-4D11-9F4E-C2500C129A37@freeswitch.org>

Why not get a provider that actually does this stuff correctly?  My blood boils ever time I hear of this stupid setup.

/b

On Apr 1, 2010, at 9:17 AM, false wrote:

> Hi Francios 
> 
> Thank you for you recommend
> 
> from your recommendation, i add the host name : sip.yeah.com into the /etc/hosts
> 
> but i am still getting error on DNS 
>  Failed with status DNS Error [503]
> 
> it seems that FS needs resolve the sip.yeah.com from the DNS server.
> the FS support register with domain - domain is not real and the parameter = register-proxy. Is there the same paramater for make call out??
> 
> i do ping sip.yeah.com successfull but when i do nslookup and the DNS server response cannot find the domain name
> 
> [root at localhost external]# nslookup sip.yeah.com
> Server:         8.8.8.8
> Address:        8.8.8.8#53
> 
> ** server can't find sip.yeah.com: NXDOMAIN
> 
> [root at localhost external]# ping sip.yeah.com
> PING sip.yeah.com (118.69.239.250) 56(84) bytes of data.
> 64 bytes from sip.yeah.com (118.69.239.250): icmp_seq=2 ttl=250 time=6.84 ms
> 
> Thank you
> Ha`

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From 12ukwn at gmail.com  Thu Apr  1 07:50:05 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Thu, 1 Apr 2010 16:50:05 +0200
Subject: [Freeswitch-users] :  need help on call to gateway
In-Reply-To: <384781.44441.qm@web76212.mail.sg1.yahoo.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
	<c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
	<384781.44441.qm@web76212.mail.sg1.yahoo.com>
Message-ID: <20100401165005.43ad63d1@anubis.defcon1>

Le Thu, 1 Apr 2010 22:17:48 +0800 (SGT),
false <cucku.cucku at yahoo.com.vn> a ?crit :

> Thank you for you recommend
> 
> from your recommendation, i add the host name : sip.yeah.com into the 
> /etc/hosts
> 
> but i am still getting error on DNS 
>  Failed with status DNS Error [503]

This is because /etc/host.conf isn't configured to first pick a name
into /etc/hosts before a DNS resolution occurs (man host.conf)

But it is usually better to create a specific DNS zone, such as:

; Zone for *.fusionpbx.set
$TTL	1D
$ORIGIN	fusionpbx.set.
;
@		   IN   SOA   fusionpbx.set.   hostmaster.fusionpbx.set. ( 
			2010030401	; serial
				8H	; refresh
				4H	; retry
				4W	; expire
				1D);
; DNS names
		   IN   NS   ns1.fusionpbx.set. 
		   IN   NS   ns2.fusionpbx.set.
; MX
		   IN   MX   10   mail.fusionpbx.set.
; Virtual Machines & Hosts
		   IN    A   192.168.1.25
ns1		   IN    A   192.168.1.1
ns2		   IN    A   192.168.1.2
mail		   IN    A   192.168.1.50
*.fusionpbx.set.   IN    A   192.168.1.25

This way, you can address any subdomain (but may be that's not what you're
looking for?)


-- 
"Take that, you hostile sons-of-bitches!"
		-- James Coburn, in the finale of _The_President's_Analyst_


From msc at freeswitch.org  Thu Apr  1 09:53:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Apr 2010 09:53:23 -0700
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
	<g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
Message-ID: <m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>

Not sure if you have a broken source tree or what. Try doing:
svn up && ./bootstrap.sh && ./configure && make install

That's a bit drastic I know but I've used it before to clear out goofy build
errors.
-MC

On Thu, Apr 1, 2010 at 2:33 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:

> same issue for me ( also with error in updating openzap directory), I just
> used :
>
> make clean
> make
>
> -- afshin
>
> On Thu, Apr 1, 2010 at 12:06 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hello,
>>
>> Trying to update the svn source.
>>
>> svn up
>>
>> make current
>>
>> I got the following Errors.
>>
>>
>> cd libs/openzap && autoconf
>> configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
>>       If this token and others are legitimate, please use
>> m4_pattern_allow.
>>       See the Autoconf documentation.
>> configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
>> configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
>> configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
>> make: *** [libs/openzap/Makefile] Error 1
>>
>>
>> How to fix the make error.
>>
>> Thanks
>> Lloyd
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Thu Apr  1 10:12:29 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 1 Apr 2010 13:12:29 -0400
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
	<g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
	<m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
Message-ID: <x2r8a19bf2e1004011012t44bca219kccb244b56edf55b7@mail.gmail.com>

Hi Michael,

Yes I follow the same steps to solve my problem.

BTW I have another question, When the will be the migration to Git complete?

SVN or GIT which one we need to choose?

Thanks
Lloyd


On Thu, Apr 1, 2010 at 12:53 PM, Michael Collins <msc at freeswitch.org> wrote:

> Not sure if you have a broken source tree or what. Try doing:
> svn up && ./bootstrap.sh && ./configure && make install
>
> That's a bit drastic I know but I've used it before to clear out goofy
> build errors.
> -MC
>
>
> On Thu, Apr 1, 2010 at 2:33 AM, afshin afzali <a.afzali2003 at gmail.com>wrote:
>
>> same issue for me ( also with error in updating openzap directory), I just
>> used :
>>
>> make clean
>> make
>>
>> -- afshin
>>
>> On Thu, Apr 1, 2010 at 12:06 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> Hello,
>>>
>>> Trying to update the svn source.
>>>
>>> svn up
>>>
>>> make current
>>>
>>> I got the following Errors.
>>>
>>>
>>> cd libs/openzap && autoconf
>>> configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
>>>       If this token and others are legitimate, please use
>>> m4_pattern_allow.
>>>       See the Autoconf documentation.
>>> configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
>>> configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
>>> configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
>>> make: *** [libs/openzap/Makefile] Error 1
>>>
>>>
>>> How to fix the make error.
>>>
>>> Thanks
>>> Lloyd
>>>
>>> _______________________________________________
>>> 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
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Apr  1 10:38:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Apr 2010 10:38:46 -0700
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <x2r8a19bf2e1004011012t44bca219kccb244b56edf55b7@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
	<g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
	<m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
	<x2r8a19bf2e1004011012t44bca219kccb244b56edf55b7@mail.gmail.com>
Message-ID: <v2o87f2f3b91004011038m4c45a3c6mc272a36ede1545a7@mail.gmail.com>

On Thu, Apr 1, 2010 at 10:12 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi Michael,
>
> Yes I follow the same steps to solve my problem.
>
> BTW I have another question, When the will be the migration to Git
> complete?
>
> SVN or GIT which one we need to choose?
>
> Thanks
> Lloyd
>
> You really only need git if you plan on having commit access. There is an
SVN mirror, so you don't really have to change anything if all you ever do
is download FS to update your system.
-MC
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From fs-list at communicatefreely.net  Thu Apr  1 18:52:29 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 01 Apr 2010 21:52:29 -0400
Subject: [Freeswitch-users] mod_local_stream path based on rate
In-Reply-To: <191c3a031003290916o3b524d95p51eb637255afec20@mail.gmail.com>
References: <4BB0CF2F.7070700@communicatefreely.net>
	<191c3a031003290916o3b524d95p51eb637255afec20@mail.gmail.com>
Message-ID: <4BB54DDD.5010806@communicatefreely.net>

That did it!

Thanks, I'll try to get this in the Wiki this week for anyone else that is interested.

-Tim

Anthony Minessale wrote:
> the default config works this way
> it runs a stream on each rate called moh/8000 moh/16000 moh/32000 and
> moh/48000
> if you try to run stream "moh" it will pick the right one.
> 
> 
> On Mon, Mar 29, 2010 at 11:02 AM, Tim St. Pierre
> <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>>
> wrote:
> 
>     Hello,
> 
>     I'm using local_stream:// as our music on hold source.  I have both
>     8000 KHz and 16000 KHz files
>     encoded so that I can play them at the native rate.  We have some
>     endpoints that can only do PCMU
>     and others that can do higher rates.
> 
>     I want to set music on hold for a channel and have it automatically
>     pick the appropriate stream
>     based on the channel's native rate, just like the voice prompts do.
>      I tried creating a stream for
>     each rate, like rock-8000 and rock-16000 and then set the channel's
>     music on hold variable to
>     local_stream://rock-${read_rate} but the read rate hasn't been
>     negotiated when this is set in the
>     dialplan.
> 
>     Any suggestions?
> 
>     -Tim
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> 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
> http://www.freeswitch.org



From nagalenoj at gmail.com  Thu Apr  1 21:49:11 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Fri, 2 Apr 2010 10:19:11 +0530
Subject: [Freeswitch-users] Grouping channels in span
In-Reply-To: <i2mc4d05cbe1004010604n3119b859rc6afdda32f643fff@mail.gmail.com>
References: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>
	<i2mc4d05cbe1004010604n3119b859rc6afdda32f643fff@mail.gmail.com>
Message-ID: <l2pfe67a3341004012149j9b11a0a7g2e6bf2488460dc63@mail.gmail.com>

Moises,
      In all cases, I should use openzap/1/a/123 and there is no chance for
configuring like openzap/1 and openzap/2. Am I right?

      The difficulty I face here is, In dialplan I'm unable to route the
calls based on spans, like,
   <condition field="chan_name" expression="^OpenZAP/1.*/(.*)">
       <action application="bridge" data="openzap/3/a/$1"/>
   </condition>
   <condition field="chan_name" expression="^OpenZAP/2.*/(.*)">
       <action application="playback" data="under_maintenance.wav"/>
   </condition>

       Is there any other way to right the expression to match the spans?

On Thu, Apr 1, 2010 at 6:34 PM, Moises Silva <moises.silva at gmail.com> wrote:

> There can be only 1 boost span currently. All b-channels must be added to
> that span. Then, in /etc/wanpipe/smg_pri.conf you configure which channels
> belong to which group. You can see how to configure spans, channels and
> groups here:
>
> http://wiki.sangoma.com/wanpipe-pri-advanced-options
>
> <http://wiki.sangoma.com/wanpipe-pri-advanced-options>Then when dialing
> from FreeSWITCH use the syntax requested:
>
> openzap/1/1234 at g1 to dial using group 1 defined in smg_pri.conf or
> openzap/1/1234 at g2 to dial using group 2 and so on.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
> On Thu, Apr 1, 2010 at 6:37 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Dear friends,
>>     I'm using Sangoma A102 and I've configured it with wanpipe. I've
>> question with regard to grouping the spans. There are 2 spans in the card
>> and I would want to group the spans separately, so when I want to make call
>> I could specify as either openzap/1/ or openzap/2/...
>>     But, Now I'm unable to make calls like this. It is asking me to give
>> the dial string as openzap/1/a/123 at g1, openzap/1/a/123 at g2..
>>
>>     I don't want this way., I would want to group the spans differently.
>> Kindly help me.
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
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From infos at madovsky.org  Thu Apr  1 10:17:43 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 1 Apr 2010 13:17:43 -0400
Subject: [Freeswitch-users] svn up then make current - Errors
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com><g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
	<m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
Message-ID: <85CCB9D6A5514599801186E0D6558000@MOBILEE1705>

same issue for me.
needed to comment out openzap in modules.conf (don't need it)


  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, April 01, 2010 12:53 PM
  Subject: Re: [Freeswitch-users] svn up then make current - Errors


  Not sure if you have a broken source tree or what. Try doing:
  svn up && ./bootstrap.sh && ./configure && make install

  That's a bit drastic I know but I've used it before to clear out goofy build errors.
  -MC


  On Thu, Apr 1, 2010 at 2:33 AM, afshin afzali <a.afzali2003 at gmail.com> wrote:

    same issue for me ( also with error in updating openzap directory), I just used :

    make clean
    make

    -- afshin


    On Thu, Apr 1, 2010 at 12:06 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com> wrote:

      Hello,


      Trying to update the svn source.


      svn up


      make current


      I got the following Errors.




      cd libs/openzap && autoconf
      configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
            If this token and others are legitimate, please use m4_pattern_allow.
            See the Autoconf documentation.
      configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
      configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
      configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
      make: *** [libs/openzap/Makefile] Error 1




      How to fix the make error.


      Thanks
      Lloyd

      _______________________________________________
      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
      http://www.freeswitch.org




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






------------------------------------------------------------------------------


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From grsingh750 at gmail.com  Thu Apr  1 13:52:08 2010
From: grsingh750 at gmail.com (guru singh)
Date: Fri, 2 Apr 2010 02:22:08 +0530
Subject: [Freeswitch-users] PSTN Integration and real deployments
Message-ID: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>

Hi,
After long nights and lots of coffee =) , I think I've largely understood
FreeSwitch. I've been playing with it and have managed most fancy things it
can do. But I've done this on my LAN using SIP softphones. Here's my problem
now, I know nothing about PSTN integration and real deployments. Here are my
questions, mostly based on what I read on wikipedia.

PSTN integration:

I have an ADSL internet connection, with a split-box? installed by my ISP
which splits the incoming line to two, one for the phone provided and one
for the adsl modem. I want to handle incoming PSTN calls via FreeSwitch and
also be able to outbound calls to PSTN/VoIP phones via an SIP client
registered with my FS server through an external gateway or the PSTN line.

0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
seamlessly converts SIP-PSTN traffic both ways or does it require
configuration? What are the ATA's that work best with FS?

1) I should register with a VoIP/SIP/DID? provider for making outbound
calls? Will I be provided with an incoming number reachable by normal PSTN
numbers? If yes, where will the number reside, as in will PSTN numbers
calling me be charged extra?

Real Deployments:

Supposing I'm to do a real deployment for a client. What are the options
that I have for hardware?

0) Get IP phones that talk SIP? Is this the most expensive option?

1) Suppose the client has a traditional plain intercom installment(think
hotels etc). with phones connecting via RJ11 connectors. Is it possible to
have something like an ATA with lots of ports working as a hub/switch, So
that all phones can be plugged into ATA and managed via FS?

Thanks

PS: If the above hardly makes sense, pardon me, you can understand my
confusion =). FS has really got me hooked and I'm itching to do more with
it.
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From nandy1925 at gmail.com  Thu Apr  1 22:33:52 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 2 Apr 2010 13:33:52 +0800
Subject: [Freeswitch-users] PSTN Integration and real deployments
In-Reply-To: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
References: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
Message-ID: <q2s7d0bfd8c1004012233u6750bd89n2cab7cbc36dca0f2@mail.gmail.com>

hi guru,

re pstn integration:
0) get an ATA. connect the FXS port to an analog phone and the FXO port to
your phone line. set the FXO port to dial (send INVITE) to Freeswitch after
1 or 2 rings. you can create an Auto-Attendant on FS to handle this call.
you can control the route of outgoing calls (PSTN or VoIP) via dialplan.

i hv tried Grandstream HT-503 but the FXO port has problems. i'd like to get
Audiocodes. grab their User Manual to give you an idea.

1) you hv to register w/ a VoIP provider if u route your calls via Internet.
the incoming number is usually offered as an option. re charges. it depends
on the plan you get.

re real deployments:

0) you can mix the client phones - IP hardphones or via FXS gateways (4/8/24
ports) to connect analog phones
1) yes. that's multi-port FXS gateways. there are multiport FXO gateways
where you can place FS to handle PSTN calls for the legacy PABX

PSTN <==> FS <==> PABX

i hope it clears up a bit from your cloud confusion :-)

-nandy

On Fri, Apr 2, 2010 at 4:52 AM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
> After long nights and lots of coffee =) , I think I've largely understood
> FreeSwitch. I've been playing with it and have managed most fancy things it
> can do. But I've done this on my LAN using SIP softphones. Here's my problem
> now, I know nothing about PSTN integration and real deployments. Here are my
> questions, mostly based on what I read on wikipedia.
>
> PSTN integration:
>
> I have an ADSL internet connection, with a split-box? installed by my ISP
> which splits the incoming line to two, one for the phone provided and one
> for the adsl modem. I want to handle incoming PSTN calls via FreeSwitch and
> also be able to outbound calls to PSTN/VoIP phones via an SIP client
> registered with my FS server through an external gateway or the PSTN line.
>
> 0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
> seamlessly converts SIP-PSTN traffic both ways or does it require
> configuration? What are the ATA's that work best with FS?
>
> 1) I should register with a VoIP/SIP/DID? provider for making outbound
> calls? Will I be provided with an incoming number reachable by normal PSTN
> numbers? If yes, where will the number reside, as in will PSTN numbers
> calling me be charged extra?
>
> Real Deployments:
>
> Supposing I'm to do a real deployment for a client. What are the options
> that I have for hardware?
>
> 0) Get IP phones that talk SIP? Is this the most expensive option?
>
> 1) Suppose the client has a traditional plain intercom installment(think
> hotels etc). with phones connecting via RJ11 connectors. Is it possible to
> have something like an ATA with lots of ports working as a hub/switch, So
> that all phones can be plugged into ATA and managed via FS?
>
> Thanks
>
> PS: If the above hardly makes sense, pardon me, you can understand my
> confusion =). FS has really got me hooked and I'm itching to do more with
> it.
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Thu Apr  1 22:42:14 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 2 Apr 2010 16:42:14 +1100
Subject: [Freeswitch-users] svn up then make current - Errors
In-Reply-To: <v2o87f2f3b91004011038m4c45a3c6mc272a36ede1545a7@mail.gmail.com>
References: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>
	<g2ydebff5a81004010233r80373b5awadc98c1dec34d6fd@mail.gmail.com>
	<m2y87f2f3b91004010953p925231aez6934a7e94e2451a7@mail.gmail.com>
	<x2r8a19bf2e1004011012t44bca219kccb244b56edf55b7@mail.gmail.com>
	<v2o87f2f3b91004011038m4c45a3c6mc272a36ede1545a7@mail.gmail.com>
Message-ID: <20100402054214.GA15808@jdc.jasonjgw.net>

Michael Collins <msc at freeswitch.org> wrote:
> > You really only need git if you plan on having commit access. There is an
> SVN mirror, so you don't really have to change anything if all you ever do
> is download FS to update your system.

Correct. If, however, you modify packaging or make other changes before
building, it should be easier with Git as you can establish your own private
branches, merge changes and access the entire history of the repository.

I've cloned the git repository but I haven't tried building FreeSWITCH from it
yet. I modify the Debian package files before building to enable certain
modules, set the version number appropriately, and so on.



From casteven at gmail.com  Thu Apr  1 23:24:15 2010
From: casteven at gmail.com (Campbell Steven)
Date: Fri, 02 Apr 2010 19:24:15 +1300
Subject: [Freeswitch-users] PSTN Integration and real deployments
In-Reply-To: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
References: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
Message-ID: <1270189455.2845.4055.camel@macmini>

Hi Guru,

If you have a bunch of analog extensions from say an old PBX
installation that you want to reuse (I *think* this is what you are
getting at?) then you can use a channelbank like these:

http://www.patton.com/products/pe_products.asp?category=406

Which essentially are an up to 32 port ATA, like Nandy says, an ATA's
purpose is to allow you to use an analogue handset on a VoIP system.
This would allow you to have each individual analogue extension
registering to it's own SIP account.

Campbell


On Fri, 2010-04-02 at 02:22 +0530, guru singh wrote:

> Hi,
> 
> After long nights and lots of coffee =) , I think I've largely
> understood FreeSwitch. I've been playing with it and have managed most
> fancy things it can do. But I've done this on my LAN using SIP
> softphones. Here's my problem now, I know nothing about PSTN
> integration and real deployments. Here are my questions, mostly based
> on what I read on wikipedia.
> 
> 
> PSTN integration:
> 
> 
> I have an ADSL internet connection, with a split-box? installed by my
> ISP which splits the incoming line to two, one for the phone provided
> and one for the adsl modem. I want to handle incoming PSTN calls via
> FreeSwitch and also be able to outbound calls to PSTN/VoIP phones via
> an SIP client registered with my FS server through an external gateway
> or the PSTN line.
> 
> 
> 0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
> seamlessly converts SIP-PSTN traffic both ways or does it require
> configuration? What are the ATA's that work best with FS?
> 
> 
> 1) I should register with a VoIP/SIP/DID? provider for making outbound
> calls? Will I be provided with an incoming number reachable by normal
> PSTN numbers? If yes, where will the number reside, as in will PSTN
> numbers calling me be charged extra?
> 
> 
> Real Deployments:
> 
> 
> Supposing I'm to do a real deployment for a client. What are the
> options that I have for hardware?
> 
> 
> 0) Get IP phones that talk SIP? Is this the most expensive option?
> 
> 
> 1) Suppose the client has a traditional plain intercom
> installment(think hotels etc). with phones connecting via RJ11
> connectors. Is it possible to have something like an ATA with lots of
> ports working as a hub/switch, So that all phones can be plugged into
> ATA and managed via FS?
> 
> 
> Thanks
> 
> 
> PS: If the above hardly makes sense, pardon me, you can understand my
> confusion =). FS has really got me hooked and I'm itching to do more
> with it.
> 
> _______________________________________________
> 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
> http://www.freeswitch.org


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From nandy1925 at gmail.com  Fri Apr  2 00:44:42 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 2 Apr 2010 15:44:42 +0800
Subject: [Freeswitch-users] PSTN Integration and real deployments
In-Reply-To: <1270189455.2845.4055.camel@macmini>
References: <x2w7b463d951004011352ke29533cnceb9099e599a7353@mail.gmail.com>
	<1270189455.2845.4055.camel@macmini>
Message-ID: <m2l7d0bfd8c1004020044q35e01de7jca433d843437a598@mail.gmail.com>

IP Channel Bank is new to me. i thought, all the while, that channel banks
uses expensive  E1/T1 interface.  now, it widens my options. tks for this
info.


On Fri, Apr 2, 2010 at 2:24 PM, Campbell Steven <casteven at gmail.com> wrote:

>  Hi Guru,
>
> If you have a bunch of analog extensions from say an old PBX installation
> that you want to reuse (I *think* this is what you are getting at?) then you
> can use a channelbank like these:
>
> http://www.patton.com/products/pe_products.asp?category=406
>
> Which essentially are an up to 32 port ATA, like Nandy says, an ATA's
> purpose is to allow you to use an analogue handset on a VoIP system. This
> would allow you to have each individual analogue extension registering to
> it's own SIP account.
>
> Campbell
>
>
>
> On Fri, 2010-04-02 at 02:22 +0530, guru singh wrote:
>
> Hi,
>
>  After long nights and lots of coffee =) , I think I've largely understood
> FreeSwitch. I've been playing with it and have managed most fancy things it
> can do. But I've done this on my LAN using SIP softphones. Here's my problem
> now, I know nothing about PSTN integration and real deployments. Here are my
> questions, mostly based on what I read on wikipedia.
>
>
>
>  PSTN integration:
>
>
>
>  I have an ADSL internet connection, with a split-box? installed by my ISP
> which splits the incoming line to two, one for the phone provided and one
> for the adsl modem. I want to handle incoming PSTN calls via FreeSwitch and
> also be able to outbound calls to PSTN/VoIP phones via an SIP client
> registered with my FS server through an external gateway or the PSTN line.
>
>
>
>  0) I should get an ATA to do this? Is an ATA just a dumb adaptor that
> seamlessly converts SIP-PSTN traffic both ways or does it require
> configuration? What are the ATA's that work best with FS?
>
>
>
>  1) I should register with a VoIP/SIP/DID? provider for making outbound
> calls? Will I be provided with an incoming number reachable by normal PSTN
> numbers? If yes, where will the number reside, as in will PSTN numbers
> calling me be charged extra?
>
>
>
>  Real Deployments:
>
>
>
>  Supposing I'm to do a real deployment for a client. What are the options
> that I have for hardware?
>
>
>
>  0) Get IP phones that talk SIP? Is this the most expensive option?
>
>
>
>  1) Suppose the client has a traditional plain intercom installment(think
> hotels etc). with phones connecting via RJ11 connectors. Is it possible to
> have something like an ATA with lots of ports working as a hub/switch, So
> that all phones can be plugged into ATA and managed via FS?
>
>
>
>  Thanks
>
>
>
>  PS: If the above hardly makes sense, pardon me, you can understand my
> confusion =). FS has really got me hooked and I'm itching to do more with
> it.
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From vfclists at googlemail.com  Fri Apr  2 02:12:25 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 2 Apr 2010 10:12:25 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on Freeswitch
Message-ID: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>

I am just trialling Freeswitch with Linksys adapters, whose default
codec I have set to G729 with 'Use Pref Codec Only:' set to no.
When I change that setting to 'yes' the calls don't go through. I am
using the latest Windows SVN.

What configuration changes do I need to allow freeswitch-codec-passthru-g729.

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From jason at jasonjgw.net  Fri Apr  2 02:23:10 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 2 Apr 2010 20:23:10 +1100
Subject: [Freeswitch-users] How to configure g729 pass though on
 Freeswitch
In-Reply-To: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
Message-ID: <20100402092310.GA18680@jdc.jasonjgw.net>

Frank Church <vfclists at googlemail.com> wrote:
> I am just trialling Freeswitch with Linksys adapters, whose default
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> When I change that setting to 'yes' the calls don't go through. I am
> using the latest Windows SVN.

FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
need it.

My recommendation would be to use a codec other than G.729 unless you have a
compelling reason, for example a carrier that only supports G.729.



From gavin.henry at gmail.com  Fri Apr  2 02:43:51 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 10:43:51 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT phones on
	FreeSWITCH Version 1.0.trunk (17177)
Message-ID: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>

Hi,

We've just upgraded (switched back to building from svn in case this
was a git migration issue) and all our phones behind various networks
on NAT won't registered anymore, whereas on version dated 2010-02-22
20:06 (don't have the revision) all these phones were fine and handled
the the FS NAT features. No NDLB features on enabled or rport. They
weren't on the 2010-02-22 20:06 version either.


freeswitch at internal> recv 524 bytes from udp/[external_ip]:43078 at
09:45:02.413013:
   ------------------------------------------------------------------------
   REGISTER sip:pbx1.xxxx.co.uk SIP/2.0
   Via: SIP/2.0/UDP internal_ip:5062;branch=z9hG4bK892744790
   From: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk>;tag=706763496
   To: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk>
   Call-ID: 1553007081 at internal_ip
   CSeq: 24 REGISTER
   Contact: <sip:609 at internal_ip:5062>
   Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY,
REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
   Max-Forwards: 70
   User-Agent: Yealink SIP-T28P 2.43.0.50
   Expires: 3600
   Content-Length: 0

   ------------------------------------------------------------------------
send 641 bytes to udp/[external_ip]:5062 at 09:45:02.413206:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP
internal_ip:5062;branch=z9hG4bK892744790;received=external_ip
   From: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk>;tag=706763496
   To: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk>;tag=gtDayy31FDN9K
   Call-ID: 1553007081 at internal_ip
   CSeq: 24 REGISTER
   User-Agent: Our test PBX 1.0
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="pbx1.xxxx.co.uk",
nonce="cf9aaa61-89bf-4d9d-adbb-51a068b23440", algorithm=MD5,
qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------


What's changed?

Thanks,

Gavin.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From jaybinks at gmail.com  Fri Apr  2 02:52:49 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 2 Apr 2010 19:52:49 +1000
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
Message-ID: <q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>

I know its not much help..
but I think the first thing you need to do here is identify the version you
were on before ..

and see if you can easily narrow it down to a SVN revision that broke it.

J



On Fri, Apr 2, 2010 at 7:43 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi,
>
> We've just upgraded (switched back to building from svn in case this
> was a git migration issue) and all our phones behind various networks
> on NAT won't registered anymore, whereas on version dated 2010-02-22
> 20:06 (don't have the revision) all these phones were fine and handled
> the the FS NAT features. No NDLB features on enabled or rport. They
> weren't on the 2010-02-22 20:06 version either.
>
>
> freeswitch at internal> recv 524 bytes from udp/[external_ip]:43078 at
> 09:45:02.413013:
>   ------------------------------------------------------------------------
>   REGISTER sip:pbx1.xxxx.co.uk SIP/2.0
>   Via: SIP/2.0/UDP internal_ip:5062;branch=z9hG4bK892744790
>   From: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk <sip%3A609 at pbx1.xxxx.co.uk>
> >;tag=706763496
>   To: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk <sip%3A609 at pbx1.xxxx.co.uk>>
>   Call-ID: 1553007081 at internal_ip
>   CSeq: 24 REGISTER
>   Contact: <sip:609 at internal_ip:5062>
>   Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY,
> REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
>   Max-Forwards: 70
>   User-Agent: Yealink SIP-T28P 2.43.0.50
>   Expires: 3600
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
> send 641 bytes to udp/[external_ip]:5062 at 09:45:02.413206:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   Via: SIP/2.0/UDP
> internal_ip:5062;branch=z9hG4bK892744790;received=external_ip
>   From: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk <sip%3A609 at pbx1.xxxx.co.uk>
> >;tag=706763496
>   To: "Gavin Henry" <sip:609 at pbx1.xxxx.co.uk <sip%3A609 at pbx1.xxxx.co.uk>
> >;tag=gtDayy31FDN9K
>   Call-ID: 1553007081 at internal_ip
>   CSeq: 24 REGISTER
>   User-Agent: Our test PBX 1.0
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   WWW-Authenticate: Digest realm="pbx1.xxxx.co.uk",
> nonce="cf9aaa61-89bf-4d9d-adbb-51a068b23440", algorithm=MD5,
> qop="auth"
>   Content-Length: 0
>
>   ------------------------------------------------------------------------
>
>
> What's changed?
>
> Thanks,
>
> Gavin.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.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
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From gavin.henry at gmail.com  Fri Apr  2 03:10:44 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 11:10:44 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
Message-ID: <k2h13ca621c1004020310h82de3e35t7f08bd6bf381fe44@mail.gmail.com>

On 2 April 2010 10:52, jay binks <jaybinks at gmail.com> wrote:
> I know its not much help..
> but I think the first thing you need to do here is identify the version you
> were on before ..
> and see if you can easily narrow it down to a SVN revision that broke it.
> J

Yeah, I wish there was more than just an svn revision. Some kind of
other tag within the code or configs....

Thanks.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From gavin.henry at gmail.com  Fri Apr  2 03:18:42 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 11:18:42 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
Message-ID: <w2m13ca621c1004020318hb5c7d65dl433429082a3dbdc2@mail.gmail.com>

On 2 April 2010 10:52, jay binks <jaybinks at gmail.com> wrote:
> I know its not much help..
> but I think the first thing you need to do here is identify the version you
> were on before ..
> and see if you can easily narrow it down to a SVN revision that broke it.
> J

The only phone that seems to get through is:

Agent:      	snom300/7.3.14

Where as all the others did too:

Yealink T28
Cisco 7940G
Polycom IP330
Aastra 31i
Aastra 51i
Aastra 53i
Aastra 55i
Aastra 57i


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From jason at jasonjgw.net  Fri Apr  2 03:22:33 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 2 Apr 2010 21:22:33 +1100
Subject: [Freeswitch-users] Registrations stop working for all NAT
 phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <k2h13ca621c1004020310h82de3e35t7f08bd6bf381fe44@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<k2h13ca621c1004020310h82de3e35t7f08bd6bf381fe44@mail.gmail.com>
Message-ID: <20100402102233.GA19211@jdc.jasonjgw.net>

Gavin Henry <gavin.henry at gmail.com> wrote:
 
> Yeah, I wish there was more than just an svn revision. Some kind of
> other tag within the code or configs....

When the migration to Git is settled, you'll be able to run git bisect, which
is designed for the purpose you're discussing. It isn't supposed to be a
substitute for debugging the code, but it's reputed to be used to track down
bugs which are hard to identify otherwise, in the Linux kernel in particular.




From gavin.henry at gmail.com  Fri Apr  2 03:56:12 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 11:56:12 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <20100402102233.GA19211@jdc.jasonjgw.net>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<k2h13ca621c1004020310h82de3e35t7f08bd6bf381fe44@mail.gmail.com>
	<20100402102233.GA19211@jdc.jasonjgw.net>
Message-ID: <v2p13ca621c1004020356l67be38dwd8fb36545e65df1@mail.gmail.com>

On 2 April 2010 11:22, Jason White <jason at jasonjgw.net> wrote:
> Gavin Henry <gavin.henry at gmail.com> wrote:
>
>> Yeah, I wish there was more than just an svn revision. Some kind of
>> other tag within the code or configs....
>
> When the migration to Git is settled, you'll be able to run git bisect, which
> is designed for the purpose you're discussing. It isn't supposed to be a
> substitute for debugging the code, but it's reputed to be used to track down
> bugs which are hard to identify otherwise, in the Linux kernel in particular.

OK, sounds good!


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From vfclists at googlemail.com  Fri Apr  2 05:03:19 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 2 Apr 2010 13:03:19 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <20100402092310.GA18680@jdc.jasonjgw.net>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
Message-ID: <g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>

On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
> Frank Church <vfclists at googlemail.com> wrote:
>> I am just trialling Freeswitch with Linksys adapters, whose default
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>> When I change that setting to 'yes' the calls don't go through. I am
>> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
> compelling reason, for example a carrier that only supports G.729.
>
>

The carrier insists on G729, although they can accept G711. I think
their call volume does not make it easy on them and their customers as
well.

I did some googling and came up with freeswitch-codec-passthru-g729. I
have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
http://wiki.freeswitch.org/wiki/Bypass_Media.
In my module.conf.xml there is also <load module="mod_g729"/>.

Does that mean that my installlation is configured for pass thru if I
make the right adjustments?

I have looked at
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
and http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
which speak of modifying the dialplan.

This is a basic freeswitch setup using the defaults. I just added the
extensions to conf/directory/default and changed the provider in
vars.xml and I want to be able to do the same in
conf/dialplan/default.xml.

In conf/dialplan/default.xml the extension is matched by the
destination. Is there an option for not falling through to other
extensions if they also match?

> _______________________________________________
> 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
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From cucku.cucku at yahoo.com.vn  Fri Apr  2 05:14:10 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Fri, 2 Apr 2010 20:14:10 +0800 (SGT)
Subject: [Freeswitch-users] =?utf-8?b?VuG7gTogIDogIG5lZWQgaGVscCBvbiBjYWxs?=
 =?utf-8?q?_to_gateway?=
In-Reply-To: <20100401165005.43ad63d1@anubis.defcon1>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
	<c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>
	<384781.44441.qm@web76212.mail.sg1.yahoo.com>
	<20100401165005.43ad63d1@anubis.defcon1>
Message-ID: <925478.50757.qm@web76215.mail.sg1.yahoo.com>

Hi Jean

yeah, your way solves my issue

Thank you





________________________________
T?: Jean-Yves F. Barbier <12ukwn at gmail.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 21:50:05, Th? N?m, 1 th?ng 4 2010 
Ch? ??: Re: [Freeswitch-users] :  need help on call to gateway

Le Thu, 1 Apr 2010 22:17:48 +0800 (SGT),
false <cucku.cucku at yahoo.com.vn> a ?crit :

> Thank you for you recommend
> 
> from your recommendation, i add the host name : sip.yeah.com into the 
> /etc/hosts
> 
> but i am still getting error on DNS 
>  Failed with status DNS Error [503]

This is because /etc/host.conf isn't configured to first pick a name
into /etc/hosts before a DNS resolution occurs (man host.conf)

But it is usually better to create a specific DNS zone, such as:

; Zone for *.fusionpbx.set
$TTL    1D
$ORIGIN    fusionpbx.set.
;
@           IN   SOA   fusionpbx.set.   hostmaster.fusionpbx.set. ( 
            2010030401    ; serial
                8H    ; refresh
                4H    ; retry
                4W    ; expire
                1D);
; DNS names
           IN   NS   ns1.fusionpbx.set. 
           IN   NS   ns2.fusionpbx.set.
; MX
           IN   MX   10   mail.fusionpbx.set.
; Virtual Machines & Hosts
           IN    A   192.168.1.25
ns1           IN    A   192.168.1.1
ns2           IN    A   192.168.1.2
mail           IN    A   192.168.1.50
*.fusionpbx.set.   IN    A   192.168.1.25

This way, you can address any subdomain (but may be that's not what you're
looking for?)


-- 
"Take that, you hostile sons-of-bitches!"
        -- James Coburn, in the finale of _The_President's_Analyst_

_______________________________________________
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
http://www.freeswitch.org



      Yahoo! Mail nay NHANH H?N - Th? ngay! http://vn.mail.yahoo.com
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From david.ponzone at gmail.com  Fri Apr  2 05:30:51 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 2 Apr 2010 14:30:51 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
Message-ID: <EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>

Frank,

mod_g729 needs to be loaded, and then G729 needs to be negotiated on  
both legs.
I really recommend you enable G729 on the Linksys, enable SIP trace on  
FS console:
sofia profile external siptrace on
sofia profile internal siptrace on

then make a (failing) call, capture the log on FS console and paste  
that to:
http://pastebin:freeswitch at pastebin.freeswitch.org/

Then, send us back the link to your paste.

You can also join us on #freeswitch (irc.freenode.net), for some live  
help.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 02/04/2010 ? 14:03, Frank Church a ?crit :

> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>> Frank Church <vfclists at googlemail.com> wrote:
>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>> When I change that setting to 'yes' the calls don't go through. I am
>>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you really
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>
> The carrier insists on G729, although they can accept G711. I think
> their call volume does not make it easy on them and their customers as
> well.
>
> I did some googling and came up with freeswitch-codec-passthru-g729. I
> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
> http://wiki.freeswitch.org/wiki/Bypass_Media.
> In my module.conf.xml there is also <load module="mod_g729"/>.
>
> Does that mean that my installlation is configured for pass thru if I
> make the right adjustments?
>
> I have looked at
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
> and http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
> which speak of modifying the dialplan.
>
> This is a basic freeswitch setup using the defaults. I just added the
> extensions to conf/directory/default and changed the provider in
> vars.xml and I want to be able to do the same in
> conf/dialplan/default.xml.
>
> In conf/dialplan/default.xml the extension is matched by the
> destination. Is there an option for not falling through to other
> extensions if they also match?
>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
> http://www.freeswitch.org

-------------- next part --------------
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From david.ponzone at gmail.com  Fri Apr  2 05:46:21 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 2 Apr 2010 14:46:21 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
Message-ID: <06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>

Frank,

sorry, I completely forgot one important detail:
in the default conf, G729 is not allowed on any SIP profiles, so you  
have to modify vars.xml.
You will find the following lines:
   <X-PRE-PROCESS cmd="set"  
data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/>

For now, I recommend you replace them by:
   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/>

Then in FS console, do:
sofia profile external restart reloadxml
sofia profile internal restart reloadxml

It should then work far better.

What we did there is to make G729 an accepted codec in inbound INVITEs  
and a proposed codec for outbound INVITEs.
Look in external.xml or internal.xml, and look at the variables  
inbound-codec-prefs and outbound-codec-prefs.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 02/04/2010 ? 14:30, David Ponzone a ?crit :

> Frank,
>
> mod_g729 needs to be loaded, and then G729 needs to be negotiated on  
> both legs.
> I really recommend you enable G729 on the Linksys, enable SIP trace  
> on FS console:
> sofia profile external siptrace on
> sofia profile internal siptrace on
>
> then make a (failing) call, capture the log on FS console and paste  
> that to:
> http://pastebin:freeswitch at pastebin.freeswitch.org/
>
> Then, send us back the link to your paste.
>
> You can also join us on #freeswitch (irc.freenode.net), for some  
> live help.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>
>> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>>> Frank Church <vfclists at googlemail.com> wrote:
>>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>>> When I change that setting to 'yes' the calls don't go through. I  
>>>> am
>>>> using the latest Windows SVN.
>>>
>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>>> bypass media
>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>>> you really
>>> need it.
>>>
>>> My recommendation would be to use a codec other than G.729 unless  
>>> you have a
>>> compelling reason, for example a carrier that only supports G.729.
>>>
>>>
>>
>> The carrier insists on G729, although they can accept G711. I think
>> their call volume does not make it easy on them and their customers  
>> as
>> well.
>>
>> I did some googling and came up with freeswitch-codec-passthru- 
>> g729. I
>> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
>> http://wiki.freeswitch.org/wiki/Bypass_Media.
>> In my module.conf.xml there is also <load module="mod_g729"/>.
>>
>> Does that mean that my installlation is configured for pass thru if I
>> make the right adjustments?
>>
>> I have looked at
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
>> and http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
>> which speak of modifying the dialplan.
>>
>> This is a basic freeswitch setup using the defaults. I just added the
>> extensions to conf/directory/default and changed the provider in
>> vars.xml and I want to be able to do the same in
>> conf/dialplan/default.xml.
>>
>> In conf/dialplan/default.xml the extension is matched by the
>> destination. Is there an option for not falling through to other
>> extensions if they also match?
>>
>>> _______________________________________________
>>> 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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> -- 
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>> http://www.freeswitch.org
>

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From lloyd.aloysius at gmail.com  Fri Apr  2 07:35:36 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 2 Apr 2010 10:35:36 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <191c3a031003231529w5e940113y899bcbfeff10d0eb@mail.gmail.com>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
	<49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
	<191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>
	<F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
	<191c3a031003200958t6e31f055g499be1153e7c4433@mail.gmail.com>
	<A643303450404497B14F79175BEC0EF4@MOBILEE1705>
	<005801cacad5$8a009820$9e01c860$@fr.eu.org>
	<191c3a031003231529w5e940113y899bcbfeff10d0eb@mail.gmail.com>
Message-ID: <u2s8a19bf2e1004020735kcb6be322o86556cdbb1a3f12b@mail.gmail.com>

After update the most recent version ,  voice mail to email stop working.

Here is the FreeSWITCH Environment

CentOS 5.4

FreeSWITCH
freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (17152)

Exim

---

Initially there is a sendmail problem then I start to using the Exim . Now
Exim also stop working.

What is the reliable method to deliver the Voice mail to Email.

Thanks in advance.

Lloyd




On Tue, Mar 23, 2010 at 6:29 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> lets open a jira about it as a reminder.
> we will figure out what to do there and come up with something.
>
>
>
> On Tue, Mar 23, 2010 at 5:09 PM, <devel at thom.fr.eu.org> wrote:
>
>>  So I did continue to track that problem.
>>
>>
>>
>> I found that when running sendmail from freeswitch, it seems that the
>> setgid bit on sendmail is not honored (I get messages like sendmail[13534]:
>> NOQUEUE: SYSERR(freeswitch): can not
>>
>>  chdir(/var/spool/mqueue-client/): Permission denied or NOQUEUE:
>> SYSERR(UID101): can not wri
>>
>> te to queue directory /var/spool/mqueue-client/ (RunAsGid=102,
>> required=107): Pe
>>
>> rmission denied when I gave freeswitch the permission on
>> /var/spool/mqueue-client).
>>
>> In any case that ends up to sendmail segfault, maybe due to a stack
>> problem.
>>
>>
>>
>> I could solve this by installing msmtp (it can live aside sendmail without
>> breaking my messaging setup, at least on debian)
>>
>>
>>
>> Fran?ois
>>
>>
>>
>> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Madovsky
>> *Envoy? :* samedi 20 mars 2010 18:13
>> *? :* freeswitch-users at lists.freeswitch.org
>> *Objet :* Re: [Freeswitch-users] voicemail options [temporarly solved]
>>
>>
>>
>> ok irght.
>>
>> So for now I will continue to use my patch to make it work.
>>
>> I will give to you the status of any system upgrade that resolves the
>> problem
>>
>>  ----- Original Message -----
>>
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>>
>> *Sent:* Saturday, March 20, 2010 12:58 PM
>>
>> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>>
>>
>>
>> I am willing to change anything we have to as long as it works on all
>> platforms and we can give proper warnings to others.
>> Maybe we will do something like another param to say if your mailer app
>> needs a pipe or can accept the filename so we can do both
>> and leave the defaults to the current way.
>>
>> Maybe we can also have another option to make a simple relay code
>> internally that can deliver the file to a mail relay if you set that option.
>>
>>
>>  On Sat, Mar 20, 2010 at 11:47 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> Ok I undertand.
>>
>> buy Why to continue to use pipe and STDIN since to use "filename" path
>> works ?
>>
>>  ----- Original Message -----
>>
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>>
>> *Sent:* Saturday, March 20, 2010 12:12 PM
>>
>> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>>
>>
>>
>> I think the problem is that fs sets the stack size for the core and new
>> threads to 240k so when the child is born from system() the new process
>> inherits the small stack size and sendmail needs more than 240k
>>
>> If you are root, it has the priveledge to raise the stack size to the
>> default size of 8m and sets it back down when the command completes.
>>
>> If you are not root you lack these privs and the stack size remains small.
>>
>> This has been a long time problem how to make a solution that is cross
>> platform and will not break working configurations.
>>
>> On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:
>>
>> I tried also with sudo and freeswitch user in sudoers with the same
>> result, segmentation fault.
>>
>> Anyway my patch works perfectly now, even with sendmail (the only
>> difference without patch is the "From" is taken from the voicemail.tpl and
>> not the user variable)
>>
>>
>> >
>> > ----- Original Message -----
>> > From: Fran?ois Legal
>> > To: freeswitch-users at lists.freeswitch.org
>>
>>
>> > Sent: Friday, March 19, 2010 6:51 AM
>> > Subject: Re: [Freeswitch-users] voicemail options [tempora...
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>> ...
>>
>>   ------------------------------
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>  ------------------------------
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From 12ukwn at gmail.com  Fri Apr  2 08:19:58 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 17:19:58 +0200
Subject: [Freeswitch-users] default_gateway
Message-ID: <20100402171958.2baecaaf@anubis.defcon1>

Hi list,

I'm making tests for a residential conf that use an ATA to bridge with
PSTN, can I define 'default_gateway' to a SIP extension (something like:
2222.xml), or am I oblige to use transfer (or something else)?

-- 
It's the RINSE CYCLE!!  They've ALL IGNORED the RINSE CYCLE!!


From brian at freeswitch.org  Fri Apr  2 08:26:10 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 2 Apr 2010 10:26:10 -0500
Subject: [Freeswitch-users] default_gateway
In-Reply-To: <20100402171958.2baecaaf@anubis.defcon1>
References: <20100402171958.2baecaaf@anubis.defcon1>
Message-ID: <8060AA8A-7FE1-48F5-B7CC-11D80129A7B0@freeswitch.org>

Thats just a gateway name based on a gateway you have setup on your system.

/b

On Apr 2, 2010, at 10:19 AM, Jean-Yves F. Barbier wrote:

> I'm making tests for a residential conf that use an ATA to bridge with
> PSTN, can I define 'default_gateway' to a SIP extension (something like:
> 2222.xml), or am I oblige to use transfer (or something else)?



From 12ukwn at gmail.com  Fri Apr  2 08:48:08 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 17:48:08 +0200
Subject: [Freeswitch-users] default_gateway
In-Reply-To: <8060AA8A-7FE1-48F5-B7CC-11D80129A7B0@freeswitch.org>
References: <20100402171958.2baecaaf@anubis.defcon1>
	<8060AA8A-7FE1-48F5-B7CC-11D80129A7B0@freeswitch.org>
Message-ID: <20100402174808.4305f53b@anubis.defcon1>

Le Fri, 2 Apr 2010 10:26:10 -0500,
Brian West <brian at freeswitch.org> a ?crit :


Could you point me to a doc that would help me to configure my ATA as a
GW? (it is a ht488, so it has FXO & FXS on the same IP)

> Thats just a gateway name based on a gateway you have setup on your
> system.

-- 
Obviously the only rational solution to your problem is suicide.


From 12ukwn at gmail.com  Fri Apr  2 11:03:39 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 20:03:39 +0200
Subject: [Freeswitch-users] difference
Message-ID: <20100402200339.26cde9a8@anubis.defcon1>

Hi,

What is exactly the difference(s) between extensions and endpoints?

-- 
Why do we have two eyes?  To watch 3-D movies with.


From clint at 42lines.net  Fri Apr  2 09:38:55 2010
From: clint at 42lines.net (Clint Popetz)
Date: Fri, 2 Apr 2010 10:38:55 -0600
Subject: [Freeswitch-users] mod_conference lag
Message-ID: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>

Hi,

I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with
mod_skypopen, and when I call the echo test with skype it is _beautiful_ and
has no lag, and the same is true for my coworker, but when we both dial a
mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on
the machine is nil.  Any ideas?

Thanks,
-Clint

-- 
Clint Popetz
http://42lines.net
Scalable Web Application Development
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From vfclists at googlemail.com  Fri Apr  2 11:52:36 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 2 Apr 2010 19:52:36 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
	<06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
Message-ID: <s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>

On 2 April 2010 13:46, David Ponzone <david.ponzone at gmail.com> wrote:
> Frank,
> sorry, I completely forgot one important detail:
> in the default conf, G729 is not allowed on any SIP profiles, so you have to
> modify vars.xml.
> You will find the following lines:
> ??<X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> ??<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/>
> For now, I recommend you replace them by:
> ??<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
> ??<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/>
> Then in FS console, do:
> sofia profile external restart reloadxml
> sofia profile internal restart reloadxml
> It should then work far better.
> What we did there is to make G729 an accepted codec in inbound INVITEs and a
> proposed codec for outbound INVITEs.
> Look in external.xml or internal.xml, and look at the variables
> inbound-codec-prefs and outbound-codec-prefs.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 14:30, David Ponzone a ?crit :
>
> Frank,
> mod_g729 needs to be loaded, and then G729 needs to be negotiated on both
> legs.
> I really recommend you enable G729 on the Linksys, enable SIP trace on FS
> console:
> sofia profile external siptrace on
> sofia profile internal siptrace on
> then make a (failing) call, capture the log on FS console and paste that to:
> http://pastebin:freeswitch at pastebin.freeswitch.org/

This the pastebin link

http://pastebin.freeswitch.org/12616

There is also one below it with the successful call.

I have set the default provider in vars.xml rather than under the
sip_profiles/external. Is there a section in the default dialplan that
handles the default context?

Can the options below be used in the default for the dialplan and sip_profiles?

In the dial plan
<action application="set" data="proxy_media=true"/>

In the sip profile
<param name="inbound-proxy-media" value="true"/>

 The Linksys accounts are in the default contexts.

> Then, send us back the link to your paste.
> You can also join us on #freeswitch (irc.freenode.net), for some live help.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>
> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>
> Frank Church <vfclists at googlemail.com> wrote:
>
> I am just trialling Freeswitch with Linksys adapters, whose default
>
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>
> When I change that setting to 'yes' the calls don't go through. I am
>
> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
>
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
> really
>
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
>
> compelling reason, for example a carrier that only supports G.729.
>
>
>
> The carrier insists on G729, although they can accept G711. I think
> their call volume does not make it easy on them and their customers as
> well.
>
> I did some googling and came up with freeswitch-codec-passthru-g729. I
> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
> http://wiki.freeswitch.org/wiki/Bypass_Media.
> In my module.conf.xml there is also <load module="mod_g729"/>.
>
> Does that mean that my installlation is configured for pass thru if I
> make the right adjustments?
>
> I have looked at
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
> and
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
> which speak of modifying the dialplan.
>
> This is a basic freeswitch setup using the defaults. I just added the
> extensions to conf/directory/default and changed the provider in
> vars.xml and I want to be able to do the same in
> conf/dialplan/default.xml.
>
> In conf/dialplan/default.xml the extension is matched by the
> destination. Is there an option for not falling through to other
> extensions if they also match?
>
> _______________________________________________
>
> 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
>
> http://www.freeswitch.org
>
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From gavin.henry at gmail.com  Fri Apr  2 12:03:29 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 20:03:29 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
Message-ID: <m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>

On 2 April 2010 10:52, jay binks <jaybinks at gmail.com> wrote:
> I know its not much help..
> but I think the first thing you need to do here is identify the version you
> were on before ..
> and see if you can easily narrow it down to a SVN revision that broke it.

I've worked out what version I was on before upgrading, so just
testing now. I remember,
as when compiling I raised this JIRA ticket:

http://jira.freeswitch.org/browse/MODLANG-157

so it was revision 16718


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From msc at freeswitch.org  Fri Apr  2 12:16:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Apr 2010 12:16:01 -0700
Subject: [Freeswitch-users] Grouping channels in span
In-Reply-To: <l2pfe67a3341004012149j9b11a0a7g2e6bf2488460dc63@mail.gmail.com>
References: <n2pfe67a3341004010337xd6281b48mc4b99b306013ca38@mail.gmail.com>
	<i2mc4d05cbe1004010604n3119b859rc6afdda32f643fff@mail.gmail.com>
	<l2pfe67a3341004012149j9b11a0a7g2e6bf2488460dc63@mail.gmail.com>
Message-ID: <m2u87f2f3b91004021216tb8834e00h7260e3cd352443d5@mail.gmail.com>

I'm not sure why you are matching on "chan_name" but even so, you should be
able to do it with the groups:
<condition field="chan_name" expression="OpenZAP/1/(\d+)\@g1">
... stuff for group 1
</condition>
<condition field="chan_name" expression="OpenZAP/1/(\d+)\@g2">
... stuff for group 2
</condition>

-MC

On Thu, Apr 1, 2010 at 9:49 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Moises,
>       In all cases, I should use openzap/1/a/123 and there is no chance for
> configuring like openzap/1 and openzap/2. Am I right?
>
>       The difficulty I face here is, In dialplan I'm unable to route the
> calls based on spans, like,
>    <condition field="chan_name" expression="^OpenZAP/1.*/(.*)">
>        <action application="bridge" data="openzap/3/a/$1"/>
>    </condition>
>    <condition field="chan_name" expression="^OpenZAP/2.*/(.*)">
>        <action application="playback" data="under_maintenance.wav"/>
>    </condition>
>
>        Is there any other way to right the expression to match the spans?
>
> On Thu, Apr 1, 2010 at 6:34 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> There can be only 1 boost span currently. All b-channels must be added to
>> that span. Then, in /etc/wanpipe/smg_pri.conf you configure which channels
>> belong to which group. You can see how to configure spans, channels and
>> groups here:
>>
>> http://wiki.sangoma.com/wanpipe-pri-advanced-options
>>
>> <http://wiki.sangoma.com/wanpipe-pri-advanced-options>Then when dialing
>> from FreeSWITCH use the syntax requested:
>>
>> openzap/1/1234 at g1 to dial using group 1 defined in smg_pri.conf or
>> openzap/1/1234 at g2 to dial using group 2 and so on.
>>
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>>
>> On Thu, Apr 1, 2010 at 6:37 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>>
>>> Dear friends,
>>>     I'm using Sangoma A102 and I've configured it with wanpipe. I've
>>> question with regard to grouping the spans. There are 2 spans in the card
>>> and I would want to group the spans separately, so when I want to make call
>>> I could specify as either openzap/1/ or openzap/2/...
>>>     But, Now I'm unable to make calls like this. It is asking me to give
>>> the dial string as openzap/1/a/123 at g1, openzap/1/a/123 at g2..
>>>
>>>     I don't want this way., I would want to group the spans differently.
>>> Kindly help me.
>>>
>>> --
>>> Regards,
>>> Nagalenoj H.
>>>
>>> _______________________________________________
>>> 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
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Apr  2 12:23:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Apr 2010 12:23:18 -0700
Subject: [Freeswitch-users] difference
In-Reply-To: <20100402200339.26cde9a8@anubis.defcon1>
References: <20100402200339.26cde9a8@anubis.defcon1>
Message-ID: <q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>

On Fri, Apr 2, 2010 at 11:03 AM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> Hi,
>
> What is exactly the difference(s) between extensions and endpoints?
>
Ah, a philosophical question. :)

In FreeSWITCH an "extension" is something in the dialplan, whereas an
endpoint usually means a physical endpoint, like a phone. Sometimes we throw
the words around loosely, so don't let that confuse you.

Just curious - what lead up to this question?
-MC
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From msc at freeswitch.org  Fri Apr  2 12:25:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 2 Apr 2010 12:25:55 -0700
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
Message-ID: <k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>

On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:

> Hi,
>
> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with
> mod_skypopen, and when I call the echo test with skype it is _beautiful_ and
> has no lag, and the same is true for my coworker, but when we both dial a
> mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on
> the machine is nil.  Any ideas?
>

Do you mean that when you speak, it takes 10-12 seconds before the audio is
heard by someone else in the conference? Also, can you try the same exercise
with a soft phone like x-lite? I'm curious to know if this happens only on
Skype calls or on any calls made to a conference.
-MC
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From david.ponzone at gmail.com  Fri Apr  2 12:36:14 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 2 Apr 2010 21:36:14 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
	<06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
	<s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>
Message-ID: <1CB9F2BA-9575-4AB2-A1D2-F7FD48F8A316@gmail.com>

Frank,

re-read my second mail.
You have to enable G729 in FS prefs (vars.xml).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 02/04/2010 ? 20:52, Frank Church a ?crit :

> On 2 April 2010 13:46, David Ponzone <david.ponzone at gmail.com> wrote:
>> Frank,
>> sorry, I completely forgot one important detail:
>> in the default conf, G729 is not allowed on any SIP profiles, so  
>> you have to
>> modify vars.xml.
>> You will find the following lines:
>>   <X-PRE-PROCESS cmd="set"
>> data 
>> ="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>   <X-PRE-PROCESS cmd="set"  
>> data="outbound_codec_prefs=PCMU,PCMA,G729"/>
>> For now, I recommend you replace them by:
>>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
>>   <X-PRE-PROCESS cmd="set"  
>> data="outbound_codec_prefs=G729,PCMU,PCMA"/>
>> Then in FS console, do:
>> sofia profile external restart reloadxml
>> sofia profile internal restart reloadxml
>> It should then work far better.
>> What we did there is to make G729 an accepted codec in inbound  
>> INVITEs and a
>> proposed codec for outbound INVITEs.
>> Look in external.xml or internal.xml, and look at the variables
>> inbound-codec-prefs and outbound-codec-prefs.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 14:30, David Ponzone a ?crit :
>>
>> Frank,
>> mod_g729 needs to be loaded, and then G729 needs to be negotiated  
>> on both
>> legs.
>> I really recommend you enable G729 on the Linksys, enable SIP trace  
>> on FS
>> console:
>> sofia profile external siptrace on
>> sofia profile internal siptrace on
>> then make a (failing) call, capture the log on FS console and paste  
>> that to:
>> http://pastebin:freeswitch at pastebin.freeswitch.org/
>
> This the pastebin link
>
> http://pastebin.freeswitch.org/12616
>
> There is also one below it with the successful call.
>
> I have set the default provider in vars.xml rather than under the
> sip_profiles/external. Is there a section in the default dialplan that
> handles the default context?
>
> Can the options below be used in the default for the dialplan and  
> sip_profiles?
>
> In the dial plan
> <action application="set" data="proxy_media=true"/>
>
> In the sip profile
> <param name="inbound-proxy-media" value="true"/>
>
> The Linksys accounts are in the default contexts.
>
>> Then, send us back the link to your paste.
>> You can also join us on #freeswitch (irc.freenode.net), for some  
>> live help.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>>
>> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>>
>> The carrier insists on G729, although they can accept G711. I think
>> their call volume does not make it easy on them and their customers  
>> as
>> well.
>>
>> I did some googling and came up with freeswitch-codec-passthru- 
>> g729. I
>> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
>> http://wiki.freeswitch.org/wiki/Bypass_Media.
>> In my module.conf.xml there is also <load module="mod_g729"/>.
>>
>> Does that mean that my installlation is configured for pass thru if I
>> make the right adjustments?
>>
>> I have looked at
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
>> and
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
>> which speak of modifying the dialplan.
>>
>> This is a basic freeswitch setup using the defaults. I just added the
>> extensions to conf/directory/default and changed the provider in
>> vars.xml and I want to be able to do the same in
>> conf/dialplan/default.xml.
>>
>> In conf/dialplan/default.xml the extension is matched by the
>> destination. Is there an option for not falling through to other
>> extensions if they also match?
>>
>> _______________________________________________
>>
>> 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
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
> http://www.freeswitch.org

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From 12ukwn at gmail.com  Fri Apr  2 12:43:07 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 21:43:07 +0200
Subject: [Freeswitch-users] difference
In-Reply-To: <q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>
References: <20100402200339.26cde9a8@anubis.defcon1>
	<q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>
Message-ID: <20100402214307.744fb47a@anubis.defcon1>

Le Fri, 2 Apr 2010 12:23:18 -0700,
Michael Collins <msc at freeswitch.org> a ?crit :

Well, I'm greping the wiki's docs trying to understand how I could
configure FS to reach my ATA as a GW, or just have an output route to it
(but didn't succeed yet), and found 'endpoint'; so at first glance, I though
it was the term for what I was looking after.

If I understand right, an '9000' is an extension, but 1001 is an endpoint.

JY

> On Fri, Apr 2, 2010 at 11:03 AM, Jean-Yves F. Barbier
> <12ukwn at gmail.com>wrote:
> 
> > Hi,
> >
> > What is exactly the difference(s) between extensions and endpoints?
> >
> Ah, a philosophical question. :)
> 
> In FreeSWITCH an "extension" is something in the dialplan, whereas an
> endpoint usually means a physical endpoint, like a phone. Sometimes we
> throw the words around loosely, so don't let that confuse you.
> 
> Just curious - what lead up to this question?
> -MC




-- 
... Logically incoherent, semantically incomprehensible, and legally ...
impeccable!


From brian at freeswitch.org  Fri Apr  2 12:49:14 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 2 Apr 2010 14:49:14 -0500
Subject: [Freeswitch-users] difference
In-Reply-To: <20100402214307.744fb47a@anubis.defcon1>
References: <20100402200339.26cde9a8@anubis.defcon1>
	<q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>
	<20100402214307.744fb47a@anubis.defcon1>
Message-ID: <804BE35D-BF34-4F24-96FD-4F318857BE9E@freeswitch.org>


On Apr 2, 2010, at 2:43 PM, Jean-Yves F. Barbier wrote:

> Le Fri, 2 Apr 2010 12:23:18 -0700,
> Michael Collins <msc at freeswitch.org> a ?crit :
> 
> 
> If I understand right, an '9000' is an extension, but 1001 is an endpoint.
> 

1001 is an extension pointed at an registered sip user 1001 which happens to be same number.

/b


> JY

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From lcm at marshap.com  Fri Apr  2 12:51:43 2010
From: lcm at marshap.com (Larry Marshall)
Date: Fri, 2 Apr 2010 12:51:43 -0700
Subject: [Freeswitch-users] Error in 'svn up'
Message-ID: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>

The current version of FS I'm using is 17135. I tried to 'make current' and
it errored out in the svn up portion:

 

svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a
different repository

 

Should I just delete the libs/openzap directory?

 

openzap is commented out in modules.conf.xml.

 

Thanks, Lars

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From 12ukwn at gmail.com  Fri Apr  2 13:01:02 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 2 Apr 2010 22:01:02 +0200
Subject: [Freeswitch-users] difference
In-Reply-To: <804BE35D-BF34-4F24-96FD-4F318857BE9E@freeswitch.org>
References: <20100402200339.26cde9a8@anubis.defcon1>
	<q2z87f2f3b91004021223zc74b7bc3h682c56bf9605a96e@mail.gmail.com>
	<20100402214307.744fb47a@anubis.defcon1>
	<804BE35D-BF34-4F24-96FD-4F318857BE9E@freeswitch.org>
Message-ID: <20100402220102.729a8251@anubis.defcon1>

Le Fri, 2 Apr 2010 14:49:14 -0500,
Brian West <brian at freeswitch.org> a ?crit :

> 
> On Apr 2, 2010, at 2:43 PM, Jean-Yves F. Barbier wrote:
> 
> > Le Fri, 2 Apr 2010 12:23:18 -0700,
> > Michael Collins <msc at freeswitch.org> a ?crit :
> > 
> > 
> > If I understand right, an '9000' is an extension, but 1001 is an
> > endpoint.
> > 
> 
> 1001 is an extension pointed at an registered sip user 1001 which happens
> to be same number.

Raaaahhhhhh, I think I'll call all of'em extensions

-- 
Debian Hint #6: There is no hint #6. Submit a hint today !


From clint at 42lines.net  Fri Apr  2 13:04:55 2010
From: clint at 42lines.net (Clint Popetz)
Date: Fri, 2 Apr 2010 14:04:55 -0600
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
	<k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>
Message-ID: <u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>

On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:
>
>> Hi,
>>
>> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with
>> mod_skypopen, and when I call the echo test with skype it is _beautiful_ and
>> has no lag, and the same is true for my coworker, but when we both dial a
>> mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on
>> the machine is nil.  Any ideas?
>>
>
> Do you mean that when you speak, it takes 10-12 seconds before the audio is
> heard by someone else in the conference?
>

Correct.


> Also, can you try the same exercise with a soft phone like x-lite? I'm
> curious to know if this happens only on Skype calls or on any calls made to
> a conference.
>

Sip is a whole other can of worms that's not working either.  When
connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo
fine, but it won't listen to DTMF numbers to change the menu.  When running
xopier on linux, I can't hear anything and DTMF doesn't get through.  Both
those softphones work fine with asterisk, including dtmf.  I was punting on
getting sip working correctly in freeswitch until I determined whether
skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.

(It's entirely possible/probably I just have sip misconfigured...I'm just
using the default configuration in that regard.)

-Clint


-MC
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


-- 
Clint Popetz
http://42lines.net
Scalable Web Application Development
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From anthony.minessale at gmail.com  Fri Apr  2 13:17:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Apr 2010 14:17:11 -0600
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
	<k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>
	<u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
Message-ID: <m2j191c3a031004021317kbae46d3fp4dfde28584852317@mail.gmail.com>

Try calling the public conference at
sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
that is a properly setup box and can accept any registrations.

The sip and the conference all work fine so it's for sure a problem on your
end.




On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <clint at 42lines.net> wrote:

>
>
> On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:
>>
>>> Hi,
>>>
>>> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with
>>> mod_skypopen, and when I call the echo test with skype it is _beautiful_ and
>>> has no lag, and the same is true for my coworker, but when we both dial a
>>> mod_conference bridge with skype, we get a 10-12 second lag.  CPU usage on
>>> the machine is nil.  Any ideas?
>>>
>>
>> Do you mean that when you speak, it takes 10-12 seconds before the audio
>> is heard by someone else in the conference?
>>
>
> Correct.
>
>
>> Also, can you try the same exercise with a soft phone like x-lite? I'm
>> curious to know if this happens only on Skype calls or on any calls made to
>> a conference.
>>
>
> Sip is a whole other can of worms that's not working either.  When
> connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo
> fine, but it won't listen to DTMF numbers to change the menu.  When running
> xopier on linux, I can't hear anything and DTMF doesn't get through.  Both
> those softphones work fine with asterisk, including dtmf.  I was punting on
> getting sip working correctly in freeswitch until I determined whether
> skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.
>
> (It's entirely possible/probably I just have sip misconfigured...I'm just
> using the default configuration in that regard.)
>
> -Clint
>
>
>  -MC
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Clint Popetz
> http://42lines.net
> Scalable Web Application Development
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From fraserredmond at gmail.com  Fri Apr  2 13:27:54 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 2 Apr 2010 21:27:54 +0100
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com> 
	<k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com> 
	<u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
Message-ID: <m2hc32ee1cf1004021327h9bf17f06nc380bc56e736c0c0@mail.gmail.com>

> Sip is a whole other can of worms that's not working either.  When
> connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo
> fine, but it won't listen to DTMF numbers to change the menu.
>

By coincidence, for the last couple of hours I've also been trying to work
out why I can't get DTMF on FreeSwitch on EC2 from Bria. Can't even think
how to debug it - the Bria diagnostic logs show the dtmf being prepared and
sent, but dtmf's don't seem to show on the server if  sofia loglevel all 9
is turned on.

So if anyone has any ideas on how to work it out, there's two of us that
could do with some help :-)

In my case I had set things up on a local dev server (with DTMF working, so
it's not a pure-Bria problem), then have transferred the same setup to the
EC2 server. I'm calling out from behind a NAT, so it could be something
related to that maybe. (The differences between the local and remote servers
are all related to being on internal vs external networks.

Cheers,
Fraser
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From gavin.henry at gmail.com  Fri Apr  2 13:42:49 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 21:42:49 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
Message-ID: <p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>

On 2 April 2010 20:03, Gavin Henry <gavin.henry at gmail.com> wrote:
> On 2 April 2010 10:52, jay binks <jaybinks at gmail.com> wrote:
>> I know its not much help..
>> but I think the first thing you need to do here is identify the version you
>> were on before ..
>> and see if you can easily narrow it down to a SVN revision that broke it.
>
> I've worked out what version I was on before upgrading, so just
> testing now. I remember,
> as when compiling I raised this JIRA ticket:
>
> http://jira.freeswitch.org/browse/MODLANG-157
>
> so it was revision 16718

OK, for some reason it works now with:

<param name="NDLB-force-rport" value="true"/>

when I'm positive it wasn't enabled before.


-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From clint at 42lines.net  Fri Apr  2 13:49:42 2010
From: clint at 42lines.net (Clint Popetz)
Date: Fri, 2 Apr 2010 14:49:42 -0600
Subject: [Freeswitch-users] mod_conference lag
In-Reply-To: <m2j191c3a031004021317kbae46d3fp4dfde28584852317@mail.gmail.com>
References: <y2qce06a7161004020938m1d9246a4g6797ba65ff2866f6@mail.gmail.com>
	<k2m87f2f3b91004021225icd4b69daj3fb8c3b84995d809@mail.gmail.com>
	<u2yce06a7161004021304h592f98a5mfc0649fe0e0d21b@mail.gmail.com>
	<m2j191c3a031004021317kbae46d3fp4dfde28584852317@mail.gmail.com>
Message-ID: <j2xce06a7161004021349i6f6dc55qf55f6957b9896ea@mail.gmail.com>

On Fri, Apr 2, 2010 at 2:17 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
>
> Try calling the public conference at sip:888 at conference.freeswitch.org
> that is a properly setup box and can accept any registrations.
>

Yup, that works, as does calling my existing asterisk pbx.

>
> The sip and the conference all work fine so it's for sure a problem on your end.

Well, this is with a fresh install using:

cd /usr/src?; wget http://www.freeswitch.org/eg/Makefile?; make
make all
cd freeswitch.trunk
make install
make cd-sounds-install
make cd-moh-install

the only thing I'v changed is to add <load module="mod_skypopen"/> and
	<action application="set"
data="skype_add_outband_dtmf_also_when_bridged=true"/>

and to install skype and make it listen to skypopen.   The rest is the
default config.

-Clint


>
>
>
> On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <clint at 42lines.net> wrote:
>>
>>
>> On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>
>>> On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint at 42lines.net> wrote:
>>>>
>>>> Hi,
>>>> I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. ?CPU usage on the machine is nil. ?Any ideas?
>>>
>>> Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference?
>>
>> Correct.
>>
>>>
>>> Also, can you try the same exercise with a soft phone like x-lite? I'm curious to know if this happens only on Skype calls or on any calls made to a conference.
>>
>> Sip is a whole other can of worms that's not working either. ?When connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo fine, but it won't listen to DTMF numbers to change the menu. ?When running xopier on linux, I can't hear anything and DTMF doesn't get through. ?Both those softphones work fine with asterisk, including dtmf. ?I was punting on getting sip working correctly in freeswitch until I determined whether skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.
>> (It's entirely possible/probably I just have sip misconfigured...I'm just using the default configuration in that regard.)
>> -Clint
>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> 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
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Clint Popetz
>> http://42lines.net
>> Scalable Web Application Development
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>



--
Clint Popetz
http://42lines.net
Scalable Web Application Development


From gavin.henry at gmail.com  Fri Apr  2 13:53:41 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 2 Apr 2010 21:53:41 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
Message-ID: <y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>

> OK, for some reason it works now with:
>
> <param name="NDLB-force-rport" value="true"/>
>
> when I'm positive it wasn't enabled before.

Shouldn't FS NAT features detect that the phones are registering from
an internal IP and do the usual fs_nat=yes stuff in the contact header
without forcing rport above and send back responses to the port the
REGISTER came from?

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From vfclists at googlemail.com  Fri Apr  2 13:53:36 2010
From: vfclists at googlemail.com (Frank Church)
Date: Fri, 2 Apr 2010 21:53:36 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <1CB9F2BA-9575-4AB2-A1D2-F7FD48F8A316@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
	<06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
	<s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>
	<1CB9F2BA-9575-4AB2-A1D2-F7FD48F8A316@gmail.com>
Message-ID: <q2j7abab2411004021353sec91c94mff3954fea571dcc@mail.gmail.com>

On 2 April 2010 20:36, David Ponzone <david.ponzone at gmail.com> wrote:
> Frank,
> re-read my second mail.

http://pastebin.freeswitch.org/12617

> You have to enable G729 in FS prefs (vars.xml).
Was that obvious from the sip trace? I missed it the first time.

> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 20:52, Frank Church a ?crit :
>
> On 2 April 2010 13:46, David Ponzone <david.ponzone at gmail.com> wrote:
>
> Frank,
>
> sorry, I completely forgot one important detail:
>
> in the default conf, G729 is not allowed on any SIP profiles, so you have to
>
> modify vars.xml.
>
> You will find the following lines:
>
> ??<X-PRE-PROCESS cmd="set"
>
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>
> ??<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729"/>
>
> For now, I recommend you replace them by:
>
> ??<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
>
> ??<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA"/>
>
> Then in FS console, do:
>
> sofia profile external restart reloadxml
>
> sofia profile internal restart reloadxml
>
> It should then work far better.
>
> What we did there is to make G729 an accepted codec in inbound INVITEs and a
>
> proposed codec for outbound INVITEs.
>
> Look in external.xml or internal.xml, and look at the variables
>
> inbound-codec-prefs and outbound-codec-prefs.
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>
> non autoris?e est interdite. Tout message ?lectronique est susceptible
>
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 14:30, David Ponzone a ?crit :
>
> Frank,
>
> mod_g729 needs to be loaded, and then G729 needs to be negotiated on both
>
> legs.
>
> I really recommend you enable G729 on the Linksys, enable SIP trace on FS
>
> console:
>
> sofia profile external siptrace on
>
> sofia profile internal siptrace on
>
> then make a (failing) call, capture the log on FS console and paste that to:
>
> http://pastebin:freeswitch at pastebin.freeswitch.org/
>
> This the pastebin link
>
> http://pastebin.freeswitch.org/12616
>
> There is also one below it with the successful call.
>
> I have set the default provider in vars.xml rather than under the
> sip_profiles/external. Is there a section in the default dialplan that
> handles the default context?
>
> Can the options below be used in the default for the dialplan and
> sip_profiles?
>
> In the dial plan
> <action application="set" data="proxy_media=true"/>
>
> In the sip profile
> <param name="inbound-proxy-media" value="true"/>
>
> The Linksys accounts are in the default contexts.
>
> Then, send us back the link to your paste.
>
> You can also join us on #freeswitch (irc.freenode.net), for some live help.
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
> Service Client?IPeva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>
> non autoris?e est interdite. Tout message ?lectronique est susceptible
>
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>
> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>
> Frank Church <vfclists at googlemail.com> wrote:
>
> I am just trialling Freeswitch with Linksys adapters, whose default
>
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>
> When I change that setting to 'yes' the calls don't go through. I am
>
> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
>
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
>
> really
>
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
>
> compelling reason, for example a carrier that only supports G.729.
>
>
>
> The carrier insists on G729, although they can accept G711. I think
>
> their call volume does not make it easy on them and their customers as
>
> well.
>
> I did some googling and came up with freeswitch-codec-passthru-g729. I
>
> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
>
> http://wiki.freeswitch.org/wiki/Bypass_Media.
>
> In my module.conf.xml there is also <load module="mod_g729"/>.
>
> Does that mean that my installlation is configured for pass thru if I
>
> make the right adjustments?
>
> I have looked at
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
>
> and
>
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
>
> which speak of modifying the dialplan.
>
> This is a basic freeswitch setup using the defaults. I just added the
>
> extensions to conf/directory/default and changed the provider in
>
> vars.xml and I want to be able to do the same in
>
> conf/dialplan/default.xml.
>
> In conf/dialplan/default.xml the extension is matched by the
>
> destination. Is there an option for not falling through to other
>
> extensions if they also match?
>
> _______________________________________________
>
> 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
>
> http://www.freeswitch.org
>
>
>
>
> --
>
> Frank Church
>
> =======================
>
> http://devblog.brahmancreations.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
>
> http://www.freeswitch.org
>
>
>
> _______________________________________________
>
> 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
>
> http://www.freeswitch.org
>
>
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
> http://www.freeswitch.org
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From lon at kickasspixels.com  Fri Apr  2 14:24:51 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 2 Apr 2010 14:24:51 -0700
Subject: [Freeswitch-users] CLI status?
Message-ID: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>

Hi gang!

I'm trying to determine the best way to periodically check the status
of production Freeswitch deployments.

While Freeswitch has been incredibly stable in production, I would
like to ping it periodically to check that its not hung in someway.

Has anyone considered or done this yet?

My thought it so issue a command via fs_cli periodically and compare
the out. But am trying to determine what a good command would be.

thinking of using show channels or show calls, in addition to status.

Any thoughts? Is this a wrong-headed approach?

Thanks for any feedback.

Lon


From vetali100 at gmail.com  Fri Apr  2 14:37:49 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 3 Apr 2010 00:37:49 +0300
Subject: [Freeswitch-users]  Calling in context public instead of default,
	mod_xml_curl
Message-ID: <k2ma858e7691004021437s5d64917cqb80c456b2bb9e119@mail.gmail.com>

Hi,

I am using mod_xml_curl to generate user directory data.
It produces data like this:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="$${domain}">
      <params>
        <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>
      <groups>
        <group name="default">
          <users>
            <user id="20001">
              <params>
                <param name="a1-hash"
value="d8c32787fd9f00b5ad4246e93f757360"/>
                <param name="vm-a1-hash"
value="a53b5e8b5edf34208274cbaa00412097"/>
              </params>
              <variables>
                *<variable name="user_context" value="**default**"/>*
                <variable name="accountcode" value="20001"/>
              </variables>
            </user>
          </users>
        </group>
      </groups>
    </domain>
  </section>
</document>

Both internal and external sofia profiles have context = "public".
But it should be overridden here in the user settings to  DEFAULT.


Wwhen I call to any number using this subscriber 20001 it calls in *context
PUBLIC instead of DEFAULT.*
2010-04-02 17:27:32.384472 [INFO] mod_dialplan_xml.c:418 Processing
20001->001...[removed]... in context *public*

I need it to call in context default.

When I change sofia profile settings to context="default", it works fine,
but shouldn't it work like this as well?

Please any hints where else should I change the context?

Thank you,
vIT
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From sos at sokhapkin.dyndns.org  Fri Apr  2 14:40:22 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 2 Apr 2010 17:40:22 -0400
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
Message-ID: <201004021740.22716.sos@sokhapkin.dyndns.org>

To me monit works the best. http://mmonit.com/monit/

On Friday 02 April 2010, Lon Baker wrote:
> Hi gang!
> 
> I'm trying to determine the best way to periodically check the status
> of production Freeswitch deployments.
> 
> While Freeswitch has been incredibly stable in production, I would
> like to ping it periodically to check that its not hung in someway.
> 
> Has anyone considered or done this yet?
> 
> My thought it so issue a command via fs_cli periodically and compare
> the out. But am trying to determine what a good command would be.
> 
> thinking of using show channels or show calls, in addition to status.
> 
> Any thoughts? Is this a wrong-headed approach?
> 
> Thanks for any feedback.
> 
> Lon
> 
> _______________________________________________
> 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
> http://www.freeswitch.org
> 



From troy at tlainvestments.com  Fri Apr  2 14:46:21 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 2 Apr 2010 14:46:21 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
Message-ID: <B2428CEC-A0E4-4224-9968-74D86DC444C6@tlainvestments.com>

One way I've contemplated is to use munin/munin node on the FS box and nagios on a central server.  That would provide notification if the service were to go offline.  I have the components in place, but haven't gotten around to writing a FS nagios plugin (which could use fs_cli -x status, for example).

-Troy


On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:

> Hi gang!
> 
> I'm trying to determine the best way to periodically check the status
> of production Freeswitch deployments.
> 
> While Freeswitch has been incredibly stable in production, I would
> like to ping it periodically to check that its not hung in someway.
> 
> Has anyone considered or done this yet?
> 
> My thought it so issue a command via fs_cli periodically and compare
> the out. But am trying to determine what a good command would be.
> 
> thinking of using show channels or show calls, in addition to status.
> 
> Any thoughts? Is this a wrong-headed approach?
> 
> Thanks for any feedback.
> 
> Lon
> 
> _______________________________________________
> 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
> http://www.freeswitch.org



From spencer at 5ninesolutions.com  Fri Apr  2 14:54:48 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 2 Apr 2010 14:54:48 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
Message-ID: <90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>

We run monit on each local machine sending out an options ping every  
minute.  If it doesn't get a response then monit restarts the  
freeswitch process.  The nice thing about this is that you can  
actually verify that Freeswitch is responding to SIP requests.


On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:

> Hi gang!
>
> I'm trying to determine the best way to periodically check the status
> of production Freeswitch deployments.
>
> While Freeswitch has been incredibly stable in production, I would
> like to ping it periodically to check that its not hung in someway.
>
> Has anyone considered or done this yet?
>
> My thought it so issue a command via fs_cli periodically and compare
> the out. But am trying to determine what a good command would be.
>
> thinking of using show channels or show calls, in addition to status.
>
> Any thoughts? Is this a wrong-headed approach?
>
> Thanks for any feedback.
>
> Lon
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>



From troy at tlainvestments.com  Fri Apr  2 14:58:02 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Fri, 2 Apr 2010 14:58:02 -0700
Subject: [Freeswitch-users] make fails
Message-ID: <5EB95B30-AF81-43D6-8583-6FEE7A967C92@tlainvestments.com>

Help.  I've compiled FS successfully on this box many times, but now, it's failing.  It must be something I've done as it even fails when I check out an svn revision that compiled in the past on this box.  I'm hopeful someone here can point me in the right direction.  Here's the output:

make
make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/chronos_client/freeswitch.svn/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /home/chronos_client/freeswitch.svn/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
Making all in .
gcc -I/home/chronos_client/freeswitch.svn/src/include -I/home/chronos_client/freeswitch.svn/src/include -I/home/chronos_client/freeswitch.svn/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/home/chronos_client/freeswitch.svn/libs/apr/include -I/home/chronos_client/freeswitch.svn/libs/apr-util/include -I/home/chronos_client/freeswitch.svn/libs/apr-util/xml/expat/lib -I/home/chronos_client/freeswitch.svn/libs/stfu -I/home/chronos_client/freeswitch.svn/libs/sqlite -I/home/chronos_client/freeswitch.svn/libs/pcre -I/home/chronos_client/freeswitch.svn/libs/speex/include -Ilibs/speex/include -I/home/chronos_client/freeswitch.svn/libs/srtp/include -I/home/chronos_client/freeswitch.svn/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -DSWITCH_HAVE_ODBC -I/usr/include -I/home/chronos_client/freeswitch.svn/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch freeswitch-switch.o  -pthread -lm ./.libs/libfreeswitch.so libs/apr/.libs/libapr-1.a libs/libedit/src/.libs/libedit.a -L/home/chronos_client/freeswitch.svn/libs/apr-util/xml/expat/lib -lrt -ldl -lcrypt -lpthread -lssl -lcrypto -lncurses -Wl,--rpath -Wl,/usr/local/freeswitch/lib
./.libs/libfreeswitch.so: undefined reference to `XML_ErrorString'
./.libs/libfreeswitch.so: undefined reference to `XML_SetUserData'
./.libs/libfreeswitch.so: undefined reference to `XML_ParserFree'
./.libs/libfreeswitch.so: undefined reference to `XML_SetElementHandler'
./.libs/libfreeswitch.so: undefined reference to `XML_SetCharacterDataHandler'
./.libs/libfreeswitch.so: undefined reference to `XML_GetErrorCode'
./.libs/libfreeswitch.so: undefined reference to `XML_ParserCreate'
./.libs/libfreeswitch.so: undefined reference to `XML_Parse'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1

Thanks!
Troy



From lon at kickasspixels.com  Fri Apr  2 17:23:22 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 2 Apr 2010 17:23:22 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com> 
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
Message-ID: <v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>

Would you care to share an example of the config for monit? I will
document the practice on the wiki.

On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
<spencer at 5ninesolutions.com> wrote:
> We run monit on each local machine sending out an options ping every
> minute. ?If it doesn't get a response then monit restarts the
> freeswitch process. ?The nice thing about this is that you can
> actually verify that Freeswitch is responding to SIP requests.
>
>
> On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
>
>> Hi gang!
>>
>> I'm trying to determine the best way to periodically check the status
>> of production Freeswitch deployments.
>>
>> While Freeswitch has been incredibly stable in production, I would
>> like to ping it periodically to check that its not hung in someway.
>>
>> Has anyone considered or done this yet?
>>
>> My thought it so issue a command via fs_cli periodically and compare
>> the out. But am trying to determine what a good command would be.
>>
>> thinking of using show channels or show calls, in addition to status.
>>
>> Any thoughts? Is this a wrong-headed approach?
>>
>> Thanks for any feedback.
>>
>> Lon
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>


From sos at sokhapkin.dyndns.org  Fri Apr  2 17:55:11 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 2 Apr 2010 20:55:11 -0400
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
	<v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
Message-ID: <201004022055.12057.sos@sokhapkin.dyndns.org>

check process freeswitch
  with pidfile /opt/freeswitch/run/freeswitch.pid
  start program = "/etc/init.d/freeswitch restart"
  stop program =  "/etc/init.d/freeswitch stop"
  if totalmem > 1000.0 MB for 5 cycles then alert
  if totalmem > 1500.0 MB for 5 cycles then alert
  if totalmem > 2000.0 MB for 5 cycles then restart
  if cpu > 60% for 5 cycles then alert
  if failed host HOST_IP_ADDRESS port 5060 type udp protocol sip within 2 
cycles
  then restart
  if 5 restarts within 5 cycles then timeout


On Friday 02 April 2010, Lon Baker wrote:
> Would you care to share an example of the config for monit? I will
> document the practice on the wiki.
> 
> On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
> 
> <spencer at 5ninesolutions.com> wrote:
> > We run monit on each local machine sending out an options ping every
> > minute. ?If it doesn't get a response then monit restarts the
> > freeswitch process. ?The nice thing about this is that you can
> > actually verify that Freeswitch is responding to SIP requests.
> >
> > On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
> >> Hi gang!
> >>
> >> I'm trying to determine the best way to periodically check the status
> >> of production Freeswitch deployments.
> >>
> >> While Freeswitch has been incredibly stable in production, I would
> >> like to ping it periodically to check that its not hung in someway.
> >>
> >> Has anyone considered or done this yet?
> >>
> >> My thought it so issue a command via fs_cli periodically and compare
> >> the out. But am trying to determine what a good command would be.
> >>
> >> thinking of using show channels or show calls, in addition to status.
> >>
> >> Any thoughts? Is this a wrong-headed approach?
> >>
> >> Thanks for any feedback.
> >>
> >> Lon
> >>
> >> _______________________________________________
> >> 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
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> 
> _______________________________________________
> 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
> http://www.freeswitch.org
> 



From anthony.minessale at gmail.com  Fri Apr  2 21:18:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Apr 2010 22:18:33 -0600
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
Message-ID: <w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>

Nothing is wrong with it,  there is no revision that broke it or there would
be 100,000 people saying it was broken.
Change you perspective to looking for the problem on your end and you will
have more luck finding your problem.

Backup your config by moving it out of the way, re-install and try the
defaults.
We use FS with nat like this 12 hours a day.


On Fri, Apr 2, 2010 at 2:53 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> > OK, for some reason it works now with:
> >
> > <param name="NDLB-force-rport" value="true"/>
> >
> > when I'm positive it wasn't enabled before.
>
> Shouldn't FS NAT features detect that the phones are registering from
> an internal IP and do the usual fs_nat=yes stuff in the contact header
> without forcing rport above and send back responses to the port the
> REGISTER came from?
>
> Thanks.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.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
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lon at kickasspixels.com  Fri Apr  2 23:16:59 2010
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 2 Apr 2010 23:16:59 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <201004022055.12057.sos@sokhapkin.dyndns.org>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
	<v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
	<201004022055.12057.sos@sokhapkin.dyndns.org>
Message-ID: <48507247-44CC-411D-BA03-AB1EAACB48F7@kickasspixels.com>

Thank you!

On Apr 2, 2010, at 5:55 PM, Sergey Okhapkin wrote:

> check process freeswitch
>  with pidfile /opt/freeswitch/run/freeswitch.pid
>  start program = "/etc/init.d/freeswitch restart"
>  stop program =  "/etc/init.d/freeswitch stop"
>  if totalmem > 1000.0 MB for 5 cycles then alert
>  if totalmem > 1500.0 MB for 5 cycles then alert
>  if totalmem > 2000.0 MB for 5 cycles then restart
>  if cpu > 60% for 5 cycles then alert
>  if failed host HOST_IP_ADDRESS port 5060 type udp protocol sip within 2 
> cycles
>  then restart
>  if 5 restarts within 5 cycles then timeout
> 
> 
> On Friday 02 April 2010, Lon Baker wrote:
>> Would you care to share an example of the config for monit? I will
>> document the practice on the wiki.
>> 
>> On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
>> 
>> <spencer at 5ninesolutions.com> wrote:
>>> We run monit on each local machine sending out an options ping every
>>> minute. ?If it doesn't get a response then monit restarts the
>>> freeswitch process. ?The nice thing about this is that you can
>>> actually verify that Freeswitch is responding to SIP requests.
>>> 
>>> On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
>>>> Hi gang!
>>>> 
>>>> I'm trying to determine the best way to periodically check the status
>>>> of production Freeswitch deployments.
>>>> 
>>>> While Freeswitch has been incredibly stable in production, I would
>>>> like to ping it periodically to check that its not hung in someway.
>>>> 
>>>> Has anyone considered or done this yet?
>>>> 
>>>> My thought it so issue a command via fs_cli periodically and compare
>>>> the out. But am trying to determine what a good command would be.
>>>> 
>>>> thinking of using show channels or show calls, in addition to status.
>>>> 
>>>> Any thoughts? Is this a wrong-headed approach?
>>>> 
>>>> Thanks for any feedback.
>>>> 
>>>> Lon
>>>> 
>>>> _______________________________________________
>>>> 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
>>>> http://www.freeswitch.org
>>> 
>>> _______________________________________________
>>> 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
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>> 
> 



From david.ponzone at gmail.com  Fri Apr  2 23:41:18 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 3 Apr 2010 08:41:18 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <q2j7abab2411004021353sec91c94mff3954fea571dcc@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<g2z7abab2411004020503x3e716cb4lcf86bbcb95202c01@mail.gmail.com>
	<EC2B96D5-F51D-45DA-A54C-6FC0874E0A1F@gmail.com>
	<06FEED52-70C4-4438-8279-921C61DB51F0@gmail.com>
	<s2p7abab2411004021152tec16b2b5qd4ce598e49cff120@mail.gmail.com>
	<1CB9F2BA-9575-4AB2-A1D2-F7FD48F8A316@gmail.com>
	<q2j7abab2411004021353sec91c94mff3954fea571dcc@mail.gmail.com>
Message-ID: <B4FE4E64-39EF-43F0-9C1B-6157807AFEB7@gmail.com>

Yes, in the lines:

2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup sofia/internal/1001 at 192.168.1.133 
  [CS_NEW] [INCOMPATIBLE_DESTINATION]
send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000:
     
------------------------------------------------------------------------
    SIP/2.0 488 Not Acceptable Here
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=bff390fd4255c0f9o0
    To: <sip:02074379497 at 192.168.4.156>;tag=m006c20Fg5Spa
    Call-ID: 1c27844d-e299e5e9 at 192.168.4.154

That is the answer from FS to the phone, just after receiving the  
INVITE that contains only G729 and NSE (??) in the SDP.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 02/04/2010 ? 22:53, Frank Church a ?crit :

> On 2 April 2010 20:36, David Ponzone <david.ponzone at gmail.com> wrote:
>> Frank,
>> re-read my second mail.
>
> http://pastebin.freeswitch.org/12617
>
>> You have to enable G729 in FS prefs (vars.xml).
> Was that obvious from the sip trace? I missed it the first time.
>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 20:52, Frank Church a ?crit :
>>
>> On 2 April 2010 13:46, David Ponzone <david.ponzone at gmail.com> wrote:
>>
>> Frank,
>>
>> sorry, I completely forgot one important detail:
>>
>> in the default conf, G729 is not allowed on any SIP profiles, so  
>> you have to
>>
>> modify vars.xml.
>>
>> You will find the following lines:
>>
>>   <X-PRE-PROCESS cmd="set"
>>
>> data 
>> ="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>
>>   <X-PRE-PROCESS cmd="set"  
>> data="outbound_codec_prefs=PCMU,PCMA,G729"/>
>>
>> For now, I recommend you replace them by:
>>
>>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,PCMA"/>
>>
>>   <X-PRE-PROCESS cmd="set"  
>> data="outbound_codec_prefs=G729,PCMU,PCMA"/>
>>
>> Then in FS console, do:
>>
>> sofia profile external restart reloadxml
>>
>> sofia profile internal restart reloadxml
>>
>> It should then work far better.
>>
>> What we did there is to make G729 an accepted codec in inbound  
>> INVITEs and a
>>
>> proposed codec for outbound INVITEs.
>>
>> Look in external.xml or internal.xml, and look at the variables
>>
>> inbound-codec-prefs and outbound-codec-prefs.
>>
>> David Ponzone  Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel:      01 74 03 18 97
>>
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>>
>> tel:      0811 46 26 26
>>
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>>
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>>
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>>
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>>
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>>
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 14:30, David Ponzone a ?crit :
>>
>> Frank,
>>
>> mod_g729 needs to be loaded, and then G729 needs to be negotiated  
>> on both
>>
>> legs.
>>
>> I really recommend you enable G729 on the Linksys, enable SIP trace  
>> on FS
>>
>> console:
>>
>> sofia profile external siptrace on
>>
>> sofia profile internal siptrace on
>>
>> then make a (failing) call, capture the log on FS console and paste  
>> that to:
>>
>> http://pastebin:freeswitch at pastebin.freeswitch.org/
>>
>> This the pastebin link
>>
>> http://pastebin.freeswitch.org/12616
>>
>> There is also one below it with the successful call.
>>
>> I have set the default provider in vars.xml rather than under the
>> sip_profiles/external. Is there a section in the default dialplan  
>> that
>> handles the default context?
>>
>> Can the options below be used in the default for the dialplan and
>> sip_profiles?
>>
>> In the dial plan
>> <action application="set" data="proxy_media=true"/>
>>
>> In the sip profile
>> <param name="inbound-proxy-media" value="true"/>
>>
>> The Linksys accounts are in the default contexts.
>>
>> Then, send us back the link to your paste.
>>
>> You can also join us on #freeswitch (irc.freenode.net), for some  
>> live help.
>>
>> David Ponzone  Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel:      01 74 03 18 97
>>
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>>
>> tel:      0811 46 26 26
>>
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>>
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>>
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>>
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>>
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>>
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 02/04/2010 ? 14:03, Frank Church a ?crit :
>>
>> On 2 April 2010 10:23, Jason White <jason at jasonjgw.net> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>>
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>>
>> The carrier insists on G729, although they can accept G711. I think
>>
>> their call volume does not make it easy on them and their customers  
>> as
>>
>> well.
>>
>> I did some googling and came up with freeswitch-codec-passthru- 
>> g729. I
>>
>> have also read http://wiki.freeswitch.org/wiki/Proxy_Media and
>>
>> http://wiki.freeswitch.org/wiki/Bypass_Media.
>>
>> In my module.conf.xml there is also <load module="mod_g729"/>.
>>
>> Does that mean that my installlation is configured for pass thru if I
>>
>> make the right adjustments?
>>
>> I have looked at
>>
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg12303.html
>>
>> and
>>
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg20949.html
>>
>> which speak of modifying the dialplan.
>>
>> This is a basic freeswitch setup using the defaults. I just added the
>>
>> extensions to conf/directory/default and changed the provider in
>>
>> vars.xml and I want to be able to do the same in
>>
>> conf/dialplan/default.xml.
>>
>> In conf/dialplan/default.xml the extension is matched by the
>>
>> destination. Is there an option for not falling through to other
>>
>> extensions if they also match?
>>
>> _______________________________________________
>>
>> 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
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>>
>> Frank Church
>>
>> =======================
>>
>> http://devblog.brahmancreations.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
>>
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>>
>> 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
>>
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.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
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Fri Apr  2 23:45:32 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 3 Apr 2010 08:45:32 +0200
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <48507247-44CC-411D-BA03-AB1EAACB48F7@kickasspixels.com>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
	<v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
	<201004022055.12057.sos@sokhapkin.dyndns.org>
	<48507247-44CC-411D-BA03-AB1EAACB48F7@kickasspixels.com>
Message-ID: <31C0F8F8-AE63-4D13-9BA7-49DCE237A170@gmail.com>

I would recommend to, obviously, ping all profiles of FS.
I recently had an issue where one profile only stopped working.
It happened only once, in a previous svn, but who knows...

Another way to monitor is to send a test call with sipp through a  
registered user to a gateway.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 03/04/2010 ? 08:16, Lon Baker a ?crit :

> Thank you!
>
> On Apr 2, 2010, at 5:55 PM, Sergey Okhapkin wrote:
>
>> check process freeswitch
>> with pidfile /opt/freeswitch/run/freeswitch.pid
>> start program = "/etc/init.d/freeswitch restart"
>> stop program =  "/etc/init.d/freeswitch stop"
>> if totalmem > 1000.0 MB for 5 cycles then alert
>> if totalmem > 1500.0 MB for 5 cycles then alert
>> if totalmem > 2000.0 MB for 5 cycles then restart
>> if cpu > 60% for 5 cycles then alert
>> if failed host HOST_IP_ADDRESS port 5060 type udp protocol sip  
>> within 2
>> cycles
>> then restart
>> if 5 restarts within 5 cycles then timeout
>>
>>
>> On Friday 02 April 2010, Lon Baker wrote:
>>> Would you care to share an example of the config for monit? I will
>>> document the practice on the wiki.
>>>
>>> On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
>>>
>>> <spencer at 5ninesolutions.com> wrote:
>>>> We run monit on each local machine sending out an options ping  
>>>> every
>>>> minute. ?If it doesn't get a response then monit restarts the
>>>> freeswitch process. ?The nice thing about this is that you can
>>>> actually verify that Freeswitch is responding to SIP requests.
>>>>
>>>> On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
>>>>> Hi gang!
>>>>>
>>>>> I'm trying to determine the best way to periodically check the  
>>>>> status
>>>>> of production Freeswitch deployments.
>>>>>
>>>>> While Freeswitch has been incredibly stable in production, I would
>>>>> like to ping it periodically to check that its not hung in  
>>>>> someway.
>>>>>
>>>>> Has anyone considered or done this yet?
>>>>>
>>>>> My thought it so issue a command via fs_cli periodically and  
>>>>> compare
>>>>> the out. But am trying to determine what a good command would be.
>>>>>
>>>>> thinking of using show channels or show calls, in addition to  
>>>>> status.
>>>>>
>>>>> Any thoughts? Is this a wrong-headed approach?
>>>>>
>>>>> Thanks for any feedback.
>>>>>
>>>>> Lon
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> 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
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> 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
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org

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From abid_freeswitch at live.com  Sat Apr  3 00:02:55 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Sat, 3 Apr 2010 12:02:55 +0500
Subject: [Freeswitch-users] Radius/Authentication/Authorization
Message-ID: <COL109-W11819E0B21AF7600E30CD4F41B0@phx.gbl>



Dear Tihomir,
Any update and help on the below please.
Abid
From: abid_freeswitch at live.com
To: tculjaga at gmail.com
CC: neal at wanlink.com
Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization
Date: Wed, 31 Mar 2010 19:35:32 +0500








Dear Tihomir,
Thank you very much for the configuration example but in which files to place these configurations. Please bear with me because I am new to FreeSwitch and if you could provide complete steps.
Also when I compile FS and load the module mod_rad_auth in conf/autoload_configs/modules.conf.xml, I get an error while starting FS as follows.
2010-03-31 19:32:29.399466 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so**/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str**
Please advise. Thanks.
Regards-----------Abid Saleem
--Forwarded Message Attachment--
From: tculjaga at gmail.com
CC: neal at wanlink.com
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 30 Mar 2010 21:42:03 +0200
Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization

hello,

here is an example



<configuration name="rad_auth.conf" description="radius authentification module">
  <settings>
     <!-- backward compatibility to allow radiusclient config file instead of an embedded config -->

     <!--  <param name="radius_config" value="/usr/local/etc/radiusclient/radiusclient.conf"/>   -->
  </settings>

  <client>
    <param name="authserver" value="10.1.1.10:1812:gateway"/>

    <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/>
    <param name="seqfile" value="/var/run/radius.seq"/>
    <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/>

    <param name="default_realm" value=""/>
    <param name="radius_timeout" value="3"/>
    <param name="radius_retries" value="2"/>
    <param name="radius_deadtime" value="0"/>

    <param name="bindaddr" value="*"/>
  </client>

  <vsas>
    <!-- 
         name:       just a description
         value:      direct input or variable 
         pec:        vendor ID (0 for default, 9 for cisco...)

         expr:       1 for channel variable, 2 for direct input
         direction:  in for radius-request, out for radius-response
    -->

    <!-- mappings for radius request message; input attributes -->

    <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1" direction="in"/>
    <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9" expr="1" direction="in"/>

    <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9" expr="1" direction="in"/>
    <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0" expr="1" direction="in"/>

    <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0" direction="in"/>
    <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0" expr="0" direction="in"/>

    <param name="Login-User" id="1" value="1" pec="0" expr="0" direction="in"/>

    <!-- mappings for radius-response message; output values from returning outributes -->

    <param name="BILING_MODEL" id="109" value="billing_model" pec="9" expr="0" direction="out"/>
    <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9" expr="0" direction="out"/>

    <param name="CURRENCY" id="110" value="currency" pec="9" expr="0" direction="out"/>
    <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9" expr="0" direction="out"/>

    <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0" direction="out"/>
    <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration" pec="9" expr="0" direction="out"/>

    <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9" expr="0" direction="out"/>
    <!-- expr param is to be ignored here-->

  </vsas>
</configuration>




in the dialplan you need to trigger auth as:

      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



there are two behaviours:

1. authorize the call according to username&pass and dialed number - if authorized, the radius server returns credit time towards the dialed number
2. authorize the call according to username&pass - if authorized, the radius server returns the current account balance







  <extension name="unitest_rad-ANI-auth">
    <condition field="destination_number" expression="^601$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>


      <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>

      <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
<!--      <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> -->

      <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>

<!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
      <action inline="true" application="set" data="PASSWD=003282"/>

<!--      <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>  -->

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO  billing_model=${billing_model}"/>
      <action application="log" data="INFO  credit_amount=${credit_amount}"/>

      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO  preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>

      <action application="log" data="INFO  h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/> -->

    </condition>
  </extension>


  <extension name="unitest_rad-ANI-balance">
    <condition field="destination_number" expression="^602$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>


      <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>

      <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
<!--      <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> -->

      <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>

<!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
      <action inline="true" application="set" data="PASSWD=003282"/>

      <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>



      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO  billing_model=${billing_model}"/>
      <action application="log" data="INFO  credit_amount=${credit_amount}"/>

      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO  preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>

      <action application="log" data="INFO  h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/> -->

    </condition>
  </extension>


will update the wiki by the end of the week.


you have enough information for now.

Tihomir.



On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch at live.com> wrote:






Hi Neal and other Contributors to FS,
I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration.

If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation.

Regards----------Abid SaleemSr. Product Manager
 		 	   		  
Hotmail: Powerful Free email with security by Microsoft. Get it now. 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From jayesh.voip at gmail.com  Sat Apr  3 00:43:38 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Sat, 3 Apr 2010 13:13:38 +0530
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
References: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
Message-ID: <x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>

I expected at east one reply saying that the question is stupid, and the
solution is simple !!
Any folks who can help me understand only how to achieve this in FS which is
acheivable in asterisk as follows:
domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
the call goes to context "mydomain" in dialplan)
domain = yourdomain.com, yourdomain (if any call has domain as "
yourdomain.com", the call goes to context "yourdomain" in dialplan)

These calls can come from anywhere, in my case it comes from an Opensips
instance !!

Thanks for any replies :)

--- Jayesh

On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> Hi All,
> I am quite very new to freeswitch and I am kind of playing with it to
> understand it better.
> I am primarily using Opensips as registrar and SIP Proxy and intend to use
> FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My
> Opensips is a multi-domain setup and I wish to have all the configuration of
> media-capabilties segregated domain-wise in the FS too.
>
> For eg: When a call for user at domain1.com needs to go to voicemail, I
> redirect that call to FS IP address keeping the URI intact. I add the
> mailbox number as a header as X-Mailbox and have FS extract it and go to
> appropriate mailbox. Similarly when a call for user at domain2.com needs to
> go to voicemail I do the same thing.
> The requirement is I want to maintain the dialplans for each domains
> separately. Thus if call from Opensips comes to FS with domain as domain1,
> the call should go to dialplan context domain1 and similarly if call from
> Opensips comes to FS with domain2 the call handling should be mentioned in
> the domain2 context.
>
> The problem is; I am not able to send the calls to respective contexts
> according to their domains when they come from Opensips. I've read the
> examples on multi-domain setup and have tried taking some help from that
> example, but whenever the call comes from Opensips to FS, it tries to go
> into the context that is defined in the SIP Profile. If i don't mention
> anything in the SIP Profile, it tries to search for default context.
> I have tried the following:
> 1) Created file domain1.xml and domain2.xml in the directory folder.
> 2) mentioned parameters in domain1.xml as follows:
> <include>
> <domain name = "domain1.com"
> <params>
> <param name="user_context" value "domain1.com"/>
> </params>
> </domain>
> </include>
> 3) Similarly done for file domain2.xml.
>
> But I am just not able to get the calls to the required context according
> to the domain value in the r-uri. In asterisk something like this in
> sip.conf worked fine for me:
> domain=domain1.com, domain1.com
> domain=domain2.com, domain2.com
> Can someone please help me understanding where I am going wrong or have I
> mis-understood something?
>
> Thanks in advance !!
>
> --- Jayesh
>
>
>
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From hungngm at bkav.com.vn  Sat Apr  3 01:05:36 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Sat, 03 Apr 2010 15:05:36 +0700
Subject: [Freeswitch-users] Some question about mod_fifo ??
Message-ID: <C80CF6A5C8F04344D308F1B135076B4D890DC624@hungngm>



 

Hi Seven Du.

Thanks to yours suggetion.

I have an ideal, it is: when the call between caller and agent is set, the
caller_id is determined. So, i want to edit code to sent the agent information
(the call_id and call_id_number) which will be displayed againt in the agent's
softphone (as Xlite..) when the call is happening.

I read some documents but i still can't determine: It's maybe yes or maybe to do
this and where to do this.

Can you give me some comments.

Best Regard.

 

Seven Du [dujinfang at gmail.com]

 

??As discussed in the list, it's not a freeswitch problem but a reality of
life.

Think about customer A and B calls in one after another, then if
FreeSWITCH call agent X with caller id A and Y with caller id B, and
angent Y answers before X, then

1) if bridge Y with A with the FIFO rule, then the caller id is wrong
2) if bridge Y with B, the caller id is right but it breaks the rule
of FIFO - A should be served before B!! And what even worse is that
if X never answer A then A never can be served which is really
unfair!!

Of course you don't want 1), and you don't need mod_fifo if you want
behavior 2), you just need some dialplan trick or some simple Lua
script I think. Also FreeSWITCH is designed to be easily extended with
almost any languages so feel free to implement anything.

2010/3/31 Nguy??n M???nh H??ng 

:
> Hi Mike and Seven Du.
> Thanks to yours help.
> I known the mechanism of mod_fifo.
>>>http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.ht
ml.
> What a pity, It can't solve this problem. I can't use freeswitch for my call
> center.
> Hope new version can solve this !!!

_______________________________________________
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
http://www.freeswitch.org

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From red.rain.seven at gmail.com  Sat Apr  3 02:30:52 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 3 Apr 2010 02:30:52 -0700
Subject: [Freeswitch-users] CLI status?
In-Reply-To: <201004022055.12057.sos@sokhapkin.dyndns.org>
References: <x2i5d3e0dc61004021424n7be3ee37n4fed5da9a97fd605@mail.gmail.com>
	<90F06993-1E5C-4A30-85E7-0EC98D9AA324@5ninesolutions.com>
	<v2u5d3e0dc61004021723i976322cdj59258ef9b263973a@mail.gmail.com>
	<201004022055.12057.sos@sokhapkin.dyndns.org>
Message-ID: <m2m59ad9ca11004030230h6af1ccfet3b22ca4b914dc953@mail.gmail.com>

Sergey:

Thank you for sharing , I am just recently looking to use monit to monitor
our freeswitch and other production servers.

Henry

Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com
Contact Me [image:
Linkedin]<http://www.linkedin.com/pub/henry-huang/1/654/578>[image:
Facebook] <http://www.facebook.com/profile.php?id=595148342>[image:
Twitter]<http://twitter.com/unicsolution>


On Fri, Apr 2, 2010 at 5:55 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:

> check process freeswitch
>  with pidfile /opt/freeswitch/run/freeswitch.pid
>  start program = "/etc/init.d/freeswitch restart"
>  stop program =  "/etc/init.d/freeswitch stop"
>  if totalmem > 1000.0 MB for 5 cycles then alert
>  if totalmem > 1500.0 MB for 5 cycles then alert
>  if totalmem > 2000.0 MB for 5 cycles then restart
>  if cpu > 60% for 5 cycles then alert
>  if failed host HOST_IP_ADDRESS port 5060 type udp protocol sip within 2
> cycles
>  then restart
>  if 5 restarts within 5 cycles then timeout
>
>
> On Friday 02 April 2010, Lon Baker wrote:
> > Would you care to share an example of the config for monit? I will
> > document the practice on the wiki.
> >
> > On Fri, Apr 2, 2010 at 2:54 PM, Spencer Thomason
> >
> > <spencer at 5ninesolutions.com> wrote:
> > > We run monit on each local machine sending out an options ping every
> > > minute. ?If it doesn't get a response then monit restarts the
> > > freeswitch process. ?The nice thing about this is that you can
> > > actually verify that Freeswitch is responding to SIP requests.
> > >
> > > On Apr 2, 2010, at 2:24 PM, Lon Baker wrote:
> > >> Hi gang!
> > >>
> > >> I'm trying to determine the best way to periodically check the status
> > >> of production Freeswitch deployments.
> > >>
> > >> While Freeswitch has been incredibly stable in production, I would
> > >> like to ping it periodically to check that its not hung in someway.
> > >>
> > >> Has anyone considered or done this yet?
> > >>
> > >> My thought it so issue a command via fs_cli periodically and compare
> > >> the out. But am trying to determine what a good command would be.
> > >>
> > >> thinking of using show channels or show calls, in addition to status.
> > >>
> > >> Any thoughts? Is this a wrong-headed approach?
> > >>
> > >> Thanks for any feedback.
> > >>
> > >> Lon
> > >>
> > >> _______________________________________________
> > >> 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
> > >> http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > 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
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From lloydie.t at googlemail.com  Sat Apr  3 04:10:30 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Sat, 3 Apr 2010 12:10:30 +0100
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
References: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
	<x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
Message-ID: <n2x8b61bd671004030410jb93443f9med5f79bbac6e81a@mail.gmail.com>

On your DNS server does mydomain.com and yourdomain point to your FS server
ip address. I suspect you may not get much response with showing the errors
for the fs cli at least. Someone more qualified maybe able to help with this
info

Lloydie T

On 3 April 2010 08:43, Jayesh Nambiar <jayesh.voip at gmail.com> wrote:

> I expected at east one reply saying that the question is stupid, and the
> solution is simple !!
> Any folks who can help me understand only how to achieve this in FS which
> is acheivable in asterisk as follows:
> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
> the call goes to context "mydomain" in dialplan)
> domain = yourdomain.com, yourdomain (if any call has domain as "
> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>
> These calls can come from anywhere, in my case it comes from an Opensips
> instance !!
>
> Thanks for any replies :)
>
> --- Jayesh
>
>
> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> Hi All,
>> I am quite very new to freeswitch and I am kind of playing with it to
>> understand it better.
>> I am primarily using Opensips as registrar and SIP Proxy and intend to use
>> FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My
>> Opensips is a multi-domain setup and I wish to have all the configuration of
>> media-capabilties segregated domain-wise in the FS too.
>>
>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>> redirect that call to FS IP address keeping the URI intact. I add the
>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>> go to voicemail I do the same thing.
>> The requirement is I want to maintain the dialplans for each domains
>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>> the call should go to dialplan context domain1 and similarly if call from
>> Opensips comes to FS with domain2 the call handling should be mentioned in
>> the domain2 context.
>>
>> The problem is; I am not able to send the calls to respective contexts
>> according to their domains when they come from Opensips. I've read the
>> examples on multi-domain setup and have tried taking some help from that
>> example, but whenever the call comes from Opensips to FS, it tries to go
>> into the context that is defined in the SIP Profile. If i don't mention
>> anything in the SIP Profile, it tries to search for default context.
>> I have tried the following:
>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>> 2) mentioned parameters in domain1.xml as follows:
>> <include>
>> <domain name = "domain1.com"
>> <params>
>> <param name="user_context" value "domain1.com"/>
>> </params>
>> </domain>
>> </include>
>> 3) Similarly done for file domain2.xml.
>>
>> But I am just not able to get the calls to the required context according
>> to the domain value in the r-uri. In asterisk something like this in
>> sip.conf worked fine for me:
>> domain=domain1.com, domain1.com
>> domain=domain2.com, domain2.com
>>  Can someone please help me understanding where I am going wrong or have I
>> mis-understood something?
>>
>> Thanks in advance !!
>>
>> --- Jayesh
>>
>>
>>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From lloydie.t at googlemail.com  Sat Apr  3 04:16:44 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Sat, 3 Apr 2010 12:16:44 +0100
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <n2x8b61bd671004030410jb93443f9med5f79bbac6e81a@mail.gmail.com>
References: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
	<x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
	<n2x8b61bd671004030410jb93443f9med5f79bbac6e81a@mail.gmail.com>
Message-ID: <t2m8b61bd671004030416k17298698h48daf41f5e957fa1@mail.gmail.com>

with showing = without showing

On 3 April 2010 12:10, lloyd thomas <lloydie.t at googlemail.com> wrote:

> On your DNS server does mydomain.com and yourdomain point to your FS
> server ip address. I suspect you may not get much response with showing the
> errors for the fs cli at least. Someone more qualified maybe able to help
> with this info
>
> Lloydie T
>
> On 3 April 2010 08:43, Jayesh Nambiar <jayesh.voip at gmail.com> wrote:
>
>> I expected at east one reply saying that the question is stupid, and the
>> solution is simple !!
>> Any folks who can help me understand only how to achieve this in FS which
>> is acheivable in asterisk as follows:
>> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
>> the call goes to context "mydomain" in dialplan)
>> domain = yourdomain.com, yourdomain (if any call has domain as "
>> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>>
>> These calls can come from anywhere, in my case it comes from an Opensips
>> instance !!
>>
>> Thanks for any replies :)
>>
>> --- Jayesh
>>
>>
>> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>>
>>> Hi All,
>>> I am quite very new to freeswitch and I am kind of playing with it to
>>> understand it better.
>>> I am primarily using Opensips as registrar and SIP Proxy and intend to
>>> use FS as media server handling voicemails, IVR, Announcements, MeetMe etc.
>>> My Opensips is a multi-domain setup and I wish to have all the configuration
>>> of media-capabilties segregated domain-wise in the FS too.
>>>
>>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>>> redirect that call to FS IP address keeping the URI intact. I add the
>>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>>> go to voicemail I do the same thing.
>>> The requirement is I want to maintain the dialplans for each domains
>>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>>> the call should go to dialplan context domain1 and similarly if call from
>>> Opensips comes to FS with domain2 the call handling should be mentioned in
>>> the domain2 context.
>>>
>>> The problem is; I am not able to send the calls to respective contexts
>>> according to their domains when they come from Opensips. I've read the
>>> examples on multi-domain setup and have tried taking some help from that
>>> example, but whenever the call comes from Opensips to FS, it tries to go
>>> into the context that is defined in the SIP Profile. If i don't mention
>>> anything in the SIP Profile, it tries to search for default context.
>>> I have tried the following:
>>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>>> 2) mentioned parameters in domain1.xml as follows:
>>> <include>
>>> <domain name = "domain1.com"
>>> <params>
>>> <param name="user_context" value "domain1.com"/>
>>> </params>
>>> </domain>
>>> </include>
>>> 3) Similarly done for file domain2.xml.
>>>
>>> But I am just not able to get the calls to the required context according
>>> to the domain value in the r-uri. In asterisk something like this in
>>> sip.conf worked fine for me:
>>> domain=domain1.com, domain1.com
>>> domain=domain2.com, domain2.com
>>>  Can someone please help me understanding where I am going wrong or have
>>> I mis-understood something?
>>>
>>> Thanks in advance !!
>>>
>>> --- Jayesh
>>>
>>>
>>>
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
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From anthony.minessale at gmail.com  Sat Apr  3 08:11:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 09:11:50 -0600
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
References: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>
	<x2v4fe8b8271004030043v16ad0ecveee7cef1dbaedf52@mail.gmail.com>
Message-ID: <i2v191c3a031004030811q2ed9458dsdf05e10c51dd31f7@mail.gmail.com>

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

The first one transfers the call to the desired exten/dialplan/context and
the 2nd one executes the specified extension in a similar manner and returns
to the same point in the dp.

You make one inbound context and use routing logic from there to decide
which context to transfer to.

Next time, please have a little more patience, I don't like it when people
reply to themselves on the list asking why nobody answered when their
question is only unanswered for 2 days especially during a holiday weekend.



On Sat, Apr 3, 2010 at 1:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> I expected at east one reply saying that the question is stupid, and the
> solution is simple !!
> Any folks who can help me understand only how to achieve this in FS which
> is acheivable in asterisk as follows:
> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
> the call goes to context "mydomain" in dialplan)
> domain = yourdomain.com, yourdomain (if any call has domain as "
> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>
> These calls can come from anywhere, in my case it comes from an Opensips
> instance !!
>
> Thanks for any replies :)
>
> --- Jayesh
>
>
> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> Hi All,
>> I am quite very new to freeswitch and I am kind of playing with it to
>> understand it better.
>> I am primarily using Opensips as registrar and SIP Proxy and intend to use
>> FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My
>> Opensips is a multi-domain setup and I wish to have all the configuration of
>> media-capabilties segregated domain-wise in the FS too.
>>
>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>> redirect that call to FS IP address keeping the URI intact. I add the
>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>> go to voicemail I do the same thing.
>> The requirement is I want to maintain the dialplans for each domains
>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>> the call should go to dialplan context domain1 and similarly if call from
>> Opensips comes to FS with domain2 the call handling should be mentioned in
>> the domain2 context.
>>
>> The problem is; I am not able to send the calls to respective contexts
>> according to their domains when they come from Opensips. I've read the
>> examples on multi-domain setup and have tried taking some help from that
>> example, but whenever the call comes from Opensips to FS, it tries to go
>> into the context that is defined in the SIP Profile. If i don't mention
>> anything in the SIP Profile, it tries to search for default context.
>> I have tried the following:
>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>> 2) mentioned parameters in domain1.xml as follows:
>> <include>
>> <domain name = "domain1.com"
>> <params>
>> <param name="user_context" value "domain1.com"/>
>> </params>
>> </domain>
>> </include>
>> 3) Similarly done for file domain2.xml.
>>
>> But I am just not able to get the calls to the required context according
>> to the domain value in the r-uri. In asterisk something like this in
>> sip.conf worked fine for me:
>> domain=domain1.com, domain1.com
>> domain=domain2.com, domain2.com
>>  Can someone please help me understanding where I am going wrong or have I
>> mis-understood something?
>>
>> Thanks in advance !!
>>
>> --- Jayesh
>>
>>
>>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Sat Apr  3 08:15:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 09:15:19 -0600
Subject: [Freeswitch-users] Some question about mod_fifo ??
In-Reply-To: <C80CF6A5C8F04344D308F1B135076B4D890DC624@hungngm>
References: <C80CF6A5C8F04344D308F1B135076B4D890DC624@hungngm>
Message-ID: <x2y191c3a031004030815s34dbf8e0s430c28efcac5e1cc@mail.gmail.com>

We already do it.
X-Lite does not support it.
If you try it with a phone like snom or polycom you will see it works just
like that.


2010/4/3 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>

>  Hi Seven Du.
>
> Thanks to yours suggetion.
>
> I have an ideal, it is: when the call between caller and agent is set, the
> caller_id is determined. So, i want to edit code to sent the agent
> information (the call_id and  call_id_number) which will be displayed
> againt in the agent's softphone (as Xlite..) when the call is happening.
>
> I read some documents but i still can't determine: It's maybe yes or maybe
> to do this and where to do this.
>
> Can you give me some comments.
>
> Best Regard.
>
> Seven Du <dujinfang at gmail.com> [dujinfang at gmail.com]
>
>
> ?As discussed in the list, it's not a freeswitch problem but a reality of
> life.
>
>
> Think about customer A and B calls in one after another, then if
> FreeSWITCH call agent X with caller id A and Y with caller id B, and
> angent Y answers before X, then
>
> 1) if bridge Y with A with the FIFO rule, then the caller id is wrong
> 2) if bridge Y with B, the caller id is right but it breaks the rule
> of FIFO - A should be served before B!! And what even worse is that
> if X never answer A then A never can be served which is really
> unfair!!
>
> Of course you don't want 1), and you don't need mod_fifo if you want
> behavior 2), you just need some dialplan trick or some simple Lua
> script I think. Also FreeSWITCH is designed to be easily extended with
> almost any languages so feel free to implement anything.
>
> 2010/3/31 Nguy?n M?nh H?ng :
>
> > Hi Mike and Seven Du.
> > Thanks to yours help.
> > I known the mechanism of mod_fifo.
> >>>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
> .
> > What a pity, It can't solve this problem. I can't use freeswitch for my
> call
> > center.
> > Hope new version can solve this !!!
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Sat Apr  3 08:20:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 09:20:28 -0600
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>
Message-ID: <r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>

yes, it was merged from external to internal.
delete it and update again.


On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:

>  The current version of FS I?m using is 17135. I tried to ?make current?
> and it errored out in the svn up portion:
>
>
>
> svn: UUID mismatch: existing directory 'libs/openzap' was checked out from
> a different repository
>
>
>
> Should I just delete the libs/openzap directory?
>
>
>
> openzap is commented out in modules.conf.xml.
>
>
>
> Thanks, Lars
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From fraserredmond at gmail.com  Sat Apr  3 08:49:21 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Sat, 3 Apr 2010 16:49:21 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
Message-ID: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>

I've got a FreeSwitch server up on Amazon EC2, ports wide open for my office
external-IP, server iptables disabled, and changed the FreeSwitch ACL
domains to "allow", so it's all wide open for now.

In the office I'm trying to connect to the server from Bria/X-lite. I've
entered a stun server (stun.freeswitch.org) and I can now call to the
server, but not from the server. I read this page:
   http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc
which suggested adding this variable to the user config:

http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunction

With that on I can now call to and from the server. However with or without
that although I can hear audio from the server, audio to the server isn't
arriving (doesn't appear in recordings), and dtmf doesn't get received
either.

When I hang up from the client, I see in the CLI that it gets that
instruction, so it hasn't started the call and lost all contact with the
softphone, it's receiving some instructions, but not the audio and dtmf.

The problem is that both the server and client are each behind NAT, so
either could be having the problem (on EC2 the auto-NAT doesn't work, so
I've specified the external rtp and sip ip's.. I've also turned on
aggressive-NAT in case that helps. Also I'm connecting to the server by a
sub-domain (A-name) rather than IP.)

I've got almost the same setup working fine on the internal network (same
dialplan and directory, and all the config is the same if it can be), so
it's got to be something to do with the NAT's.

Any suggestions on what the problem might be, or how to find it?

Cheers,
Fraser
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From larclap at yahoo.com  Sat Apr  3 08:56:22 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 3 Apr 2010 08:56:22 -0700
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>
	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>
Message-ID: <008801cad346$3511a240$9f34e6c0$@com>

I deleted the openzap directory. I ran 'svn up' to make sure I was current.
Then ran 'make current' and got:

 

making uninstall mod_vmd

 

making uninstall mod_xml_curl

 

making uninstall mod_xml_ldap

 

WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....

 

making uninstall mod_yaml

cd libs/openzap && autoconf

/bin/sh: line 0: cd: libs/openzap: No such file or directory

make: *** [libs/openzap/Makefile] Error 1

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 8:20 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

yes, it was merged from external to internal.
delete it and update again.



On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:

The current version of FS I'm using is 17135. I tried to 'make current' and
it errored out in the svn up portion:

 

svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a
different repository

 

Should I just delete the libs/openzap directory?

 

openzap is commented out in modules.conf.xml.

 

Thanks, Lars


_______________________________________________
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
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From riedinger at sns.eu  Sat Apr  3 08:14:59 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Sat, 03 Apr 2010 17:14:59 +0200
Subject: [Freeswitch-users] Conference Dial In Isn't Working (mod_conference)
Message-ID: <4BB75B73.7040501@sns.eu>

If I dial in to a conference from my Cisco IP Phone 7940 all is working 
as it should.

However, if I dial in via a voip carrier, the calls are disconnect with 
disconnect cause "facility rejected" by the voip carrier. It seems that 
Freeswitch tries to change some parameters of the call setup, which is 
refused. You find below the console log output. Do you have an idea, 
what I have to change to get it working?

I know that it is working in prinicple, because I can setup outgoing 
conferences via the voip carrier.

Thank you very much in advance
     Jan



=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2010.04.03 16:56:10 
=~=~=~=~=~=~=~=~=~=~=~=
2010-04-03 16:56:07.593334 [NOTICE] switch_channel.c:669 New 
Channel sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
[27446b50-dc2a-4088-83f2-7b7cc8f014e8]
2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [received][100]
2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4164 Remote SDP:
v=0

o=- 1270306567 1270306567 IN IP4 XXX.XX.XX.X6

s=-

c=IN IP4 XXX.XX.XX.X6

t=0 0

m=audio 19188 RTP/AVP 18 4 8 0 101

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[G7221:115:32000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[G7221:107:16000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[G722:9:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[PCMU:0:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[PCMA:8:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G729:18:8000:20]/[GSM:3:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[G7221:115:32000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[G7221:107:16000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[G722:9:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[PCMU:0:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[PCMA:8:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [G723:4:8000:20]/[GSM:3:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[G7221:115:32000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[G7221:107:16000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[G722:9:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[PCMU:0:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec 
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:2354 Set Codec 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 PCMA/8000 20 ms 160 samples
2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3524 Set 2833 
dtmf send/recv payload to 101
2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4310 
(sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) State Change CS_NEW -> CS_INIT
2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_INIT
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State INIT
2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:83 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA INIT
2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:117 
(sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) State Change CS_INIT -> CS_ROUTING
2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State INIT going to sleep
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_ROUTING
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State ROUTING
2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:140 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA ROUTING
2010-04-03 16:56:07.593334 [DEBUG] 
switch_core_state_machine.c:77 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard ROUTING
2010-04-03 16:56:07.593334 [INFO] mod_dialplan_xml.c:418 
Processing 49XXXXXXX6->49331YYYYYY in context public
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->unloop] continue=false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (PASS) 
[unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL) 
[unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->outside_call] continue=true
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Absolute 
Condition [outside_call]
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action 
set(outside_call=true)
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->call_debug] continue=true
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL) 
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
...
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->sns_conference] continue=false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (PASS) 
[sns_conference] destination_number(49331YYYYYY) =~ /^(49331YYYYYY)$/ 
break=on-false
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action answer()
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action 
set(conference_enforce_security=false)
Dialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action 
conference(49331YYYYYY-${domain_name}@default)
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:119 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State Change CS_ROUTING -> CS_EXECUTE
2010-04-03 16:56:07.596335 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State ROUTING going to sleep
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_EXECUTE
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State EXECUTE
2010-04-03 16:56:07.596335 [DEBUG] mod_sofia.c:226 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA EXECUTE
2010-04-03 16:56:07.596335 [DEBUG] 
switch_core_state_machine.c:157 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard EXECUTE
EXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
set(outside_call=true)
2010-04-03 16:56:07.596335 [DEBUG] mod_dptools.c:816 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SET [outside_call]=[true]
EXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 answer()
2010-04-03 16:56:07.596335 [DEBUG] sofia_glue.c:2594 AUDIO RTP 
[sofia/external/49XXXXXXX6 at XXX.XX.XX.X6] XXX.XX.XX.X7 port 17138 -> 
XXX.XX.XX.X6 port 19188 codec: 8 ms: 20
2010-04-03 16:56:07.596335 [DEBUG] switch_rtp.c:1182 Starting 
timer [soft] 160 bytes per 20ms
2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2774 Set 2833 
dtmf send payload to 101
2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2779 Set 2833 
dtmf receive payload to 101
2010-04-03 16:56:07.599335 [DEBUG] mod_sofia.c:636 Local SDP 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6:
v=0
o=FreeSWITCH 1270289429 1270289430 IN IP4 XXX.XX.XX.X7
s=FreeSWITCH
c=IN IP4 XXX.XX.XX.X7
t=0 0
m=audio 17138 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2010-04-03 16:56:07.599335 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [completed][200]
2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.599335 [NOTICE] mod_dptools.c:719 Channel 
[sofia/external/49XXXXXXX6 at XXX.XX.XX.X6] has been answered
EXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
set(conference_enforce_security=false)
2010-04-03 16:56:07.599335 [DEBUG] mod_dptools.c:816 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SET 
[conference_enforce_security]=[false]
EXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
conference(49331YYYYYY-XXX.XX.XX.X7 at default)
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'mute' bound to '0'.
2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1616 max len 1
2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1620 min len 1
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'deaf mute' bound to '*'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'energy up' bound to '9'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'energy equ' bound to '8'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'energy dn' bound to '7'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol talk up' bound to '3'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol talk zero' bound to '2'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol talk dn' bound to '1'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol listen up' bound to '6'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol listen zero' bound to '5'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'vol listen dn' bound to '4'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671 
Installing default caller control action 'hangup' bound to '#'.
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:4990 Raw 
Codec Activation Success L16 at 8000hz 1 channel 20ms
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5035 Raw 
Codec Activation Success L16 at 8000hz 1 channel 20ms
2010-04-03 16:56:07.599335 [DEBUG] switch_core_codec.c:122 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Push codec L16:10
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:994 Setup 
timer success interval: 20  samples: 160
2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:2202 Setup 
timer soft success interval: 20  samples: 160
2010-04-03 16:56:07.617340 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [ready][200]
2010-04-03 16:56:07.839332 [NOTICE] sofia.c:481 Hangup 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [CS_EXECUTE] [FACILITY_REJECTED]
2010-04-03 16:56:07.839332 [DEBUG] switch_channel.c:2071 Send 
signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [KILL]
2010-04-03 16:56:07.839332 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.848341 [DEBUG] mod_conference.c:2473 Channel 
leaving conference, cause: FACILITY_REJECTED
2010-04-03 16:56:07.851332 [DEBUG] switch_core_codec.c:146 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Restore previous codec PCMA:8.
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State EXECUTE going to sleep
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_HANGUP
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State HANGUP
2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:408 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Overriding SIP cause 501 with 200 
from the other leg
2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:414 Channel 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 hanging up, cause: FACILITY_REJECTED
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:46 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard HANGUP, cause: FACILITY_REJECTED
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State HANGUP going to sleep
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:333 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State Change CS_HANGUP -> CS_REPORTING
2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_REPORTING
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State REPORTING
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:53 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard REPORTING, cause: FACILITY_REJECTED
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State REPORTING going to sleep
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:327 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State Change CS_REPORTING -> CS_DESTROY
2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018 
Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1161 
Session 7 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) Locked, Waiting on 
external entities
2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1179 
Session 7 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) Ended
2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1181 
Close Channel sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [CS_DESTROY]
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:428 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
Running State Change CS_DESTROY
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State DESTROY
2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:341 
sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA DESTROY
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:60 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 
Standard DESTROY
2010-04-03 16:56:07.851332 [DEBUG] 
switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) 
State DESTROY going to sleep
2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1361 Write 
Lock ON
2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1364 Write 
Lock OFF


-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                        E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From anthony.minessale at gmail.com  Sat Apr  3 09:16:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 10:16:14 -0600
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <008801cad346$3511a240$9f34e6c0$@com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>
	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>
	<008801cad346$3511a240$9f34e6c0$@com>
Message-ID: <w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>

if you actually want to use openzap after you delete it and svn up, issue

make oz-reconf

then make as usual.


On Sat, Apr 3, 2010 at 9:56 AM, Lars Zeb <larclap at yahoo.com> wrote:

>  I deleted the openzap directory. I ran ?svn up? to make sure I was
> current. Then ran ?make current? and got:
>
>
>
> making uninstall mod_vmd
>
>
>
> making uninstall mod_xml_curl
>
>
>
> making uninstall mod_xml_ldap
>
>
>
> WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....
>
>
>
> making uninstall mod_yaml
>
> cd libs/openzap && autoconf
>
> /bin/sh: line 0: cd: libs/openzap: No such file or directory
>
> make: *** [libs/openzap/Makefile] Error 1
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Saturday, April 03, 2010 8:20 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Error in 'svn up'
>
>
>
> yes, it was merged from external to internal.
> delete it and update again.
>
>  On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:
>
> The current version of FS I?m using is 17135. I tried to ?make current? and
> it errored out in the svn up portion:
>
>
>
> svn: UUID mismatch: existing directory 'libs/openzap' was checked out from
> a different repository
>
>
>
> Should I just delete the libs/openzap directory?
>
>
>
> openzap is commented out in modules.conf.xml.
>
>
>
> Thanks, Lars
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From brian at freeswitch.org  Sat Apr  3 09:19:04 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 11:19:04 -0500
Subject: [Freeswitch-users] Conference Dial In Isn't Working
	(mod_conference)
In-Reply-To: <4BB75B73.7040501@sns.eu>
References: <4BB75B73.7040501@sns.eu>
Message-ID: <E18056B6-90B7-4D7D-AB9C-1B979F7042C6@freeswitch.org>

sofia profile xxxxx siptrace on (replace xxx with the profile)

Then try again.

/b

On Apr 3, 2010, at 10:14 AM, Jan Riedinger wrote:

> However, if I dial in via a voip carrier, the calls are disconnect with 
> disconnect cause "facility rejected" by the voip carrier. It seems that 
> Freeswitch tries to change some parameters of the call setup, which is 
> refused. You find below the console log output. Do you have an idea, 
> what I have to change to get it working?



From anthony.minessale at gmail.com  Sat Apr  3 09:23:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 10:23:36 -0600
Subject: [Freeswitch-users] Conference Dial In Isn't Working
	(mod_conference)
In-Reply-To: <4BB75B73.7040501@sns.eu>
References: <4BB75B73.7040501@sns.eu>
Message-ID: <l2h191c3a031004030923x6668e77ena2996e9a3006daaa@mail.gmail.com>

run sofia profile internal siptrace on
and repeat so you can see the sip traffic.


On Sat, Apr 3, 2010 at 9:14 AM, Jan Riedinger <riedinger at sns.eu> wrote:

> If I dial in to a conference from my Cisco IP Phone 7940 all is working
> as it should.
>
> However, if I dial in via a voip carrier, the calls are disconnect with
> disconnect cause "facility rejected" by the voip carrier. It seems that
> Freeswitch tries to change some parameters of the call setup, which is
> refused. You find below the console log output. Do you have an idea,
> what I have to change to get it working?
>
> I know that it is working in prinicple, because I can setup outgoing
> conferences via the voip carrier.
>
> Thank you very much in advance
>     Jan
>
>
>
> =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2010.04.03 16:56:10
> =~=~=~=~=~=~=~=~=~=~=~=
>  [36m2010-04-03 16:56:07.593334 [NOTICE] switch_channel.c:669 New
> Channel sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> [27446b50-dc2a-4088-83f2-7b7cc8f014e8]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4153 Channel
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [received][100]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4164 Remote SDP:
> v=0
>
> o=- 1270306567 1270306567 IN IP4 XXX.XX.XX.X6
>
> s=-
>
> c=IN IP4 XXX.XX.XX.X6
>
> t=0 0
>
> m=audio 19188 RTP/AVP 18 4 8 0 101
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-15
>
>
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[G7221:115:32000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[G7221:107:16000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[G722:9:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[PCMU:0:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[PCMA:8:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G729:18:8000:20]/[GSM:3:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[G7221:115:32000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[G7221:107:16000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[G722:9:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[PCMU:0:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[PCMA:8:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [G723:4:8000:20]/[GSM:3:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[G7221:115:32000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[G7221:107:16000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[G722:9:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMU:0:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3585 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:2354 Set Codec
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 PCMA/8000 20 ms 160 samples
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia_glue.c:3524 Set 2833
> dtmf send/recv payload to 101
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] sofia.c:4310
> (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) State Change CS_NEW -> CS_INIT
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_INIT
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State INIT
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:83
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA INIT
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:117
> (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) State Change CS_INIT ->
> CS_ROUTING
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:338 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State INIT going to sleep
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_ROUTING
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State ROUTING
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG] mod_sofia.c:140
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA ROUTING
>  [m [33m2010-04-03 16:56:07.593334 [DEBUG]
> switch_core_state_machine.c:77 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard ROUTING
>  [m [32m2010-04-03 16:56:07.593334 [INFO] mod_dialplan_xml.c:418
> Processing 49XXXXXXX6->49331YYYYYY in context public
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing
> [public->unloop] continue=false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (PASS)
> [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL)
> [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing
> [public->outside_call] continue=true
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Absolute
> Condition [outside_call]
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action
> set(outside_call=true)
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing
> [public->call_debug] continue=true
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL)
> [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> ...
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 parsing
> [public->sns_conference] continue=false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Regex (PASS)
> [sns_conference] destination_number(49331YYYYYY) =~ /^(49331YYYYYY)$/
> break=on-false
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action answer()
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action
> set(conference_enforce_security=false)
>  [m [33mDialplan: sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Action
> conference(49331YYYYYY-${domain_name}@default)
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:119 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State Change CS_ROUTING -> CS_EXECUTE
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:341 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State ROUTING going to sleep
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_EXECUTE
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State EXECUTE
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] mod_sofia.c:226
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA EXECUTE
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG]
> switch_core_state_machine.c:157 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard EXECUTE
>  [m [33mEXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> set(outside_call=true)
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] mod_dptools.c:816
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SET [outside_call]=[true]
>  [m [33mEXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 answer()
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] sofia_glue.c:2594 AUDIO RTP
> [sofia/external/49XXXXXXX6 at XXX.XX.XX.X6] XXX.XX.XX.X7 port 17138 ->
> XXX.XX.XX.X6 port 19188 codec: 8 ms: 20
>  [m [33m2010-04-03 16:56:07.596335 [DEBUG] switch_rtp.c:1182 Starting
> timer [soft] 160 bytes per 20ms
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2774 Set 2833
> dtmf send payload to 101
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] sofia_glue.c:2779 Set 2833
> dtmf receive payload to 101
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_sofia.c:636 Local SDP
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6:
> v=0
> o=FreeSWITCH 1270289429 1270289430 IN IP4 XXX.XX.XX.X7
> s=FreeSWITCH
> c=IN IP4 XXX.XX.XX.X7
> t=0 0
> m=audio 17138 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] sofia.c:4153 Channel
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [completed][200]
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [36m2010-04-03 16:56:07.599335 [NOTICE] mod_dptools.c:719 Channel
> [sofia/external/49XXXXXXX6 at XXX.XX.XX.X6] has been answered
>  [m [33mEXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> set(conference_enforce_security=false)
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_dptools.c:816
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SET
> [conference_enforce_security]=[false]
>  [m [33mEXECUTE sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> conference(49331YYYYYY-XXX.XX.XX.X7 at default)
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'mute' bound to '0'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1616 max len 1
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_ivr.c:1620 min len 1
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'deaf mute' bound to '*'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'energy up' bound to '9'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'energy equ' bound to '8'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'energy dn' bound to '7'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol talk up' bound to '3'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol talk zero' bound to '2'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol talk dn' bound to '1'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol listen up' bound to '6'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol listen zero' bound to '5'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'vol listen dn' bound to '4'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5671
> Installing default caller control action 'hangup' bound to '#'.
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:4990 Raw
> Codec Activation Success L16 at 8000hz 1 channel 20ms
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:5035 Raw
> Codec Activation Success L16 at 8000hz 1 channel 20ms
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_codec.c:122
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Push codec L16:10
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:994 Setup
> timer success interval: 20  samples: 160
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] switch_core_session.c:638
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.599335 [DEBUG] mod_conference.c:2202 Setup
> timer soft success interval: 20  samples: 160
>  [m [33m2010-04-03 16:56:07.617340 [DEBUG] sofia.c:4153 Channel
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 entering state [ready][200]
>  [m [36m2010-04-03 16:56:07.839332 [NOTICE] sofia.c:481 Hangup
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [CS_EXECUTE] [FACILITY_REJECTED]
>  [m [33m2010-04-03 16:56:07.839332 [DEBUG] switch_channel.c:2071 Send
> signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [KILL]
>  [m [33m2010-04-03 16:56:07.839332 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.848341 [DEBUG] mod_conference.c:2473 Channel
> leaving conference, cause: FACILITY_REJECTED
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_codec.c:146
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Restore previous codec PCMA:8.
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:348 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State EXECUTE going to sleep
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_HANGUP
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State HANGUP
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:408
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 Overriding SIP cause 501 with 200
> from the other leg
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:414 Channel
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 hanging up, cause:
> FACILITY_REJECTED
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:46 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard HANGUP, cause: FACILITY_REJECTED
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:499 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State HANGUP going to sleep
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:333 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State Change CS_HANGUP -> CS_REPORTING
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:314 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_REPORTING
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State REPORTING
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:53 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard REPORTING, cause: FACILITY_REJECTED
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:590 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State REPORTING going to sleep
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:327 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State Change CS_REPORTING -> CS_DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1018
> Send signal sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [BREAK]
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] switch_core_session.c:1161
> Session 7 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) Locked, Waiting on
> external entities
>  [m [36m2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1179
> Session 7 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6) Ended
>  [m [36m2010-04-03 16:56:07.851332 [NOTICE] switch_core_session.c:1181
> Close Channel sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 [CS_DESTROY]
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:428 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> Running State Change CS_DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG] mod_sofia.c:341
> sofia/external/49XXXXXXX6 at XXX.XX.XX.X6 SOFIA DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:60 sofia/external/49XXXXXXX6 at XXX.XX.XX.X6
> Standard DESTROY
>  [m [33m2010-04-03 16:56:07.851332 [DEBUG]
> switch_core_state_machine.c:439 (sofia/external/49XXXXXXX6 at XXX.XX.XX.X6)
> State DESTROY going to sleep
>  [m [33m2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1361 Write
> Lock ON
>  [m [33m2010-04-03 16:56:07.866331 [DEBUG] mod_conference.c:1364 Write
> Lock OFF
>
>
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                        E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From frank at carmickle.com  Sat Apr  3 09:33:05 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Sat, 3 Apr 2010 12:33:05 -0400
Subject: [Freeswitch-users] Conference Dial In Isn't
	Working	(mod_conference)
In-Reply-To: <4BB75B73.7040501@sns.eu>
References: <4BB75B73.7040501@sns.eu>
Message-ID: <20100403163305.GE5746@base.carmickle.com>

On Sat, Apr 03, Jan Riedinger wrote:
> If I dial in to a conference from my Cisco IP Phone 7940 all is working 
> as it should.
> 
> However, if I dial in via a voip carrier, the calls are disconnect with 
> disconnect cause "facility rejected" by the voip carrier. It seems that 
> Freeswitch tries to change some parameters of the call setup, which is 
> refused. You find below the console log output. Do you have an idea, 
> what I have to change to get it working?

Make sure you answer the channel first.  But with out full logging we don't know what's going on.  Do as BKW says.

--FC


From riedinger at sns.eu  Sat Apr  3 10:00:34 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Sat, 03 Apr 2010 19:00:34 +0200
Subject: [Freeswitch-users] Conference Dial In Isn't
	Working	(mod_conference)
In-Reply-To: <E18056B6-90B7-4D7D-AB9C-1B979F7042C6@freeswitch.org>
References: <4BB75B73.7040501@sns.eu>
	<E18056B6-90B7-4D7D-AB9C-1B979F7042C6@freeswitch.org>
Message-ID: <4BB77432.8080508@sns.eu>

Here is the missing debug information.

Thanks
    Jan

freeswitch at ...> recv 839 bytes from udp/[XXX.XX.XX.X6]:5060 at 
16:48:12.118650:
   ------------------------------------------------------------------------
   INVITE sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone SIP/2.0
   Via: SIP/2.0/UDP 
XXX.XX.XX.X6:5060;branch=z9hG4bK-34ff11004c71ff10ff000024ff0c6dff
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 1 INVITE
   Contact: <sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>
   Max-Forwards: 10
   User-Agent: MERA MSIP v.1.0.2
   Cisco-Guid: 2962728880-1061163487-2855600256-2188435832
   Content-Type: application/sdp
   Content-Length:   264

   v=0
   o=- 1270313292 1270313292 IN IP4 XXX.XX.XX.X6
   s=-
   c=IN IP4 XXX.XX.XX.X6
   t=0 0
   m=audio 22284 RTP/AVP 18 4 8 0 101
   a=rtpmap:18 G729/8000
   a=rtpmap:4 G723/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   ------------------------------------------------------------------------
send 392 bytes to udp/[XXX.XX.XX.X6]:5060 at 16:48:12.118936:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 
XXX.XX.XX.X6:5060;branch=z9hG4bK-34ff11004c71ff10ff000024ff0c6dff
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.git-exportiert
   Content-Length: 0

   ------------------------------------------------------------------------
2010-04-03 18:48:12.116629 [NOTICE] switch_channel.c:669 New Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 
[d1242848-5588-4083-acea-062fd74e5f66]
2010-04-03 18:48:12.116629 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 entering state [received][100]
2010-04-03 18:48:12.116629 [DEBUG] sofia.c:4164 Remote SDP:
v=0
o=- 1270313292 1270313292 IN IP4 XXX.XX.XX.X6
s=-
c=IN IP4 XXX.XX.XX.X6
t=0 0
m=audio 22284 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[G7221:115:32000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[G7221:107:16000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[G722:9:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[PCMU:0:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[PCMA:8:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_NEW
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G729:18:8000:20]/[GSM:3:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[G7221:115:32000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[G7221:107:16000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[G722:9:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[PCMU:0:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[PCMA:8:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[G723:4:8000:20]/[GSM:3:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[G7221:115:32000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[G7221:107:16000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[G722:9:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMU:0:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3585 Audio Codec Compare 
[PCMA:8:8000:20]/[PCMA:8:8000:20]
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:2354 Set Codec 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 PCMA/8000 20 ms 160 samples
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf 
send/recv payload to 101
2010-04-03 18:48:12.119630 [DEBUG] sofia.c:4310 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_NEW -> CS_INIT
2010-04-03 18:48:12.119630 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:320 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State NEW
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_INIT
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:338 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State INIT
2010-04-03 18:48:12.119630 [DEBUG] mod_sofia.c:83 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SOFIA INIT
2010-04-03 18:48:12.119630 [DEBUG] mod_sofia.c:117 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_INIT -> CS_ROUTING
2010-04-03 18:48:12.119630 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:338 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State INIT going to sleep
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_ROUTING
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:341 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State ROUTING
2010-04-03 18:48:12.119630 [DEBUG] mod_sofia.c:140 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SOFIA ROUTING
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:77 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard ROUTING
2010-04-03 18:48:12.119630 [INFO] mod_dialplan_xml.c:418 Processing 
49XXXXXXXX6->49331YYYYYY in context public
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->unloop] continue=false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Regex (PASS) [unloop] 
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL) [unloop] 
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->outside_call] continue=true
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Absolute Condition 
[outside_call]
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Action 
set(outside_call=true)
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->call_debug] continue=true
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Regex (FAIL) 
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
...
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 parsing 
[public->sns_conference] continue=false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Regex (PASS) 
[sns_conference] destination_number(49331YYYYYY) =~ /^(49331YYYYYY)$/ 
break=on-false
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Action answer()
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Action 
set(conference_enforce_security=false)
Dialplan: sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Action 
conference(49331YYYYYY-${domain_name}@default)
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:119 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_ROUTING -> 
CS_EXECUTE
2010-04-03 18:48:12.119630 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:341 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State ROUTING going to sleep
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_EXECUTE
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:348 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State EXECUTE
2010-04-03 18:48:12.119630 [DEBUG] mod_sofia.c:226 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SOFIA EXECUTE
2010-04-03 18:48:12.119630 [DEBUG] switch_core_state_machine.c:157 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard EXECUTE
EXECUTE sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 set(outside_call=true)
2010-04-03 18:48:12.119630 [DEBUG] mod_dptools.c:816 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SET [outside_call]=[true]
EXECUTE sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 answer()
2010-04-03 18:48:12.119630 [DEBUG] sofia_glue.c:2594 AUDIO RTP 
[sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6] XXX.XX.XX.X7 port 21986 -> 
XXX.XX.XX.X6 port 22284 codec: 8 ms: 20
2010-04-03 18:48:12.119630 [DEBUG] switch_rtp.c:1182 Starting timer 
[soft] 160 bytes per 20ms
2010-04-03 18:48:12.122631 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf send 
payload to 101
2010-04-03 18:48:12.122631 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf 
receive payload to 101
2010-04-03 18:48:12.122631 [DEBUG] mod_sofia.c:636 Local SDP 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6:
v=0
o=FreeSWITCH 1270291306 1270291307 IN IP4 XXX.XX.XX.X7
s=FreeSWITCH
c=IN IP4 XXX.XX.XX.X7
t=0 0
m=audio 21986 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2010-04-03 18:48:12.122631 [DEBUG] switch_core_session.c:638 Send signal 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.122631 [NOTICE] mod_dptools.c:719 Channel 
[sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6] has been answered
send 1068 bytes to udp/[XXX.XX.XX.X6]:5060 at 16:48:12.124920:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 
XXX.XX.XX.X6:5060;branch=z9hG4bK-34ff11004c71ff10ff000024ff0c6dff
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>;tag=Uv2Z4H5vNFyej
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 1 INVITE
   Contact: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.git-exportiert
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 247
   Remote-Party-ID: "49331YYYYYY" 
<sip:49331YYYYYY at XXX.XX.XX.X7>;party=calling;privacy=off;screen=no

   v=0
EXECUTE sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 
set(conference_enforce_security=false)
   o=FreeSWITCH 1270291306 1270291307 IN IP4 XXX.XX.XX.X7
   s=FreeSWITCH
   c=IN IP4 XXX.XX.XX.X7
   t=0 0
   m=audio 21986 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   ------------------------------------------------------------------------
2010-04-03 18:48:12.122631 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 entering state [completed][200]
2010-04-03 18:48:12.122631 [DEBUG] mod_dptools.c:816 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SET 
[conference_enforce_security]=[false]
EXECUTE sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 
conference(49331YYYYYY-XXX.XX.XX.X7 at default)
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'mute' bound to '0'.
2010-04-03 18:48:12.122631 [DEBUG] switch_ivr.c:1616 max len 1
2010-04-03 18:48:12.122631 [DEBUG] switch_ivr.c:1620 min len 1
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'deaf mute' bound to '*'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'energy up' bound to '9'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'energy equ' bound to '8'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'energy dn' bound to '7'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol talk up' bound to '3'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol talk zero' bound to '2'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol talk dn' bound to '1'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol listen up' bound to '6'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol listen zero' bound to '5'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'vol listen dn' bound to '4'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5671 Installing 
default caller control action 'hangup' bound to '#'.
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:4990 Raw Codec 
Activation Success L16 at 8000hz 1 channel 20ms
2010-04-03 18:48:12.122631 [DEBUG] mod_conference.c:5035 Raw Codec 
Activation Success L16 at 8000hz 1 channel 20ms
2010-04-03 18:48:12.125638 [DEBUG] switch_core_codec.c:122 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Push codec L16:10
2010-04-03 18:48:12.125638 [DEBUG] switch_core_session.c:638 Send signal 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.125638 [DEBUG] mod_conference.c:2202 Setup timer 
soft success interval: 20  samples: 160
2010-04-03 18:48:12.125638 [DEBUG] mod_conference.c:994 Setup timer 
success interval: 20  samples: 160
recv 445 bytes from udp/[XXX.XX.XX.X6]:5060 at 16:48:12.144223:
   ------------------------------------------------------------------------
   ACK sip:49331YYYYYY at XXX.XX.XX.X7:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 
XXX.XX.XX.X6:5060;branch=z9hG4bK-34ff11004c71ff10ff000024ff0c6dff
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>;tag=Uv2Z4H5vNFyej
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 1 ACK
   Max-Forwards: 10
   User-Agent: MERA MSIP v.1.0.2
   Content-Length: 0

   ------------------------------------------------------------------------
2010-04-03 18:48:12.143633 [DEBUG] sofia.c:4153 Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 entering state [ready][200]
recv 446 bytes from udp/[XXX.XX.XX.X6]:5060 at 16:48:12.311539:
   ------------------------------------------------------------------------
   BYE sip:49331YYYYYY at XXX.XX.XX.X7:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP XXX.XX.XX.X6:5060
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>;tag=Uv2Z4H5vNFyej
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 2 BYE
   Max-Forwards: 10
   User-Agent: MERA MSIP v.1.0.2
   Reason: Q.850;cause=29;text="Facility rejected"
   Content-Length: 0

   ------------------------------------------------------------------------
2010-04-03 18:48:12.308666 [NOTICE] sofia.c:481 Hangup 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [CS_EXECUTE] [FACILITY_REJECTED]
2010-04-03 18:48:12.308666 [DEBUG] switch_channel.c:2071 Send signal 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [KILL]
2010-04-03 18:48:12.308666 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
send 493 bytes to udp/[XXX.XX.XX.X6]:5060 at 16:48:12.311906:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP XXX.XX.XX.X6:5060
   From: 
<sip:49XXXXXXXX6 at XXX.XX.XX.X6;user=phone>;tag=44ff11004c71ff10ff000024ff0c6dff
   To: <sip:49331YYYYYY at XXX.XX.XX.X7:5060;user=phone>;tag=Uv2Z4H5vNFyej
   Call-ID: 9aef11004c71b710800000248c0c6d9e at DOMAIN_ORIG
   CSeq: 2 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.git-exportiert
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
2010-04-03 18:48:12.326660 [DEBUG] mod_conference.c:2473 Channel leaving 
conference, cause: FACILITY_REJECTED
2010-04-03 18:48:12.329664 [DEBUG] switch_core_codec.c:146 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Restore previous codec PCMA:8.
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:348 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State EXECUTE going to sleep
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_HANGUP
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:499 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State HANGUP
2010-04-03 18:48:12.329664 [DEBUG] mod_sofia.c:408 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Overriding SIP cause 501 with 
200 from the other leg
2010-04-03 18:48:12.329664 [DEBUG] mod_sofia.c:414 Channel 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 hanging up, cause: FACILITY_REJECTED
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:46 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard HANGUP, cause: 
FACILITY_REJECTED
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:499 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State HANGUP going to sleep
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:333 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_HANGUP -> 
CS_REPORTING
2010-04-03 18:48:12.329664 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:314 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_REPORTING
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:590 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State REPORTING
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:53 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard REPORTING, cause: 
FACILITY_REJECTED
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:590 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State REPORTING going to sleep
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:327 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State Change CS_REPORTING -> 
CS_DESTROY
2010-04-03 18:48:12.329664 [DEBUG] switch_core_session.c:1018 Send 
signal sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [BREAK]
2010-04-03 18:48:12.329664 [DEBUG] switch_core_session.c:1161 Session 8 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Locked, Waiting on external 
entities
2010-04-03 18:48:12.329664 [NOTICE] switch_core_session.c:1179 Session 8 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Ended
2010-04-03 18:48:12.329664 [NOTICE] switch_core_session.c:1181 Close 
Channel sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 [CS_DESTROY]
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:428 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) Running State Change CS_DESTROY
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:439 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State DESTROY
2010-04-03 18:48:12.329664 [DEBUG] mod_sofia.c:341 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 SOFIA DESTROY
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:60 
sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6 Standard DESTROY
2010-04-03 18:48:12.329664 [DEBUG] switch_core_state_machine.c:439 
(sofia/external/49XXXXXXXX6 at XXX.XX.XX.X6) State DESTROY going to sleep
2010-04-03 18:48:12.347672 [DEBUG] mod_conference.c:1361 Write Lock ON
2010-04-03 18:48:12.347672 [DEBUG] mod_conference.c:1364 Write Lock OFF

Brian West schrieb:
> sofia profile xxxxx siptrace on (replace xxx with the profile)
>
> Then try again.
>
> /b
>
> On Apr 3, 2010, at 10:14 AM, Jan Riedinger wrote:
>
>   
>> However, if I dial in via a voip carrier, the calls are disconnect with 
>> disconnect cause "facility rejected" by the voip carrier. It seems that 
>> Freeswitch tries to change some parameters of the call setup, which is 
>> refused. You find below the console log output. Do you have an idea, 
>> what I have to change to get it working?
>>     
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>   

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                        E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From brian at freeswitch.org  Sat Apr  3 10:05:27 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 12:05:27 -0500
Subject: [Freeswitch-users] Conference Dial In Isn't Working
	(mod_conference)
In-Reply-To: <4BB77432.8080508@sns.eu>
References: <4BB75B73.7040501@sns.eu>
	<E18056B6-90B7-4D7D-AB9C-1B979F7042C6@freeswitch.org>
	<4BB77432.8080508@sns.eu>
Message-ID: <FA3A06CA-92D8-4BEC-9AE9-7E881D45E3E3@freeswitch.org>

  Reason: Q.850;cause=29;text="Facility rejected"

Contact your provider their MERA switch sent you a BYE.

/b

On Apr 3, 2010, at 12:00 PM, Jan Riedinger wrote:

> Here is the missing debug information.
> 
> Thanks
>    Jan



From larclap at yahoo.com  Sat Apr  3 11:22:50 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 3 Apr 2010 11:22:50 -0700
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>
	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>
Message-ID: <00b901cad35a$ab6857e0$023907a0$@com>

Sorry to be a nuisance with this. No, I do not use openzap.

 

I tried to execute 'make oz-reconf' in an attempt to get around the error,
but no go.

 

[root at fs freeswitch]# make oz-reconf

cd libs/openzap && make clean

make[1]: Entering directory `/usr/src/freeswitch/libs/openzap'

make[1]: *** No rule to make target `clean'.  Stop.

make[1]: Leaving directory `/usr/src/freeswitch/libs/openzap'

make: *** [oz-reconf] Error 2

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 9:16 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

if you actually want to use openzap after you delete it and svn up, issue

make oz-reconf

then make as usual.



On Sat, Apr 3, 2010 at 9:56 AM, Lars Zeb <larclap at yahoo.com> wrote:

I deleted the openzap directory. I ran 'svn up' to make sure I was current.
Then ran 'make current' and got:

 

making uninstall mod_vmd

 

making uninstall mod_xml_curl

 

making uninstall mod_xml_ldap

 

WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....

 

making uninstall mod_yaml

cd libs/openzap && autoconf

/bin/sh: line 0: cd: libs/openzap: No such file or directory

make: *** [libs/openzap/Makefile] Error 1

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 8:20 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

yes, it was merged from external to internal.
delete it and update again.

On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:

The current version of FS I'm using is 17135. I tried to 'make current' and
it errored out in the svn up portion:

 

svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a
different repository

 

Should I just delete the libs/openzap directory?

 

openzap is commented out in modules.conf.xml.

 

Thanks, Lars


_______________________________________________
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
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


_______________________________________________
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
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From mrene_lists at avgs.ca  Sat Apr  3 11:25:44 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 3 Apr 2010 14:25:44 -0400
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <00b901cad35a$ab6857e0$023907a0$@com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>
	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>
	<00b901cad35a$ab6857e0$023907a0$@com>
Message-ID: <D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>

Re-run ./configure from the top level directory

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-03, at 2:22 PM, Lars Zeb wrote:

> Sorry to be a nuisance with this. No, I do not use openzap.
>
> I tried to execute ?make oz-reconf? in an attempt to get around the  
> error, but no go.
>
> [root at fs freeswitch]# make oz-reconf
> cd libs/openzap && make clean
> make[1]: Entering directory `/usr/src/freeswitch/libs/openzap'
> make[1]: *** No rule to make target `clean'.  Stop.
> make[1]: Leaving directory `/usr/src/freeswitch/libs/openzap'
> make: *** [oz-reconf] Error 2
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Anthony Minessale
> Sent: Saturday, April 03, 2010 9:16 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Error in 'svn up'
>
> if you actually want to use openzap after you delete it and svn up,  
> issue
>
> make oz-reconf
>
> then make as usual.
>
>
> On Sat, Apr 3, 2010 at 9:56 AM, Lars Zeb <larclap at yahoo.com> wrote:
> I deleted the openzap directory. I ran ?svn up? to make sure I was  
> current. Then ran ?make current? and got:
>
> making uninstall mod_vmd
>
> making uninstall mod_xml_curl
>
> making uninstall mod_xml_ldap
>
> WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping  
> it.....
>
> making uninstall mod_yaml
> cd libs/openzap && autoconf
> /bin/sh: line 0: cd: libs/openzap: No such file or directory
> make: *** [libs/openzap/Makefile] Error 1
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Anthony Minessale
> Sent: Saturday, April 03, 2010 8:20 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Error in 'svn up'
>
> yes, it was merged from external to internal.
> delete it and update again.
>
> On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com>  
> wrote:
> The current version of FS I?m using is 17135. I tried to ?make  
> current? and it errored out in the svn up portion:
>
> svn: UUID mismatch: existing directory 'libs/openzap' was checked  
> out from a different repository
>
> Should I just delete the libs/openzap directory?
>
> openzap is commented out in modules.conf.xml.
>
> Thanks, Lars
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> 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
> http://www.freeswitch.org

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From jim at k4gvo.com  Sat Apr  3 12:17:44 2010
From: jim at k4gvo.com (Jim)
Date: Sat, 03 Apr 2010 15:17:44 -0400
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
Message-ID: <4BB79458.8080601@k4gvo.com>

I obviously need to set a <variable name="toll_allow" 
value="domestic,international,local"/> somewhere but I can't figure out 
what file to put it in.  The examples all show it in the 
directory/default/xxxx.xml files but those appear to be sip only.  In 
any event creating files in that directory for my extension did nothing 
to help the problem.

The only places I have the extension mentioned is in  the openzap.conf 
file and dialplan/default/00_incoming-1.xml.  Adding a <variables ... 
element to the latter does nothing.

How do I set that variable?  Or where?

Thanks,
Jim.


From gavin.henry at gmail.com  Sat Apr  3 12:30:05 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 3 Apr 2010 20:30:05 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
Message-ID: <r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>

On 3 April 2010 05:18, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Nothing is wrong with it,? there is no revision that broke it or there would
> be 100,000 people saying it was broken.

Unless I was the first that hit a weird configuration. That's the
nature of building software yourself
and these types of e-mails should be encouraged, not frowned upon.

> Change you perspective to looking for the problem on your end and you will
> have more luck finding your problem.

Of course, I know. But sometimes many eyes help and it's easier to ask
a quick question.

> Backup your config by moving it out of the way, re-install and try the
> defaults.

Cheers.

> We use FS with nat like this 12 hours a day.

Good to know.

Thanks all!

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From anthony.minessale at gmail.com  Sat Apr  3 12:51:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Apr 2010 13:51:49 -0600
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
Message-ID: <j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>

I didn't frown upon you I am being serious and straightforward with you.
I am giving you advice I have learned the hard way from personal experience
on troubleshooting.
The facts and pointers lead to a quicker solution and eliminate variables.


On Sat, Apr 3, 2010 at 1:30 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 3 April 2010 05:18, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> > Nothing is wrong with it,  there is no revision that broke it or there
> would
> > be 100,000 people saying it was broken.
>
> Unless I was the first that hit a weird configuration. That's the
> nature of building software yourself
> and these types of e-mails should be encouraged, not frowned upon.
>
> > Change you perspective to looking for the problem on your end and you
> will
> > have more luck finding your problem.
>
> Of course, I know. But sometimes many eyes help and it's easier to ask
> a quick question.
>
> > Backup your config by moving it out of the way, re-install and try the
> > defaults.
>
> Cheers.
>
> > We use FS with nat like this 12 hours a day.
>
> Good to know.
>
> Thanks all!
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.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
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From larclap at yahoo.com  Sat Apr  3 13:09:29 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 3 Apr 2010 13:09:29 -0700
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>	<00b901cad35a$ab6857e0$023907a0$@com>
	<D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>
Message-ID: <00e501cad369$91318950$b3949bf0$@com>

 

configure: creating ./config.status

config.status: creating Makefile

config.status: creating doc/Makefile

config.status: creating test-data/Makefile

config.status: creating test-data/local/Makefile

config.status: creating test-data/itu/Makefile

config.status: creating src/Makefile

config.status: creating src/g722_1.h

config.status: creating tests/Makefile

config.status: creating g722_1.spec

config.status: creating tests/regression_tests.sh

config.status: creating src/config.h

config.status: src/config.h is unchanged

config.status: executing depfiles commands

configure: configuring in libs/silk

configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch
CONFIGURE_CFLAGS='-g -O2' CONFIGURE_CXXFLAGS='-g -O2' CONFIGURE_LDFLAGS=''
--cache-file=/dev/null --srcdir=.

./configure.gnu: line 3: ./configure: No such file or directory

configure: error: /bin/sh './configure.gnu' failed for libs/silk

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu
Rene
Sent: Saturday, April 03, 2010 11:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

Re-run ./configure from the top level directory

 

Mathieu Rene

Avant-Garde Solutions Inc

Office: + 1 (514) 664-1044 x100

Cell: +1 (514) 664-1044 x200

mrene at avgs.ca

 

 

 

 

On 2010-04-03, at 2:22 PM, Lars Zeb wrote:





Sorry to be a nuisance with this. No, I do not use openzap.

 

I tried to execute 'make oz-reconf' in an attempt to get around the error,
but no go.

 

[root at fs freeswitch]# make oz-reconf

cd libs/openzap && make clean

make[1]: Entering directory `/usr/src/freeswitch/libs/openzap'

make[1]: *** No rule to make target `clean'.  Stop.

make[1]: Leaving directory `/usr/src/freeswitch/libs/openzap'

make: *** [oz-reconf] Error 2

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 9:16 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

if you actually want to use openzap after you delete it and svn up, issue

make oz-reconf

then make as usual.




On Sat, Apr 3, 2010 at 9:56 AM, Lars Zeb <larclap at yahoo.com> wrote:

I deleted the openzap directory. I ran 'svn up' to make sure I was current.
Then ran 'make current' and got:

 

making uninstall mod_vmd

 

making uninstall mod_xml_curl

 

making uninstall mod_xml_ldap

 

WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....

 

making uninstall mod_yaml

cd libs/openzap && autoconf

/bin/sh: line 0: cd: libs/openzap: No such file or directory

make: *** [libs/openzap/Makefile] Error 1

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Saturday, April 03, 2010 8:20 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

yes, it was merged from external to internal.
delete it and update again.

On Fri, Apr 2, 2010 at 1:51 PM, Larry Marshall <lcm at marshap.com> wrote:

The current version of FS I'm using is 17135. I tried to 'make current' and
it errored out in the svn up portion:

 

svn: UUID mismatch: existing directory 'libs/openzap' was checked out from a
different repository

 

Should I just delete the libs/openzap directory?

 

openzap is commented out in modules.conf.xml.

 

Thanks, Lars


_______________________________________________
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
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


_______________________________________________
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
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

_______________________________________________
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
http://www.freeswitch.org

 

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From gavin.henry at gmail.com  Sat Apr  3 13:34:20 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 3 Apr 2010 21:34:20 +0100
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
Message-ID: <x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>

On 3 April 2010 20:51, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> I didn't frown upon you I am being serious and straightforward with you.
> I am giving you advice I have learned the hard way from personal experience
> on troubleshooting.
> The facts and pointers lead to a quicker solution and eliminate variables.

I know Anthony, and appreciate it! Just mentioning in case others get
scared of posting
things if they get shouted at :-) We have this problem in other projects.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From brian at freeswitch.org  Sat Apr  3 13:43:33 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 15:43:33 -0500
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <00e501cad369$91318950$b3949bf0$@com>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>	<00b901cad35a$ab6857e0$023907a0$@com>
	<D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>
	<00e501cad369$91318950$b3949bf0$@com>
Message-ID: <23C60E84-79D9-4EB7-9242-B1BDAEB5CA93@freeswitch.org>

You have to rebootstrap after you update... I missed the part to do libs/silk in configure.in

/b

On Apr 3, 2010, at 3:09 PM, Lars Zeb wrote:

>  
> configure: creating ./config.status
> config.status: creating Makefile
> config.status: creating doc/Makefile
> config.status: creating test-data/Makefile
> config.status: creating test-data/local/Makefile
> config.status: creating test-data/itu/Makefile
> config.status: creating src/Makefile
> config.status: creating src/g722_1.h
> config.status: creating tests/Makefile
> config.status: creating g722_1.spec
> config.status: creating tests/regression_tests.sh
> config.status: creating src/config.h
> config.status: src/config.h is unchanged
> config.status: executing depfiles commands
> configure: configuring in libs/silk
> configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch  CONFIGURE_CFLAGS='-g -O2' CONFIGURE_CXXFLAGS='-g -O2' CONFIGURE_LDFLAGS='' --cache-file=/dev/null --srcdir=.
> ./configure.gnu: line 3: ./configure: No such file or directory
> configure: error: /bin/sh './configure.gnu' failed for libs/silk
>  

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From larclap at yahoo.com  Sat Apr  3 15:08:47 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 3 Apr 2010 15:08:47 -0700
Subject: [Freeswitch-users] Error in 'svn up'
In-Reply-To: <23C60E84-79D9-4EB7-9242-B1BDAEB5CA93@freeswitch.org>
References: <00fb01cad29d$eb8c4f60$c2a4ee20$@com>	<r2v191c3a031004030820j7ed30f0t17dc5eb6fccfc806@mail.gmail.com>	<008801cad346$3511a240$9f34e6c0$@com>	<w2r191c3a031004030916qd88824chaacdd9959fa90208@mail.gmail.com>	<00b901cad35a$ab6857e0$023907a0$@com>	<D31A2180-03A1-4010-9630-E5B4B6F9F822@avgs.ca>	<00e501cad369$91318950$b3949bf0$@com>
	<23C60E84-79D9-4EB7-9242-B1BDAEB5CA93@freeswitch.org>
Message-ID: <011301cad37a$3d9fbcb0$b8df3610$@com>

That did it, thanks.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Saturday, April 03, 2010 1:44 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error in 'svn up'

 

You have to rebootstrap after you update... I missed the part to do
libs/silk in configure.in

 

/b

 

On Apr 3, 2010, at 3:09 PM, Lars Zeb wrote:





 

configure: creating ./config.status

config.status: creating Makefile

config.status: creating doc/Makefile

config.status: creating test-data/Makefile

config.status: creating test-data/local/Makefile

config.status: creating test-data/itu/Makefile

config.status: creating src/Makefile

config.status: creating src/g722_1.h

config.status: creating tests/Makefile

config.status: creating g722_1.spec

config.status: creating tests/regression_tests.sh

config.status: creating src/config.h

config.status: src/config.h is unchanged

config.status: executing depfiles commands

configure: configuring in libs/silk

configure: running /bin/sh './configure.gnu' --prefix=/usr/local/freeswitch
CONFIGURE_CFLAGS='-g -O2' CONFIGURE_CXXFLAGS='-g -O2' CONFIGURE_LDFLAGS=''
--cache-file=/dev/null --srcdir=.

./configure.gnu: line 3: ./configure: No such file or directory

configure: error: /bin/sh './configure.gnu' failed for libs/silk

 

 

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From jayesh.voip at gmail.com  Sat Apr  3 20:47:58 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Sun, 4 Apr 2010 09:17:58 +0530
Subject: [Freeswitch-users]  domain-wise context
Message-ID: <p2j4fe8b8271004032047md9a17614z65db0b4f8de52bcd@mail.gmail.com>

Hi,
Sorry for sounding so impatient, the anxiety only grew because before
posting it to the list I spent a week on reading all the documentation
available on and around this topic FS site and mailing lists. I really
appreciate and am thankful for your suggestions.
I'll try out the suggestions given by you. Is there a way that we can
compare the domain name in the dialplan using regular expressions. Is there
a value in condition tag that can be used to compare this, something like
"extension_number".
Or should I store the domain value in some variable(something like
sip_h_<domain_name>) and compare that variable to take the call to a
different context?

Thanks,

--- Jayesh

---------- Forwarded message ----------

> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 3 Apr 2010 09:11:50 -0600
> Subject: Re: [Freeswitch-users] domain-wise context
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_execute_extension
>
> The first one transfers the call to the desired exten/dialplan/context and
> the 2nd one executes the specified extension in a similar manner and returns
> to the same point in the dp.
>
> You make one inbound context and use routing logic from there to decide
> which context to transfer to.
>
> Next time, please have a little more patience, I don't like it when people
> reply to themselves on the list asking why nobody answered when their
> question is only unanswered for 2 days especially during a holiday weekend.
>
>
>
> On Sat, Apr 3, 2010 at 1:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> I expected at east one reply saying that the question is stupid, and the
>> solution is simple !!
>> Any folks who can help me understand only how to achieve this in FS which
>> is acheivable in asterisk as follows:
>> domain = mydomain.com, mydomain (If any call has domain as "mydomain.com",
>> the call goes to context "mydomain" in dialplan)
>> domain = yourdomain.com, yourdomain (if any call has domain as "
>> yourdomain.com", the call goes to context "yourdomain" in dialplan)
>>
>> These calls can come from anywhere, in my case it comes from an Opensips
>> instance !!
>>
>> Thanks for any replies :)
>>
>> --- Jayesh
>>
>>
>> On Thu, Apr 1, 2010 at 1:35 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>>
>>> Hi All,
>>> I am quite very new to freeswitch and I am kind of playing with it to
>>> understand it better.
>>> I am primarily using Opensips as registrar and SIP Proxy and intend to
>>> use FS as media server handling voicemails, IVR, Announcements, MeetMe etc.
>>> My Opensips is a multi-domain setup and I wish to have all the configuration
>>> of media-capabilties segregated domain-wise in the FS too.
>>>
>>> For eg: When a call for user at domain1.com needs to go to voicemail, I
>>> redirect that call to FS IP address keeping the URI intact. I add the
>>> mailbox number as a header as X-Mailbox and have FS extract it and go to
>>> appropriate mailbox. Similarly when a call for user at domain2.com needs to
>>> go to voicemail I do the same thing.
>>> The requirement is I want to maintain the dialplans for each domains
>>> separately. Thus if call from Opensips comes to FS with domain as domain1,
>>> the call should go to dialplan context domain1 and similarly if call from
>>> Opensips comes to FS with domain2 the call handling should be mentioned in
>>> the domain2 context.
>>>
>>> The problem is; I am not able to send the calls to respective contexts
>>> according to their domains when they come from Opensips. I've read the
>>> examples on multi-domain setup and have tried taking some help from that
>>> example, but whenever the call comes from Opensips to FS, it tries to go
>>> into the context that is defined in the SIP Profile. If i don't mention
>>> anything in the SIP Profile, it tries to search for default context.
>>> I have tried the following:
>>> 1) Created file domain1.xml and domain2.xml in the directory folder.
>>> 2) mentioned parameters in domain1.xml as follows:
>>> <include>
>>> <domain name = "domain1.com"
>>> <params>
>>> <param name="user_context" value "domain1.com"/>
>>> </params>
>>> </domain>
>>> </include>
>>> 3) Similarly done for file domain2.xml.
>>>
>>> But I am just not able to get the calls to the required context according
>>> to the domain value in the r-uri. In asterisk something like this in
>>> sip.conf worked fine for me:
>>> domain=domain1.com, domain1.com
>>> domain=domain2.com, domain2.com
>>>  Can someone please help me understanding where I am going wrong or have
>>> I mis-understood something?
>>>
>>> Thanks in advance !!
>>>
>>> --- Jayesh
>>>
>>
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From brian at freeswitch.org  Sat Apr  3 20:57:19 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 22:57:19 -0500
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <p2j4fe8b8271004032047md9a17614z65db0b4f8de52bcd@mail.gmail.com>
References: <p2j4fe8b8271004032047md9a17614z65db0b4f8de52bcd@mail.gmail.com>
Message-ID: <1A7D6A7C-B8A5-46B5-AFB1-C94ED55C8F30@freeswitch.org>

see user_context variable on the user.

/b

On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote:

> Hi,
> Sorry for sounding so impatient, the anxiety only grew because before posting it to the list I spent a week on reading all the documentation available on and around this topic FS site and mailing lists. I really appreciate and am thankful for your suggestions.
> I'll try out the suggestions given by you. Is there a way that we can compare the domain name in the dialplan using regular expressions. Is there a value in condition tag that can be used to compare this, something like "extension_number".
> Or should I store the domain value in some variable(something like sip_h_<domain_name>) and compare that variable to take the call to a different context?
> 
> Thanks,
> 
> --- Jayesh



From mayamatakeshi at gmail.com  Sat Apr  3 21:31:57 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sun, 4 Apr 2010 13:31:57 +0900
Subject: [Freeswitch-users] Contrib modules
Message-ID: <g2v15b9404e1004032131xf78c4d66qc142766c9c5198ad@mail.gmail.com>

Hi,
I have updated (svn) to trunk (r. 17188) and now I don't see the contrib
folder anymore (I was using mod_xml_odbc from there).
What happened?

regards,
takeshi
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From brian at freeswitch.org  Sat Apr  3 21:37:50 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 3 Apr 2010 23:37:50 -0500
Subject: [Freeswitch-users] Contrib modules
In-Reply-To: <g2v15b9404e1004032131xf78c4d66qc142766c9c5198ad@mail.gmail.com>
References: <g2v15b9404e1004032131xf78c4d66qc142766c9c5198ad@mail.gmail.com>
Message-ID: <21FDFE0D-4743-4B01-9F75-3C898FEBB15C@freeswitch.org>

They are being split out into a repo.  That i'm not sure is complete yet.

/b

On Apr 3, 2010, at 11:31 PM, mayamatakeshi wrote:

> Hi,
> I have updated (svn) to trunk (r. 17188) and now I don't see the contrib folder anymore (I was using mod_xml_odbc from there).
> What happened?
> 
> regards,
> takeshi



From msc at freeswitch.org  Sat Apr  3 23:29:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 3 Apr 2010 23:29:23 -0700
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
In-Reply-To: <4BB79458.8080601@k4gvo.com>
References: <4BB79458.8080601@k4gvo.com>
Message-ID: <s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>

Variables set in the directory/default/xxxx.xml files apply to users who
make authenticated calls through FS. Generally those will be SIP phones.

Let's back up a step. What problem are you trying to solve, i.e., why is it
that you need to set the toll_allow variable? What endpoint is making an
openzap call?

-MC

On Sat, Apr 3, 2010 at 12:17 PM, Jim <jim at k4gvo.com> wrote:

> I obviously need to set a <variable name="toll_allow"
> value="domestic,international,local"/> somewhere but I can't figure out
> what file to put it in.  The examples all show it in the
> directory/default/xxxx.xml files but those appear to be sip only.  In
> any event creating files in that directory for my extension did nothing
> to help the problem.
>
> The only places I have the extension mentioned is in  the openzap.conf
> file and dialplan/default/00_incoming-1.xml.  Adding a <variables ...
> element to the latter does nothing.
>
> How do I set that variable?  Or where?
>
> Thanks,
> Jim.
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From 12ukwn at gmail.com  Sun Apr  4 00:08:38 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sun, 4 Apr 2010 09:08:38 +0200
Subject: [Freeswitch-users] cepstral PB
Message-ID: <20100404090838.414c58ce@anubis.defcon1>

FS svn-17188M
Debian Lenny
=================

Hi list,

I've installed cepstral voices (in /opt/swift, and also changed its
rights recursively to freeswitch:freeswitch) but I had a complain
at FS start that some cepstral libraries were missing.

Eventually, I was obliged to simlink /opt/swift/lib/lib* into 
/usr/local/freeswitch/lib to have it working correctly, is it normal, 
or did I miss something?

(I followed the wiki, so all libs were added to ld.so.cache, path in
mod_cepstral source Makefile are good, SWIFT_HOME is defined, module
flite loading is commented, etc- so I said to myself: Nantidiou!)

-- 
Sorry never means having your say to love.


From jim at k4gvo.com  Sun Apr  4 04:48:57 2010
From: jim at k4gvo.com (Jim)
Date: Sun, 04 Apr 2010 07:48:57 -0400
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
In-Reply-To: <s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>
References: <4BB79458.8080601@k4gvo.com>
	<s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>
Message-ID: <4BB87CA9.3050403@k4gvo.com>

Michael Collins wrote:
> Variables set in the directory/default/xxxx.xml files apply to users 
> who make authenticated calls through FS. Generally those will be SIP 
> phones.
>
> Let's back up a step. What problem are you trying to solve, i.e., why 
> is it that you need to set the toll_allow variable? What endpoint is 
> making an openzap call?
>
> -MC
Hi, Michael,

I have multiple sip phone that are working fine.  When I dial a 10 digit 
number they connect with my sip provider and place the call.  The 
openzap configured phone gets dial tone and can call other extensions, 
however when I dial a 10 digit number it gives me a busy.  In the log I 
see it appears to be failing on the toll_allow test:

Dialplan: OpenZAP/1:1/7707190068 parsing [default->local_call] 
continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [local_call] 
${toll_allow}() =~ /local/ break=on-false
Dialplan: OpenZAP/1:1/7707190068 parsing [default->domestic_call] 
continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [domestic_call] 
${toll_allow}() =~ /domestic/ break=on-false
Dialplan: OpenZAP/1:1/7707190068 parsing 
[default->international.example.com] continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) 
[international.example.com] ${toll_allow}() =~ /international/ 
break=on-false

This is the area when it should be placing the call, I belive.  When 
placing a call from a sip phone I see:

Dialplan: sofia/internal/1003 at 192.168.2.51 parsing [default->local_call] 
continue=false
Dialplan: sofia/internal/1003 at 192.168.2.51 Regex (PASS) [local_call] 
${toll_allow}(domestic,international,local) =~ /local/ break=on-false
Dialplan: sofia/internal/1003 at 192.168.2.51 Regex (PASS) [local_call] 
destination_number(7707190068) =~ /^(\d{10})$/ break=on-false
Dialplan: sofia/internal/1003 at 192.168.2.51 Action 
set(effective_caller_id_number=${outbound_caller_id_number})
Dialplan: sofia/internal/1003 at 192.168.2.51 Action 
set(effective_caller_id_name=${outbound_caller_id_name})
Dialplan: sofia/internal/1003 at 192.168.2.51 Action 
set(continue_on_fail=true)
Dialplan: sofia/internal/1003 at 192.168.2.51 Action 
bridge(sofia/gateway/${default_gateway}/7707190068)

I simply want this extension to be able to dial out.  The configuration 
is 99% default. 

Thanks,
Jim.
>
> On Sat, Apr 3, 2010 at 12:17 PM, Jim <jim at k4gvo.com 
> <mailto:jim at k4gvo.com>> wrote:
>
>     I obviously need to set a <variable name="toll_allow"
>     value="domestic,international,local"/> somewhere but I can't
>     figure out
>     what file to put it in.  The examples all show it in the
>     directory/default/xxxx.xml files but those appear to be sip only.  In
>     any event creating files in that directory for my extension did
>     nothing
>     to help the problem.
>
>     The only places I have the extension mentioned is in  the openzap.conf
>     file and dialplan/default/00_incoming-1.xml.  Adding a <variables ...
>     element to the latter does nothing.
>
>     How do I set that variable?  Or where?
>
>     Thanks,
>     Jim.
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>   



From vfclists at googlemail.com  Sun Apr  4 06:16:34 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sun, 4 Apr 2010 14:16:34 +0100
Subject: [Freeswitch-users] Freeswitch for Windows - command history too
	short?
Message-ID: <n2i7abab2411004040616w5a1eef47j392099111bdcc52c@mail.gmail.com>

I am using svn 17048 for Windows and the command history is too short
and doesn't persist between restarts.

Are there some configuration settings to fix that?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From jeff at jefflenk.com  Sun Apr  4 09:07:49 2010
From: jeff at jefflenk.com (Jeff Lenk )
Date: Sun, 4 Apr 2010 16:07:49 +0000
Subject: [Freeswitch-users] Freeswitch for Windows - command history too
 short?
Message-ID: <COL117-DS119D271516A01C4A3F96FEB91A0@phx.gbl>

No there is no config setting for that yet - feature does not exist yet


-----Original Message-----
From: Frank Church
Sent: 4/4/2010 1:16:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch for Windows - command history too short?
I am using svn 17048 for Windows and the command history is too short
and doesn't persist between restarts.

Are there some configuration settings to fix that?

--
Frank Church

=======================
http://devblog.brahmancreations.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
http://www.freeswitch.org
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From lloyd.aloysius at gmail.com  Sun Apr  4 09:38:58 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 12:38:58 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
Message-ID: <g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>

Hi All,

All Aastra Phones Behind the NAT stop working after update to the most
recent version.

freeswitch at internal> version
FreeSWITCH Version 1.0.head (svn-17188)

Aastra 57i
Aastra 9133i

Also I can confirm the Xlite is working without any problem.

Thanks
Lloyd



On Sat, Apr 3, 2010 at 4:34 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> On 3 April 2010 20:51, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> > I didn't frown upon you I am being serious and straightforward with you.
> > I am giving you advice I have learned the hard way from personal
> experience
> > on troubleshooting.
> > The facts and pointers lead to a quicker solution and eliminate
> variables.
>
> I know Anthony, and appreciate it! Just mentioning in case others get
> scared of posting
> things if they get shouted at :-) We have this problem in other projects.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.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
> http://www.freeswitch.org
>
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From max.clark at gmail.com  Sun Apr  4 11:12:16 2010
From: max.clark at gmail.com (Max Clark)
Date: Sun, 4 Apr 2010 11:12:16 -0700
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <20100402092310.GA18680@jdc.jasonjgw.net>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
Message-ID: <p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>

Clarification - for G729 does freeswitch need to be in "bypass media"
or "proxy media"? My understanding was that G729 would work with
"proxy media" enabled and without the new fangled module?

-Max

On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net> wrote:
> Frank Church <vfclists at googlemail.com> wrote:
>> I am just trialling Freeswitch with Linksys adapters, whose default
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>> When I change that setting to 'yes' the calls don't go through. I am
>> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you really
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
> compelling reason, for example a carrier that only supports G.729.
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>


From lloyd.aloysius at gmail.com  Sun Apr  4 11:13:46 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 14:13:46 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
Message-ID: <t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>

The following version working without any problem.

freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (17155)

------------------------------------------------------ Additional Bug
Informations
----------------------------------------------------------------

Also in freeswitch at internal> version
FreeSWITCH Version 1.0.head (svn-17188)


Here is the Dial plan issues.

Extension 201 - Aastra 57i

Extension 202 - Xlite

Extension 202 -> 201 working great.

2010-04-04 13:54:21.012078 [INFO] mod_dialplan_xml.c:418 Processing 202->201
in context dev.abc.ca


Extension 201 -> 202

*2010-04-04 13:55:08.711996 [INFO] mod_dialplan_xml.c:418 Processing
201->202 in context public*

I do not know why it is looking into the public context.

Thanks
Lloyd



On Sun, Apr 4, 2010 at 12:38 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> All Aastra Phones Behind the NAT stop working after update to the most
> recent version.
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.head (svn-17188)
>
> Aastra 57i
> Aastra 9133i
>
> Also I can confirm the Xlite is working without any problem.
>
> Thanks
> Lloyd
>
>
>
> On Sat, Apr 3, 2010 at 4:34 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
>
>> On 3 April 2010 20:51, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>> > I didn't frown upon you I am being serious and straightforward with you.
>> > I am giving you advice I have learned the hard way from personal
>> experience
>> > on troubleshooting.
>> > The facts and pointers lead to a quicker solution and eliminate
>> variables.
>>
>> I know Anthony, and appreciate it! Just mentioning in case others get
>> scared of posting
>> things if they get shouted at :-) We have this problem in other projects.
>>
>> --
>> http://www.suretecsystems.com/services/openldap/
>> http://www.suretectelecom.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
>> http://www.freeswitch.org
>>
>
>
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From brian at freeswitch.org  Sun Apr  4 11:18:57 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 13:18:57 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<q2yec2ce17e1004020252nad4c3d25t61c7715d7ce19f2c@mail.gmail.com>
	<m2v13ca621c1004021203ida388b3bjeb32acf805a34db@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
Message-ID: <CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>

because its not authenticated.

/b

On Apr 4, 2010, at 1:13 PM, Aloysius Lloyd wrote:

> I do not know why it is looking into the public context.



From brian at freeswitch.org  Sun Apr  4 11:19:17 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 13:19:17 -0500
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
Message-ID: <3ABB861A-D39E-4574-B954-0EAC338323C3@freeswitch.org>

proxy media isn't needed.

/b

On Apr 4, 2010, at 1:12 PM, Max Clark wrote:

> Clarification - for G729 does freeswitch need to be in "bypass media"
> or "proxy media"? My understanding was that G729 would work with
> "proxy media" enabled and without the new fangled module?
> 
> -Max



From david.ponzone at gmail.com  Sun Apr  4 11:35:39 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 4 Apr 2010 20:35:39 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
Message-ID: <D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>

No, FreeSWITCH does NOT need to be in bypass media or proxy media.
You just need the regular passthrough module: mod_g729 and to allow  
G729 as inbound and outbound codecs in vars.xml.

To summarize:
-normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the  
RTP path, it relays the audio stream between endpoints, but can still  
detect DTMFs
-proxy media enabled: FreeSWITCH relays the audio stream  
transparently, DTMF detection is impossible. In this mode, FS is  
really a "dumb" transparent RTP-forwarder (this is required to get T38  
working between the 2 endpoints)
-bypass media enabled: FreeSWITCH is not in the RTP path

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 04/04/2010 ? 20:12, Max Clark a ?crit :

> Clarification - for G729 does freeswitch need to be in "bypass media"
> or "proxy media"? My understanding was that G729 would work with
> "proxy media" enabled and without the new fangled module?
>
> -Max
>
> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
> wrote:
>> Frank Church <vfclists at googlemail.com> wrote:
>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>> When I change that setting to 'yes' the calls don't go through. I am
>>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you really
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> 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
> http://www.freeswitch.org

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From lloyd.aloysius at gmail.com  Sun Apr  4 11:45:36 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 14:45:36 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
Message-ID: <s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>

There is no settings changed.

 Aastra 57i phone is registered to the freeswitch.

Aastra 9133i no more registered to freeswitch.


Is there any major change between 17155 & 17188

Thanks
Lloyd



On Sun, Apr 4, 2010 at 2:18 PM, Brian West <brian at freeswitch.org> wrote:

> because its not authenticated.
>
> /b
>
> On Apr 4, 2010, at 1:13 PM, Aloysius Lloyd wrote:
>
> > I do not know why it is looking into the public context.
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Apr  4 11:49:42 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 13:49:42 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<p2j13ca621c1004021342x199e54e0t31a39d12d2d890cd@mail.gmail.com>
	<y2z13ca621c1004021353n1de1fda6s21c033877c056eaf@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
Message-ID: <79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>

Its an aastra... I don't trust it.

/b

On Apr 4, 2010, at 1:45 PM, Aloysius Lloyd wrote:

> 
> There is no settings changed.
> 
>  Aastra 57i phone is registered to the freeswitch.
> 
> Aastra 9133i no more registered to freeswitch.
> 
> 
> Is there any major change between 17155 & 17188
> 
> Thanks
> Lloyd
> 



From lloyd.aloysius at gmail.com  Sun Apr  4 11:57:29 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 14:57:29 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
Message-ID: <z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>

There is no firmware update on Aastra. Only change in FreeSWITCH.


Snom 320 throwing the following Error.

2010-04-04 14:04:44.588816 [ERR] sofia_glue.c:3380 a=crypto in RTP/AVP,
refer to rfc3711





On Sun, Apr 4, 2010 at 2:49 PM, Brian West <brian at freeswitch.org> wrote:

> Its an aastra... I don't trust it.
>
> /b
>
> On Apr 4, 2010, at 1:45 PM, Aloysius Lloyd wrote:
>
> >
> > There is no settings changed.
> >
> >  Aastra 57i phone is registered to the freeswitch.
> >
> > Aastra 9133i no more registered to freeswitch.
> >
> >
> > Is there any major change between 17155 & 17188
> >
> > Thanks
> > Lloyd
> >
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Apr  4 12:05:03 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 14:05:03 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<w2i191c3a031004022118r8b6f3fa6q1763d8f3a1b6f004@mail.gmail.com>
	<r2w13ca621c1004031230q2dbc4754t79f2a30de8a2958e@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
Message-ID: <32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>

That snom error means you don't have the SRTP set to optional... because a=crypto in an RTP/AVP is invalid.

Again I don't trust the aastra's cuz I have personally seen them fuck up in production with no changes to FreeSWITCH.  Check the invites... mine started sending 0.0.0.0 in the sdp on initial invites.

/b

On Apr 4, 2010, at 1:57 PM, Aloysius Lloyd wrote:

> There is no firmware update on Aastra. Only change in FreeSWITCH.
> 
> 
> Snom 320 throwing the following Error.
> 
> 2010-04-04 14:04:44.588816 [ERR] sofia_glue.c:3380 a=crypto in RTP/AVP, refer to rfc3711



From lloyd.aloysius at gmail.com  Sun Apr  4 12:15:07 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 15:15:07 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<j2z191c3a031004031251i3a249f5cj3dd6c52fee759048@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
Message-ID: <y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>

Thank you for the information.

I have deployed hundreds of Aastra Phones so far no major problems and
complains. There are some firmware issues in earlier versions.

In my scenario FreeSWITCH 17155 working and 17188 is not working.

Thanks
Lloyd

On Sun, Apr 4, 2010 at 3:05 PM, Brian West <brian at freeswitch.org> wrote:

> That snom error means you don't have the SRTP set to optional... because
> a=crypto in an RTP/AVP is invalid.
>
> Again I don't trust the aastra's cuz I have personally seen them fuck up in
> production with no changes to FreeSWITCH.  Check the invites... mine started
> sending 0.0.0.0 in the sdp on initial invites.
>
> /b
>
> On Apr 4, 2010, at 1:57 PM, Aloysius Lloyd wrote:
>
> > There is no firmware update on Aastra. Only change in FreeSWITCH.
> >
> >
> > Snom 320 throwing the following Error.
> >
> > 2010-04-04 14:04:44.588816 [ERR] sofia_glue.c:3380 a=crypto in RTP/AVP,
> refer to rfc3711
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Sun Apr  4 12:19:44 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 15:19:44 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
Message-ID: <l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>

I just notice in the console log.

2010-04-04 15:04:07.403184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by acl
"domains". Falling back to Digest auth.
2010-04-04 15:04:07.566184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by acl
"domains". Falling back to Digest auth.

Snom and Aastra - I have the same behavior.

But Xlite - Not giving the same message and working.

Thanks
Lloyd





On Sun, Apr 4, 2010 at 3:15 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Thank you for the information.
>
> I have deployed hundreds of Aastra Phones so far no major problems and
> complains. There are some firmware issues in earlier versions.
>
> In my scenario FreeSWITCH 17155 working and 17188 is not working.
>
> Thanks
> Lloyd
>
>
> On Sun, Apr 4, 2010 at 3:05 PM, Brian West <brian at freeswitch.org> wrote:
>
>> That snom error means you don't have the SRTP set to optional... because
>> a=crypto in an RTP/AVP is invalid.
>>
>> Again I don't trust the aastra's cuz I have personally seen them fuck up
>> in production with no changes to FreeSWITCH.  Check the invites... mine
>> started sending 0.0.0.0 in the sdp on initial invites.
>>
>> /b
>>
>> On Apr 4, 2010, at 1:57 PM, Aloysius Lloyd wrote:
>>
>> > There is no firmware update on Aastra. Only change in FreeSWITCH.
>> >
>> >
>> > Snom 320 throwing the following Error.
>> >
>> > 2010-04-04 14:04:44.588816 [ERR] sofia_glue.c:3380 a=crypto in RTP/AVP,
>> refer to rfc3711
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>
>
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From brian at freeswitch.org  Sun Apr  4 12:24:34 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 14:24:34 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<x2h13ca621c1004031334v3864f444r54360f9b8c4722eb@mail.gmail.com>
	<g2g8a19bf2e1004040938ya94a911co69c802e76d4e31d2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
Message-ID: <1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>

I'm going to not be able to tell you unless you show me the register and reply to register.. please turn on the sip trace and give me that and I can see what is going on.

/b

On Apr 4, 2010, at 2:19 PM, Aloysius Lloyd wrote:

> I just notice in the console log.
> 
> 2010-04-04 15:04:07.403184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by acl "domains". Falling back to Digest auth.
> 2010-04-04 15:04:07.566184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by acl "domains". Falling back to Digest auth.
> 
> Snom and Aastra - I have the same behavior.
> 
> But Xlite - Not giving the same message and working.
> 
> Thanks
> Lloyd



From lloyd.aloysius at gmail.com  Sun Apr  4 13:05:26 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 16:05:26 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
Message-ID: <t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>

Brian,

Could you please guide me , how to capture the SIP Trace.

1. One Phone Register
2. Other One Not Register
3. Both cannot make any calls.
4.  Multi Tenant Environment.

Thanks
Lloyd


On Sun, Apr 4, 2010 at 3:24 PM, Brian West <brian at freeswitch.org> wrote:

> I'm going to not be able to tell you unless you show me the register and
> reply to register.. please turn on the sip trace and give me that and I can
> see what is going on.
>
> /b
>
> On Apr 4, 2010, at 2:19 PM, Aloysius Lloyd wrote:
>
> > I just notice in the console log.
> >
> > 2010-04-04 15:04:07.403184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by
> acl "domains". Falling back to Digest auth.
> > 2010-04-04 15:04:07.566184 [DEBUG] sofia.c:5847 IP A.B.C.D Rejected by
> acl "domains". Falling back to Digest auth.
> >
> > Snom and Aastra - I have the same behavior.
> >
> > But Xlite - Not giving the same message and working.
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Apr  4 13:10:42 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 15:10:42 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<t2z8a19bf2e1004041113i2d517efbkbea0de1108d7afe8@mail.gmail.com>
	<CFB15B2D-4D67-4F5E-AA8F-31FB8D5D449F@freeswitch.org>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
Message-ID: <69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>

sofia profile xxxx siptrace on

/b
PS:xxxx replace with your profilename 

On Apr 4, 2010, at 3:05 PM, Aloysius Lloyd wrote:

> Brian,
> 
> Could you please guide me , how to capture the SIP Trace. 
> 
> 1. One Phone Register
> 2. Other One Not Register
> 3. Both cannot make any calls.
> 4.  Multi Tenant Environment.
> 
> Thanks
> Lloyd



From lloyd.aloysius at gmail.com  Sun Apr  4 13:27:29 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 16:27:29 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
Message-ID: <w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>

Please find the following trace.


1. Snom 320 - Xlite Call Failed

http://pastebin.freeswitch.org/12624

<http://pastebin.freeswitch.org/12624>2. Aastra 57i - Dial 9999

http://pastebin.freeswitch.org/12626

3. Phone Not Registering.

http://pastebin.freeswitch.org/12627


On Sun, Apr 4, 2010 at 4:10 PM, Brian West <brian at freeswitch.org> wrote:

> sofia profile xxxx siptrace on
>
> /b
> PS:xxxx replace with your profilename
>
> On Apr 4, 2010, at 3:05 PM, Aloysius Lloyd wrote:
>
> > Brian,
> >
> > Could you please guide me , how to capture the SIP Trace.
> >
> > 1. One Phone Register
> > 2. Other One Not Register
> > 3. Both cannot make any calls.
> > 4.  Multi Tenant Environment.
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Apr  4 13:35:14 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 15:35:14 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<s2j8a19bf2e1004041145k8a4d42ccr71f81f138557eed8@mail.gmail.com>
	<79F3C29E-4FE7-45CC-BAA4-738D5A1291DB@freeswitch.org>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
Message-ID: <B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>


On Apr 4, 2010, at 3:27 PM, Aloysius Lloyd wrote:

> Please find the following trace.
> 
> 
> 1. Snom 320 - Xlite Call Failed
> 
> http://pastebin.freeswitch.org/12624

INCOMPATIBLE_DESTINATION (turn SRTP off or set it to optional for the SAVP option.)

Fix the option I told you about already.  This is listed in the FAQ on the wiki already.

> 
> 2. Aastra 57i - Dial 9999
> 
> http://pastebin.freeswitch.org/12626

Your phone isn't even getting autenticated sounds like you have put some ACL's in place to allow it in without auth.

201->9999 in context public


> 
> 3. Phone Not Registering.
> 
> http://pastebin.freeswitch.org/12627

Enable rport if you notice the register comes from port 1026 and we challenge to 5060 because thats what the phone put in the contact field.  Enable rport on the phone will fix this.

/b



> 
> 
> On Sun, Apr 4, 2010 at 4:10 PM, Brian West <brian at freeswitch.org> wrote:
> sofia profile xxxx siptrace on
> 
> /b
> PS:xxxx replace with your profilename
> 
> On Apr 4, 2010, at 3:05 PM, Aloysius Lloyd wrote:
> 
> > Brian,
> >
> > Could you please guide me , how to capture the SIP Trace.
> >
> > 1. One Phone Register
> > 2. Other One Not Register
> > 3. Both cannot make any calls.
> > 4.  Multi Tenant Environment.
> >
> > Thanks
> > Lloyd
> 
> 
> _______________________________________________
> 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
> http://www.freeswitch.org
> 
> _______________________________________________
> 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
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Apr  4 13:37:40 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 15:37:40 -0500
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
Message-ID: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>

Anyone else get spam from them?

/b



From lloyd.aloysius at gmail.com  Sun Apr  4 14:11:59 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 17:11:59 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
Message-ID: <z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>

Thanks Brian.

This is simple direct install and no ACL. I am going to Install From
Scratch.

Thanks
Lloyd

On Sun, Apr 4, 2010 at 4:35 PM, Brian West <brian at freeswitch.org> wrote:

>
> On Apr 4, 2010, at 3:27 PM, Aloysius Lloyd wrote:
>
> > Please find the following trace.
> >
> >
> > 1. Snom 320 - Xlite Call Failed
> >
> > http://pastebin.freeswitch.org/12624
>
> INCOMPATIBLE_DESTINATION (turn SRTP off or set it to optional for the SAVP
> option.)
>
> Fix the option I told you about already.  This is listed in the FAQ on the
> wiki already.
>
> >
> > 2. Aastra 57i - Dial 9999
> >
> > http://pastebin.freeswitch.org/12626
>
> Your phone isn't even getting autenticated sounds like you have put some
> ACL's in place to allow it in without auth.
>
> 201->9999 in context public
>
>
> >
> > 3. Phone Not Registering.
> >
> > http://pastebin.freeswitch.org/12627
>
> Enable rport if you notice the register comes from port 1026 and we
> challenge to 5060 because thats what the phone put in the contact field.
>  Enable rport on the phone will fix this.
>
> /b
>
>
>
> >
> >
> > On Sun, Apr 4, 2010 at 4:10 PM, Brian West <brian at freeswitch.org> wrote:
> > sofia profile xxxx siptrace on
> >
> > /b
> > PS:xxxx replace with your profilename
> >
> > On Apr 4, 2010, at 3:05 PM, Aloysius Lloyd wrote:
> >
> > > Brian,
> > >
> > > Could you please guide me , how to capture the SIP Trace.
> > >
> > > 1. One Phone Register
> > > 2. Other One Not Register
> > > 3. Both cannot make any calls.
> > > 4.  Multi Tenant Environment.
> > >
> > > Thanks
> > > Lloyd
> >
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Sun Apr  4 14:13:24 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 4 Apr 2010 17:13:24 -0400
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
Message-ID: <q2m8a19bf2e1004041413q960e1419ga8f2e6da60252e62@mail.gmail.com>

I recived one - Sun, Apr 4, 2010 at 4:26 PM

Lloyd

On Sun, Apr 4, 2010 at 4:37 PM, Brian West <brian at freeswitch.org> wrote:

> Anyone else get spam from them?
>
> /b
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Apr  4 14:25:46 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 16:25:46 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<z2p8a19bf2e1004041157tf4e1c362ra7734508cbb14373@mail.gmail.com>
	<32ED4209-63E9-451B-8F70-97856A389FB5@freeswitch.org>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
Message-ID: <E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>

be sure to nuke your configs... cuz we won't when you reinstall.

/b

On Apr 4, 2010, at 4:11 PM, Aloysius Lloyd wrote:

> Thanks Brian.
> 
> This is simple direct install and no ACL. I am going to Install From Scratch.
> 
> Thanks
> Lloyd



From brian at freeswitch.org  Sun Apr  4 14:28:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 16:28:12 -0500
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <q2m8a19bf2e1004041413q960e1419ga8f2e6da60252e62@mail.gmail.com>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
	<q2m8a19bf2e1004041413q960e1419ga8f2e6da60252e62@mail.gmail.com>
Message-ID: <3D15397A-8F23-4585-8EE6-801AAD427022@freeswitch.org>

Ok that confirms they are spammers and should be SHOT!

/b

On Apr 4, 2010, at 4:13 PM, Aloysius Lloyd wrote:

> I recived one - Sun, Apr 4, 2010 at 4:26 PM
> 
> Lloyd
> 
> On Sun, Apr 4, 2010 at 4:37 PM, Brian West <brian at freeswitch.org> wrote:
> Anyone else get spam from them?
> 
> /b

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From max.clark at gmail.com  Sun Apr  4 15:00:34 2010
From: max.clark at gmail.com (Max Clark)
Date: Sun, 4 Apr 2010 15:00:34 -0700
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
Message-ID: <p2t2fa1e1781004041500i773850cbsc426df3912984c53@mail.gmail.com>

How would one detect T38 and convert the session into proxy media?

On Sun, Apr 4, 2010 at 11:35 AM, David Ponzone <david.ponzone at gmail.com> wrote:
> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> You just need the regular passthrough module: mod_g729 and to allow G729 as
> inbound and outbound codecs in vars.xml.
> To summarize:
> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP
> path, it relays the audio stream between endpoints, but can still detect
> DTMFs
> -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF
> detection is impossible. In this mode, FS is really a "dumb" transparent
> RTP-forwarder (this is required to get T38 working between the 2 endpoints)
> -bypass media enabled: FreeSWITCH is not in the RTP path
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>
> Clarification - for G729 does freeswitch need to be in "bypass media"
> or "proxy media"? My understanding was that G729 would work with
> "proxy media" enabled and without the new fangled module?
>
> -Max
>
> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net> wrote:
>
> Frank Church <vfclists at googlemail.com> wrote:
>
> I am just trialling Freeswitch with Linksys adapters, whose default
>
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>
> When I change that setting to 'yes' the calls don't go through. I am
>
> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
>
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
> really
>
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
>
> compelling reason, for example a carrier that only supports G.729.
>
>
> _______________________________________________
>
> 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
>
> http://www.freeswitch.org
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


From linux4michelle at tamay-dogan.net  Sun Apr  4 15:16:47 2010
From: linux4michelle at tamay-dogan.net (Michelle Konzack)
Date: Mon, 5 Apr 2010 00:16:47 +0200
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
Message-ID: <20100404221647.GN3737@tamay-dogan.net>

Hello Brian West,

Am 2010-04-04 15:37:40, hacktest Du folgendes herunter:
> Anyone else get spam from them?

YesNo because my server/spamassassin has rejected it.

Missing spamfilter?

Thanks, Greetings and nice Day/Evening
    Michelle Konzack
    Systemadministrator
    24V Electronic Engineer
    Tamay Dogan Network
    Debian GNU/Linux Consultant

-- 
Linux-User #280138 with the Linux Counter, http://counter.li.org/
##################### Debian GNU/Linux Consultant #####################
<http://www.tamay-dogan.net/>                 Michelle Konzack
<http://www.can4linux.org/>                   Apt. 917
<http://www.flexray4linux.org/>               50, rue de Soultz
Jabber linux4michelle at jabber.ccc.de           67100 Strasbourg/France
IRC    #Debian (irc.icq.com)                  Tel. DE: +49 177 9351947
ICQ    #328449886                             Tel. FR: +33  6  61925193
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From brian at freeswitch.org  Sun Apr  4 15:23:07 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 17:23:07 -0500
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <20100404221647.GN3737@tamay-dogan.net>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
	<20100404221647.GN3737@tamay-dogan.net>
Message-ID: <B2D535ED-E3BB-4846-A0DB-557C4E38D78B@freeswitch.org>

I shouldn't have to deal with it.  They should be responsible or DIE.  It slipped thru... but it should be known they are probably a shady operation and shouldn't be trusted.

/b

On Apr 4, 2010, at 5:16 PM, Michelle Konzack wrote:

> Hello Brian West,
> 
> Am 2010-04-04 15:37:40, hacktest Du folgendes herunter:
>> Anyone else get spam from them?
> 
> YesNo because my server/spamassassin has rejected it.
> 
> Missing spamfilter?
> 
> Thanks, Greetings and nice Day/Evening
>    Michelle Konzack
>    Systemadministrator
>    24V Electronic Engineer
>    Tamay Dogan Network
>    Debian GNU/Linux Consultant



From brian at freeswitch.org  Sun Apr  4 20:07:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 22:07:05 -0500
Subject: [Freeswitch-users] azdirectroute.com == SPAMMERS
In-Reply-To: <20100404221647.GN3737@tamay-dogan.net>
References: <D8756EF2-52B1-42EE-BDCC-E1B829E0B51E@freeswitch.org>
	<20100404221647.GN3737@tamay-dogan.net>
Message-ID: <256C90FA-AE87-4D2D-9FA8-71DD9434EB89@freeswitch.org>

Well it seems they harvested our mailing list... so every please be on the look out for it.

/b

On Apr 4, 2010, at 5:16 PM, Michelle Konzack wrote:

> Hello Brian West,
> 
> Am 2010-04-04 15:37:40, hacktest Du folgendes herunter:
>> Anyone else get spam from them?
> 
> YesNo because my server/spamassassin has rejected it.
> 
> Missing spamfilter?



From msc at freeswitch.org  Sun Apr  4 20:11:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 4 Apr 2010 20:11:04 -0700
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
In-Reply-To: <4BB87CA9.3050403@k4gvo.com>
References: <4BB79458.8080601@k4gvo.com>
	<s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>
	<4BB87CA9.3050403@k4gvo.com>
Message-ID: <h2w87f2f3b91004042011m3f5e7c34pbaf94c219a72a867@mail.gmail.com>

I looked in mod_openzap.c and I didn't see any references to channel
variables. However, you have context and dialplan options. I suggest that
you create a dialplan context just for your FXS port(s). Try this. Create
conf/dialplan/fxs-ports.xml:

<include>
  <context name="fxs-ports">
    <extension name="outbound">
      <condition field="destination_number" expression="^(.*)$">
        <action application="set"
data="toll_allow=local,domestic,international"/>
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>
  </context>
</include>

Then in your openzap.conf.xml change the context for the analog span(s) with
the FXS ports:
<param name="context" value="default"/>

Restart FS after making these changes and then give it a shot. You should
see the call from the analog phone going into context "fxs-ports" and then
get transferred over to the default context where it will act like your SIP
phones because we manually set the ${toll_allow} chan var.

-MC

On Sun, Apr 4, 2010 at 4:48 AM, Jim <jim at k4gvo.com> wrote:

> Michael Collins wrote:
> > Variables set in the directory/default/xxxx.xml files apply to users
> > who make authenticated calls through FS. Generally those will be SIP
> > phones.
> >
> > Let's back up a step. What problem are you trying to solve, i.e., why
> > is it that you need to set the toll_allow variable? What endpoint is
> > making an openzap call?
> >
> > -MC
> Hi, Michael,
>
> I have multiple sip phone that are working fine.  When I dial a 10 digit
> number they connect with my sip provider and place the call.  The
> openzap configured phone gets dial tone and can call other extensions,
> however when I dial a 10 digit number it gives me a busy.  In the log I
> see it appears to be failing on the toll_allow test:
>
> Dialplan: OpenZAP/1:1/7707190068 parsing [default->local_call]
> continue=false
> Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [local_call]
> ${toll_allow}() =~ /local/ break=on-false
> Dialplan: OpenZAP/1:1/7707190068 parsing [default->domestic_call]
> continue=false
> Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [domestic_call]
> ${toll_allow}() =~ /domestic/ break=on-false
> Dialplan: OpenZAP/1:1/7707190068 parsing
> [default->international.example.com] continue=false
> Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL)
> [international.example.com] ${toll_allow}() =~ /international/
> break=on-false
>
> This is the area when it should be placing the call, I belive.  When
> placing a call from a sip phone I see:
>
> Dialplan: sofia/internal/1003 at 192.168.2.51 parsing [default->local_call]
> continue=false
> Dialplan: sofia/internal/1003 at 192.168.2.51 Regex (PASS) [local_call]
> ${toll_allow}(domestic,international,local) =~ /local/ break=on-false
> Dialplan: sofia/internal/1003 at 192.168.2.51 Regex (PASS) [local_call]
> destination_number(7707190068) =~ /^(\d{10})$/ break=on-false
> Dialplan: sofia/internal/1003 at 192.168.2.51 Action
> set(effective_caller_id_number=${outbound_caller_id_number})
> Dialplan: sofia/internal/1003 at 192.168.2.51 Action
> set(effective_caller_id_name=${outbound_caller_id_name})
> Dialplan: sofia/internal/1003 at 192.168.2.51 Action
> set(continue_on_fail=true)
> Dialplan: sofia/internal/1003 at 192.168.2.51 Action
> bridge(sofia/gateway/${default_gateway}/7707190068)
>
> I simply want this extension to be able to dial out.  The configuration
> is 99% default.
>
> Thanks,
> Jim.
> >
> > On Sat, Apr 3, 2010 at 12:17 PM, Jim <jim at k4gvo.com
> > <mailto:jim at k4gvo.com>> wrote:
> >
> >     I obviously need to set a <variable name="toll_allow"
> >     value="domestic,international,local"/> somewhere but I can't
> >     figure out
> >     what file to put it in.  The examples all show it in the
> >     directory/default/xxxx.xml files but those appear to be sip only.  In
> >     any event creating files in that directory for my extension did
> >     nothing
> >     to help the problem.
> >
> >     The only places I have the extension mentioned is in  the
> openzap.conf
> >     file and dialplan/default/00_incoming-1.xml.  Adding a <variables ...
> >     element to the latter does nothing.
> >
> >     How do I set that variable?  Or where?
> >
> >     Thanks,
> >     Jim.
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Sun Apr  4 20:16:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 4 Apr 2010 20:16:05 -0700
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <20100404090838.414c58ce@anubis.defcon1>
References: <20100404090838.414c58ce@anubis.defcon1>
Message-ID: <h2v87f2f3b91004042016wc562e5fds46911c2d9bf43652@mail.gmail.com>

On Sun, Apr 4, 2010 at 12:08 AM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> FS svn-17188M
> Debian Lenny
> =================
>
> Hi list,
>
> I've installed cepstral voices (in /opt/swift, and also changed its
> rights recursively to freeswitch:freeswitch) but I had a complain
> at FS start that some cepstral libraries were missing.
>
> Eventually, I was obliged to simlink /opt/swift/lib/lib* into
> /usr/local/freeswitch/lib to have it working correctly, is it normal,
> or did I miss something?
>
> That's "normal" even though it's not desired. Getting Cepstral working
properly again would be nice but I don't think it's high on the priority
list.
-MC


> (I followed the wiki, so all libs were added to ld.so.cache, path in
> mod_cepstral source Makefile are good, SWIFT_HOME is defined, module
> flite loading is commented, etc- so I said to myself: Nantidiou!)
>
> --
> Sorry never means having your say to love.
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Apr  4 20:19:48 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 4 Apr 2010 22:19:48 -0500
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <20100404090838.414c58ce@anubis.defcon1>
References: <20100404090838.414c58ce@anubis.defcon1>
Message-ID: <009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>

Their exists a file in /etc/ called ld.so.conf  if you happen to add  /opt/swift/lib/lib into that then run ldconfig it would work.

muhahahahaha linux is great.

/b

On Apr 4, 2010, at 2:08 AM, Jean-Yves F. Barbier wrote:

> 
> Eventually, I was obliged to simlink /opt/swift/lib/lib* into 
> /usr/local/freeswitch/lib to have it working correctly, is it normal, 
> or did I miss something?



From hungngm at bkav.com.vn  Sun Apr  4 20:27:26 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Mon, 05 Apr 2010 10:27:26 +0700
Subject: [Freeswitch-users] Some question about mod_fifo ??
Message-ID: <5A74CE1F31E29751064BEC8F43427755651603EC@hungngm>



 

Hi Anthony,

Can you discuss some details in how polycom or snom can do this and x-lite not.

If can, I want to edit some open source soffphone like officeSIP to do this.

Best Regards.

 

Anthony Minessale [anthony.minessale at gmail.com]

 

We already do it.
X-Lite does not support it.
If you try it with a phone like snom or polycom you will see it works just like
that.


 

2010/4/3 Nguy??n M???nh H??ng < hungngm at bkav.com.vn >
 

     

    Hi Seven Du.

    

    Thanks to yours suggetion.

    

    I have an ideal, it is: when the call between caller and agent is set, the
    caller_id is determined. So, i want to edit code to sent the agent
    information (the call_id and call_id_number) which will be displayed againt
    in the agent's softphone (as Xlite..) when the call is happening.

    

    I read some documents but i still can't determine: It's maybe yes or maybe
    to do this and where to do this.

    

    Can you give me some comments.

    

    Best Regard.

    

     

    Seven Du [ dujinfang at gmail.com ]

    

    

    ??As discussed in the list, it's not a freeswitch problem but a reality
    of life. 



    Think about customer A and B calls in one after another, then if
    FreeSWITCH call agent X with caller id A and Y with caller id B, and
    angent Y answers before X, then

    1) if bridge Y with A with the FIFO rule, then the caller id is wrong
    2) if bridge Y with B, the caller id is right but it breaks the rule
    of FIFO - A should be served before B!! And what even worse is that
    if X never answer A then A never can be served which is really
    unfair!!

    Of course you don't want 1), and you don't need mod_fifo if you want
    behavior 2), you just need some dialplan trick or some simple Lua
    script I think. Also FreeSWITCH is designed to be easily extended with
    almost any languages so feel free to implement anything.

    2010/3/31 Nguy??n M???nh H??ng : 


    > Hi Mike and Seven Du.
    > Thanks to yours help.
    > I known the mechanism of mod_fifo.
    >>>
   
   
   
   
    http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
    .
    > What a pity, It can't solve this problem. I can't use freeswitch for my
    call
    > center.
    > Hope new version can solve this !!!

    _______________________________________________
    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
    http://www.freeswitch.org


    _______________________________________________
    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
    http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

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From anthony.minessale at gmail.com  Sun Apr  4 21:41:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 4 Apr 2010 23:41:56 -0500
Subject: [Freeswitch-users] Some question about mod_fifo ??
In-Reply-To: <5A74CE1F31E29751064BEC8F43427755651603EC@hungngm>
References: <5A74CE1F31E29751064BEC8F43427755651603EC@hungngm>
Message-ID: <i2w191c3a031004042141g11c18e92wcd63e833fe1f3aa0@mail.gmail.com>

its done by SIP UPDATE on polycom/aastra or sip INFO packets on snom when
the call is bridged.
X-lite does not update anything when it receives them.

That's about it.


2010/4/4 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>

>  Hi Anthony,
>
> Can you discuss some details in how polycom or snom can do this and x-lite
> not.
>
> If can, I want to edit some open source soffphone like officeSIP to do
> this.
>
> Best Regards.
>
> Anthony Minessale <anthony.minessale at gmail.com> [
> anthony.minessale at gmail.com]
>
>
> We already do it.
> X-Lite does not support it.
> If you try it with a phone like snom or polycom you will see it works just
> like that.
>
>
> 2010/4/3 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>
>
>>  Hi Seven Du.
>>
>> Thanks to yours suggetion.
>>
>> I have an ideal, it is: when the call between caller and agent is set, the
>> caller_id is determined. So, i want to edit code to sent the agent
>> information (the call_id and  call_id_number) which will be displayed
>> againt in the agent's softphone (as Xlite..) when the call is happening.
>>
>> I read some documents but i still can't determine: It's maybe yes or maybe
>> to do this and where to do this.
>>
>> Can you give me some comments.
>>
>> Best Regard.
>>
>> Seven Du <dujinfang at gmail.com> [dujinfang at gmail.com]
>>
>>
>> ?As discussed in the list, it's not a freeswitch problem but a reality of
>> life.
>>
>>
>> Think about customer A and B calls in one after another, then if
>> FreeSWITCH call agent X with caller id A and Y with caller id B, and
>> angent Y answers before X, then
>>
>> 1) if bridge Y with A with the FIFO rule, then the caller id is wrong
>> 2) if bridge Y with B, the caller id is right but it breaks the rule
>> of FIFO - A should be served before B!! And what even worse is that
>> if X never answer A then A never can be served which is really
>> unfair!!
>>
>> Of course you don't want 1), and you don't need mod_fifo if you want
>> behavior 2), you just need some dialplan trick or some simple Lua
>> script I think. Also FreeSWITCH is designed to be easily extended with
>> almost any languages so feel free to implement anything.
>>
>> 2010/3/31 Nguy?n M?nh H?ng :
>>
>> > Hi Mike and Seven Du.
>> > Thanks to yours help.
>> > I known the mechanism of mod_fifo.
>> >>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
>> .
>> > What a pity, It can't solve this problem. I can't use freeswitch for my
>> call
>> > center.
>> > Hope new version can solve this !!!
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jayesh.voip at gmail.com  Sun Apr  4 22:43:26 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Mon, 5 Apr 2010 11:13:26 +0530
Subject: [Freeswitch-users] domain-wise context
Message-ID: <u2z4fe8b8271004042243jf5c79487x64e651b11fff2d05@mail.gmail.com>

Hi Brian,
I've tried user_context variable. Does user_context apply to only registered
users in freeswitch?
Because in my case the users are registered on a different software(OSips).
Also out of curiosity, does the OSips needs to be defined as a Gateway in FS
for handling the calls in a standard fashion?

Thanks,

--- Jayesh


> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 3 Apr 2010 22:57:19 -0500
> Subject: Re: [Freeswitch-users] domain-wise context
> see user_context variable on the user.
>
> /b
>
> On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote:
>
> > Hi,
> > Sorry for sounding so impatient, the anxiety only grew because before
> posting it to the list I spent a week on reading all the documentation
> available on and around this topic FS site and mailing lists. I really
> appreciate and am thankful for your suggestions.
> > I'll try out the suggestions given by you. Is there a way that we can
> compare the domain name in the dialplan using regular expressions. Is there
> a value in condition tag that can be used to compare this, something like
> "extension_number".
> > Or should I store the domain value in some variable(something like
> sip_h_<domain_name>) and compare that variable to take the call to a
> different context?
> >
> > Thanks,
> >
> > --- Jayesh
>
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From david.ponzone at gmail.com  Mon Apr  5 01:11:09 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 5 Apr 2010 10:11:09 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <p2t2fa1e1781004041500i773850cbsc426df3912984c53@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
	<p2t2fa1e1781004041500i773850cbsc426df3912984c53@mail.gmail.com>
Message-ID: <A90BAE74-B676-40EA-8B4C-D18509059905@gmail.com>

As far as I know, you can't, because T38 is not advertised at first.
T38 starts with regular codecs in the SDP, and then later, a T38  
REINVITE is negotiated.

I guess the easiest way to handle that is:
-enable proxy media based on DID for inbound fax
-enable proxy media based on CLID for outbound fax

This requires to have specific extensions in your dialplan.

For outbound, you can also have a dedicated SIP profile for T38 ATAs  
so you can enable proxy-media for the whole profile.
For inbound, you may do the same if your ITSP/gateway can send you the  
fax DIDs on a specific trunk (so to a specific SIP profile).

Be aware that T38 is a PITA, and that you really need to validate that  
your T38 device is compatible with the other endpoint, which is  
probably a gateway.
If this gateway is yours, that's fine because you control it, and you  
can rely on it for your T38 service.
If it's not yours, you could have issues if some day, your iTSP  
decices to change it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 05/04/2010 ? 00:00, Max Clark a ?crit :

> How would one detect T38 and convert the session into proxy media?
>
> On Sun, Apr 4, 2010 at 11:35 AM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>> You just need the regular passthrough module: mod_g729 and to allow  
>> G729 as
>> inbound and outbound codecs in vars.xml.
>> To summarize:
>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in  
>> the RTP
>> path, it relays the audio stream between endpoints, but can still  
>> detect
>> DTMFs
>> -proxy media enabled: FreeSWITCH relays the audio stream  
>> transparently, DTMF
>> detection is impossible. In this mode, FS is really a "dumb"  
>> transparent
>> RTP-forwarder (this is required to get T38 working between the 2  
>> endpoints)
>> -bypass media enabled: FreeSWITCH is not in the RTP path
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>
>> Clarification - for G729 does freeswitch need to be in "bypass media"
>> or "proxy media"? My understanding was that G729 would work with
>> "proxy media" enabled and without the new fangled module?
>>
>> -Max
>>
>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
>> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>> _______________________________________________
>>
>> 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
>>
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> 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
> http://www.freeswitch.org

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From vfclists at googlemail.com  Mon Apr  5 02:49:13 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 5 Apr 2010 10:49:13 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
Message-ID: <h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>

Have you reviewed http://pastebin.freeswitch.org/12617 ?
It has the G729 set in the codecs section.
In this one it seems the call does not get to the external gateway.
Freeswitch stops the call before calling the external gateway


I have checked it again a few times using the bypass_media,
proxy_media settings.

And with those settings the call ends as soon as ringing starts or as
sonn as the call is answered.

I will do another one just to confirm




On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> You just need the regular passthrough module: mod_g729 and to allow G729 as
> inbound and outbound codecs in vars.xml.
> To summarize:
> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP
> path, it relays the audio stream between endpoints, but can still detect
> DTMFs
> -proxy media enabled: FreeSWITCH relays the audio stream transparently, DTMF
> detection is impossible. In this mode, FS is really a "dumb" transparent
> RTP-forwarder (this is required to get T38 working between the 2 endpoints)
> -bypass media enabled: FreeSWITCH is not in the RTP path
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>
> Clarification - for G729 does freeswitch need to be in "bypass media"
> or "proxy media"? My understanding was that G729 would work with
> "proxy media" enabled and without the new fangled module?
>
> -Max
>
> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net> wrote:
>
> Frank Church <vfclists at googlemail.com> wrote:
>
> I am just trialling Freeswitch with Linksys adapters, whose default
>
> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>
> When I change that setting to 'yes' the calls don't go through. I am
>
> using the latest Windows SVN.
>
> FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass media
>
> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
> really
>
> need it.
>
> My recommendation would be to use a codec other than G.729 unless you have a
>
> compelling reason, for example a carrier that only supports G.729.
>
>
> _______________________________________________
>
> 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
>
> http://www.freeswitch.org
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From david.ponzone at gmail.com  Mon Apr  5 03:02:27 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 5 Apr 2010 12:02:27 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
	<h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
Message-ID: <FE3DC7CC-7FBB-4344-A495-42D210B544AF@gmail.com>

Frank,

again:

     
------------------------------------------------------------------------
recv 1101 bytes from udp/[192.168.4.154]:5060 at 20:41:04.437500:
     
------------------------------------------------------------------------
    INVITE sip:02074379497 at 192.168.4.156 SIP/2.0
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
    To: <sip:02074379497 at 192.168.4.156>
    Remote-Party-ID: Booth1 <sip: 
1001 at 192.168.4.156>;screen=yes;party=calling
    Call-ID: 68993a3f-e38f80af at 192.168.4.154
    CSeq: 102 INVITE
    Max-Forwards: 70
    Proxy-Authorization: Digest  
username="1001",realm="192.168.4.156",nonce="9567
e05b-d541-4a5e-9e47-2152eb90a199",uri="sip: 
02074379497 at 192.168.4.156",algorithm=
MD5 
,response 
="d0549f668825c7ce92e120071f1cb5ed",qop=auth,nc=00000001,cnonce="18d
a1954"
    Contact: Booth1 <sip:1001 at 192.168.4.154:5060>
    Expires: 240
    User-Agent: Linksys/SPA2102-3.2.8(d)
    Content-Length: 260
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Supported: x-sipura
    Content-Type: application/sdp

    v=0
    o=- 2612067 2612067 IN IP4 192.168.4.154
    s=-
    c=IN IP4 192.168.4.154
    t=0 0
    m=audio 16392 RTP/AVP 18 100 101
    a=rtpmap:18 G729a/8000
    a=rtpmap:100 NSE/8000
    a=fmtp:100 192-193
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv

     
------------------------------------------------------------------------
send 312 bytes to udp/[192.168.4.154]:5060 at 20:41:04.484375:
     
------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
    To: <sip:02074379497 at 192.168.4.156>
    Call-ID: 68993a3f-e38f80af at 192.168.4.154
    CSeq: 102 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-17048M
    Content-Length: 0

     
------------------------------------------------------------------------
2010-04-02 21:41:04.593750 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1001 at 192.168.4.156 
  [cc4b83a6-3d16-4da3-bd6d-5148d0f983e8]
2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup sofia/internal/1001 at 192.168.1.133 
  [CS_NEW] [INCOMPATIBLE_DESTINATION]

send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000:
     
------------------------------------------------------------------------
    SIP/2.0 488 Not Acceptable Here
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=bff390fd4255c0f9o0
    To: <sip:02074379497 at 192.168.4.156>;tag=m006c20Fg5Spa
    Call-ID: 1c27844d-e299e5e9 at 192.168.4.154

That is the answer from FS to the phone, just after receiving the  
INVITE that contains only G729 and NSE (??) in the SDP.

If you're sure you enabled G729 in vars.xml, you should check you  
restarted your SIP profile.
If you are not sure how to do it right, check my previous emails, or  
restart FS .

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 05/04/2010 ? 11:49, Frank Church a ?crit :

> Have you reviewed http://pastebin.freeswitch.org/12617 ?
> It has the G729 set in the codecs section.
> In this one it seems the call does not get to the external gateway.
> Freeswitch stops the call before calling the external gateway
>
>
> I have checked it again a few times using the bypass_media,
> proxy_media settings.
>
> And with those settings the call ends as soon as ringing starts or as
> sonn as the call is answered.
>
> I will do another one just to confirm
>
>
>
>
> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>> You just need the regular passthrough module: mod_g729 and to allow  
>> G729 as
>> inbound and outbound codecs in vars.xml.
>> To summarize:
>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in  
>> the RTP
>> path, it relays the audio stream between endpoints, but can still  
>> detect
>> DTMFs
>> -proxy media enabled: FreeSWITCH relays the audio stream  
>> transparently, DTMF
>> detection is impossible. In this mode, FS is really a "dumb"  
>> transparent
>> RTP-forwarder (this is required to get T38 working between the 2  
>> endpoints)
>> -bypass media enabled: FreeSWITCH is not in the RTP path
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>
>> Clarification - for G729 does freeswitch need to be in "bypass media"
>> or "proxy media"? My understanding was that G729 would work with
>> "proxy media" enabled and without the new fangled module?
>>
>> -Max
>>
>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
>> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>> _______________________________________________
>>
>> 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
>>
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
> http://www.freeswitch.org

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From oseslija at gmail.com  Mon Apr  5 03:34:56 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Apr 2010 12:34:56 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
	<h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
Message-ID: <n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>

G729a is invalid as a codec name. FS used to allow it but not anymore afaik.
You should change codec name to G729 (I assume you're using Linksys product;
there is a setting to change under SIP tab).

Ognjen

On Mon, Apr 5, 2010 at 11:49 AM, Frank Church <vfclists at googlemail.com>wrote:

> Have you reviewed http://pastebin.freeswitch.org/12617 ?
> It has the G729 set in the codecs section.
> In this one it seems the call does not get to the external gateway.
> Freeswitch stops the call before calling the external gateway
>
>
> I have checked it again a few times using the bypass_media,
> proxy_media settings.
>
> And with those settings the call ends as soon as ringing starts or as
> sonn as the call is answered.
>
> I will do another one just to confirm
>
>
>
>
> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> > No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> > You just need the regular passthrough module: mod_g729 and to allow G729
> as
> > inbound and outbound codecs in vars.xml.
> > To summarize:
> > -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in the RTP
> > path, it relays the audio stream between endpoints, but can still detect
> > DTMFs
> > -proxy media enabled: FreeSWITCH relays the audio stream transparently,
> DTMF
> > detection is impossible. In this mode, FS is really a "dumb" transparent
> > RTP-forwarder (this is required to get T38 working between the 2
> endpoints)
> > -bypass media enabled: FreeSWITCH is not in the RTP path
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> >
> > Clarification - for G729 does freeswitch need to be in "bypass media"
> > or "proxy media"? My understanding was that G729 would work with
> > "proxy media" enabled and without the new fangled module?
> >
> > -Max
> >
> > On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net> wrote:
> >
> > Frank Church <vfclists at googlemail.com> wrote:
> >
> > I am just trialling Freeswitch with Linksys adapters, whose default
> >
> > codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> >
> > When I change that setting to 'yes' the calls don't go through. I am
> >
> > using the latest Windows SVN.
> >
> > FreeSWITCH only supports G.729 in pass-through mode (i.e., with bypass
> media
> >
> > enabled). Apparently you can buy a G.729 licence for FreeSWITCH if you
> > really
> >
> > need it.
> >
> > My recommendation would be to use a codec other than G.729 unless you
> have a
> >
> > compelling reason, for example a carrier that only supports G.729.
> >
> >
> > _______________________________________________
> >
> > 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
> >
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
> http://www.freeswitch.org
>
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From david.ponzone at gmail.com  Mon Apr  5 03:46:38 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 5 Apr 2010 12:46:38 +0200
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<20100402092310.GA18680@jdc.jasonjgw.net>
	<p2w2fa1e1781004041112t3e2bd5d9q63cff3347d2f1a9@mail.gmail.com>
	<D715E1A3-9168-4850-877C-790527D3FAF4@gmail.com>
	<h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
Message-ID: <31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>

Ognjen,

very good point, but I used to think that for G729 (and all payload id  
smaller than 97), FS was relying on the payload id, and not the name.

Am I wrong ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :

> G729a is invalid as a codec name. FS used to allow it but not  
> anymore afaik.
> You should change codec name to G729 (I assume you're using Linksys  
> product; there is a setting to change under SIP tab).
>
> Ognjen
>
> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church  
> <vfclists at googlemail.com> wrote:
> Have you reviewed http://pastebin.freeswitch.org/12617 ?
> It has the G729 set in the codecs section.
> In this one it seems the call does not get to the external gateway.
> Freeswitch stops the call before calling the external gateway
>
>
> I have checked it again a few times using the bypass_media,
> proxy_media settings.
>
> And with those settings the call ends as soon as ringing starts or as
> sonn as the call is answered.
>
> I will do another one just to confirm
>
>
>
>
> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> > No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> > You just need the regular passthrough module: mod_g729 and to  
> allow G729 as
> > inbound and outbound codecs in vars.xml.
> > To summarize:
> > -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in  
> the RTP
> > path, it relays the audio stream between endpoints, but can still  
> detect
> > DTMFs
> > -proxy media enabled: FreeSWITCH relays the audio stream  
> transparently, DTMF
> > detection is impossible. In this mode, FS is really a "dumb"  
> transparent
> > RTP-forwarder (this is required to get T38 working between the 2  
> endpoints)
> > -bypass media enabled: FreeSWITCH is not in the RTP path
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou  
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est  
> susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
> message s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
> de ce
> > message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
> >
> >
> >
> > Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> >
> > Clarification - for G729 does freeswitch need to be in "bypass  
> media"
> > or "proxy media"? My understanding was that G729 would work with
> > "proxy media" enabled and without the new fangled module?
> >
> > -Max
> >
> > On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
> wrote:
> >
> > Frank Church <vfclists at googlemail.com> wrote:
> >
> > I am just trialling Freeswitch with Linksys adapters, whose default
> >
> > codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> >
> > When I change that setting to 'yes' the calls don't go through. I am
> >
> > using the latest Windows SVN.
> >
> > FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
> bypass media
> >
> > enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
> you
> > really
> >
> > need it.
> >
> > My recommendation would be to use a codec other than G.729 unless  
> you have a
> >
> > compelling reason, for example a carrier that only supports G.729.
> >
> >
> > _______________________________________________
> >
> > 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
> >
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.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
> http://www.freeswitch.org
>
> _______________________________________________
> 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
> http://www.freeswitch.org

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From sos at sokhapkin.dyndns.org  Mon Apr  5 03:51:50 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 5 Apr 2010 06:51:50 -0400
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <FE3DC7CC-7FBB-4344-A495-42D210B544AF@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<h2m7abab2411004050249m9f0203e6yf5fb14d282c45f3@mail.gmail.com>
	<FE3DC7CC-7FBB-4344-A495-42D210B544AF@gmail.com>
Message-ID: <201004050651.50465.sos@sokhapkin.dyndns.org>

G729a is incorrect codec name, it must be changed to G729 in SPA settings.

On Monday 05 April 2010, David Ponzone wrote:
> Frank,
> 
> again:
> 
> 
> ------------------------------------------------------------------------
> recv 1101 bytes from udp/[192.168.4.154]:5060 at 20:41:04.437500:
> 
> ------------------------------------------------------------------------
>     INVITE sip:02074379497 at 192.168.4.156 SIP/2.0
>     Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
>     From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
>     To: <sip:02074379497 at 192.168.4.156>
>     Remote-Party-ID: Booth1 <sip:
> 1001 at 192.168.4.156>;screen=yes;party=calling
>     Call-ID: 68993a3f-e38f80af at 192.168.4.154
>     CSeq: 102 INVITE
>     Max-Forwards: 70
>     Proxy-Authorization: Digest
> username="1001",realm="192.168.4.156",nonce="9567
> e05b-d541-4a5e-9e47-2152eb90a199",uri="sip:
> 02074379497 at 192.168.4.156",algorithm=
> MD5
> ,response
> ="d0549f668825c7ce92e120071f1cb5ed",qop=auth,nc=00000001,cnonce="18d
> a1954"
>     Contact: Booth1 <sip:1001 at 192.168.4.154:5060>
>     Expires: 240
>     User-Agent: Linksys/SPA2102-3.2.8(d)
>     Content-Length: 260
>     Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>     Supported: x-sipura
>     Content-Type: application/sdp
> 
>     v=0
>     o=- 2612067 2612067 IN IP4 192.168.4.154
>     s=-
>     c=IN IP4 192.168.4.154
>     t=0 0
>     m=audio 16392 RTP/AVP 18 100 101
>     a=rtpmap:18 G729a/8000
>     a=rtpmap:100 NSE/8000
>     a=fmtp:100 192-193
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-15
>     a=ptime:30
>     a=sendrecv
> 
> 
> ------------------------------------------------------------------------
> send 312 bytes to udp/[192.168.4.154]:5060 at 20:41:04.484375:
> 
> ------------------------------------------------------------------------
>     SIP/2.0 100 Trying
>     Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
>     From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
>     To: <sip:02074379497 at 192.168.4.156>
>     Call-ID: 68993a3f-e38f80af at 192.168.4.154
>     CSeq: 102 INVITE
>     User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-17048M
>     Content-Length: 0
> 
> 
> ------------------------------------------------------------------------
> 2010-04-02 21:41:04.593750 [NOTICE] switch_channel.c:669 New Channel
>  sofia/internal/1001 at 192.168.4.156 [cc4b83a6-3d16-4da3-bd6d-5148d0f983e8]
> 2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup
>  sofia/internal/1001 at 192.168.1.133 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> 
> send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000:
> 
> ------------------------------------------------------------------------
>     SIP/2.0 488 Not Acceptable Here
>     Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d
>     From: Booth1 <sip:1001 at 192.168.4.156>;tag=bff390fd4255c0f9o0
>     To: <sip:02074379497 at 192.168.4.156>;tag=m006c20Fg5Spa
>     Call-ID: 1c27844d-e299e5e9 at 192.168.4.154
> 
> That is the answer from FS to the phone, just after receiving the
> INVITE that contains only G729 and NSE (??) in the SDP.
> 
> If you're sure you enabled G729 in vars.xml, you should check you
> restarted your SIP profile.
> If you are not sure how to do it right, check my previous emails, or
> restart FS .
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
> 
> Le 05/04/2010 ? 11:49, Frank Church a ?crit :
> > Have you reviewed http://pastebin.freeswitch.org/12617 ?
> > It has the G729 set in the codecs section.
> > In this one it seems the call does not get to the external gateway.
> > Freeswitch stops the call before calling the external gateway
> >
> >
> > I have checked it again a few times using the bypass_media,
> > proxy_media settings.
> >
> > And with those settings the call ends as soon as ringing starts or as
> > sonn as the call is answered.
> >
> > I will do another one just to confirm
> >
> > On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> >> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> >> You just need the regular passthrough module: mod_g729 and to allow
> >> G729 as
> >> inbound and outbound codecs in vars.xml.
> >> To summarize:
> >> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
> >> the RTP
> >> path, it relays the audio stream between endpoints, but can still
> >> detect
> >> DTMFs
> >> -proxy media enabled: FreeSWITCH relays the audio stream
> >> transparently, DTMF
> >> detection is impossible. In this mode, FS is really a "dumb"
> >> transparent
> >> RTP-forwarder (this is required to get T38 working between the 2
> >> endpoints)
> >> -bypass media enabled: FreeSWITCH is not in the RTP path
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >> Ce message et toutes les pi?ces jointes sont confidentiels et
> >> ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou
> >> diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est
> >> susceptible
> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
> >> message s'il
> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
> >> de ce
> >> message, merci de le d?truire imm?diatement et d'avertir
> >> l'exp?diteur.
> >>
> >>
> >>
> >> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> >>
> >> Clarification - for G729 does freeswitch need to be in "bypass media"
> >> or "proxy media"? My understanding was that G729 would work with
> >> "proxy media" enabled and without the new fangled module?
> >>
> >> -Max
> >>
> >> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
> >> wrote:
> >>
> >> Frank Church <vfclists at googlemail.com> wrote:
> >>
> >> I am just trialling Freeswitch with Linksys adapters, whose default
> >>
> >> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> >>
> >> When I change that setting to 'yes' the calls don't go through. I am
> >>
> >> using the latest Windows SVN.
> >>
> >> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
> >> bypass media
> >>
> >> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
> >> you
> >> really
> >>
> >> need it.
> >>
> >> My recommendation would be to use a codec other than G.729 unless
> >> you have a
> >>
> >> compelling reason, for example a carrier that only supports G.729.
> >>
> >>
> >> _______________________________________________
> >>
> >> 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
> >>
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> 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
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> 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
> >> http://www.freeswitch.org
> >
> > =======================
> > http://devblog.brahmancreations.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
> > http://www.freeswitch.org
> 



From sos at sokhapkin.dyndns.org  Mon Apr  5 03:54:11 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 5 Apr 2010 06:54:11 -0400
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
Message-ID: <201004050654.11705.sos@sokhapkin.dyndns.org>

FS looks at codec name too.

On Monday 05 April 2010, David Ponzone wrote:
> Ognjen,
> 
> very good point, but I used to think that for G729 (and all payload id
> smaller than 97), FS was relying on the payload id, and not the name.
> 
> Am I wrong ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
> 
> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
> > G729a is invalid as a codec name. FS used to allow it but not
> > anymore afaik.
> > You should change codec name to G729 (I assume you're using Linksys
> > product; there is a setting to change under SIP tab).
> >
> > Ognjen
> >
> > On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
> > <vfclists at googlemail.com> wrote:
> > Have you reviewed http://pastebin.freeswitch.org/12617 ?
> > It has the G729 set in the codecs section.
> > In this one it seems the call does not get to the external gateway.
> > Freeswitch stops the call before calling the external gateway
> >
> >
> > I have checked it again a few times using the bypass_media,
> > proxy_media settings.
> >
> > And with those settings the call ends as soon as ringing starts or as
> > sonn as the call is answered.
> >
> > I will do another one just to confirm
> >
> > On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
> > > No, FreeSWITCH does NOT need to be in bypass media or proxy media.
> > > You just need the regular passthrough module: mod_g729 and to
> >
> > allow G729 as
> >
> > > inbound and outbound codecs in vars.xml.
> > > To summarize:
> > > -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
> >
> > the RTP
> >
> > > path, it relays the audio stream between endpoints, but can still
> >
> > detect
> >
> > > DTMFs
> > > -proxy media enabled: FreeSWITCH relays the audio stream
> >
> > transparently, DTMF
> >
> > > detection is impossible. In this mode, FS is really a "dumb"
> >
> > transparent
> >
> > > RTP-forwarder (this is required to get T38 working between the 2
> >
> > endpoints)
> >
> > > -bypass media enabled: FreeSWITCH is not in the RTP path
> > > David Ponzone  Direction Technique
> > > email: david.ponzone at ipeva.fr
> > > tel:      01 74 03 18 97
> > > gsm:   06 66 98 76 34
> > > Service Client IPeva
> > > tel:      0811 46 26 26
> > > www.ipeva.fr  -   www.ipeva-studio.com
> > > Ce message et toutes les pi?ces jointes sont confidentiels et
> >
> > ?tablis ?
> >
> > > l'intention exclusive de ses destinataires. Toute utilisation ou
> >
> > diffusion
> >
> > > non autoris?e est interdite. Tout message ?lectronique est
> >
> > susceptible
> >
> > > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
> >
> > message s'il
> >
> > > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
> >
> > de ce
> >
> > > message, merci de le d?truire imm?diatement et d'avertir
> >
> > l'exp?diteur.
> >
> > > Le 04/04/2010 ? 20:12, Max Clark a ?crit :
> > >
> > > Clarification - for G729 does freeswitch need to be in "bypass
> >
> > media"
> >
> > > or "proxy media"? My understanding was that G729 would work with
> > > "proxy media" enabled and without the new fangled module?
> > >
> > > -Max
> > >
> > > On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
> >
> > wrote:
> > > Frank Church <vfclists at googlemail.com> wrote:
> > >
> > > I am just trialling Freeswitch with Linksys adapters, whose default
> > >
> > > codec I have set to G729 with 'Use Pref Codec Only:' set to no.
> > >
> > > When I change that setting to 'yes' the calls don't go through. I am
> > >
> > > using the latest Windows SVN.
> > >
> > > FreeSWITCH only supports G.729 in pass-through mode (i.e., with
> >
> > bypass media
> >
> > > enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
> >
> > you
> >
> > > really
> > >
> > > need it.
> > >
> > > My recommendation would be to use a codec other than G.729 unless
> >
> > you have a
> >
> > > compelling reason, for example a carrier that only supports G.729.
> > >
> > >
> > > _______________________________________________
> > >
> > > 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-user
> > >s
> > >
> > > http://www.freeswitch.org
> > >
> > >
> > > _______________________________________________
> > > 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-user
> > >s http://www.freeswitch.org
> > >
> > >
> > > _______________________________________________
> > > 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-user
> > >s http://www.freeswitch.org
> >
> > --
> > Frank Church
> >
> > =======================
> > http://devblog.brahmancreations.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
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > 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
> > http://www.freeswitch.org
> 



From mrene_lists at avgs.ca  Mon Apr  5 03:59:07 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 5 Apr 2010 06:59:07 -0400
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <201004050654.11705.sos@sokhapkin.dyndns.org>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
Message-ID: <26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>

Yup, thats why we even have a param called "NDLB-allow-bad-iananame"  
in sofia profiles.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:

> FS looks at codec name too.
>
> On Monday 05 April 2010, David Ponzone wrote:
>> Ognjen,
>>
>> very good point, but I used to think that for G729 (and all payload  
>> id
>> smaller than 97), FS was relying on the payload id, and not the name.
>>
>> Am I wrong ?
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
>> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
>> pas destinataire de ce message, merci de le d?truire imm?diatement et
>> d'avertir l'exp?diteur.
>>
>> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
>>> G729a is invalid as a codec name. FS used to allow it but not
>>> anymore afaik.
>>> You should change codec name to G729 (I assume you're using Linksys
>>> product; there is a setting to change under SIP tab).
>>>
>>> Ognjen
>>>
>>> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
>>> <vfclists at googlemail.com> wrote:
>>> Have you reviewed http://pastebin.freeswitch.org/12617 ?
>>> It has the G729 set in the codecs section.
>>> In this one it seems the call does not get to the external gateway.
>>> Freeswitch stops the call before calling the external gateway
>>>
>>>
>>> I have checked it again a few times using the bypass_media,
>>> proxy_media settings.
>>>
>>> And with those settings the call ends as soon as ringing starts or  
>>> as
>>> sonn as the call is answered.
>>>
>>> I will do another one just to confirm
>>>
>>> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com>  
>>> wrote:
>>>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>>>> You just need the regular passthrough module: mod_g729 and to
>>>
>>> allow G729 as
>>>
>>>> inbound and outbound codecs in vars.xml.
>>>> To summarize:
>>>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
>>>
>>> the RTP
>>>
>>>> path, it relays the audio stream between endpoints, but can still
>>>
>>> detect
>>>
>>>> DTMFs
>>>> -proxy media enabled: FreeSWITCH relays the audio stream
>>>
>>> transparently, DTMF
>>>
>>>> detection is impossible. In this mode, FS is really a "dumb"
>>>
>>> transparent
>>>
>>>> RTP-forwarder (this is required to get T38 working between the 2
>>>
>>> endpoints)
>>>
>>>> -bypass media enabled: FreeSWITCH is not in the RTP path
>>>> David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et
>>>
>>> ?tablis ?
>>>
>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>
>>> diffusion
>>>
>>>> non autoris?e est interdite. Tout message ?lectronique est
>>>
>>> susceptible
>>>
>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
>>>
>>> message s'il
>>>
>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
>>>
>>> de ce
>>>
>>>> message, merci de le d?truire imm?diatement et d'avertir
>>>
>>> l'exp?diteur.
>>>
>>>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>>>
>>>> Clarification - for G729 does freeswitch need to be in "bypass
>>>
>>> media"
>>>
>>>> or "proxy media"? My understanding was that G729 would work with
>>>> "proxy media" enabled and without the new fangled module?
>>>>
>>>> -Max
>>>>
>>>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
>>>
>>> wrote:
>>>> Frank Church <vfclists at googlemail.com> wrote:
>>>>
>>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>>>
>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>>>
>>>> When I change that setting to 'yes' the calls don't go through. I  
>>>> am
>>>>
>>>> using the latest Windows SVN.
>>>>
>>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
>>>
>>> bypass media
>>>
>>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
>>>
>>> you
>>>
>>>> really
>>>>
>>>> need it.
>>>>
>>>> My recommendation would be to use a codec other than G.729 unless
>>>
>>> you have a
>>>
>>>> compelling reason, for example a carrier that only supports G.729.
>>>>
>>>>
>>>> _______________________________________________
>>>>
>>>> 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-user
>>>> s
>>>>
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> 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-user
>>>> s http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> 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-user
>>>> s http://www.freeswitch.org
>>>
>>> --
>>> Frank Church
>>>
>>> =======================
>>> http://devblog.brahmancreations.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
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> 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
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org



From oseslija at gmail.com  Mon Apr  5 04:14:41 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 5 Apr 2010 13:14:41 +0200
Subject: [Freeswitch-users] domain-wise context
In-Reply-To: <u2z4fe8b8271004042243jf5c79487x64e651b11fff2d05@mail.gmail.com>
References: <u2z4fe8b8271004042243jf5c79487x64e651b11fff2d05@mail.gmail.com>
Message-ID: <q2r4468a6771004050414xc4759078h58c9e434c0eb1c5d@mail.gmail.com>

user_context applies to authenticated users. That means if a user
successfully gets its INVITE authenticated either by IP or SIP auth method
its call will end up in ${user_context}.

Ognjen

On Mon, Apr 5, 2010 at 7:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:

> Hi Brian,
> I've tried user_context variable. Does user_context apply to only
> registered users in freeswitch?
> Because in my case the users are registered on a different software(OSips).
> Also out of curiosity, does the OSips needs to be defined as a Gateway in FS
> for handling the calls in a standard fashion?
>
> Thanks,
>
> --- Jayesh
>
>
>> ---------- Forwarded message ----------
>> From: Brian West <brian at freeswitch.org>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Sat, 3 Apr 2010 22:57:19 -0500
>> Subject: Re: [Freeswitch-users] domain-wise context
>> see user_context variable on the user.
>>
>> /b
>>
>> On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote:
>>
>> > Hi,
>> > Sorry for sounding so impatient, the anxiety only grew because before
>> posting it to the list I spent a week on reading all the documentation
>> available on and around this topic FS site and mailing lists. I really
>> appreciate and am thankful for your suggestions.
>> > I'll try out the suggestions given by you. Is there a way that we can
>> compare the domain name in the dialplan using regular expressions. Is there
>> a value in condition tag that can be used to compare this, something like
>> "extension_number".
>> > Or should I store the domain value in some variable(something like
>> sip_h_<domain_name>) and compare that variable to take the call to a
>> different context?
>> >
>> > Thanks,
>> >
>> > --- Jayesh
>>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From jayesh.voip at gmail.com  Mon Apr  5 05:24:05 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Mon, 5 Apr 2010 17:54:05 +0530
Subject: [Freeswitch-users] domain-wise context
Message-ID: <m2h4fe8b8271004050524kbda3bbc2hf86948274c934da7@mail.gmail.com>

>
> Hi,
>
   Now i know, why user_context is not working for me. My calls are coming
from Opensips and are not authenticated by Freeswitch.

--- Jayesh

>
> ---------- Forwarded message ----------
> From: Ognjen Seslija <oseslija at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 5 Apr 2010 13:14:41 +0200
> Subject: Re: [Freeswitch-users] domain-wise context
> user_context applies to authenticated users. That means if a user
> successfully gets its INVITE authenticated either by IP or SIP auth method
> its call will end up in ${user_context}.
>
> Ognjen
>
> On Mon, Apr 5, 2010 at 7:43 AM, Jayesh Nambiar <jayesh.voip at gmail.com>wrote:
>
>> Hi Brian,
>> I've tried user_context variable. Does user_context apply to only
>> registered users in freeswitch?
>> Because in my case the users are registered on a different
>> software(OSips). Also out of curiosity, does the OSips needs to be defined
>> as a Gateway in FS for handling the calls in a standard fashion?
>>
>> Thanks,
>>
>> --- Jayesh
>>
>>
>>> ---------- Forwarded message ----------
>>> From: Brian West <brian at freeswitch.org>
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Sat, 3 Apr 2010 22:57:19 -0500
>>> Subject: Re: [Freeswitch-users] domain-wise context
>>> see user_context variable on the user.
>>>
>>> /b
>>>
>>> On Apr 3, 2010, at 10:47 PM, Jayesh Nambiar wrote:
>>>
>>> > Hi,
>>> > Sorry for sounding so impatient, the anxiety only grew because before
>>> posting it to the list I spent a week on reading all the documentation
>>> available on and around this topic FS site and mailing lists. I really
>>> appreciate and am thankful for your suggestions.
>>> > I'll try out the suggestions given by you. Is there a way that we can
>>> compare the domain name in the dialplan using regular expressions. Is there
>>> a value in condition tag that can be used to compare this, something like
>>> "extension_number".
>>> > Or should I store the domain value in some variable(something like
>>> sip_h_<domain_name>) and compare that variable to take the call to a
>>> different context?
>>> >
>>> > Thanks,
>>> >
>>> > --- Jayesh
>>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From jim at k4gvo.com  Mon Apr  5 05:53:50 2010
From: jim at k4gvo.com (Jim)
Date: Mon, 05 Apr 2010 08:53:50 -0400
Subject: [Freeswitch-users] Openzap extension can't use outside lines.
In-Reply-To: <h2w87f2f3b91004042011m3f5e7c34pbaf94c219a72a867@mail.gmail.com>
References: <4BB79458.8080601@k4gvo.com>	<s2v87f2f3b91004032329sed101915u48f4f63f16df8250@mail.gmail.com>	<4BB87CA9.3050403@k4gvo.com>
	<h2w87f2f3b91004042011m3f5e7c34pbaf94c219a72a867@mail.gmail.com>
Message-ID: <4BB9DD5E.8030706@k4gvo.com>

I never gets around to reading that file.  It looks like it parses 
conf/dialplan/default.xml, conf/dialplan/default/*.xml and then stops.  
It seems to be matching something in the 99999_enum.xml file and never 
getting any other file.  I read up on enum but I don't really know what 
it's supposed to do.

This is the last bit of parsing he does:

Dialplan: OpenZAP/1:1/7707190068 parsing 
[default->international.example.com] continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) 
[international.example.com] ${toll_allow}() =~ /international/ 
break=on-false
Dialplan: OpenZAP/1:1/7707190068 parsing [default->enum] continue=false
Dialplan: OpenZAP/1:1/7707190068 Regex (PASS) [enum] 
${module_exists(mod_enum)}(true) =~ /true/ break=on-false
Dialplan: OpenZAP/1:1/7707190068 Regex (PASS) [enum] 
destination_number(7707190068) =~ /^(.*)$/ break=on-false
Dialplan: OpenZAP/1:1/7707190068 Action transfer(7707190068 enum)


Thanks,
Jim.
Michael Collins wrote:
> I looked in mod_openzap.c and I didn't see any references to channel 
> variables. However, you have context and dialplan options. I suggest 
> that you create a dialplan context just for your FXS port(s). Try 
> this. Create conf/dialplan/fxs-ports.xml:
>
> <include>
>   <context name="fxs-ports">
>     <extension name="outbound">
>       <condition field="destination_number" expression="^(.*)$">
>         <action application="set" 
> data="toll_allow=local,domestic,international"/>
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>     </extension>
>   </context>
> </include>
>
> Then in your openzap.conf.xml change the context for the analog 
> span(s) with the FXS ports:
> <param name="context" value="default"/>
>
> Restart FS after making these changes and then give it a shot. You 
> should see the call from the analog phone going into context 
> "fxs-ports" and then get transferred over to the default context where 
> it will act like your SIP phones because we manually set the 
> ${toll_allow} chan var.
>
> -MC
>
> On Sun, Apr 4, 2010 at 4:48 AM, Jim <jim at k4gvo.com 
> <mailto:jim at k4gvo.com>> wrote:
>
>     Michael Collins wrote:
>     > Variables set in the directory/default/xxxx.xml files apply to users
>     > who make authenticated calls through FS. Generally those will be SIP
>     > phones.
>     >
>     > Let's back up a step. What problem are you trying to solve,
>     i.e., why
>     > is it that you need to set the toll_allow variable? What endpoint is
>     > making an openzap call?
>     >
>     > -MC
>     Hi, Michael,
>
>     I have multiple sip phone that are working fine.  When I dial a 10
>     digit
>     number they connect with my sip provider and place the call.  The
>     openzap configured phone gets dial tone and can call other extensions,
>     however when I dial a 10 digit number it gives me a busy.  In the
>     log I
>     see it appears to be failing on the toll_allow test:
>
>     Dialplan: OpenZAP/1:1/7707190068 parsing [default->local_call]
>     continue=false
>     Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [local_call]
>     ${toll_allow}() =~ /local/ break=on-false
>     Dialplan: OpenZAP/1:1/7707190068 parsing [default->domestic_call]
>     continue=false
>     Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL) [domestic_call]
>     ${toll_allow}() =~ /domestic/ break=on-false
>     Dialplan: OpenZAP/1:1/7707190068 parsing
>     [default->international.example.com
>     <http://international.example.com>] continue=false
>     Dialplan: OpenZAP/1:1/7707190068 Regex (FAIL)
>     [international.example.com <http://international.example.com>]
>     ${toll_allow}() =~ /international/
>     break=on-false
>
>     This is the area when it should be placing the call, I belive.  When
>     placing a call from a sip phone I see:
>
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> parsing [default->local_call]
>     continue=false
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Regex (PASS) [local_call]
>     ${toll_allow}(domestic,international,local) =~ /local/ break=on-false
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Regex (PASS) [local_call]
>     destination_number(7707190068) =~ /^(\d{10})$/ break=on-false
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Action
>     set(effective_caller_id_number=${outbound_caller_id_number})
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Action
>     set(effective_caller_id_name=${outbound_caller_id_name})
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Action
>     set(continue_on_fail=true)
>     Dialplan: sofia/internal/1003 at 192.168.2.51
>     <mailto:1003 at 192.168.2.51> Action
>     bridge(sofia/gateway/${default_gateway}/7707190068)
>
>     I simply want this extension to be able to dial out.  The
>     configuration
>     is 99% default.
>
>     Thanks,
>     Jim.
>     >
>     > On Sat, Apr 3, 2010 at 12:17 PM, Jim <jim at k4gvo.com
>     <mailto:jim at k4gvo.com>
>     > <mailto:jim at k4gvo.com <mailto:jim at k4gvo.com>>> wrote:
>     >
>     >     I obviously need to set a <variable name="toll_allow"
>     >     value="domestic,international,local"/> somewhere but I can't
>     >     figure out
>     >     what file to put it in.  The examples all show it in the
>     >     directory/default/xxxx.xml files but those appear to be sip
>     only.  In
>     >     any event creating files in that directory for my extension did
>     >     nothing
>     >     to help the problem.
>     >
>     >     The only places I have the extension mentioned is in  the
>     openzap.conf
>     >     file and dialplan/default/00_incoming-1.xml.  Adding a
>     <variables ...
>     >     element to the latter does nothing.
>     >
>     >     How do I set that variable?  Or where?
>     >
>     >     Thanks,
>     >     Jim.
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     http://www.freeswitch.org
>     >
>     >
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>   



From 12ukwn at gmail.com  Mon Apr  5 06:07:24 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Mon, 5 Apr 2010 15:07:24 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
Message-ID: <20100405150724.214af8a2@anubis.defcon1>

Le Sun, 4 Apr 2010 22:19:48 -0500,
Brian West <brian at freeswitch.org> a ?crit :

> Their exists a file in /etc/ called ld.so.conf  if you happen to
> add  /opt/swift/lib/lib into that then run ldconfig it would work.
> 
> muhahahahaha linux is great.

As I wrote in my 1st post: I followed the wiki (and I'm not a Linux rooky),
so I asked here; may be I should reformulate my question: why having to
symlink all ceptsral libs into /usr/local/freswitch/lib while already having
them cached in the system libs cache (/etc/ld.so.cache)?

This is the thing I don't understand...

-- 
For internal use only.


From Russell.Mosemann at cune.org  Mon Apr  5 08:54:08 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Mon, 5 Apr 2010 10:54:08 -0500
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <20100405150724.214af8a2@anubis.defcon1>
References: <20100404090838.414c58ce@anubis.defcon1><009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
Message-ID: <769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>

Jean-Yves F. Barbier wrote:

> so I asked here; may be I should reformulate my question: why having to
> symlink all ceptsral libs into /usr/local/freswitch/lib while already
> having them cached in the system libs cache (/etc/ld.so.cache)?

What does "ldd freeswitch" say?

--
Russell Mosemann



From fraserredmond at gmail.com  Mon Apr  5 09:12:42 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 5 Apr 2010 17:12:42 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
Message-ID: <j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>

I've taken another stab at this one way audio problem today.

I've run a wireshark capture, and looking at the RTP analysis it only has
the down-stream, it doesn't record anything being sent upstream at all.

Below is the SIP graph, which shows RTP coming down, but none going up. But
I don't know enough about SIP to know whether something is missing.

Any suggestions of what I should try now?

Would the dtmf's be sent in the sip packets, or in the rtp?

To preempt the easy answers and save some time:
- ports are open on EC2 config,
- iptables turned off for the test,
- RTP port range uncommented in switch.conf.xml,
- softphone stun was set to stun.freeswitch.org

Cheers,
Fraser


|Time     | 192.168.1.8                           |
|         |                   | 184.73.226.197    |
|6.488    |         INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC
g711A g)          |SIP From:
sip:12610 at 184.73.226.197<sip%3A12610 at 184.73.226.197>
To:sip:12605 at 184.73.226.197 <To%3Asip%3A12605 at 184.73.226.197>
|         |(25829)  ------------------>  (5060)   |
|6.615    |         100 Trying|                   |SIP Status
|         |(25829)  <------------------  (5060)   |
|6.623    |         407 Proxy Authentication Required          |SIP Status
|         |(25829)  <------------------  (5060)   |
|6.623    |         ACK       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|6.738    |         INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC
g711A g)          |SIP From:
sip:12610 at 184.73.226.197<sip%3A12610 at 184.73.226.197>
To:sip:12605 at 184.73.226.197 <To%3Asip%3A12605 at 184.73.226.197>
|         |(25829)  ------------------>  (5060)   |
|6.869    |         100 Trying|                   |SIP Status
|         |(25829)  <------------------  (5060)   |
|7.070    |         183 Session Progress SDP ( g711U
telephone-eve...          |SIP Status
|         |(25829)  <------------------  (5060)   |
|7.264    |         RTP (g711U)                   |RTP Num packets:520
Duration:10.793s SSRC:0x5433093E
|         |(44172)  <------------------  (30432)  |
|18.090   |         200 OK SDP ( g711U telephone-event)          |SIP Status
|         |(25829)  <------------------  (5060)   |
|18.112   |         ACK       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|31.750   |         BYE       |                   |SIP Request
|         |(25829)  ------------------>  (5060)   |
|31.872   |         200 OK    |                   |SIP Status
|         |(25829)  <------------------  (5060)   |






On Sat, Apr 3, 2010 at 4:49 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> I've got a FreeSwitch server up on Amazon EC2, ports wide open for my
> office external-IP, server iptables disabled, and changed the FreeSwitch ACL
> domains to "allow", so it's all wide open for now.
>
> In the office I'm trying to connect to the server from Bria/X-lite. I've
> entered a stun server (stun.freeswitch.org) and I can now call to the
> server, but not from the server. I read this page:
>    http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc
> which suggested adding this variable to the user config:
>
> http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunction
>
> With that on I can now call to and from the server. However with or without
> that although I can hear audio from the server, audio to the server isn't
> arriving (doesn't appear in recordings), and dtmf doesn't get received
> either.
>
> When I hang up from the client, I see in the CLI that it gets that
> instruction, so it hasn't started the call and lost all contact with the
> softphone, it's receiving some instructions, but not the audio and dtmf.
>
> The problem is that both the server and client are each behind NAT, so
> either could be having the problem (on EC2 the auto-NAT doesn't work, so
> I've specified the external rtp and sip ip's.. I've also turned on
> aggressive-NAT in case that helps. Also I'm connecting to the server by a
> sub-domain (A-name) rather than IP.)
>
> I've got almost the same setup working fine on the internal network (same
> dialplan and directory, and all the config is the same if it can be), so
> it's got to be something to do with the NAT's.
>
> Any suggestions on what the problem might be, or how to find it?
>
> Cheers,
> Fraser
>
>
>
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From brian at freeswitch.org  Mon Apr  5 09:21:33 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 11:21:33 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
Message-ID: <4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>

Most likely its in the RTP stream as RFC2833 which is the reason you're not getting anything plus I need a FULL sip trace not this abbreviated trace.

/b

On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:

> I've taken another stab at this one way audio problem today.
> 
> I've run a wireshark capture, and looking at the RTP analysis it only has the down-stream, it doesn't record anything being sent upstream at all.
> 
> Below is the SIP graph, which shows RTP coming down, but none going up. But I don't know enough about SIP to know whether something is missing.
> 
> Any suggestions of what I should try now?
> 
> Would the dtmf's be sent in the sip packets, or in the rtp?
> 
> To preempt the easy answers and save some time:
> - ports are open on EC2 config, 
> - iptables turned off for the test,
> - RTP port range uncommented in switch.conf.xml,
> - softphone stun was set to stun.freeswitch.org
> 
> Cheers,
> Fraser

-------------- next part --------------
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From 12ukwn at gmail.com  Mon Apr  5 09:24:14 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Mon, 05 Apr 2010 18:24:14 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
References: <20100404090838.414c58ce@anubis.defcon1><009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
Message-ID: <4BBA0EAE.4020003@gmail.com>

Russell Mosemann wrote:
>> so I asked here; may be I should reformulate my question: why having to
>> symlink all ceptsral libs into /usr/local/freswitch/lib while already
>> having them cached in the system libs cache (/etc/ld.so.cache)?
> 
> What does "ldd freeswitch" say?
linux-gate.so.1 =>  (0xb7faf000)
libm.so.6 => /lib/i686/cmov/libm.so.6 (0xb7f5d000)
libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1 (0xb7db3000)
libuuid.so.1 => /lib/libuuid.so.1 (0xb7dae000)
librt.so.1 => /lib/i686/cmov/librt.so.1 (0xb7da5000)
libdl.so.2 => /lib/i686/cmov/libdl.so.2 (0xb7da1000)
libcrypt.so.1 => /lib/i686/cmov/libcrypt.so.1 (0xb7d6f000)
libpthread.so.0 => /lib/i686/cmov/libpthread.so.0 (0xb7d56000)
libssl.so.0.9.8 => /usr/lib/i686/cmov/libssl.so.0.9.8 (0xb7d0f000)
libcrypto.so.0.9.8 => /usr/lib/i686/cmov/libcrypto.so.0.9.8 (0xb7bbb000)
libncurses.so.5 => /lib/libncurses.so.5 (0xb7b89000)
libc.so.6 => /lib/i686/cmov/libc.so.6 (0xb7a2e000)
/lib/ld-linux.so.2 (0xb7fb0000)
libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0xb7940000)
libgcc_s.so.1 => /lib/libgcc_s.so.1 (0xb7933000)
libodbc.so.1 => /usr/lib/libodbc.so.1 (0xb78d2000)
libz.so.1 => /usr/lib/libz.so.1 (0xb78bd000)
libltdl.so.3 => /usr/lib/libltdl.so.3 (0xb78b6000)

-- 
We don't have to protect the environment -- the Second Coming is at hand.
		-- James Watt, noted ecologist


From Russell.Mosemann at cune.org  Mon Apr  5 09:44:06 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Mon, 5 Apr 2010 11:44:06 -0500
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <4BBA0EAE.4020003@gmail.com>
References: <20100404090838.414c58ce@anubis.defcon1><009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>	<20100405150724.214af8a2@anubis.defcon1><769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
Message-ID: <08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>

Jean-Yves F. Barbier wrote:
> >> so I asked here; may be I should reformulate my question: why having to
> >> symlink all ceptsral libs into /usr/local/freswitch/lib while already
> >> having them cached in the system libs cache (/etc/ld.so.cache)?

When you say that the libs are in the system libs cache, do you mean that the libs are configured in /etc/ld.conf (or /etc/ld.conf.d/, as appropriate), and when you enter "ldconfig -v", the libs are listed in the output under their home directory (was it /opt/something?)?

On rare occasion, I have seen software not find a library for some strange reason when there is a libsomething.so.2.0 but there is no symbolic link for libsomething.so or libsomething.so.0. Adding a symbolic link worked. I don't know if this is the same kind of situation.

--
Russell Mosemann



From vfclists at googlemail.com  Mon Apr  5 10:00:19 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 5 Apr 2010 18:00:19 +0100
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
Message-ID: <q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>

I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
were completely garbled, even the ringing. It could be affecting
something on the provider end.

Could it be that the provider has a different G729 codec that is not
compatible with the actual G729a the Linksys is sending?



On 5 April 2010 11:59, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Yup, thats why we even have a param called "NDLB-allow-bad-iananame"
> in sofia profiles.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-04-05, at 6:54 AM, Sergey Okhapkin wrote:
>
>> FS looks at codec name too.
>>
>> On Monday 05 April 2010, David Ponzone wrote:
>>> Ognjen,
>>>
>>> very good point, but I used to think that for G729 (and all payload
>>> id
>>> smaller than 97), FS was relying on the payload id, and not the name.
>>>
>>> Am I wrong ?
>>>
>>> David Ponzone ?Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel: ? ? ?01 74 03 18 97
>>> gsm: ? 06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel: ? ? ?0811 46 26 26
>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou
>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
>>> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
>>> pas destinataire de ce message, merci de le d?truire imm?diatement et
>>> d'avertir l'exp?diteur.
>>>
>>> Le 05/04/2010 ? 12:34, Ognjen Seslija a ?crit :
>>>> G729a is invalid as a codec name. FS used to allow it but not
>>>> anymore afaik.
>>>> You should change codec name to G729 (I assume you're using Linksys
>>>> product; there is a setting to change under SIP tab).
>>>>
>>>> Ognjen
>>>>
>>>> On Mon, Apr 5, 2010 at 11:49 AM, Frank Church
>>>> <vfclists at googlemail.com> wrote:
>>>> Have you reviewed http://pastebin.freeswitch.org/12617 ?
>>>> It has the G729 set in the codecs section.
>>>> In this one it seems the call does not get to the external gateway.
>>>> Freeswitch stops the call before calling the external gateway
>>>>
>>>>
>>>> I have checked it again a few times using the bypass_media,
>>>> proxy_media settings.
>>>>
>>>> And with those settings the call ends as soon as ringing starts or
>>>> as
>>>> sonn as the call is answered.
>>>>
>>>> I will do another one just to confirm
>>>>
>>>> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com>
>>>> wrote:
>>>>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>>>>> You just need the regular passthrough module: mod_g729 and to
>>>>
>>>> allow G729 as
>>>>
>>>>> inbound and outbound codecs in vars.xml.
>>>>> To summarize:
>>>>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in
>>>>
>>>> the RTP
>>>>
>>>>> path, it relays the audio stream between endpoints, but can still
>>>>
>>>> detect
>>>>
>>>>> DTMFs
>>>>> -proxy media enabled: FreeSWITCH relays the audio stream
>>>>
>>>> transparently, DTMF
>>>>
>>>>> detection is impossible. In this mode, FS is really a "dumb"
>>>>
>>>> transparent
>>>>
>>>>> RTP-forwarder (this is required to get T38 working between the 2
>>>>
>>>> endpoints)
>>>>
>>>>> -bypass media enabled: FreeSWITCH is not in the RTP path
>>>>> David Ponzone ?Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel: ? ? ?01 74 03 18 97
>>>>> gsm: ? 06 66 98 76 34
>>>>> Service Client IPeva
>>>>> tel: ? ? ?0811 46 26 26
>>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et
>>>>
>>>> ?tablis ?
>>>>
>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>>
>>>> diffusion
>>>>
>>>>> non autoris?e est interdite. Tout message ?lectronique est
>>>>
>>>> susceptible
>>>>
>>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce
>>>>
>>>> message s'il
>>>>
>>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire
>>>>
>>>> de ce
>>>>
>>>>> message, merci de le d?truire imm?diatement et d'avertir
>>>>
>>>> l'exp?diteur.
>>>>
>>>>> Le 04/04/2010 ? 20:12, Max Clark a ?crit :
>>>>>
>>>>> Clarification - for G729 does freeswitch need to be in "bypass
>>>>
>>>> media"
>>>>
>>>>> or "proxy media"? My understanding was that G729 would work with
>>>>> "proxy media" enabled and without the new fangled module?
>>>>>
>>>>> -Max
>>>>>
>>>>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>
>>>>
>>>> wrote:
>>>>> Frank Church <vfclists at googlemail.com> wrote:
>>>>>
>>>>> I am just trialling Freeswitch with Linksys adapters, whose default
>>>>>
>>>>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>>>>
>>>>> When I change that setting to 'yes' the calls don't go through. I
>>>>> am
>>>>>
>>>>> using the latest Windows SVN.
>>>>>
>>>>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with
>>>>
>>>> bypass media
>>>>
>>>>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if
>>>>
>>>> you
>>>>
>>>>> really
>>>>>
>>>>> need it.
>>>>>
>>>>> My recommendation would be to use a codec other than G.729 unless
>>>>
>>>> you have a
>>>>
>>>>> compelling reason, for example a carrier that only supports G.729.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> 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-user
>>>>> s
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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-user
>>>>> s http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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-user
>>>>> s http://www.freeswitch.org
>>>>
>>>> --
>>>> Frank Church
>>>>
>>>> =======================
>>>> http://devblog.brahmancreations.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
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> 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
>>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> 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
>> http://www.freeswitch.org
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From brian at freeswitch.org  Mon Apr  5 10:09:10 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 12:09:10 -0500
Subject: [Freeswitch-users] How to configure g729 pass though on
	Freeswitch
In-Reply-To: <q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
References: <v2g7abab2411004020212ze3af6b6axed960cae28fae35c@mail.gmail.com>
	<n2u4468a6771004050334s3481f92xbb3a9026e7f3737e@mail.gmail.com>
	<31E32D33-7B6B-4EEB-8C13-A27BE765F452@gmail.com>
	<201004050654.11705.sos@sokhapkin.dyndns.org>
	<26A3E189-DDD9-4D20-B66D-3C8ACEADDA6F@avgs.ca>
	<q2k7abab2411004051000pceb439delb97d1470dd3d58ea@mail.gmail.com>
Message-ID: <6602B9C0-96E3-4BC6-A76E-05DD3555D46D@freeswitch.org>

Visit www.freeswitch.org, Click G729 at the top and buy some licenses.  Problem solved ;)  It supports the project and solves a problem all at the same time.

Thanks,
Brian
PS: In depth install instructions are coming shortly for G729.

On Apr 5, 2010, at 12:00 PM, Frank Church wrote:

> I changed the 'G729a Codec Name' in the Linksys to G729 and the calls
> were completely garbled, even the ringing. It could be affecting
> something on the provider end.
> 
> Could it be that the provider has a different G729 codec that is not
> compatible with the actual G729a the Linksys is sending?
> 



From fraserredmond at gmail.com  Mon Apr  5 10:28:17 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 5 Apr 2010 18:28:17 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com> 
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com> 
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
Message-ID: <h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>

Thanks Brian. Sorry, should have done a full sip trace before, but here is
one now: <http://pastebin.freeswitch.org/12632>

Calling an IVR dialplan:
http://pastebin.freeswitch.org/12634

Calling from one extn to another.
http://pastebin.freeswitch.org/12633
(With this one, the source/calling softphone gets a message on it saying put
on hold by the other user - not sure if that helps.)

For what it's worth, at a couple of points when I was running the trace I
was pressing keys to generate dtmf, and nothing changed on the screen - no
activity at all.

Also, I've been able to remote desktop into a computer on another network,
and install x-lite and it can connect to our internal server and works fine,
but it can't do dtmf on the EC2 server either (so it's definitely a problem
on the server end somehow, not my local network's NAT.)

Cheers,
Fraser




On Mon, Apr 5, 2010 at 5:21 PM, Brian West <brian at freeswitch.org> wrote:

> Most likely its in the RTP stream as RFC2833 which is the reason you're not
> getting anything plus I need a FULL sip trace not this abbreviated trace.
>
> /b
>
> On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:
>
> I've taken another stab at this one way audio problem today.
>
> I've run a wireshark capture, and looking at the RTP analysis it only has
> the down-stream, it doesn't record anything being sent upstream at all.
>
> Below is the SIP graph, which shows RTP coming down, but none going up. But
> I don't know enough about SIP to know whether something is missing.
>
> Any suggestions of what I should try now?
>
> Would the dtmf's be sent in the sip packets, or in the rtp?
>
> To preempt the easy answers and save some time:
> - ports are open on EC2 config,
> - iptables turned off for the test,
> - RTP port range uncommented in switch.conf.xml,
> - softphone stun was set to stun.freeswitch.org
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Apr  5 10:34:09 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 12:34:09 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
Message-ID: <7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>

sofia loglevel all 0
sofia profile xx siptrace on

replace xx with profile.  What you have provided is NOT a sip trace.

Thanks,
Brian

On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:

> Thanks Brian. Sorry, should have done a full sip trace before, but here is one now:
> 
> Calling an IVR dialplan:
> http://pastebin.freeswitch.org/12634
> 
> Calling from one extn to another.
> http://pastebin.freeswitch.org/12633
> (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)
> 
> For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.
> 
> Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)
> 
> Cheers,
> Fraser

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From brian at freeswitch.org  Mon Apr  5 10:34:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 12:34:36 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
Message-ID: <A1E647FA-30C1-49FA-9D87-BDE8B353A53F@freeswitch.org>

Also have you checked your firewall?  service iptables stop and test .

/b

On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:

> Thanks Brian. Sorry, should have done a full sip trace before, but here is one now:
> 
> Calling an IVR dialplan:
> http://pastebin.freeswitch.org/12634
> 
> Calling from one extn to another.
> http://pastebin.freeswitch.org/12633
> (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)
> 
> For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.
> 
> Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)
> 
> Cheers,
> Fraser
> 

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From lloyd.aloysius at gmail.com  Mon Apr  5 10:46:14 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 5 Apr 2010 13:46:14 -0400
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
Message-ID: <s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>

Brian,

I reinstall the OS , FreeSWITCH and Setup the configurations.

It is same problem.
.

1. Multi Tenant Setup is not working [ Aastra Phones , 57i Register and
9143is Not Registering and No way to make a calls ] .

But Xlite Working without any problem.

2. When I use the Aasta Phones connecting Default Extension 1000 it is
working.

3. My Local Router LinkSys WRT54GL + Tomato 1.27

Multi Tenant Setup working fine before. Please let me know if you need a SSH
access.

Thanks
Lloyd






On Sun, Apr 4, 2010 at 5:25 PM, Brian West <brian at freeswitch.org> wrote:

> be sure to nuke your configs... cuz we won't when you reinstall.
>
> /b
>
> On Apr 4, 2010, at 4:11 PM, Aloysius Lloyd wrote:
>
> > Thanks Brian.
> >
> > This is simple direct install and no ACL. I am going to Install From
> Scratch.
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
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From 12ukwn at gmail.com  Mon Apr  5 10:53:54 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Mon, 5 Apr 2010 19:53:54 +0200
Subject: [Freeswitch-users] cepstral PB
In-Reply-To: <08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
References: <20100404090838.414c58ce@anubis.defcon1>
	<009A194C-F8AD-43C2-AD23-BB890DDA12BC@freeswitch.org>
	<20100405150724.214af8a2@anubis.defcon1>
	<769C4F3CB97C48CC8D3C3AE2ED5059D9@cune.pri>
	<4BBA0EAE.4020003@gmail.com>
	<08A6C95C326A4ABE8284E8C066E2E5E6@cune.pri>
Message-ID: <20100405195354.376ea693@anubis.defcon1>

Le Mon, 5 Apr 2010 11:44:06 -0500,
"Russell Mosemann" <Russell.Mosemann at cune.org> a ?crit :

> When you say that the libs are in the system libs cache, do you mean that
> the libs are configured in /etc/ld.conf (or /etc/ld.conf.d/, as
> appropriate), and when you enter "ldconfig -v", the libs are listed in
> the output under their home directory (was it /opt/something?)?

Yep (/etc/ld.so.conf.d/CEPSTRAL.conf, containing '/opt/swift/lib' +
ldconfig), and what is even stranger is that it was like that for a week,
and the last rebuild put everything back in place (I just tested and removed
the symlinks.)

> On rare occasion, I have seen software not find a library for some
> strange reason when there is a libsomething.so.2.0 but there is no
> symbolic link for libsomething.so or libsomething.so.0. Adding a symbolic
> link worked. I don't know if this is the same kind of situation.

May be, may be not, on rare occasions I've seen uncanny interactions into
Linux (but tons less than in m$ (all alpha) products:)

Thanks anyway!

> Jean-Yves F. Barbier wrote:
> > >> so I asked here; may be I should reformulate my question: why having
> > >> to symlink all ceptsral libs into /usr/local/freswitch/lib while
> > >> already having them cached in the system libs cache
> > >> (/etc/ld.so.cache)?
 

-- 
Old programmers never die, they just branch to a new address.


From fraserredmond at gmail.com  Mon Apr  5 10:53:41 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 5 Apr 2010 18:53:41 +0100
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com> 
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com> 
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
	<7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
Message-ID: <r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>

Ok, sorry, try this:
http://pastebin.freeswitch.org/12635

And, yes, as I said in my earlier message, the firewall is off - checked it
over and over again :-)

Amazon EC2 has it's own firewall system as well, which can't be turned off,
but I've set it to enable all ports (0-65535)

Cheers,
Fraser




On Mon, Apr 5, 2010 at 6:34 PM, Brian West <brian at freeswitch.org> wrote:

> sofia loglevel all 0
> sofia profile xx siptrace on
>
> replace xx with profile.  What you have provided is NOT a sip trace.
>
> Thanks,
> Brian
>
> On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:
>
> Thanks Brian. Sorry, should have done a full sip trace before, but here is
> one now: <http://pastebin.freeswitch.org/12632>
>
> Calling an IVR dialplan:
> http://pastebin.freeswitch.org/12634
>
> Calling from one extn to another.
> http://pastebin.freeswitch.org/12633
> (With this one, the source/calling softphone gets a message on it saying
> put on hold by the other user - not sure if that helps.)
>
> For what it's worth, at a couple of points when I was running the trace I
> was pressing keys to generate dtmf, and nothing changed on the screen - no
> activity at all.
>
> Also, I've been able to remote desktop into a computer on another network,
> and install x-lite and it can connect to our internal server and works fine,
> but it can't do dtmf on the EC2 server either (so it's definitely a problem
> on the server end somehow, not my local network's NAT.)
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> 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
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Apr  5 10:56:45 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 12:56:45 -0500
Subject: [Freeswitch-users] Registrations stop working for all NAT
	phones on FreeSWITCH Version 1.0.trunk (17177)
In-Reply-To: <s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
References: <v2n13ca621c1004020243nc702d9e4k9d79e04bf60056f2@mail.gmail.com>
	<y2p8a19bf2e1004041215se9efe4e6w623a33515f2c9af7@mail.gmail.com>
	<l2l8a19bf2e1004041219hcacb8f41r68925a40207bc54@mail.gmail.com>
	<1C66D627-72CB-4808-AA5F-F974DDE6A351@freeswitch.org>
	<t2m8a19bf2e1004041305kf447ed8cl1411933c52b8ab79@mail.gmail.com>
	<69E55F79-9CB8-45BC-BFE1-FD25425AF186@freeswitch.org>
	<w2q8a19bf2e1004041327i8b5cd40fr8657eb9acd837b16@mail.gmail.com>
	<B89953AC-780A-4C41-989A-603A220FBC35@freeswitch.org>
	<z2k8a19bf2e1004041411ubd8371bds5997545ae49f84fd@mail.gmail.com>
	<E61C9D52-6599-474A-AC1F-76F68C77559E@freeswitch.org>
	<s2p8a19bf2e1004051046y5705d08cybe4393ae9b438a8e@mail.gmail.com>
Message-ID: <A4891738-E7CA-4656-BB59-E69B70758C5D@freeswitch.org>

Then you're config is clearly wrong.

/b

On Apr 5, 2010, at 12:46 PM, Aloysius Lloyd wrote:

> Brian,
> 
> I reinstall the OS , FreeSWITCH and Setup the configurations.
> 
> It is same problem.
> .
> 
> 1. Multi Tenant Setup is not working [ Aastra Phones , 57i Register and 9143is Not Registering and No way to make a calls ] . 
> 
> But Xlite Working without any problem.
> 
> 2. When I use the Aasta Phones connecting Default Extension 1000 it is working.
> 
> 3. My Local Router LinkSys WRT54GL + Tomato 1.27
> 
> Multi Tenant Setup working fine before. Please let me know if you need a SSH access.
> 
> Thanks
> Lloyd
> 
> 



From brian at freeswitch.org  Mon Apr  5 11:08:16 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 5 Apr 2010 13:08:16 -0500
Subject: [Freeswitch-users] No audio/dtmf from softphone behind NAT
In-Reply-To: <r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>
References: <w2lc32ee1cf1004030849ybeb9c6ebr7e7ebb3b23e48087@mail.gmail.com>
	<j2xc32ee1cf1004050912md7345545sea39866fca3074d@mail.gmail.com>
	<4387FF74-FDB8-4602-A34F-5BC8CCA48DCF@freeswitch.org>
	<h2ic32ee1cf1004051028o1b4a4dedufb0b556f162c98f2@mail.gmail.com>
	<7C8E0491-F9D7-4E4B-B31C-B2FE6A6B12EE@freeswitch.org>
	<r2mc32ee1cf1004051053pc936651bgf95e3ecc88200f6b@mail.gmail.com>
Message-ID: <19F7C568-0391-4F6F-82AA-F4745F9BCDCB@freeswitch.org>

While its valid to ha