Re: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user

2009-03-24 Thread Raul Fragoso 
You should use the id value of the user, not accountcode:

user_exists id gigaset toyos.nl

Regards,

Raul

On Tue, 2009-03-24 at 15:55 +0100, Leon de Rooij wrote:
> Hi,
> 
> I'm trying to get some cli commands working in combination with xml- 
> curl.
> 
> Endpoints are parsed properly for SIP registrations and invites, but  
> when I use the CLI command "user_exists" it returns false, while I do  
> return an  endpoint (same syntax as for a sofia_reg_parse_auth event)  
> on the webserver.
> 
> Should switch_xml_locate_user event receive a different syntax ?
> 
> freeswi...@internal> user_exists accountcode gigaset toyos.nl
> false
> 
> XML returned from webserver:
> 
> 
> 
> 
>
>  
>
>   value="a211336e29c4f3756e6af343ce6da27b"/>
>  
>  
>  
>
>
>  
>  
>  
>  
>  
>  
>  
>  
>  
>
>  
>
> 
> 
> 
> thanks & regards,
> 
> Leon
> 
> 
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Re: [Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user

2009-03-24 Thread Mathieu Rene 
As Raul said, the user_exists function will look for a user  
*attribute* so unless you write your user as , you need to use id (and the later wouldnt set  
the accountcode variable, no)

Math

On 25-Mar-09, at 1:53 AM, Raul Fragoso wrote:

> You should use the id value of the user, not accountcode:
>
> user_exists id gigaset toyos.nl
>
> Regards,
>
> Raul
>
> On Tue, 2009-03-24 at 15:55 +0100, Leon de Rooij wrote:
>> Hi,
>>
>> I'm trying to get some cli commands working in combination with xml-
>> curl.
>>
>> Endpoints are parsed properly for SIP registrations and invites, but
>> when I use the CLI command "user_exists" it returns false, while I do
>> return an  endpoint (same syntax as for a sofia_reg_parse_auth event)
>> on the webserver.
>>
>> Should switch_xml_locate_user event receive a different syntax ?
>>
>> freeswi...@internal> user_exists accountcode gigaset toyos.nl
>> false
>>
>> XML returned from webserver:
>>
>> 
>> 
>> 
>>   
>> 
>>   
>> > value="a211336e29c4f3756e6af343ce6da27b"/>
>> 
>> 
>> 
>>   
>>   
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>>   
>> 
>>   
>> 
>> 
>>
>> thanks & regards,
>>
>> Leon
>>
>>
>> ___
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>
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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Mitul Limbani 
MC,

I don't have much exp with mediawiki but have some with drupal.

Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/


On 25-Mar-09, at 0:35, Michael Collins  wrote:

> 2009/3/24 Mitul Limbani :
>> Brian,
>> I can help with website, wiki and testing, tell me what's next step  
>> forward.
>>
>
> Mitul, do you have any experience with MediaWiki?
> -MC
>
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[Freeswitch-users] ANNOUNCEMENT: ClueCon 2009 - Register Today!

2009-03-24 Thread Michael Collins 
The FreeSWITCH Team is pleased to announce that all may register for
ClueCon 2009 immediately!

ClueCon is the Telephony Developers Conference by developers, for
developers. This year's event will be held at the beautiful Wyndham
Hotel in Chicago, August 4-6. Special room rates have been secured and
you will enjoy staying at this luxurious hotel. The conference
attendee price is only $499 per person, but hurry! This early bird
special price is valid only until June 30th.

The schedule is still being finalized, however you can expect to hear
from notable names in the world of telephony. You will also have
opportunities to spend one-on-one time with developers, vendors, and
other attendees. One special highlight for this year is the great
MacBook Pro giveaway! On Thursday at the end of the conference a 15"
MacBook Pro will be given away to one lucky attendee. This laptop will
be laser engraved with the logos of all of the sponsors for ClueCon
2009 and will be used as the presenters' laptop for the entire
conference.

Start making your travel plans right away. Visit
http://www.cluecon.com and sign up today. We look forward to seeing
you in Chicago this August!

Michael S Collins
877-742-CLUE

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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Brian West 

On Mar 24, 2009, at 7:47 PM, dujinfang wrote:

>
> On Mar 25, 2009, at 2:42 AM, Brian West wrote:
>> How about we all work together and work on the FreeSWITCH.org  
>> infrastructure instead of spreading the resources thinner and  
>> thinner till nobody is doing really much of anything.
>>
>
> I agree with this and would like to work with the  
> wiki.freeswitch.org on the EN and CN language part. However, I don't  
> think we can maintain a Chinese version of IRC, seems still we need  
> a Chinese BBS.

Someone tried to do a spanish IRC channel it doesn't have anyone in it  
but me most of the time... hrm.. I'm not trying to discount  
anything here but it seems language specific anything seems to not  
take off.

We need more cheerleaders btw cluecon is coming up remember to  
register early! ;)


>> We need people to step up and help out with the website, wiki,  
>> jira, irc, testing and various other things that currently are  
>> spread thin.  ;)  Please!!!  I'm not asking everyone to step up and  
>> code in C...
>>
>
> Take me in :)

What do you want to be involved in?  #freeswitch-web is the correct  
channel to be in if you want to help out on IRC.  ;)


>
>> /b


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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread dujinfang 


On Mar 25, 2009, at 2:42 AM, Brian West wrote:

How about we all work together and work on the FreeSWITCH.org  
infrastructure instead of spreading the resources thinner and  
thinner till nobody is doing really much of anything.




I agree with this and would like to work with the wiki.freeswitch.org  
on the EN and CN language part. However, I don't think we can maintain  
a Chinese version of IRC, seems still we need a Chinese BBS.



We need people to step up and help out with the website, wiki, jira,  
irc, testing and various other things that currently are spread  
thin.  ;)  Please!!!  I'm not asking everyone to step up and code in  
C...




Take me in :)


/b

On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote:


Hello,

I can provide you with the hosting on our box in US let me know.

Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/



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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread dujinfang 

On Mar 25, 2009, at 2:03 AM, Michael Collins wrote:

>> While FS official site has plenty of documentation, obviously we  
>> don't
>> want to translate word by word.  Then where should we start from? A
>> forum?
>
> I would start by finding as many people as possible who are literate
> in both English and Chinese and who are willing to help out. Once you
> have your group assembled you will have a better idea of what your
> goals should be. More people helping will make things easier to
> accomplish.

Ppl not only literate in both English and Chinse, but also need to has  
good knowledge of VoIP and FreeSWITCH.


> There was a big argue between a forum and mailing list in the
>> last few weeks, and finally an English forum and an Italian one is
>> out.  While I can find server in China, the major pain is that  
>> running
>> any kind of BBS in China mainland need to get some kind of  
>> permissions
>> by the government first.
>>
>> Any idea, suggestion? Anyone want to help or cooperate about this?
>
> I am looking into creating a multi-language wiki at
> wiki.freeswitch.org. Anyone with experience in setting up multiple
> languages with MediaWiki software please contact me. So far we have
> people willing to help create documentation in French, Spanish,
> Portugese, Russian, and now Chinese. I think we have some Italians out
> there as well! (ciao bella)
>
> Please email me off list if you are in a position to assist with the
> wiki and languages other than English.
>

How's the plan of a multi-language wiki?  I think mediaWiKi do support  
multi-language. So, maybe we can start from wiki.freeswitch.org/zh_CN/  
or some structure else. While it is possible to follow the change log  
of the EN wiki to change Chinese version accordingly, it would be
challenge to make sure the consistency. And regardlessly what if  
Chinese version change first?

I'd like to help with this. But I cannot translate all hundreds of  
pages and make it consistency by my self. But I'm pretty sure that we  
would get a group of ppl to do this. So the start maybe either  
translate some key pages and make the consistency, or translate as  
much pages ...

Email me off list on how to start this. :)

>>
>> And, if I can get this runing, can I use the FreeSWITCH logo and 
>> www.freeswitch.org.cn
>>  ?
>
> Please direct questions about the FreeSWITCH logo and domain names to
> consult...@freeswitch.org. FreeSWITCH and the logo are trademarks so
> it would be best to get permission from Anthony before doing anything.
>
> -MC
>
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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Michael Collins 
2009/3/24 Mitul Limbani :
> Brian,
> I can help with website, wiki and testing, tell me what's next step forward.
>

Mitul, do you have any experience with MediaWiki?
-MC

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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Mitul Limbani 

Brian,

I can help with website, wiki and testing, tell me what's next step  
forward.


Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/


On 25-Mar-09, at 0:12, Brian West  wrote:

How about we all work together and work on the FreeSWITCH.org  
infrastructure instead of spreading the resources thinner and  
thinner till nobody is doing really much of anything.


We need people to step up and help out with the website, wiki, jira,  
irc, testing and various other things that currently are spread  
thin.  ;)  Please!!!  I'm not asking everyone to step up and code in  
C...


/b

On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote:


Hello,

I can provide you with the hosting on our box in US let me know.

Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/



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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward 
Ah ha!

Thanks for finding that.

I updated my * server and I'm all set.

Many thanks for all the feedback and help.

2009/3/24 Andy Spitzer 

> Woof!
>
> Appears to be a recently fixed * bug:
>
> 0014431: Bad branch parameter value in CANCEL request
> http://bugs.digium.com/view.php?id=14431
>
> --Woof!
>
>
>
>
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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Brian West 
How about we all work together and work on the FreeSWITCH.org  
infrastructure instead of spreading the resources thinner and thinner  
till nobody is doing really much of anything.


We need people to step up and help out with the website, wiki, jira,  
irc, testing and various other things that currently are spread  
thin.  ;)  Please!!!  I'm not asking everyone to step up and code in  
C...


/b

On Mar 24, 2009, at 1:32 PM, Mitul Limbani wrote:


Hello,

I can provide you with the hosting on our box in US let me know.

Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/



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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Mitul Limbani 
Hello,

I can provide you with the hosting on our box in US let me know.

Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt Ltd,
The Enterprise Linux Company(r),
http://www.enterux.com/


On 24-Mar-09, at 23:00, dujinfang  wrote:

> Hi all,
>
> In about a year of playing with FreeSWITCH, I really like it. Not only
> the software is great but also the community. As people keep asking me
> where can find some documentation in Chinese, I told them if they want
> to play in deep they need to know English better. However, the fact
> is, even one can read English without problem, he still prefer find
> some information in his native language first. So, I'd like to start a
> Chinese community hoping FreeSWITCH can get more popular  in China. As
> we saw big telecom companies merged in the last year, I bet VoIP
> technology and application will go more and more faster in the near
> future...
>
> While FS official site has plenty of documentation, obviously we don't
> want to translate word by word.  Then where should we start from? A
> forum? There was a big argue between a forum and mailing list in the
> last few weeks, and finally an English forum and an Italian one is
> out.  While I can find server in China, the major pain is that running
> any kind of BBS in China mainland need to get some kind of permissions
> by the government first.
>
> Any idea, suggestion? Anyone want to help or cooperate about this?
>
> And, if I can get this runing, can I use the FreeSWITCH logo and 
> www.freeswitch.org.cn
>  ?
>
> Best regards,
> Seven.
>
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Re: [Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread Michael Collins 
> While FS official site has plenty of documentation, obviously we don't
> want to translate word by word.  Then where should we start from? A
> forum?

I would start by finding as many people as possible who are literate
in both English and Chinese and who are willing to help out. Once you
have your group assembled you will have a better idea of what your
goals should be. More people helping will make things easier to
accomplish.

There was a big argue between a forum and mailing list in the
> last few weeks, and finally an English forum and an Italian one is
> out.  While I can find server in China, the major pain is that running
> any kind of BBS in China mainland need to get some kind of permissions
> by the government first.
>
> Any idea, suggestion? Anyone want to help or cooperate about this?

I am looking into creating a multi-language wiki at
wiki.freeswitch.org. Anyone with experience in setting up multiple
languages with MediaWiki software please contact me. So far we have
people willing to help create documentation in French, Spanish,
Portugese, Russian, and now Chinese. I think we have some Italians out
there as well! (ciao bella)

Please email me off list if you are in a position to assist with the
wiki and languages other than English.

>
> And, if I can get this runing, can I use the FreeSWITCH logo and 
> www.freeswitch.org.cn
>  ?

Please direct questions about the FreeSWITCH logo and domain names to
consult...@freeswitch.org. FreeSWITCH and the logo are trademarks so
it would be best to get permission from Anthony before doing anything.

-MC

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[Freeswitch-users] FreeSWITCH Chinese Community

2009-03-24 Thread dujinfang 
Hi all,

In about a year of playing with FreeSWITCH, I really like it. Not only  
the software is great but also the community. As people keep asking me  
where can find some documentation in Chinese, I told them if they want  
to play in deep they need to know English better. However, the fact  
is, even one can read English without problem, he still prefer find  
some information in his native language first. So, I'd like to start a  
Chinese community hoping FreeSWITCH can get more popular  in China. As  
we saw big telecom companies merged in the last year, I bet VoIP  
technology and application will go more and more faster in the near  
future...

While FS official site has plenty of documentation, obviously we don't  
want to translate word by word.  Then where should we start from? A  
forum? There was a big argue between a forum and mailing list in the  
last few weeks, and finally an English forum and an Italian one is  
out.  While I can find server in China, the major pain is that running  
any kind of BBS in China mainland need to get some kind of permissions  
by the government first.

Any idea, suggestion? Anyone want to help or cooperate about this?

And, if I can get this runing, can I use the FreeSWITCH logo and 
www.freeswitch.org.cn 
  ?

Best regards,
Seven.

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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Andy Spitzer 
Woof!

Appears to be a recently fixed * bug:

0014431: Bad branch parameter value in CANCEL request
http://bugs.digium.com/view.php?id=14431

--Woof!




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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward 
Mike,

Thanks for taking the time to look at this - I appreciate it.  I'll go ahead
and test it out on the current svn trunk.

- Steve

2009/3/24 Michael Jerris 

> This appears to be a bug in FreeSWITCH.  Can you please test this on
> current svn trunk and if it is still a problem, please report this as a bug
> to http://jira.freeswitch.org.
> MIke
>
>  On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote:
>
>  I note that its missing the to tag from the 180 sent 5 seconds earlier (I
> think thats okay) but the via branch tag is also different, which seems
> wrong.  Can anyone else chime in, I can't recall the dialog matching rules
> of early dialog like this.
> Mike
>
>  On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:
>
>  Here it is:
>
> freeswi...@b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at
> 13:53:07.644865:
>
>OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
>Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
>From: "Unknown" 
> >;tag=as5adee8f4
>To: 
>Contact: >
>Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
>CSeq: 102 OPTIONS
>User-Agent: Asterisk PBX
>Max-Forwards: 70
>Date: Tue, 24 Mar 2009 13:53:07 GMT
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>Supported: replaces
>Content-Length: 0
>
> send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:
>
>SIP/2.0 200 OK
>Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
>From: "Unknown" 
> >;tag=as5adee8f4
>To: ;tag=DytraHp3K84aD
>Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
>CSeq: 102 OPTIONS
>Contact: 
>User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
>Accept: application/sdp
>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>Supported: 100rel, timer, precondition, path, replaces
>Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>Content-Length: 0
>
> recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:
>
>INVITE 
> sip:70...@b-pbx-lab-1.mynet.netSIP/2.0
>Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
>From: "Steve" 
> >;tag=as4863e49a
>To: 
> >
>Contact: >
>Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
>CSeq: 102 INVITE
>User-Agent: Asterisk PBX
>Max-Forwards: 70
>Date: Tue, 24 Mar 2009 13:53:11 GMT
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>Supported: replaces
>Content-Type: application/sdp
>Content-Length: 258
>v=0
>o=root 4756 4756 IN IP4 10.1.21.44
>s=session
>c=IN IP4 10.1.21.44
>t=0 0
>m=audio 17956 RTP/AVP 0 8 101
>a=rtpmap:0 PCMU/8000
>a=rtpmap:8 PCMA/8000
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-16
>a=silenceSupp:off - - - -
>a=ptime:20
>a=sendrecv
>
> send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:
>
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
>From: "Steve" 
> >;tag=as4863e49a
>To: 
> >
>Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
>CSeq: 102 INVITE
>User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
>Content-Length: 0
>
> send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:
>
>SIP/2.0 407 Proxy Authentication Required
>Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
>From: "Steve" 
> >;tag=as4863e49a
>To: 
> >;tag=e7KHcc76gHUXr
>Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
>CSeq: 102 INVITE
>User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
>Accept: application/sdp
>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>Supported: 100rel, timer, precondition, path, replaces
>Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>Proxy-Authenticate: Digest realm="10.1.21.44",
> nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth"
>Content-Length: 0
>
> recv 407 bytes from udp/[10.1.21.

Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Michael Jerris 
This appears to be a bug in FreeSWITCH.  Can you please test this on  
current svn trunk and if it is still a problem, please report this as  
a bug to http://jira.freeswitch.org.


MIke

On Mar 24, 2009, at 10:54 AM, Michael Jerris wrote:

I note that its missing the to tag from the 180 sent 5 seconds  
earlier (I think thats okay) but the via branch tag is also  
different, which seems wrong.  Can anyone else chime in, I can't  
recall the dialog matching rules of early dialog like this.


Mike

On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:


Here it is:

freeswi...@b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060  
at 13:53:07.644865:



   OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
   From: "Unknown" ;tag=as5adee8f4
   To: 
   Contact: 
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:07 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Length: 0



send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:



   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
   From: "Unknown" ;tag=as5adee8f4
   To: ;tag=DytraHp3K84aD
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   Contact: 
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH

   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include- 
session-description, presence.winfo, message-summary, refer

   Content-Length: 0



recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:



   INVITE sip:70...@b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" ;tag=as4863e49a
   To: 
   Contact: 
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4756 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv



send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:



   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" ;tag=as4863e49a
   To: 
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0



send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:



   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" ;tag=as4863e49a
   To: ;tag=e7KHcc76gHUXr
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH

   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include- 
session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="10.1.21.44",  
nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5,  
qop="auth"

   Content-Length: 0



recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:



   ACK sip:70...@b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" ;tag=as4863e49a
   To: ;tag=e7KHcc76gHUXr
   Contact: 
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102

[Freeswitch-users] XML (curl) returned for Event-Calling-Function = switch_xml_locate_user

2009-03-24 Thread Leon de Rooij 
Hi,

I'm trying to get some cli commands working in combination with xml- 
curl.

Endpoints are parsed properly for SIP registrations and invites, but  
when I use the CLI command "user_exists" it returns false, while I do  
return an  endpoint (same syntax as for a sofia_reg_parse_auth event)  
on the webserver.

Should switch_xml_locate_user event receive a different syntax ?

freeswi...@internal> user_exists accountcode gigaset toyos.nl
false

XML returned from webserver:




   
 
   
 
 
 
 
   
   
 
 
 
 
 
 
 
 
 
   
 
   



thanks & regards,

Leon


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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Michael Jerris 
I note that its missing the to tag from the 180 sent 5 seconds earlier  
(I think thats okay) but the via branch tag is also different, which  
seems wrong.  Can anyone else chime in, I can't recall the dialog  
matching rules of early dialog like this.


Mike

On Mar 24, 2009, at 9:57 AM, Steven Ward wrote:


Here it is:

freeswi...@b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at  
13:53:07.644865:



   OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
   From: "Unknown" ;tag=as5adee8f4
   To: 
   Contact: 
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:07 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Length: 0



send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:



   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
   From: "Unknown" ;tag=as5adee8f4
   To: ;tag=DytraHp3K84aD
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   Contact: 
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH

   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include- 
session-description, presence.winfo, message-summary, refer

   Content-Length: 0



recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:



   INVITE sip:70...@b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" ;tag=as4863e49a
   To: 
   Contact: 
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4756 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv



send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:



   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" ;tag=as4863e49a
   To: 
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0



send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:



   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" ;tag=as4863e49a
   To: ;tag=e7KHcc76gHUXr
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH

   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include- 
session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="10.1.21.44",  
nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5,  
qop="auth"

   Content-Length: 0



recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:



   ACK sip:70...@b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" ;tag=as4863e49a
   To: ;tag=e7KHcc76gHUXr
   Contact: 
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 ACK
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0



recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306:

-

Re: [Freeswitch-users] Sip for Skype - g.729 requirement

2009-03-24 Thread Derrick Edwards 
 It seems to me that they are just offering SIP trunking like
voicepulse. Very interested to see what there rates would be like since
they have a large infrastructure. 



On Tue, 2009-03-24 at 08:54 -0400, David Sugar wrote:
> They require one use g.729, which is patent encumbered as well as rather
> computationally intensive.
> 
> Dan wrote:
> > You probably already saw this but
> > 
> > http://www.skypeforsip.com/
> > 
> > Skype is supporting sip for business users.
> > 
> > 
> > 
> > 
> > ___
> > 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
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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Even André Fiskvik 

Please see documentation at:
http://wiki.freeswitch.org/wiki/Fs_cli
and
http://wiki.freeswitch.org/wiki/Mod_commands


Best regards,
Even André

On 24. mars. 2009, at 15.23, Will Smith wrote:


Thank you all for your help.
I use : /usr/local/freeswitch/bin/fs_cli  to open a FS instance. But  
then cannot use "shutdown" command. Then I used 'fsctl shutdown  
asap'  (directed by Szymon). It works perfectly.


Again, thank you. Have a great day you all.

Will

--- On Tue, 3/24/09, Leon de Rooij  wrote:
From: Leon de Rooij 
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:47 AM

Just make sure you have mod_event socket loaded in conf/ 
autoload_modules/modules.conf.xml :




And have it configured in conf/autoload_modules/event_socket.conf.xml

Then you can use bin/fs_cli to connect to running FS instance.

I don't think you can reconnect to a process of which you have  
disconnected the terminal (without using screen)..


regards,

Leon

On Mar 24, 2009, at 2:35 PM, Will Smith wrote:


No, I started FS with this:
/usr/local/freeswitch/bin/freeswitch . What can I do in this case ?

thank you for help

--- On Tue, 3/24/09, Saeed Ahmed  wrote:
From: Saeed Ahmed 
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:25 AM

Did you started FS with –nc option?
with this option you can connect to FS using ./fs_cli
OR
use screen!

From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Will Smith

Sent: Tuesday, March 24, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Console Window


Hi,

I closed the console window that ran FS without shutting down FS. I  
had to restart the server to get back to the console window. I just  
wonder if there is a way to bring that window up without restart  
server. In the file /log/freeswitch.pid, I found a number, is that  
a seesion id?




Thank you.


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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Szymon Olko 
Will Smith pisze:
> Thank you all for your help.
> I use : /usr/local/freeswitch/bin/fs_cli  to open a FS instance. But
> then cannot use "shutdown" command. Then I used 'fsctl shutdown asap' 
> (directed by Szymon). It works perfectly.
>  
> Again, thank you. Have a great day you all.
>  
Remeber there are other flags to shutdown freeswitch

freeswi...@vertux> fsctl
API CALL [fsctl()] output:
-USAGE: [send_sighup|hupall|pause|resume|shutdown
[cancel|elegant|asap|restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration
 [num]|loglevel [level]]

Look for descriptions in wiki.

> Will
> 
> --- On *Tue, 3/24/09, Leon de Rooij //* wrote:
> 
> From: Leon de Rooij 
> Subject: Re: [Freeswitch-users] Console Window
> To: freeswitch-users@lists.freeswitch.org
> Date: Tuesday, March 24, 2009, 6:47 AM
> 
> Just make sure you have mod_event socket loaded in
> conf/autoload_modules/modules.conf.xml :
> 
> 
> 
> And have it configured in conf/autoload_modules/event_socket.conf.xml
> 
> Then you can use bin/fs_cli to connect to running FS instance.
> 
> I don't think you can reconnect to a process of which you have
> disconnected the terminal (without using screen)..
> 
> regards,
> 
> Leon 
> 
> On Mar 24, 2009, at 2:35 PM, Will Smith wrote:
> 
>> No, I started FS with this:
>> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ?
>>  
>> thank you for help
>>
>> --- On *Tue, 3/24/09, Saeed Ahmed /> >/* wrote:
>>
>> From: Saeed Ahmed > >
>> Subject: Re: [Freeswitch-users] Console Window
>> To: freeswitch-users@lists.freeswitch.org
>> 
>> Date: Tuesday, March 24, 2009, 6:25 AM
>>
>> Did you started FS with –nc option?
>> with this option you can connect to FS using ./fs_cli
>> OR 
>> use /screen/!
>>
>> 
>> 
>>
>> *From:* freeswitch-users-boun...@lists.freeswitch.org
>> 
>> [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On
>> Behalf Of *Will Smith
>> *Sent:* Tuesday, March 24, 2009 1:52 PM
>> *To:* freeswitch-users@lists.freeswitch.org
>> 
>> *Subject:* [Freeswitch-users] Console Window
>>
>>  
>>
>> Hi,
>>
>> I closed the console window that ran FS without shutting down
>> FS. I had to restart the server to get back to the console
>> window. I just wonder if there is a way to bring that window
>> up without restart server. In the file /log/freeswitch.pid, I
>> found a number, is that a seesion id?
>>
>>  
>>
>> Thank you.
>>
>>  
>>
>> ___
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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Diego Viola 
I put /usr/local/freeswitch/bin in my path, like this.

export PATH=/usr/local/freeswitch/bin:$PATH on my ~/.bashrc, you can
put it on your /etc/profile if you want it to be global.

Then I just do `freeswitch -nc' when I need to start it, `fs_cli' to
connect, and `fsctl shutdown asap' to shut it down, or `freeswitch
-stop'.

Hope that helps.

Diego

2009/3/24 Will Smith :
> Thank you all for your help.
> I use : /usr/local/freeswitch/bin/fs_cli  to open a FS instance. But then
> cannot use "shutdown" command. Then I used 'fsctl shutdown asap'  (directed
> by Szymon). It works perfectly.
>
> Again, thank you. Have a great day you all.
>
> Will
>
> --- On Tue, 3/24/09, Leon de Rooij  wrote:
>
> From: Leon de Rooij 
> Subject: Re: [Freeswitch-users] Console Window
> To: freeswitch-users@lists.freeswitch.org
> Date: Tuesday, March 24, 2009, 6:47 AM
>
> Just make sure you have mod_event socket loaded in
> conf/autoload_modules/modules.conf.xml :
>     
> And have it configured in conf/autoload_modules/event_socket.conf.xml
> Then you can use bin/fs_cli to connect to running FS instance.
> I don't think you can reconnect to a process of which you have disconnected
> the terminal (without using screen)..
> regards,
> Leon
> On Mar 24, 2009, at 2:35 PM, Will Smith wrote:
>
> No, I started FS with this:
> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ?
>
> thank you for help
>
> --- On Tue, 3/24/09, Saeed Ahmed  wrote:
>
> From: Saeed Ahmed 
> Subject: Re: [Freeswitch-users] Console Window
> To: freeswitch-us...@lists.freeswitch.org
> Date: Tuesday, March 24, 2009, 6:25 AM
>
> Did you started FS with –nc option?
> with this option you can connect to FS using ./fs_cli
> OR
> use screen!
>
> 
>
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] on Behalf Of Will
> Smith
> Sent: Tuesday, March 24, 2009 1:52 PM
> To: freeswitch-us...@lists.freeswitch.org
> Subject: [Freeswitch-users] Console Window
>
>
>
> Hi,
>
> I closed the console window that ran FS without shutting down FS. I had to
> restart the server to get back to the console window. I just wonder if there
> is a way to bring that window up without restart server. In the file
> /log/freeswitch.pid, I found a number, is that a seesion id?
>
>
>
> Thank you.
>
>
>
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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Will Smith 
Thank you all for your help.
I use : /usr/local/freeswitch/bin/fs_cli  to open a FS instance. But then 
cannot use "shutdown" command. Then I used 'fsctl shutdown asap'  (directed by 
Szymon). It works perfectly.
 
Again, thank you. Have a great day you all.
 
Will

--- On Tue, 3/24/09, Leon de Rooij  wrote:

From: Leon de Rooij 
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:47 AM


Just make sure you have mod_event socket loaded in 
conf/autoload_modules/modules.conf.xml : 



    


And have it configured in conf/autoload_modules/event_socket.conf.xml


Then you can use bin/fs_cli to connect to running FS instance.


I don't think you can reconnect to a process of which you have disconnected the 
terminal (without using screen)..


regards,


Leon 



On Mar 24, 2009, at 2:35 PM, Will Smith wrote:







No, I started FS with this:
/usr/local/freeswitch/bin/freeswitch . What can I do in this case ?
 
thank you for help

--- On Tue, 3/24/09, Saeed Ahmed  wrote:

From: Saeed Ahmed 
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-us...@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:25 AM





Did you started FS with –nc option?
with this option you can connect to FS using ./fs_cli
OR 
use screen!




From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] on Behalf Of Will Smith
Sent: Tuesday, March 24, 2009 1:52 PM
To: freeswitch-us...@lists.freeswitch.org
Subject: [Freeswitch-users] Console Window
  





Hi,

I closed the console window that ran FS without shutting down FS. I had to 
restart the server to get back to the console window. I just wonder if there is 
a way to bring that window up without restart server. In the file 
/log/freeswitch.pid, I found a number, is that a seesion id?

 

Thank you.
 
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Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward 
Here it is:

freeswi...@b-pbx-lab-1> recv 517 bytes from udp/[10.1.21.44]:5060 at
13:53:07.644865:
   
   OPTIONS sip:b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport
   From: "Unknown" 
>;tag=as5adee8f4
   To: 
   Contact: >
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:07 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Length: 0
   
send 694 bytes to udp/[10.1.21.44]:5060 at 13:53:07.646132:
   
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK1b372b8d;rport=5060
   From: "Unknown" 
>;tag=as5adee8f4
   To: ;tag=DytraHp3K84aD
   Call-ID: 2e6222b16df27200056f742a070f0...@10.1.21.44
   CSeq: 102 OPTIONS
   Contact: 
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
recv 812 bytes from udp/[10.1.21.44]:5060 at 13:53:11.661169:
   
   INVITE 
sip:70...@b-pbx-lab-1.mynet.netSIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" 
>;tag=as4863e49a
   To: >
   Contact: >
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Date: Tue, 24 Mar 2009 13:53:11 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 258
   v=0
   o=root 4756 4756 IN IP4 10.1.21.44
   s=session
   c=IN IP4 10.1.21.44
   t=0 0
   m=audio 17956 RTP/AVP 0 8 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   
send 333 bytes to udp/[10.1.21.44]:5060 at 13:53:11.662467:
   
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" 
>;tag=as4863e49a
   To: >
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Content-Length: 0
   
send 815 bytes to udp/[10.1.21.44]:5060 at 13:53:11.682660:
   
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport=5060
   From: "Steve" 
>;tag=as4863e49a
   To: 
>;tag=e7KHcc76gHUXr
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: 100rel, timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="10.1.21.44",
nonce="1d23f0ec-187b-11de-8c60-ad87768304bc", algorithm=MD5, qop="auth"
   Content-Length: 0
   
recv 407 bytes from udp/[10.1.21.44]:5060 at 13:53:11.684103:
   
   ACK sip:70...@b-pbx-lab-1.mynet.net
SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK0231224c;rport
   From: "Steve" 
>;tag=as4863e49a
   To: 
>;tag=e7KHcc76gHUXr
   Contact: >
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 102 ACK
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0
   
recv 1089 bytes from udp/[10.1.21.44]:5060 at 13:53:11.685306:
   
   INVITE 
sip:70...@b-pbx-lab-1.mynet.netSIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK7858c13c;rport
   From: "Steve" 
>;tag=as4863e49a
   To: >
   Contact: >
   Call-ID: 4db5b31f3f9d99c436804e4b54277...@10.1.21.44
   CSeq: 103 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 7

Re: [Freeswitch-users] sip cancel request fails

2009-03-24 Thread Michael Jerris 
This means we could not match the cancel to a current call dialog.  I  
would need to see the full sip trace of the call to know why, but  
typically this is because of not matching call Id or to or from tags


Mike

On Mar 24, 2009, at 9:43 AM, Steven Ward  wrote:

A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b- 
lab-1) while the call is still ringing does not work.


Why is this request resulting in a 481?

I appreciate the help - I'm still just starting to learn SIP & FS.   
The CANCEL request and 481 response appear as follows on my FS  
console:



recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:

--- 
-

   CANCEL sip:70...@b-pbx-lab-1.mynet.net SIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: "Steve" ;tag=as7f6965ea
   To: 
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

--- 
-

send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:

--- 
-

   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: "Steve" ;tag=as7f6965ea
   To: ;tag=71m745HKHKyjc
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   Content-Length: 0

   --



Thanks.  - SW
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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Leon de Rooij 
Just make sure you have mod_event socket loaded in conf/ 
autoload_modules/modules.conf.xml :




And have it configured in conf/autoload_modules/event_socket.conf.xml

Then you can use bin/fs_cli to connect to running FS instance.

I don't think you can reconnect to a process of which you have  
disconnected the terminal (without using screen)..


regards,

Leon

On Mar 24, 2009, at 2:35 PM, Will Smith wrote:


No, I started FS with this:
/usr/local/freeswitch/bin/freeswitch . What can I do in this case ?

thank you for help

--- On Tue, 3/24/09, Saeed Ahmed  wrote:
From: Saeed Ahmed 
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:25 AM

Did you started FS with –nc option?
with this option you can connect to FS using ./fs_cli
OR
use screen!

From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Will Smith

Sent: Tuesday, March 24, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Console Window


Hi,

I closed the console window that ran FS without shutting down FS. I  
had to restart the server to get back to the console window. I just  
wonder if there is a way to bring that window up without restart  
server. In the file /log/freeswitch.pid, I found a number, is that a  
seesion id?




Thank you.



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Re: [Freeswitch-users] setInputCallback not working with Javascript?

2009-03-24 Thread Brian West 
Javascript doesn't use the Core Session constructor.  Its not the same  
as the other languages.


/b

On Mar 24, 2009, at 1:40 AM, mszla...@aol.com wrote:


I'm getting in build 12653M:

[ERR] notify.js:130 mod_spidermonkey()  TypeError:  
session.setInputCallback is not a function


The wiki says this function should work in Javascript.

http://wiki.freeswitch.org/wiki/ 
CoreSession_Constructor#session:setInputCallback


Also, has there been changes to session.collectInput with  
type="event"? I get dtmf type events with my callback function but  
can't seem to get type="event" with speech events.


Mark.



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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Szymon Olko 
Will Smith pisze:
> No, I started FS with this:
> /usr/local/freeswitch/bin/freeswitch . What can I do in this case ?
>  
run command

fs_cli

it is almost the same as normal freeswitch console. You just cannot shut it 
down with '...' command.

Just use this one 'fsctl shutdown asap'.



> thank you for help
> 
> --- On *Tue, 3/24/09, Saeed Ahmed //* wrote:
> 
> From: Saeed Ahmed 
> Subject: Re: [Freeswitch-users] Console Window
> To: freeswitch-users@lists.freeswitch.org
> Date: Tuesday, March 24, 2009, 6:25 AM
> 
> Did you started FS with –nc option?
> with this option you can connect to FS using ./fs_cli
> OR
> use /screen/!
> 
> 
> 
> *From:* freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of
> *Will Smith
> *Sent:* Tuesday, March 24, 2009 1:52 PM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* [Freeswitch-users] Console Window
> 
>  
> 
> Hi,
> 
> I closed the console window that ran FS without shutting down FS. I
> had to restart the server to get back to the console window. I just
> wonder if there is a way to bring that window up without restart
> server. In the file /log/freeswitch.pid, I found a number, is that a
> seesion id?
> 
>  
> 
> Thank you.
> 
>  
> 
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[Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward 
A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while
the call is still ringing does not work.

Why is this request resulting in a 481?

I appreciate the help - I'm still just starting to learn SIP & FS.  The
CANCEL request and 481 response appear as follows on my FS console:


recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
   
   CANCEL 
sip:70...@b-pbx-lab-1.mynet.netSIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: "Steve" 
>;tag=as7f6965ea
   To: >
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

   
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
   
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: "Steve" 
>;tag=as7f6965ea
   To: 
>;tag=71m745HKHKyjc
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   Content-Length: 0
   --



Thanks.  - SW
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[Freeswitch-users] sip cancel request fails

2009-03-24 Thread Steven Ward 
A CANCEL request sent from my Asterisk UA (10.1.21.44) to FS (b-lab-1) while
the call is still ringing does not work.

Why is this request resulting in a 481?

I appreciate the help - I'm still just starting to learn SIP & FS.  The
CANCEL request and 481 response appear as follows on my FS console:


recv 362 bytes from udp/[10.1.21.44]:5060 at 13:30:23.291616:
   
   CANCEL 
sip:70...@b-pbx-lab-1.mynet.netSIP/2.0
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport
   From: "Steve" 
>;tag=as7f6965ea
   To: >
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   User-Agent: Asterisk PBX
   Max-Forwards: 70
   Content-Length: 0

   
send 327 bytes to udp/[10.1.21.44]:5060 at 13:30:23.292235:
   
   SIP/2.0 481 Call/Transaction Does Not Exist
   Via: SIP/2.0/UDP 10.1.21.44:5060;branch=z9hG4bK6f7f35ab;rport=5060
   From: "Steve" 
>;tag=as7f6965ea
   To: 
>;tag=71m745HKHKyjc
   Call-ID: 237598fd102b739a03b4a4047bf69...@10.1.21.44
   CSeq: 103 CANCEL
   Content-Length: 0
   --



Thanks.  - SW
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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Will Smith 
No, I started FS with this:
/usr/local/freeswitch/bin/freeswitch . What can I do in this case ?
 
thank you for help

--- On Tue, 3/24/09, Saeed Ahmed  wrote:

From: Saeed Ahmed 
Subject: Re: [Freeswitch-users] Console Window
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, March 24, 2009, 6:25 AM









Did you started FS with –nc option?
with this option you can connect to FS using ./fs_cli
OR 
use screen!




From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Will Smith
Sent: Tuesday, March 24, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Console Window
 





Hi,

I closed the console window that ran FS without shutting down FS. I had to 
restart the server to get back to the console window. I just wonder if there is 
a way to bring that window up without restart server. In the file 
/log/freeswitch.pid, I found a number, is that a seesion id?

 

Thank you.
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Re: [Freeswitch-users] Console Window

2009-03-24 Thread Saeed Ahmed 
Did you started FS with -nc option?
with this option you can connect to FS using ./fs_cli
OR 
use screen!

  _  

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Will
Smith
Sent: Tuesday, March 24, 2009 1:52 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Console Window

 


Hi,

I closed the console window that ran FS without shutting down FS. I had to
restart the server to get back to the console window. I just wonder if there
is a way to bring that window up without restart server. In the file
/log/freeswitch.pid, I found a number, is that a seesion id?

 

Thank you.

 

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Re: [Freeswitch-users] Sip for Skype - g.729 requirement

2009-03-24 Thread David Sugar 
They require one use g.729, which is patent encumbered as well as rather
computationally intensive.

Dan wrote:
> You probably already saw this but
> 
> http://www.skypeforsip.com/
> 
> Skype is supporting sip for business users.
> 
> 
> 
> 
> ___
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> http://www.freeswitch.org
begin:vcard
fn:David Sugar
n:Sugar;David
org:GNU Telephony
email;internet:dy...@gnutelephony.org
tel;work:+1 609 465 5336
url:http://www.gnutelephony.org
version:2.1
end:vcard

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[Freeswitch-users] Console Window

2009-03-24 Thread Will Smith 
Hi,
I closed the console window that ran FS without shutting down FS. I had to 
restart the server to get back to the console window. I just wonder if there is 
a way to bring that window up without restart server. In the file 
/log/freeswitch.pid, I found a number, is that a seesion id?
 
Thank you.


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[Freeswitch-users] [SUSE Enterprise/FS 1.0.3] ODBC-related compile error

2009-03-24 Thread Gilles 
Brian West > We changed this to autodetect it means you have the libs 
but no devel headers.. either disable it or install the dev headers.

Thanks guys, that was it: The stock SLED 10 that I have on this MSI 
host did have unixODBC.rpm but not its unixODBC-devel.rpm counterpart.

I re-read the "Download & Installation Guide" page, which doesn't say 
that Freeswitch will compile for ODBC instead of SQLite if it detects 
unixODBC... but doesn't check that its counterpart unixODBC-devel is 
also installed :-/

I suggest that either the config/make script be updated to check for 
this, or the wiki be edited to tell users to check for both packages 
(if N.A. : yast install unixODBC-devel OR rpm -UVH 
unixODBC-devel.rpm) or configure Freeswitch to use SQLite instead of 
ODBC (good enough to get newbies started).

Cheers,


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