Re: [Freeswitch-users] reload from managed

[srinivasula reddy]

i tried switch_api_execute its working fine, thanks rupa

On Tue, Oct 27, 2009 at 8:48 PM, Rupa Schomaker  wrote:

> Is it necessary for you to call the reload function directly?  Why not
> make an api call using switch_api_execute and the command "reloadxml".
>
>
> On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy
>  wrote:
> > Hi,
> >
> > how can i call reload from managed code? and i called
> > Csharp_switch_xml_open_root, this is working first time, then onwards it
> is
> > not working.
> > is this the correct function to call reload?
> >
> > --
> > Srinivasula Reddy K
> >
> > ___
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> >
> >
>
>
>
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>
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Re: [Freeswitch-users] switch_xml_open_root

[Michael Jerris]

just api execute reloadxml

Mike

On Oct 28, 2009, at 12:37 AM, srinivasula reddy wrote:

> Hi mike,
>
> thank for your reply.
> i am trying to call that function from swig.cs.   its working fine  
> first time with the warning information, then onwards it is not  
> working.
> is there other way can i call reload function from  
> freeswitch.managed code. any help
>

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[Freeswitch-users] Retrieve conference member state using cli?

[Lon Baker]

Just like you can call conference list for conferences, is there a way to
retrieve the profile state of a conference member using the cli or xml/rpc?

Lon
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Re: [Freeswitch-users] switch_xml_open_root

[srinivasula reddy]

Hi mike,

thank for your reply.
i am trying to call that function from swig.cs.   its working fine first
time with the warning information, then onwards it is not working.
is there other way can i call reload function from freeswitch.managed code.
any help

srinivas



On Wed, Oct 28, 2009 at 3:01 AM, Michael Jerris  wrote:

> You probably should not be calling that function, what are you trying to
> do?
>
> Mike
>
> On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
>
> *
> *
> Hi,
>
> when i am calling  switch_xml_open_root(1,err) . i am getting this warning
> message.
> HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 0016,
> 100E9FD0 ). can any know. please help me.
>
>
>
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Re: [Freeswitch-users] how to config FS with two net interface?

[Brian West]

Ok then something is broken badly... and that makes NO sense.  Because  
it works on my box.


/b

On Oct 27, 2009, at 8:34 PM, Lei Tang wrote:

Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when  I  
set the ext-rtp-ip, it works fine.


2009/10/28 Brian West 
you should have just needed to set the rtp-ip and the sip-ip on both
profiles to their values and it would have worked fine... their would
have been no need to set the ext-*-ip equiv.

/b

On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:

> Thanks Eliot, It works.


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Re: [Freeswitch-users] how to config FS with two net interface?

[Lei Tang]

Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when  I set the
ext-rtp-ip, it works fine.

2009/10/28 Brian West 

> you should have just needed to set the rtp-ip and the sip-ip on both
> profiles to their values and it would have worked fine... their would
> have been no need to set the ext-*-ip equiv.
>
> /b
>
> On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:
>
> > Thanks Eliot, It works.
>
>
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Re: [Freeswitch-users] how to config FS with two net interface?

[Brian West]

you should have just needed to set the rtp-ip and the sip-ip on both  
profiles to their values and it would have worked fine... their would  
have been no need to set the ext-*-ip equiv.

/b

On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:

> Thanks Eliot, It works.


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Re: [Freeswitch-users] how to config FS with two net interface?

[Lei Tang]

Thanks Eliot, It works.

2009/10/27 Eliot Gable

>

> Try setting ext-rtp-ip and ext-sip-ip on both profiles.
>
> On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang  wrote:
> > Hi all, I run FS on a machine with two net interface, each interface has
> a
> > ip addr, one of the them connect to public network(has ip addr A), the
> > other  connect to a private network(has ip addr B), FS server as a SIP
> > server for public through A, all outbound call will bridge to a
> softswitch
> > in private network through B. here is my sofia config file and diaplan
> > config:
> >
> > sofia internal.xml
> > 
> > 
> > 
> >  
> >
> > sofia external.xml
> > 
> > 
> > 
> > 
> >
> > dialplan
> > ..
> > 
> > 
> > 
> >  >
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
> >  > data="effective_caller_id_number=xxx"/>  
> >  > data="sofia/external/${destination_numb...@x"/>
> >   
> > 
> > .
> >
> > then call seq is
> > sipAgent --> [internal -->(bridge)-->external] -->softswith
> >   FREESWITCH
> >
> > the question is, when sipAgent make a outbound call, FS can't recevie the
> > caller's up audio stream, I traced the SIP packets, found that FS has
> return
> > addr B in SDP when ack the invite request from sipAgent, the ack packet
> is
> > ===
> > SIP/2.0 183 Session Progress
> > Via: SIP/2.0/UDP
> >
> x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> > From: "1000" ;tag=cb4d3c4e
> > To: "65960581" ;tag=DtvSc0QX01yKN
> > Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> > CSeq: 2 INVITE
> > Contact: 
> > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> > Accept: application/sdp
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY,
> > REFER, UPDATE, REGISTER, INFO, PUBLISH
> > Supported: timer, precondition, path, replaces
> > Allow-Events: talk, presence, dialog, call-info, sla,
> > include-session-description, presence.winfo, message-summary, refer
> > Content-Type: application/sdp
> > Content-Disposition: session
> > Content-Length: 245
> >
> > v=0
> > o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;wrong this is the ip
> addr
> > of the adapter connect to the private network
> > s=FreeSWITCH
> > c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
> > private network
> > t=0 0
> > m=audio 31066 RTP/AVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > 
> > I think FS should return A in SDP, not the external binding addr (B),
> does
> > somebody known how to solve this problem?
> >
> > --
> > Lei.Tang
> > lei.tl...@gmail.com
> >
> > ___
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> >
> >
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect

[Georgiewskiy Yuriy]

On 2009-10-23 15:14 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:


TC>
TC> so the real solution is to implement a check for CallProceeding , Progress
TC>and Facility message whether it has a faststart element included. It it is
TC>true than you might start pre_answer.
TC>
TC>
TC>also, i don't see any handling of Call Proceeding ... what if there is a
TC>fastStart element in CallProceeding message? :)

Handling of fastStart in CallProceeding is commented out in h323plus library,
this is exploration from h323plus developers about this:


Yes that should be mera.

The problem is that Callproceeding does not always come from the remote it
may be generated by the gatekeeper. MERA where sending fast start elements
in the Call proceeding and connect. The call proceeding where not valid and
causing the media to fail. Normally (although valid) EP's do not set Fast
Start in Call proceeding so the code was disabled to resolve the MERA issue.

if you wont read "bugs" file in mod_h323, there is explaned how to enable it.

C уважением   With Best Regards
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Re: [Freeswitch-users] switch_xml_open_root

[Michael Collins]

On Tue, Oct 27, 2009 at 2:31 PM, Michael Jerris  wrote:

> You probably should not be calling that function, what are you trying to
> do?
>
> Mike
>
> On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:
>

You might also want to move this conversation to the -dev list because it's
a little intense for the -users list.
-MC
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Re: [Freeswitch-users] switch_xml_open_root

[Michael Jerris]

You probably should not be calling that function, what are you trying  
to do?


Mike

On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote:




Hi,

when i am calling  switch_xml_open_root(1,err) . i am getting this  
warning message.
HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap 
( 0016, 100E9FD0 ). can any know. please help me.


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Re: [Freeswitch-users] FS Training

[Michael Collins]

On Tue, Oct 27, 2009 at 12:17 PM, Jerry Richards  wrote:

>
> Did the voting booth close?  I was unable to vote.  I'm not sure what link
> to click and I have had some strange issues with my FS account today.
>
> I would be interested in paid training.  Do you have plans for offering a
> training session at your locale?  Or would you travel onsite to provide
> training?
>
> Jerry,

Yes the poll closed last Thursday. I'm glad to hear that you are interested
in paid training. We are still evaluating all the options. I will be
contacting people off list to discuss specifics. This is still in the
incubation stage, so no ETA on anything just yet.

-MC
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Re: [Freeswitch-users] Estimating Call Capacity

[Shelby Ramsey]

Cliff,

Try using xml_rpc ... status or show channels will give you what you need.

SDR

Cliff Wells wrote:
> A little off-topic, but since call-capacity is the subject, what are
> people using to analyze their CDR's to discover this?   I'm handling
> about 30k calls per day but have only a bandwidth-based guesstimate of
> the peak number of concurrent calls I'm handling.
>
> If there's an open source solution, I'd appreciate a pointer.
>
> Regards,
> Cliff
>
> On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
>   
>> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
>> server from FS with FS on a Dell R710 with their current best
>> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
>> GB memory. The web app server is less than half the power of the R710.
>> I maxed the web app server at 300 calls per second (both setting up
>> and tearing down) and the R710 running FS was 65% idle. No audio was
>> being proxied through FS, though. If I were running the web app server
>> on an equivalent R710, they probably would have been on-par with each
>> other in performance. Extrapolating, I expect that in such a case I
>> should be able to get at least 650 CPS out of FS, though for
>> production I would probably limit it to 400 CPS or less so I leave
>> room for miscellaneous tasks. I maxed out the R710 at over 16,000
>> simultaneous calls (again, no audio proxying) but the only reason I
>> couldn't do more was because I hit some sort of thread creation limit
>> in Linux. There was about 17 GB of memory used for this many calls.
>> This should give you some ballpark idea of what you can accomplish
>> with FS.
>>
>> At some point, I will track down and resolve the thread creation
>> issue, at which time I believe call limits will be limited either by a
>> complex combination of available memory, the speed of the processor,
>> the cost of thread context switching, calls per second setup rate, and
>> call duration.
>>
>> --
>> Eliot Gable
>>
>> 
>>> -Original Message-
>>>
>>> From: freeswitch-users-boun...@lists.freeswitch.org 
>>> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of 
>>> Giovanni Maruzzelli
>>>
>>> Sent: Monday, October 26, 2009 4:56 PM
>>>
>>> To: freeswitch-users@lists.freeswitch.org
>>>
>>> Subject: Re: [Freeswitch-users] Estimating Call Capacity
>>>
>>>
>>>
>>> On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
>>>
>>>  wrote:
>>>
>>>   
 Here are a few benchmarks that I had stumbled upon.
 
 http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
 
>>>
>>> Please remember NO benchmarks are endorsed by the FS community or
>>>
>>> developers, because there are just too many variables, and a simple
>>>
>>> figure is just useful for marketing hype, not for real dimensioning.
>>>
>>>
>>>
>>> You MUST do your own benchmarking, so you get an idea about how to
>>>
>>> dimension for your own use case and hardware.
>>>
>>>
>>>
>>>
>>>
>>>   
 Thanks,
 
 Vinuth.
 
 On Tue, Oct 27, 2009 at 1:43 AM, Brian West  wrote:
 
> I highly doubt it... You can wait for someone to post their results
>   
> but in the end you'll have to do your own load testing because not
>   
> everyone's numbers will jive with your use case.  Which is the reason
>   
> the project never posts or endorses a set call count.
>   
> /b
>   
> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
>   
>> Are there any benchmarking test results available publicly?
>> 
>> 
>> 
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>>>
>>>
>>>
>>>
>>>
>>> --
>>>
>>> Sincerely,
>>>
>>>
>>>
>>> Giovanni Maruzzelli
>>>
>>> Cell : +39-347-2665618
>>>
>>>
>>>
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Re: [Freeswitch-users] Estimating Call Capacity

[Gregory Boehnlein]

I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
the mirrors, tracking all sorts of good data.

You can install it on a Centos box, and get a free trial.

http://www.manageengine.com/products/vqmanager/index.html

What is really cool is that it actually monitors the RTP/RTCP as well as all
of the SIP headers and archives the calls, so you can look at calls from
several days ago and see EXACTLY what happened on them. I have used this
extensively to pinpoint bad Level 3 and X/O media gateways.. Much better
than trying to sniff packets in real-time and MAYBE catch a problem..

I've also used it to find/fix several SIP issues w/ odd endpoints.. Very
easy to see..

> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-
> users-boun...@lists.freeswitch.org] On Behalf Of Cliff Wells
> Sent: Tuesday, October 27, 2009 3:52 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Estimating Call Capacity
> 
> A little off-topic, but since call-capacity is the subject, what are
> people using to analyze their CDR's to discover this?   I'm handling
> about 30k calls per day but have only a bandwidth-based guesstimate of
> the peak number of concurrent calls I'm handling.
> 
> If there's an open source solution, I'd appreciate a pointer.
> 
> Regards,
> Cliff
> 
> On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> > Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> > server from FS with FS on a Dell R710 with their current best
> > processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and
> 32
> > GB memory. The web app server is less than half the power of the
> R710.
> > I maxed the web app server at 300 calls per second (both setting up
> > and tearing down) and the R710 running FS was 65% idle. No audio was
> > being proxied through FS, though. If I were running the web app
> server
> > on an equivalent R710, they probably would have been on-par with each
> > other in performance. Extrapolating, I expect that in such a case I
> > should be able to get at least 650 CPS out of FS, though for
> > production I would probably limit it to 400 CPS or less so I leave
> > room for miscellaneous tasks. I maxed out the R710 at over 16,000
> > simultaneous calls (again, no audio proxying) but the only reason I
> > couldn't do more was because I hit some sort of thread creation limit
> > in Linux. There was about 17 GB of memory used for this many calls.
> > This should give you some ballpark idea of what you can accomplish
> > with FS.
> >
> > At some point, I will track down and resolve the thread creation
> > issue, at which time I believe call limits will be limited either by
> a
> > complex combination of available memory, the speed of the processor,
> > the cost of thread context switching, calls per second setup rate,
> and
> > call duration.
> >
> > --
> > Eliot Gable
> >
> > > -Original Message-
> > >
> > > From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> Giovanni Maruzzelli
> > >
> > > Sent: Monday, October 26, 2009 4:56 PM
> > >
> > > To: freeswitch-users@lists.freeswitch.org
> > >
> > > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> > >
> > >
> > >
> > > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> > >
> > >  wrote:
> > >
> > > > Here are a few benchmarks that I had stumbled upon.
> > >
> > > >
> http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+
> x2200+M2
> > >
> > >
> > >
> > > Please remember NO benchmarks are endorsed by the FS community or
> > >
> > > developers, because there are just too many variables, and a simple
> > >
> > > figure is just useful for marketing hype, not for real
> dimensioning.
> > >
> > >
> > >
> > > You MUST do your own benchmarking, so you get an idea about how to
> > >
> > > dimension for your own use case and hardware.
> > >
> > >
> > >
> > >
> > >
> > > > Thanks,
> > >
> > > > Vinuth.
> > >
> > > >
> > >
> > > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West
>  wrote:
> > >
> > > >>
> > >
> > > >> I highly doubt it... You can wait for someone to post their
> results
> > >
> > > >> but in the end you'll have to do your own load testing because
> not
> > >
> > > >> everyone's numbers will jive with your use case.  Which is the
> reason
> > >
> > > >> the project never posts or endorses a set call count.
> > >
> > > >>
> > >
> > > >> /b
> > >
> > > >>
> > >
> > > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> > >
> > > >>
> > >
> > > >> > Are there any benchmarking test results available publicly?
> > >
> > > >> > 
> > >
> > > >>
> > >
> > > >>
> > >
> > > >> ___
> > >
> > > >> FreeSWITCH-users mailing list
> > >
> > > >> FreeSWITCH-users@lists.freeswitch.org
> > >
> > > >>

Re: [Freeswitch-users] Estimating Call Capacity

[Cliff Wells]

A little off-topic, but since call-capacity is the subject, what are
people using to analyze their CDR's to discover this?   I'm handling
about 30k calls per day but have only a bandwidth-based guesstimate of
the peak number of concurrent calls I'm handling.

If there's an open source solution, I'd appreciate a pointer.

Regards,
Cliff

On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote:
> Although, FYI, I just benchmarked mod_xml_curl on a separate web app
> server from FS with FS on a Dell R710 with their current best
> processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32
> GB memory. The web app server is less than half the power of the R710.
> I maxed the web app server at 300 calls per second (both setting up
> and tearing down) and the R710 running FS was 65% idle. No audio was
> being proxied through FS, though. If I were running the web app server
> on an equivalent R710, they probably would have been on-par with each
> other in performance. Extrapolating, I expect that in such a case I
> should be able to get at least 650 CPS out of FS, though for
> production I would probably limit it to 400 CPS or less so I leave
> room for miscellaneous tasks. I maxed out the R710 at over 16,000
> simultaneous calls (again, no audio proxying) but the only reason I
> couldn't do more was because I hit some sort of thread creation limit
> in Linux. There was about 17 GB of memory used for this many calls.
> This should give you some ballpark idea of what you can accomplish
> with FS.
> 
> At some point, I will track down and resolve the thread creation
> issue, at which time I believe call limits will be limited either by a
> complex combination of available memory, the speed of the processor,
> the cost of thread context switching, calls per second setup rate, and
> call duration.
> 
> --
> Eliot Gable
> 
> > -Original Message-
> >
> > From: freeswitch-users-boun...@lists.freeswitch.org 
> > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of 
> > Giovanni Maruzzelli
> >
> > Sent: Monday, October 26, 2009 4:56 PM
> >
> > To: freeswitch-users@lists.freeswitch.org
> >
> > Subject: Re: [Freeswitch-users] Estimating Call Capacity
> >
> >
> >
> > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur
> >
> >  wrote:
> >
> > > Here are a few benchmarks that I had stumbled upon.
> >
> > > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2
> >
> >
> >
> > Please remember NO benchmarks are endorsed by the FS community or
> >
> > developers, because there are just too many variables, and a simple
> >
> > figure is just useful for marketing hype, not for real dimensioning.
> >
> >
> >
> > You MUST do your own benchmarking, so you get an idea about how to
> >
> > dimension for your own use case and hardware.
> >
> >
> >
> >
> >
> > > Thanks,
> >
> > > Vinuth.
> >
> > >
> >
> > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West  wrote:
> >
> > >>
> >
> > >> I highly doubt it... You can wait for someone to post their results
> >
> > >> but in the end you'll have to do your own load testing because not
> >
> > >> everyone's numbers will jive with your use case.  Which is the reason
> >
> > >> the project never posts or endorses a set call count.
> >
> > >>
> >
> > >> /b
> >
> > >>
> >
> > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote:
> >
> > >>
> >
> > >> > Are there any benchmarking test results available publicly?
> >
> > >> > 
> >
> > >>
> >
> > >>
> >
> > >> ___
> >
> > >> FreeSWITCH-users mailing list
> >
> > >> FreeSWITCH-users@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@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,
> >
> >
> >
> > Giovanni Maruzzelli
> >
> > Cell : +39-347-2665618
> >
> >
> >
> > ___
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users@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@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://www.google.com/sea

[Freeswitch-users] FS Training

[Jerry Richards]

Did the voting booth close?  I was unable to vote.  I'm not sure what link
to click and I have had some strange issues with my FS account today.

I would be interested in paid training.  Do you have plans for offering a
training session at your locale?  Or would you travel onsite to provide
training?

Best Regards,
Jerry


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Re: [Freeswitch-users] Forcing endpoint registration

[Brian West]

From FreeSWITCH you can't force and endpoint to register... thats the  
endpoints job... what you're doing is just flushing the reg out of  
FreeSWITCH so now it can't find the endpoint...  Try adding a reboot  
on there maybe i'll reboot the device and cause it to register again...


/b

On Oct 27, 2009, at 12:33 PM, Lars Zeb wrote:

Is the following the correct command to force registration of an  
endpoint below which is not showing up in ‘sofia status profile  
internal’ currently, but when it does it looks like the following?


Call-ID:mjkg8c3zlqwjzfd.hrt5
User:   1...@192.168.10.29
Contact:"user" 
Agent:  snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1)
Status: Registered(UDP)(unknown) EXP(2009-10-27 10:46:07)
Host:   fs
IP: 192.168.10.103
Port:   5060
Auth-User:  1019
Auth-Realm: 192.168.10.29
MWI-Account:1...@192.168.10.29

sofia profile internal  flush_inbound_reg 1...@192.168.10.29

Thanks Lars


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[Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda - Oct 30th

[Michael Collins]

Greetings all!

I just wanted to say thank you to those who have been contributing to the
documentation and cleanup efforts. Please keep up the good work. FYI, we
still have lots of janitorial stuff to do. Just keep checking the latest
conf call agenda for updates. This week's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30

We are always looking for people to help out with documentation and other
little projects so please feel free to ask me if you would like to assist.
Also, we are working on having members of the community give brief
presentations during the weekly conference calls. We want people to join the
conference and share with the group information about how they're using
FreeSWITCH in production, or they can give little tutorials on how to use
various parts of FS. We are also working on getting a web-cast enabled so
that we can have a little video to go with the audio. If you have something
you'd like to discuss with group please let me know.

Thanks for being such a great community!
-Michael
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[Freeswitch-users] Forcing endpoint registration

[Lars Zeb]

Is the following the correct command to force registration of an endpoint
below which is not showing up in 'sofia status profile internal' currently,
but when it does it looks like the following?

 

Call-ID:mjkg8c3zlqwjzfd.hrt5

User:   1...@192.168.10.29

Contact:"user" 

Agent:  snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1)

Status: Registered(UDP)(unknown) EXP(2009-10-27 10:46:07)

Host:   fs

IP: 192.168.10.103

Port:   5060

Auth-User:  1019

Auth-Realm: 192.168.10.29

MWI-Account:1...@192.168.10.29

 

sofia profile internal  flush_inbound_reg  
1...@192.168.10.29

 

Thanks Lars

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Re: [Freeswitch-users] SIP provider with extern rtp server

[Eliot Gable]

No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can you send a Wireshark capture taken on the
FreeSWITCH server of both call legs? Or, if you can, pastebin a debug
log from FreeSWITCH console with sofia loglevel set to 9 and siptrace
on for any Sofia SIP profiles involved.

On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold  wrote:
> The server is on a public IP, so there is no nat issue here.
>
> I can also see the rtp messages on wireshark starting just after the
> 183 Session Progress message on the server, but just in one direction,
> coming in to the server.
> So it looks like Freeswitch is stopping the rtp.
> Is this because the rtp originates from another ip than the  sip
> provider ip?
>
> Ivan
>
> Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:
>
>> Make sure you let their media IPs through your firewall. Also, if you
>> are behind a NAT, check you have things passing to the correct
>> internal address.
>>
>> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold 
>> wrote:
>>> I have used a SIP provider for more than a year. A few days ago, he
>>> said he was moving to a new server, and asked me to reconfigure. I
>>> did, and everything seemed to work fine, until I did an outgoing call
>>> to an external telephone. I found out I had no audio, in neither
>>> direction. Incoming calls was working fine.
>>>
>>> My provider said that the rtp is not going through the sip server, as
>>> it did earlier, but now through several other IP's.
>>>
>>> Do I have to do some special configuration to handle that?
>>>
>>> Ivan
>>>
>>> ___
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Eliot Gable
>>
>> "We do not inherit the Earth from our ancestors: we borrow it from our
>> children." ~David Brower
>>
>> "I decided the words were too conservative for me. We're not borrowing
>> from our children, we're stealing from them--and it's not even
>> considered to be a crime." ~David Brower
>>
>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
>> live; not live to eat.) ~Marcus Tullius Cicero
>>
>> ___
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users@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
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] Problem with inbound call answered but no sound

[Brian West]

Let me review that and see I can't off hand think of anything that  
would cause that.

/b

On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:


Thanks for the reply, Brian.

Did something in FS change between v15183 and v15225 to make this  
occur? I ask because this same configuration worked OK in the  
earlier version.


Lars


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Re: [Freeswitch-users] Problem with inbound call answered but no sound

[Lars Zeb]

Thanks for the reply, Brian.

 

Did something in FS change between v15183 and v15225 to make this occur? I
ask because this same configuration worked OK in the earlier version.

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, October 27, 2009 8:26 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem with inbound call answered but no
sound

 

The issue is you're not behind nat-pmp or upnp so it can't figure out what
your public IP is... you'll have to fill that in manually or enable those
options on your router if possible.

 

/b

 

On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:





I haven't changed anything since v15183, where it worked OK.

 

In conf/sip_profiles/external:

 

 

 

 

And

 

 

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Re: [Freeswitch-users] SIP provider with extern rtp server

[Ivan C Myrvold]

The server is on a public IP, so there is no nat issue here.

I can also see the rtp messages on wireshark starting just after the  
183 Session Progress message on the server, but just in one direction,  
coming in to the server.
So it looks like Freeswitch is stopping the rtp.
Is this because the rtp originates from another ip than the  sip  
provider ip?

Ivan

Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable:

> Make sure you let their media IPs through your firewall. Also, if you
> are behind a NAT, check you have things passing to the correct
> internal address.
>
> On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold   
> wrote:
>> I have used a SIP provider for more than a year. A few days ago, he
>> said he was moving to a new server, and asked me to reconfigure. I
>> did, and everything seemed to work fine, until I did an outgoing call
>> to an external telephone. I found out I had no audio, in neither
>> direction. Incoming calls was working fine.
>>
>> My provider said that the rtp is not going through the sip server, as
>> it did earlier, but now through several other IP's.
>>
>> Do I have to do some special configuration to handle that?
>>
>> Ivan
>>
>> ___
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> ___
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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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Re: [Freeswitch-users] Problem with inbound call answered but no sound

[Brian West]

The issue is you're not behind nat-pmp or upnp so it can't figure out  
what your public IP is... you'll have to fill that in manually or  
enable those options on your router if possible.


/b

On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:


I haven’t changed anything since v15183, where it worked OK.

In conf/sip_profiles/external:

 
 

And



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Re: [Freeswitch-users] reload from managed

[Rupa Schomaker]

Is it necessary for you to call the reload function directly?  Why not
make an api call using switch_api_execute and the command "reloadxml".


On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy
 wrote:
> Hi,
>
> how can i call reload from managed code? and i called
> Csharp_switch_xml_open_root, this is working first time, then onwards it is
> not working.
> is this the correct function to call reload?
>
> --
> Srinivasula Reddy K
>
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>
>



-- 
-Rupa

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Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."

[Anthony Minessale]

won't compile or won't run?
maybe you should try rebuilding it.


On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz <
maciej.aniserow...@gmail.com> wrote:

> Sorry, trunk does not compile on win7, here are the details:
>
>
> rev.15247
>
> ---
> Microsoft Visual C++ Debug Library
> ---
> Debug Assertion Failed!
>
> Program: ...ev\Projects\External\FreeSWITCH\Original\Debug\FreeSwitch.exe
> File: f:\dd\vctools\crt_bld\self_x86\crt\src\dbgheap.c
> Line: 1317
>
> Expression: _CrtIsValidHeapPointer(pUserData)
>
> For information on how your program can cause an assertion
> failure, see the Visual C++ documentation on asserts.
>
> (Press Retry to debug the application)
> ---
> Abort   Retry   Ignore
> ---
>
> VS Call stack:
>
>   ntdll.dll!77ccfadc()
>   [Frames below may be incorrect and/or missing, no symbols loaded for
> ntdll.dll]
>   ntdll.dll!77c9272c()
>   ntdll.dll!77c5e1ef()
>   msvcr90d.dll!_free_dbg_nolock(void * pUserData=0x00664b88, int
> nBlockUse=1)  Line 1317 + 0x9 bytes C++
>   msvcr90d.dll!_free_dbg(void * pUserData=0x00664b88, int nBlockUse=1)
> Line 1258 + 0xd bytes C++
>   msvcr90d.dll!free(void * pUserData=0x00664b88)  Line 49 + 0xb bytes C++
> > FreeSwitch.dll!switch_core_mime_add_type(const char * type=0x003bcd1c,
> const char * ext=0x003bcd37)  Line 748 + 0xc bytes C
>   FreeSwitch.dll!load_mime_types()  Line 791 C
>   FreeSwitch.dll!switch_core_init(unsigned int flags=65, switch_bool_t
> console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1244 C
>   FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=65,
> switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1454
> + 0x11 bytes C
>   FreeSwitch.exe!main(int argc=1, char * * argv=0x02144c40)  Line 764 +
> 0x23 bytes C
>   FreeSwitch.exe!__tmainCRTStartup()  Line 586 + 0x19 bytes C
>   FreeSwitch.exe!mainCRTStartup()  Line 403 C
>   kernel32.dll!77713677()
>   ntdll.dll!77c39d72()
>   ntdll.dll!77c39d45()
>
>
> Error occurs in :
>
> SWITCH_DECLARE(switch_status_t) switch_core_mime_add_type(const char *type,
> const char *ext)
> {
>  const char *check;
>  switch_status_t status = SWITCH_STATUS_FALSE;
>
>  switch_assert(type);
>  switch_assert(ext);
>
>  check = (const char *) switch_core_hash_find(runtime.mime_types, ext);
>
>  if (!check) {
>   char *ptype = switch_core_permanent_strdup(type);
>   char *ext_list = strdup(ext);
>   int argc = 0;
>   char *argv[20] = { 0 };
>   int x;
>
>   switch_assert(ext_list);
>
>   if ((argc = switch_separate_string(ext_list, ' ', argv, (sizeof(argv) /
> sizeof(argv[0]) {
>
>for (x = 0; x < argc; x++) {
> if (argv[x] && ptype) {
>  switch_core_hash_insert(runtime.mime_types, argv[x], ptype);
> }
>}
>
>status = SWITCH_STATUS_SUCCESS;
>   }
>
>   free(ext_list);  // <--- HERE
>  }
>
>  return status;
> }
>
>
>
> - Original Message -
> *From:* [hidden 
> email]
> *To:* [hidden 
> email]
> *Sent:* Monday, October 26, 2009 10:32 PM
> *Subject:* Re: [Freeswitch-users] "Can not record session. Media not
> enabled on channel."
>
>
>
> On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden 
> email]
> > wrote:
>
>>
>> Yes, I can confirm - this exact error occurs each time when I start
>> recording
>> before the call is answered (just after sending ORIGINATE command) - but I
>> think that's completely understandable that media is not ready for an
>> unanswered call.
>> But... is there any other event that guarantees media to be ready?
>>
>> Update to latest SVN and try again.
> -MC
>
>
> ___
> FreeSWITCH-users mailing list
> [hidden 
> email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> View this message in context: Re: [Freeswitch-users] "Can not record
> session. Media not enabled on 
> channel."
>
> Sent from the freeswitch-users mailing list 
> archiveat Nabble.com.
>
> ___
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>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_

Re: [Freeswitch-users] "Can not record session. Media not enabled on channel."

[Maciej Aniserowicz]

Sorry, trunk does not compile on win7, here are the details:


rev.15247

---
Microsoft Visual C++ Debug Library
---
Debug Assertion Failed!

Program: ...ev\Projects\External\FreeSWITCH\Original\Debug\FreeSwitch.exe
File: f:\dd\vctools\crt_bld\self_x86\crt\src\dbgheap.c
Line: 1317

Expression: _CrtIsValidHeapPointer(pUserData)

For information on how your program can cause an assertion
failure, see the Visual C++ documentation on asserts.

(Press Retry to debug the application)
---
Abort   Retry   Ignore   
---

VS Call stack:

  ntdll.dll!77ccfadc()  
  [Frames below may be incorrect and/or missing, no symbols loaded for 
ntdll.dll] 
  ntdll.dll!77c9272c()  
  ntdll.dll!77c5e1ef()  
  msvcr90d.dll!_free_dbg_nolock(void * pUserData=0x00664b88, int nBlockUse=1)  
Line 1317 + 0x9 bytes C++
  msvcr90d.dll!_free_dbg(void * pUserData=0x00664b88, int nBlockUse=1)  Line 
1258 + 0xd bytes C++
  msvcr90d.dll!free(void * pUserData=0x00664b88)  Line 49 + 0xb bytes C++
> FreeSwitch.dll!switch_core_mime_add_type(const char * type=0x003bcd1c, const 
> char * ext=0x003bcd37)  Line 748 + 0xc bytes C
  FreeSwitch.dll!load_mime_types()  Line 791 C
  FreeSwitch.dll!switch_core_init(unsigned int flags=65, switch_bool_t 
console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1244 C
  FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=65, 
switch_bool_t console=SWITCH_TRUE, const char * * err=0x003bf68c)  Line 1454 + 
0x11 bytes C
  FreeSwitch.exe!main(int argc=1, char * * argv=0x02144c40)  Line 764 + 0x23 
bytes C
  FreeSwitch.exe!__tmainCRTStartup()  Line 586 + 0x19 bytes C
  FreeSwitch.exe!mainCRTStartup()  Line 403 C
  kernel32.dll!77713677()  
  ntdll.dll!77c39d72()  
  ntdll.dll!77c39d45()  


Error occurs in :

SWITCH_DECLARE(switch_status_t) switch_core_mime_add_type(const char *type, 
const char *ext)
{
 const char *check;
 switch_status_t status = SWITCH_STATUS_FALSE;

 switch_assert(type);
 switch_assert(ext);

 check = (const char *) switch_core_hash_find(runtime.mime_types, ext);

 if (!check) {
  char *ptype = switch_core_permanent_strdup(type);
  char *ext_list = strdup(ext);
  int argc = 0;
  char *argv[20] = { 0 };
  int x;

  switch_assert(ext_list);

  if ((argc = switch_separate_string(ext_list, ' ', argv, (sizeof(argv) / 
sizeof(argv[0]) {

   for (x = 0; x < argc; x++) {
if (argv[x] && ptype) {
 switch_core_hash_insert(runtime.mime_types, argv[x], ptype);
}
   }

   status = SWITCH_STATUS_SUCCESS;
  }

  free(ext_list);  // <--- HERE
 }

 return status;
}


  - Original Message - 
  From: mercutioviz [via freeswitch-users] 
  To: Maciej Aniserowicz 
  Sent: Monday, October 26, 2009 10:32 PM
  Subject: Re: [Freeswitch-users] "Can not record session. Media not enabled on 
channel."





  On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz <[hidden email]> wrote:


Yes, I can confirm - this exact error occurs each time when I start 
recording
before the call is answered (just after sending ORIGINATE command) - but I
think that's completely understandable that media is not ready for an
unanswered call.
But... is there any other event that guarantees media to be ready?



  Update to latest SVN and try again.
  -MC



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http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3895104.html
 
  To unsubscribe from Re: "Can not record session. Media not enabled on 
channel.", click here. 


-- 
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http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] SIP provider with extern rtp server

[Eliot Gable]

Make sure you let their media IPs through your firewall. Also, if you
are behind a NAT, check you have things passing to the correct
internal address.

On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold  wrote:
> I have used a SIP provider for more than a year. A few days ago, he
> said he was moving to a new server, and asked me to reconfigure. I
> did, and everything seemed to work fine, until I did an outgoing call
> to an external telephone. I found out I had no audio, in neither
> direction. Incoming calls was working fine.
>
> My provider said that the rtp is not going through the sip server, as
> it did earlier, but now through several other IP's.
>
> Do I have to do some special configuration to handle that?
>
> Ivan
>
> ___
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>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] how to config FS with two net interface?

[Eliot Gable]

Try setting ext-rtp-ip and ext-sip-ip on both profiles.

On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang  wrote:
> Hi all, I run FS on a machine with two net interface, each interface has a
> ip addr, one of the them connect to public network(has ip addr A), the
> other  connect to a private network(has ip addr B), FS server as a SIP
> server for public through A, all outbound call will bridge to a softswitch
> in private network through B. here is my sofia config file and diaplan
> config:
>
> sofia internal.xml
> 
> 
> 
>  
>
> sofia external.xml
> 
> 
> 
> 
>
> dialplan
> ..
> 
>     
>     
>      data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
>      data="effective_caller_id_number=xxx"/>  
>          data="sofia/external/${destination_numb...@x"/>
>   
>     
> .
>
> then call seq is
> sipAgent --> [internal -->(bridge)-->external] -->softswith
>   FREESWITCH
>
> the question is, when sipAgent make a outbound call, FS can't recevie the
> caller's up audio stream, I traced the SIP packets, found that FS has return
> addr B in SDP when ack the invite request from sipAgent, the ack packet is
> ===
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP
> x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> From: "1000" ;tag=cb4d3c4e
> To: "65960581" ;tag=DtvSc0QX01yKN
> Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> CSeq: 2 INVITE
> Contact: 
> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
> REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 245
>
> v=0
> o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;wrong this is the ip addr
> of the adapter connect to the private network
> s=FreeSWITCH
> c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
> private network
> t=0 0
> m=audio 31066 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> 
> I think FS should return A in SDP, not the external binding addr (B), does
> somebody known how to solve this problem?
>
> --
> Lei.Tang
> lei.tl...@gmail.com
>
> ___
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>



-- 
Eliot Gable

"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower

"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower

"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero

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Re: [Freeswitch-users] Playing Background music as well as a file

[lakshmanan ganapathy]

I tried the uuid_displace by using the Event Socket Outbound.
Not tried within dialplan, and I'm not sure whether that could be done in
dialplan itself.


On Tue, Oct 27, 2009 at 12:37 PM, Dome Charoenyost  wrote:

> 2009/10/27 lakshmanan ganapathy :
> > Hi all,
> > I've done experimenting with the uuid_displace and mux. What mux does
> is
> > playing a file when the conversation is also happening. But I've a
> different
> > requirement.
>
> Is posible to use uuid_displace in dialplan ?  i want to do music
> background base on called id
>
> > I need to play a background music to a UUID, that will get played
> > continuously and also I need to play some other voice message to that
> uuid.
> >
> > Is it possible in freeswitch? If so please guide me on how to do that!
> >
> >
> >
> > ___
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@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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[Freeswitch-users] switch_xml_open_root

[srinivasula reddy]

**

Hi,

when i am calling  switch_xml_open_root(1,err) . i am getting this warning
message.
HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 0016,
100E9FD0 ). can any know. please help me.

Thanks
Srinivasula Reddy K
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[Freeswitch-users] reload from managed

[srinivasula reddy]

Hi,

how can i call reload from managed code? and i called
Csharp_switch_xml_open_root, this is working first time, then onwards it is
not working.
is this the correct function to call reload?

-- 
Srinivasula Reddy K
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[Freeswitch-users] how to config FS with two net interface?

[Lei Tang]

Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other  connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a softswitch
in private network through B. here is my sofia config file and diaplan
config:

sofia internal.xml



 

sofia external.xml





dialplan
..




  

  

.

then call seq is
sipAgent --> [internal -->(bridge)-->external] -->softswith
  FREESWITCH

the question is, when sipAgent make a outbound call, FS can't recevie the
caller's up audio stream, I traced the SIP packets, found that FS has return
addr B in SDP when ack the invite request from sipAgent, the ack packet is
===
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
From: "1000" ;tag=cb4d3c4e
To: "65960581" ;tag=DtvSc0QX01yKN
Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
CSeq: 2 INVITE
Contact: 
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 245

v=0
o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;wrong this is the ip addr
of the adapter connect to the private network
s=FreeSWITCH
c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
private network
t=0 0
m=audio 31066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

I think FS should return A in SDP, not the external binding addr (B), does
somebody known how to solve this problem?

-- 
Lei.Tang
lei.tl...@gmail.com
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Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

[Peter Olsson]

Thanks Brian, it works now. I'll try to learn to say NO next time :)

/Peter


-Ursprungligt meddelande-
Från: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West
Skickat: den 26 oktober 2009 19:40
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

At some point we'll have to NO NO NO fix your broken crap.  :P  The  
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with  
me... "NO!"

/b

On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:

> I understand your frustration :) We deal with SIP integration with  
> about 10 different PBX vendors today, And it's always something that  
> doesn't work as it should. Right now I don't have anything more  
> connected to FS though.
>
> /Peter


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!DSPAM:4ae5ef4a32936548940180!


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Re: [Freeswitch-users] How to load user account from databse ?

[Henry Huang]

Sorry, guys, I need to fix my earlier statement about xml_odbc. I just
talked to my developer and xml_odbc IS loading registered user on real time,
not from the beginning of the FS start.

So Lei Tang, you can use xml_odbc as well

On Tue, Oct 27, 2009 at 2:41 PM, Lei Tang  wrote:

> Hi  noob and Michael, thanks for your answers,  I'll try to use
> mod_xml_curl.
> Hi  Henry,  http://wiki.freeswitch.org/wiki/Mod_xml_curl has mentioned, FS
> will post a request to webserver when it get a registration request. you can
> refer to the doc for more detail.
>
>
>
>
>
> Lei.Tang
> lei.tl...@gmail.com
>
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>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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Re: [Freeswitch-users] Playing Background music as well as a file

[Dome Charoenyost]

2009/10/27 lakshmanan ganapathy :
> Hi all,
>     I've done experimenting with the uuid_displace and mux. What mux does is
> playing a file when the conversation is also happening. But I've a different
> requirement.

Is posible to use uuid_displace in dialplan ?  i want to do music
background base on called id

> I need to play a background music to a UUID, that will get played
> continuously and also I need to play some other voice message to that uuid.
>
> Is it possible in freeswitch? If so please guide me on how to do that!
>
>
>
> ___
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>

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