Re: [Freeswitch-users] any way ring fifo members one by one?

[Antonio Gallo]

seven ha scritto:
> oh, thank you Antonio. I think it would be better to collect more  
> ideas before open a bounty. And I more interested in playing(including  
> patching the code) with that than use the function.
>   
I was working on other stuff yesterday and just looked at the wiki:
- it seems there is already a bounty for something like that;
- there is a wiki page about how to implement it with Javascript, ofc 
you need to tailor it to your own needs;

AgX



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[Freeswitch-users] TLS initialization failures

[Jason White]

I know this isn't the place to report bugs; unfortunately, the Jira Web
interface isn't working for me due to accessibility issues. (If there is an
alternative way to submit reports that could be efficiently handled by the
developers, let me know).

A few weeks ago I reported problems with the initialization of multiple SIP
profiles for which TLS was enabled. At the time, I suspected the underlying
cause was an attempt to bind to port 5061 under IPv4 and IPv6 separately.

It turns out, however, that even if the TLS ports in all of the profiles are
distinct, one or more of the profiles often fails to start when FreeSWITCH is
first loaded. Running sofia profile start  thereafter always
succeeds; thus I suspect a subtle timing problem. I am having trouble
reproducing this with the Sofia debugging variables enabled.

I also have a core file that was generated when FreeSWITCH crashed while
attempting to initialize TLS during startup. I can supply a backtrace if this
would help. The segfault is not reliably reproducable, but it has happened
more than once, most recently under rev. 13081, for which I kept the core
file.

I am running Debian Sid. A colleague with Fedora 10 has also been able to
reproduce the TLS initialization bug.


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Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?

[Ron McCarthy]

That's really I want that looks perfect, anyways I can get stats for a
gateway? Far as the calls-in and calls out?

If not the heartbeat looks like I could writ esomething up and use that to.


On Tue, Apr 28, 2009 at 7:33 PM, Brian West  wrote:

>
> On Apr 28, 2009, at 9:19 PM, Ron McCarthy wrote:
>
> will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU?
>
>
> No its just silly in my opinion if you want that then hook on event socket
> using ESL and write something to register callbacks for each event and
> collect stats.  Then you have realtime stats collection you can query or
> dump out any way you see fit.
>
>
> Only reason id like to hit every second so I could get some sort of attempt
> on the call setups and call transactions.
>
>
> we already collect those stats on the sofia profiles.
>
> 14
> 11
> 2
> 3
>
> sofia xmlstatus profile 
>
> ;)
>
>
>
> Has anyone even done anything like this before?
>
> Thanks
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?

[Brian West]

On Apr 28, 2009, at 9:19 PM, Ron McCarthy wrote:

will hitting it every 10 or 20 seconds hurt performance / eat a lot  
of CPU?


No its just silly in my opinion if you want that then hook on event  
socket using ESL and write something to register callbacks for each  
event and collect stats.  Then you have realtime stats collection you  
can query or dump out any way you see fit.




Only reason id like to hit every second so I could get some sort of  
attempt on the call setups and call transactions.


we already collect those stats on the sofia profiles.

14
11
2
3

sofia xmlstatus profile 

;)




Has anyone even done anything like this before?

Thanks


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?

[SP]

Connect to the socket and subscribe to the heartbeat.  The heartbeat
event fires every 20 seconds and contains the session count.

On Tue, Apr 28, 2009 at 21:19, Ron McCarthy  wrote:
> will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU?
>
> Only reason id like to hit every second so I could get some sort of attempt
> on the call setups and call transactions.
>
> Has anyone even done anything like this before?
>
> Thanks
>
> On Tue, Apr 28, 2009 at 7:16 PM, Brian West  wrote:
>>
>> every second is way too high of a resolution.  10 seconds 20 seconds are
>> more sane!
>> /b
>> On Apr 28, 2009, at 9:11 PM, Ron McCarthy wrote:
>>
>> lso, if I have to make some custom plugin that montiors output the of show
>> channels /show calls via one of the API's, anyway we can get a realtime
>> stream so I don't have to query it every second to get correct data?
>>
>> Brian West
>> br...@freeswitch.org
>> -- Meet us at ClueCon!  http://www.cluecon.com
>>
>>
>>
>>
>>
>> ___
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-- 
Shannon

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Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?

[Ron McCarthy]

will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU?

Only reason id like to hit every second so I could get some sort of attempt
on the call setups and call transactions.

Has anyone even done anything like this before?

Thanks

On Tue, Apr 28, 2009 at 7:16 PM, Brian West  wrote:

> every second is way too high of a resolution.  10 seconds 20 seconds are
> more sane!
> /b
>
> On Apr 28, 2009, at 9:11 PM, Ron McCarthy wrote:
>
> lso, if I have to make some custom plugin that montiors output the of show
> channels /show calls via one of the API's, anyway we can get a realtime
> stream so I don't have to query it every second to get correct data?
>
>
> Brian West
> br...@freeswitch.org
>
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
> ___
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Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?

[Brian West]

every second is way too high of a resolution.  10 seconds 20 seconds  
are more sane!


/b

On Apr 28, 2009, at 9:11 PM, Ron McCarthy wrote:

lso, if I have to make some custom plugin that montiors output the  
of show channels /show calls via one of the API's, anyway we can get  
a realtime stream so I don't have to query it every second to get  
correct data?


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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[Freeswitch-users] SNMP or Cacti to graph perrformance?

[Ron McCarthy]

Hi List,

I have searched and searched and have found nothing. Id like to be able to
keep track of current calls and the CPS and then chart these via Cacti. Is
their some a SNMP plugin that I have not seen for FS that can do some of
these things?

Also, if I have to make some custom plugin that montiors output the of show
channels /show calls via one of the API's, anyway we can get a realtime
stream so I don't have to query it every second to get correct data?

Thanks!
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Re: [Freeswitch-users] any way ring fifo members one by one?

[seven]

oh, thank you Antonio. I think it would be better to collect more  
ideas before open a bounty. And I more interested in playing(including  
patching the code) with that than use the function.

On Apr 28, 2009, at 11:39 PM, Antonio Gallo wrote:

> dujinfang ha scritto:
>> Yeah but FS is more than a *, it should be able to do a call center
>> like job.
> I think if you open a bounty you can get some nice feedback about  
> this.
>
> Antonio
>
>
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Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available

[Mike Fedyk]

Are the pre relerses a specific revision of think or is there a branch
for the 1.0 series?

On 4/28/09, Michael Collins  wrote:
> The latest pre-release of 1.0.4 is available. More information here:
> http://www.freeswitch.org/node/177
>
> The FreeSWITCH development team would like to thank everyone for their help
> in reporting bugs and testing features. Please keep the feedback coming and
> spread the word about what a cool project FreeSWITCH really is!
>
> -Michael
>
> See you at ClueCon - http://www.cluecon.com
>

-- 
Sent from my mobile device

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Re: [Freeswitch-users] Voice mail: skip announcement

[Brian West]

You could just record a greeting and copy a file thats 1ms long over  
the top of it.


/b

On Apr 28, 2009, at 3:46 PM, paul.degt wrote:


But is there a way to keep original macro as well and trigger them on
demand from a dial plan?


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available

[Brian West]

Fixed.

/b

On Apr 28, 2009, at 3:38 PM, Alex Rambau wrote:


The link in the post refers to pre5, not pre6.


Brian West
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-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Voice mail: skip announcement

[paul.degt]

But is there a way to keep original macro as well and trigger them on 
demand from a dial plan?

Brian West wrote:
> This macro:
>
>  
>   
>
>
>   
>
>
>   
> 
>  
>   
> 
>  type="name_spelled"/> 
>  
>  data="voicemail/vm-not_available.wav"/>   
>  
>
>   
>
>
>   
>
>
>
> in conf/lang/en/vm/sounds.xml
>
> This is what you would modify to do that.
>
> On Apr 28, 2009, at 3:00 PM, paul.degt wrote:
>
>> Is there a way to skip "the person at extension..." announcement when 
>> forwarding a call to voice mail, so that it starts recording 
>> immediately, or just says something like "start recording..."?
>> Also, is it possible to turn off voice mail play back feature, meaning 
>> when somebody tries to leave a voice mail one needs to press a button  
>> to end message and then VM suddenly plays it back?
>> I see there's sounds.xml but not sure it help me to do what I need , is 
>> there any write up on how to customize it?
>> Would appreciate some directions.
>
> Brian West
> br...@freeswitch.org 
>
> -- Meet us at ClueCon!  http://www.cluecon.com 
>
>
>
>
> 
>
> ___
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Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available

[Alex Rambau]

The link in the post refers to pre5, not pre6.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, April 28, 2009 12:44 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available

 

The latest pre-release of 1.0.4 is available. More information here:
http://www.freeswitch.org/node/177

The FreeSWITCH development team would like to thank everyone for their help
in reporting bugs and testing features. Please keep the feedback coming and
spread the word about what a cool project FreeSWITCH really is!

-Michael

See you at ClueCon - http://www.cluecon.com

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Re: [Freeswitch-users] Voice mail: skip announcement

[Brian West]

This macro:

  

  

type="name_spelled"/>


  

  

in conf/lang/en/vm/sounds.xml

This is what you would modify to do that.

On Apr 28, 2009, at 3:00 PM, paul.degt wrote:


Is there a way to skip "the person at extension..." announcement when
forwarding a call to voice mail, so that it starts recording
immediately, or just says something like "start recording..."?
Also, is it possible to turn off voice mail play back feature, meaning
when somebody tries to leave a voice mail one needs to press a button
to end message and then VM suddenly plays it back?
I see there's sounds.xml but not sure it help me to do what I need ,  
is

there any write up on how to customize it?
Would appreciate some directions.


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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[Freeswitch-users] Voice mail: skip announcement

[paul.degt]

Is there a way to skip "the person at extension..." announcement when 
forwarding a call to voice mail, so that it starts recording 
immediately, or just says something like "start recording..."?
Also, is it possible to turn off voice mail play back feature, meaning 
when somebody tries to leave a voice mail one needs to press a button  
to end message and then VM suddenly plays it back?
I see there's sounds.xml but not sure it help me to do what I need , is 
there any write up on how to customize it?
Would appreciate some directions.

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[Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available

[Michael Collins]

The latest pre-release of 1.0.4 is available. More information here:
http://www.freeswitch.org/node/177

The FreeSWITCH development team would like to thank everyone for their help
in reporting bugs and testing features. Please keep the feedback coming and
spread the word about what a cool project FreeSWITCH really is!

-Michael

See you at ClueCon - http://www.cluecon.com
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Re: [Freeswitch-users] DID Provider

[Saeed Ahmed]

Yup, I would need it for inbound call center services.

Thanks for your detailed reply.

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jon
Radel
Sent: Tuesday, April 28, 2009 8:07 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DID Provider

The best quality and cheapest for all applications serving all markets 
around the world, all at the same time?  Tall order.

It's easier for people to give meaningful answers if you give some hints 
as to little details such as what country you're talking about (though 
the working assumption of the U.S. is probably relatively safe...). 
However, since you're talking about 50 channels per DID, there's a 
chance that you're talking about an application of the type that is 
starting to drive telcos a bit nuts.

For example, Verizon, a perfectly fine provider of SIP trunking that I 
happen to use, just announced on the 22nd that "high volume predictive 
auto-dialers and mass calling applications" have resulted in such an 
increase in traffic so far in 2009 that they've put their wholesale 
customers on notice that certain calling patterns are liable to get your 
contract canceled.  For example, an average call length per account over 
a given month of less than 78 seconds is no longer allowed.  The ratio 
of concurrent channels used to call attempts per second also has to meet 
their new standards.

A long way of saying that if you're talking about any type of call 
center or mass calling you're much better off doing your own, detailed 
research, especially on cost.

-- 

--Jon Radel
j...@radel.com




Paul wrote:
> 
> Go with Verizon. They're reliable and available anywhere in the US.
> 
> 
> - Original Message 
> From: Saeed Ahmed 
> To: freeswitch-users@lists.freeswitch.org
> Sent: Tuesday, April 28, 2009 1:21:46 PM
> Subject: Re: [Freeswitch-users] DID Provider
> 
> Hi,
> 
> But which one would you prefer? In terms of cost and quality.
> 
> Thanks.
> 
> Paul wrote:
>> FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH
works with any major telco.
>>
>>
>> - Original Message 
>> From: Saeed Ahmed 
>> To: freeswitch-users@lists.freeswitch.org
>> Sent: Tuesday, April 28, 2009 1:09:42 PM
>> Subject: [Freeswitch-users] DID Provider
>>
>> Hi All,
>>
>> Could you please refer me to a best DID provider which also works 
>> perfectly with FreeSWITCH. I would need 50 channels per DID number.
>>
>> Thanks.




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Re: [Freeswitch-users] DID Provider

[Jon Radel]

The best quality and cheapest for all applications serving all markets 
around the world, all at the same time?  Tall order.


It's easier for people to give meaningful answers if you give some hints 
as to little details such as what country you're talking about (though 
the working assumption of the U.S. is probably relatively safe...). 
However, since you're talking about 50 channels per DID, there's a 
chance that you're talking about an application of the type that is 
starting to drive telcos a bit nuts.


For example, Verizon, a perfectly fine provider of SIP trunking that I 
happen to use, just announced on the 22nd that "high volume predictive 
auto-dialers and mass calling applications" have resulted in such an 
increase in traffic so far in 2009 that they've put their wholesale 
customers on notice that certain calling patterns are liable to get your 
contract canceled.  For example, an average call length per account over 
a given month of less than 78 seconds is no longer allowed.  The ratio 
of concurrent channels used to call attempts per second also has to meet 
their new standards.


A long way of saying that if you're talking about any type of call 
center or mass calling you're much better off doing your own, detailed 
research, especially on cost.


--

--Jon Radel
j...@radel.com




Paul wrote:


Go with Verizon. They're reliable and available anywhere in the US.


- Original Message 
From: Saeed Ahmed 
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, April 28, 2009 1:21:46 PM
Subject: Re: [Freeswitch-users] DID Provider

Hi,

But which one would you prefer? In terms of cost and quality.

Thanks.

Paul wrote:

FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH works 
with any major telco.


- Original Message 
From: Saeed Ahmed 
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, April 28, 2009 1:09:42 PM
Subject: [Freeswitch-users] DID Provider

Hi All,

Could you please refer me to a best DID provider which also works 
perfectly with FreeSWITCH. I would need 50 channels per DID number.


Thanks.





smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [Freeswitch-users] DID Provider

[Paul]

Go with Verizon. They're reliable and available anywhere in the US.


- Original Message 
From: Saeed Ahmed 
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, April 28, 2009 1:21:46 PM
Subject: Re: [Freeswitch-users] DID Provider

Hi,

But which one would you prefer? In terms of cost and quality.

Thanks.

Paul wrote:
> FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH 
> works with any major telco.
>
>
> - Original Message 
> From: Saeed Ahmed 
> To: freeswitch-users@lists.freeswitch.org
> Sent: Tuesday, April 28, 2009 1:09:42 PM
> Subject: [Freeswitch-users] DID Provider
>
> Hi All,
>
> Could you please refer me to a best DID provider which also works 
> perfectly with FreeSWITCH. I would need 50 channels per DID number.
>
> Thanks.
>
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>  
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Re: [Freeswitch-users] DID Provider

[Geoff Love]

https://teliax.com/?referral_code=11

Teliax can provide 50 Channels per did.

Contact me for more information.

Thanks,

Geoff Love
303-629-8304

On Tue, Apr 28, 2009 at 11:09 AM, Saeed Ahmed wrote:

> Hi All,
>
> Could you please refer me to a best DID provider which also works
> perfectly with FreeSWITCH. I would need 50 channels per DID number.
>
> Thanks.
>
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-- 
Geoff Love
Sales Engineer
gl...@teliax.com
303-629-8304
Referral Code 11
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Re: [Freeswitch-users] DID Provider

[Saeed Ahmed]

Hi,

But which one would you prefer? In terms of cost and quality.

Thanks.

Paul wrote:
> FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH 
> works with any major telco.
>
>
> - Original Message 
> From: Saeed Ahmed 
> To: freeswitch-users@lists.freeswitch.org
> Sent: Tuesday, April 28, 2009 1:09:42 PM
> Subject: [Freeswitch-users] DID Provider
>
> Hi All,
>
> Could you please refer me to a best DID provider which also works 
> perfectly with FreeSWITCH. I would need 50 channels per DID number.
>
> Thanks.
>
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>
>
>   
>
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Re: [Freeswitch-users] DID Provider

[Paul]

Please check here for more information.

http://www.voip-info.org/


- Original Message 
From: Saeed Ahmed 
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, April 28, 2009 1:09:42 PM
Subject: [Freeswitch-users] DID Provider

Hi All,

Could you please refer me to a best DID provider which also works 
perfectly with FreeSWITCH. I would need 50 channels per DID number.

Thanks.

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Re: [Freeswitch-users] DID Provider

[Paul]

FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH works 
with any major telco.


- Original Message 
From: Saeed Ahmed 
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, April 28, 2009 1:09:42 PM
Subject: [Freeswitch-users] DID Provider

Hi All,

Could you please refer me to a best DID provider which also works 
perfectly with FreeSWITCH. I would need 50 channels per DID number.

Thanks.

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[Freeswitch-users] DID Provider

[Saeed Ahmed]

Hi All,

Could you please refer me to a best DID provider which also works 
perfectly with FreeSWITCH. I would need 50 channels per DID number.

Thanks.

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[Freeswitch-users] Fwd: FreeSWITCH In Blogosphere

[Michael Collins]

FYI, here's the correct link to the DIDX interview story on the main page:
http://www.freeswitch.org/node/176

Sorry for the broken link.  (Thanks moy!)
-MC

-- Forwarded message --
From: Michael Collins 
Date: Tue, Apr 28, 2009 at 9:27 AM
Subject: FreeSWITCH In Blogosphere
To: "freeswitch-users@lists.freeswitch.org" <
freeswitch-users@lists.freeswitch.org>, freeswitch-...@lists.freeswitch.org


I just wanted to let everyone know that there are a few blog posts out there
that you FreeSWITCHers might find interesting. Please check out these two
stories on the main FreeSWITCH page:

Another Interesting Use For FreeSWITCH:
OpenSim

DIDX Interview With Anthony Minessale and Brian
West

Be sure to leave comments where appropriate. I will followup with digg links
shortly.
-MC
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[Freeswitch-users] FreeSWITCH In Blogosphere

[Michael Collins]

I just wanted to let everyone know that there are a few blog posts out there
that you FreeSWITCHers might find interesting. Please check out these two
stories on the main FreeSWITCH page:

Another Interesting Use For FreeSWITCH:
OpenSim

DIDX Interview With Anthony Minessale and Brian
West

Be sure to leave comments where appropriate. I will followup with digg links
shortly.
-MC
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Re: [Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?

[Mikael Bjerkeland]

Thanks! I'll notify them of the problem and see if there's a way around it.



2009/4/28 Anthony Minessale 

> as soon as FS sees 183 it expects media.
>
> if they send 183 and no media it will most certainly timeout
>
> On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander Bjerkeland <
> mik...@bjerkeland.com> wrote:
>
>> The scenario I was referring to was actually an outbound call from a
>> locally registered SIP phone to a cellphone. The same thing happens
>> whether I use a SIP or PRI trunk. After 6 s it hangs up.
>>
>>
>> I get SDP on 183 no matter whether I'm calling a cellphone or a fixed
>> line. I also get ringing indication. The 183+sdp is passed on to the
>> Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks
>> claim to send early media but there seems to be no audio/RTP. If I
>> answer the call in 6 s it's not dropped because the media path was
>> established before RTP timeout.
>>
>> The same thing happens on latest trunk.
>> I added the debug line at 1520 and did make && /etc/init.d/freeswitch
>> stop && make install && /etc/init.d/freeswitch start but the debug line
>> didn't show up anywhere in the CLI.
>>
>> Is my upstream provider doing something wrong in sending early media in
>> these cases? Seems pretty odd. It can be avoided by setting a higher
>> rtp-timeout-sec but it will still be an absolute timeout on ringing.
>>
>>
>> A transcript of the log:
>>
>> send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865:
>>
>> 
>>   INVITE 
>> sip:21651...@domain.appsvrslip11.prigw.comSIP/2.0
>>   Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK
>>   Route: >
>>   Max-Forwards: 69
>>   From: "someone" 
>> >;tag=m2SepeSZ63e3g
>>   To: 
>> 
>> >
>>   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
>>   CSeq: 114345142 INVITE
>>   Contact: 
>>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M
>>   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: 383
>>   P-Asserted-Identity: "someone" 
>> 
>> >
>>
>>   v=0
>>   o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2
>>   s=FreeSWITCH
>>   c=IN IP4 2.2.2.2
>>   t=0 0
>>   m=audio 52706 RTP/AVP 9 8 0 3 101 13
>>   a=rtpmap:9 G722/8000
>>   a=rtpmap:8 PCMA/8000
>>   a=rtpmap:0 PCMU/8000
>>   a=rtpmap:3 GSM/8000
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-16
>>   a=rtpmap:13 CN/8000
>>   a=ptime:20
>>   m=video 52752 RTP/AVP 99
>>   a=rtpmap:99 H264/9
>>
>> 
>> 2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()
>> Channel
>> sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=
>> sip:21651...@1.1.1.1  entering state [calling][0]
>> recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864:
>>
>> 
>>   SIP/2.0 100 Trying
>>   From: "someone"
>> >;tag=m2SepeSZ63e3g
>>   To: 
>> 
>> >
>>   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
>>   CSeq: 114345142 INVITE
>>   Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK
>>   Content-Length: 0
>>
>>
>> 
>> recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906:
>>
>> 
>>   SIP/2.0 183 Session Progress
>>   From: "someone"
>> >;tag=m2SepeSZ63e3g
>>   To:
>> 
>> >;tag=20134330840200942815366
>>   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
>>   CSeq: 114345142 INVITE
>>   Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK
>>   content-type: application/sdp
>>   contact: >
>> +~K!-.f0vfc830~P68.cio~H9zwgW0VyisWTdcaM26c610Xbo1.nfS.5NQt3mO~~70!-.f0vft815;nt_server_host=1.1.1.1>
>>   supported: 100rel
>>   x-nt-party-id: -/
>>   allow: ACK
>>   allow: BYE
>>   allow: CANCEL
>>   allow: INVITE
>>   allow: OPTIONS
>>   allow: INFO
>>   allow: SUBSCRIBE
>>   allow: REFER
>>   allow: NOTIFY
>>   allow: PRACK
>>   server:  CS2000_NGSS/9.0
>>   Content-Length: 300
>>
>>   v=0
>>   o=IWSPM 573585738 573585738 IN IP4 84.20.97.100
>>   s=-
>>   e=unkn...@invalid.net
>>   t=0 0
>>   m=audio 45954 RTP/AVP 8 0 18 101
>>   c=IN IP4 84.20.97.100
>>   a=ptime:20
>>   a=fmtp:18 annexb=no
>>   a=rtpmap:101 telephone-event/8000
>>   a=fmtp:101 0-15
>>   m=video 0 RTP/AVP 99
>>   c=IN IP4 2.2.2.2
>>   a=rtpmap:99 H264/9
>>
>> 
>> 2009-04-28 15:55:58 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()
>> Channel
>> sofia/external-eth1/21651...@domain.appsvrs

Re: [Freeswitch-users] any way ring fifo members one by one?

[Antonio Gallo]

dujinfang ha scritto:
> Yeah but FS is more than a *, it should be able to do a call center 
> like job.
I think if you open a bounty you can get some nice feedback about this.

Antonio


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Re: [Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?

[Anthony Minessale]

as soon as FS sees 183 it expects media.

if they send 183 and no media it will most certainly timeout

On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander Bjerkeland <
mik...@bjerkeland.com> wrote:

> The scenario I was referring to was actually an outbound call from a
> locally registered SIP phone to a cellphone. The same thing happens
> whether I use a SIP or PRI trunk. After 6 s it hangs up.
>
>
> I get SDP on 183 no matter whether I'm calling a cellphone or a fixed
> line. I also get ringing indication. The 183+sdp is passed on to the
> Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks
> claim to send early media but there seems to be no audio/RTP. If I
> answer the call in 6 s it's not dropped because the media path was
> established before RTP timeout.
>
> The same thing happens on latest trunk.
> I added the debug line at 1520 and did make && /etc/init.d/freeswitch
> stop && make install && /etc/init.d/freeswitch start but the debug line
> didn't show up anywhere in the CLI.
>
> Is my upstream provider doing something wrong in sending early media in
> these cases? Seems pretty odd. It can be avoided by setting a higher
> rtp-timeout-sec but it will still be an absolute timeout on ringing.
>
>
> A transcript of the log:
>
> send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865:
>
> 
>   INVITE 
> sip:21651...@domain.appsvrslip11.prigw.comSIP/2.0
>   Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK
>   Route: >
>   Max-Forwards: 69
>   From: "someone" 
> >;tag=m2SepeSZ63e3g
>   To: 
> 
> >
>   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
>   CSeq: 114345142 INVITE
>   Contact: 
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M
>   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: 383
>   P-Asserted-Identity: "someone" 
> >
>
>   v=0
>   o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2
>   s=FreeSWITCH
>   c=IN IP4 2.2.2.2
>   t=0 0
>   m=audio 52706 RTP/AVP 9 8 0 3 101 13
>   a=rtpmap:9 G722/8000
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:0 PCMU/8000
>   a=rtpmap:3 GSM/8000
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=rtpmap:13 CN/8000
>   a=ptime:20
>   m=video 52752 RTP/AVP 99
>   a=rtpmap:99 H264/9
>
> 
> 2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()
> Channel
> sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=
> sip:21651...@1.1.1.1  entering state [calling][0]
> recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864:
>
> 
>   SIP/2.0 100 Trying
>   From: "someone"
> >;tag=m2SepeSZ63e3g
>   To: 
> 
> >
>   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
>   CSeq: 114345142 INVITE
>   Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK
>   Content-Length: 0
>
>
> 
> recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906:
>
> 
>   SIP/2.0 183 Session Progress
>   From: "someone"
> >;tag=m2SepeSZ63e3g
>   To:
> 
> >;tag=20134330840200942815366
>   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
>   CSeq: 114345142 INVITE
>   Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK
>   content-type: application/sdp
>   contact: 
> +~K!-.f0vfc830~P68.cio~H9zwgW0VyisWTdcaM26c610Xbo1.nfS.5NQt3mO~~70!-.f0vft815;nt_server_host=1.1.1.1>
>   supported: 100rel
>   x-nt-party-id: -/
>   allow: ACK
>   allow: BYE
>   allow: CANCEL
>   allow: INVITE
>   allow: OPTIONS
>   allow: INFO
>   allow: SUBSCRIBE
>   allow: REFER
>   allow: NOTIFY
>   allow: PRACK
>   server:  CS2000_NGSS/9.0
>   Content-Length: 300
>
>   v=0
>   o=IWSPM 573585738 573585738 IN IP4 84.20.97.100
>   s=-
>   e=unkn...@invalid.net
>   t=0 0
>   m=audio 45954 RTP/AVP 8 0 18 101
>   c=IN IP4 84.20.97.100
>   a=ptime:20
>   a=fmtp:18 annexb=no
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-15
>   m=video 0 RTP/AVP 99
>   c=IN IP4 2.2.2.2
>   a=rtpmap:99 H264/9
>
> 
> 2009-04-28 15:55:58 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()
> Channel
> sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=
> sip:21651...@1.1.1.1  entering state
> [proceeding][183]
> 2009-04-28 15:55:58 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state()
> Remote SDP:
> v=0
> o=IWSPM 573585738 573585738 IN IP4 84.20.97.100
> s=-
> e=unkn...@invalid.net
> t=0 0
> 

Re: [Freeswitch-users] any way ring fifo members one by one?

[dujinfang]

Thank you for the detailed implementation, comments follows.


On Apr 28, 2009, at 10:24 PM, Anthony Minessale wrote:

keep in mind mod_fifo is not a call center app. it's a simple *fifo*  
queue hence the name.




Yeah but FS is more than a *, it should be able to do a call center  
like job.


On Tue, Apr 28, 2009 at 8:30 AM, dujinfang   
wrote:
Ah, right, that works. I had thought the purpose of members is for  
sequential hunting. looks I was wrong.


However, add a | sep'ed dial string is hard to do round robin  
hunting, as we don't want the first agent always busy while others  
have nothing to do. It is possible to add/delete members using  
another script to set different dialstring to emulate a round robin  
hunt, but why not implement it in the queue logic?


Other strategies to place calls could be added with a patch to the  
code.  The goal of the module was to be basic and have

most of the control and logic remain outside the module.



Then maybe better to keep it simple, and make another mod other than  
patch the code. What's your suggestion? And, do you think it would be  
easy to control my logic outside the module using lua/event_socket ?





questions:

1) What's the purpose for members? just for simultaneous ring?

They are essentially on-hook agents for a setup where you don't need  
people waiting on the phone.


2) What's the best use case of agents dial in a fifo to wait  
callers? They just listening to music and waiting if no caller? I  
guess that would be for very busy call centers.


Yes its so you can call in an pop all the calls off the fifo in the  
same one call.



3) In my test, other members keep ring after one answered, some  
times it even ring a long time after the caller hangup.


It's not ring-all, there is exactly one outbound call generated for  
every one person in the queue who is waiting so sometimes there is  
collateral damage.


Will do more test on this.





I'm currently using trixbox. when a call comes in, it just play a  
greeting and ring one free agent and fail over to other agents if no  
answer or playing moh if all agents are busy. I just want to  
implement the same logic in FS and replace it as * causing a lot of  
problems.


I know there are already rules of how to pull a call out from a  
fifo, and guess it would be possible to add some params to do  
sequential/round robin hunting for members, and by using dp tools to  
dynamically add/delete members it would me more powerful.


You could post it as a bounty, a change like that is a lot to do as  
a wish request.




Maybe I can try to play that, will look the code to see how hard will  
be. Thank you again.






Thanks.

On Apr 28, 2009, at 8:33 PM, Anthony Minessale wrote:

learn to think 4th dimensionally =D

Add one  with a | sep list in the dial string.


On Mon, Apr 27, 2009 at 11:35 PM, seven  wrote:
Hi,

I'm on trunk 13174, and route a call to fifo, but two members ring at
the same time. I want it ring one by one in a round robin manner,
what's wrong with me?

here is fifo.xml


  {fifo_member_wait=wait}user/1...@$${domain}
  {fifo_member_wait=wait}user/1...@$${domain}


We want to implement a call center where agents register to waiting
customers, when a customer calls in, it will drop in a queue and
search one available agent(in round robin manner). Most fifo  
functions

seems implemented for scenarios where agents dial in and waiting
callers, which is unnecessary on our condition.


Thanks.

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sip

Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP?

[Ognjen Seslija]

Hi,

there is a "edit online" option to use to change the option on the phone.
You have to first install SIP Voip Settings Tool from the Nokia forum, and
then change the option "Secure call" to prefered from NCT.

Ognjen

On Tue, Apr 28, 2009 at 4:32 PM, Peter P GMX  wrote:

> I also installed the Nokia Configuration Tool (V3).
>
> Which settings did you apply for getting the crypto line?
>
> So far I only got TLS to work, there is no crypto line so far.
>
> Best regards
> Peter
>
> Ognjen Seslija schrieb:
> > Hi,
> >
> > after installing Nokia Configuration Tool I managed to get E61i to
> > offer srtp only it sends crypto line in the RTP/AVP which is not rfc
> > behaviour:
> >
> > 2009-04-27 12:11:12 [ERR] sofia_glue.c:2785 sofia_glue_negotiate_sdp()
> > a=crypto in RTP/AVP, refer to rfc3711
> >
> > Can anything be done with this?
> >
> > Ognjen
> >
> >
> > On Mon, Apr 27, 2009 at 11:22 AM, Hostinsky Miroslav
> >  > > wrote:
> >
> > I think SRTP on S60 is not supported in native SIP/IMS client bundled
> > with phone.
> >
> > There is pdf document about RTP/SRTP implementation on the
> forum-nokia
> > website
> >
> >
> http://nds2.fds-forum.nokia.com/fdp/interface?fid=A1A1BNIMDIYB&st=vmeOzm
> >
> volzmnda95395bfe31cd07f895a9bd39b45e007033b8900c4f7df925ac9de383011968dc
> > <
> http://nds2.fds-forum.nokia.com/fdp/interface?fid=A1A1BNIMDIYB&st=vmeOzm%0Avolzmnda95395bfe31cd07f895a9bd39b45e007033b8900c4f7df925ac9de383011968dc
> >
> >
> d3b65f0b52fa4cb08fab3705157dc42e3af9022012ab2ac111cbeb00d9079d531933758c
> >
> b95b6eab7016953987e9489c30075d9e08c51f2857cec41f60409d58ce806fbae1127e2c
> >
> 87f420efbe962c28eecf16b7c108d6127de0310ea9d5d34c07076ce111ba72f7c91d795e
> > dc2adaa57534d08024eb1061a70aa6ba0c12be59ae7a6c354cd177&lid=FN
> >
> > I didn't find any other information about SRTP and S60...
> >
> > --
> > Miro
> >
> > -Original Message-
> > From: freeswitch-users-boun...@lists.freeswitch.org
> > 
> > [mailto:freeswitch-users-boun...@lists.freeswitch.org
> > ] On Behalf Of
> > Peter P GMX
> > Sent: Saturday, April 25, 2009 7:01 PM
> > To: freeswitch-users@lists.freeswitch.org
> > 
> > Subject: Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60
> > 3rd) with Pjsip and TLS/SRTP?
> >
> > Thanks Jason,
> >
> > I do TLS/SRTP with some other Phones
> (Snom/X-Lite/Other_Freeswitches).
> > This works nad I have those crypto lines.
> > But I couldn't get it work with my E71 and I am struggling whether I
> > should invest more time into it. I am sure it's on the Symbian side.
> >
> > This was just to ask if anybody manged to get it to work. I you
> > all tell
> > me: "Peter I tried, it doesn't work, please do not invest any
> > time", or
> > "it works", this would be very helpful information for me.
> >
> > Best regards
> > Peter
> >
> > Jason White schrieb:
> > > Peter P GMX  > > wrote:
> > >
> > >> has anybody tried successfully to setup a Nokia E71 (or similar
> > symbian
> > >> S60 3rd phone) with Pjsip and TLS/SRTP?
> > >> TLS seems to work but what about the SRTP part?
> > >>
> > >
> > > Do you have log entries like this?
> > >
> > > 2009-04-24 11:05:19 [INFO] switch_rtp.c:782
> > switch_rtp_add_crypto_key()
> > > Activati
> > > ng Secure RTP SEND
> > > 2009-04-24 11:05:19 [INFO] switch_rtp.c:762
> > switch_rtp_add_crypto_key()
> > > Activati
> > > ng Secure RTP RECV
> > >
> > > If so, this establishes that SRTP is being used. Of course, you
> > could
> > also
> > > intercept the packets just to be sure, but I think the above log
> > entries
> > > should be a reliable indicator.
> > >
> > >
> > > ___
> > > Freeswitch-users mailing list
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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
> > >
> > >
> >
> > ___
> > 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
> > 

Re: [Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?

[Mikael Aleksander Bjerkeland]

The scenario I was referring to was actually an outbound call from a
locally registered SIP phone to a cellphone. The same thing happens
whether I use a SIP or PRI trunk. After 6 s it hangs up.


I get SDP on 183 no matter whether I'm calling a cellphone or a fixed
line. I also get ringing indication. The 183+sdp is passed on to the
Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks
claim to send early media but there seems to be no audio/RTP. If I
answer the call in 6 s it's not dropped because the media path was
established before RTP timeout.

The same thing happens on latest trunk.
I added the debug line at 1520 and did make && /etc/init.d/freeswitch
stop && make install && /etc/init.d/freeswitch start but the debug line
didn't show up anywhere in the CLI.

Is my upstream provider doing something wrong in sending early media in
these cases? Seems pretty odd. It can be avoided by setting a higher
rtp-timeout-sec but it will still be an absolute timeout on ringing.


A transcript of the log:

send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865:


   INVITE sip:21651...@domain.appsvrslip11.prigw.com SIP/2.0
   Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK
   Route: 
   Max-Forwards: 69
   From: "someone" ;tag=m2SepeSZ63e3g
   To: 
   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
   CSeq: 114345142 INVITE
   Contact: 
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M
   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: 383
   P-Asserted-Identity: "someone" 
   
   v=0
   o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2
   s=FreeSWITCH
   c=IN IP4 2.2.2.2
   t=0 0
   m=audio 52706 RTP/AVP 9 8 0 3 101 13
   a=rtpmap:9 G722/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   m=video 52752 RTP/AVP 99
   a=rtpmap:99 H264/9


2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()
Channel
sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=sip:21651...@1.1.1.1
 entering state [calling][0]
recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864:


   SIP/2.0 100 Trying
   From: "someone";tag=m2SepeSZ63e3g
   To: 
   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
   CSeq: 114345142 INVITE
   Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK
   Content-Length: 0
   


recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906:


   SIP/2.0 183 Session Progress
   From: "someone";tag=m2SepeSZ63e3g
   To:
;tag=20134330840200942815366
   Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
   CSeq: 114345142 INVITE
   Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK
   content-type: application/sdp
   contact: 
   supported: 100rel
   x-nt-party-id: -/
   allow: ACK
   allow: BYE
   allow: CANCEL
   allow: INVITE
   allow: OPTIONS
   allow: INFO
   allow: SUBSCRIBE
   allow: REFER
   allow: NOTIFY
   allow: PRACK
   server:  CS2000_NGSS/9.0
   Content-Length: 300
   
   v=0
   o=IWSPM 573585738 573585738 IN IP4 84.20.97.100
   s=-
   e=unkn...@invalid.net
   t=0 0
   m=audio 45954 RTP/AVP 8 0 18 101
   c=IN IP4 84.20.97.100
   a=ptime:20
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   m=video 0 RTP/AVP 99
   c=IN IP4 2.2.2.2
   a=rtpmap:99 H264/9


2009-04-28 15:55:58 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()
Channel
sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=sip:21651...@1.1.1.1
 entering state [proceeding][183]
2009-04-28 15:55:58 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state()
Remote SDP:
v=0
o=IWSPM 573585738 573585738 IN IP4 84.20.97.100
s=-
e=unkn...@invalid.net
t=0 0
m=audio 45954 RTP/AVP 8 0 18 101
c=IN IP4 84.20.97.100
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 0 RTP/AVP 99
c=IN IP4 2.2.2.2
a=rtpmap:99 H264/9

2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20]
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:1912
sofia_glue_tech_set

Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP?

[Peter P GMX]

I also installed the Nokia Configuration Tool (V3).

Which settings did you apply for getting the crypto line?

So far I only got TLS to work, there is no crypto line so far.

Best regards
Peter

Ognjen Seslija schrieb:
> Hi,
>
> after installing Nokia Configuration Tool I managed to get E61i to
> offer srtp only it sends crypto line in the RTP/AVP which is not rfc
> behaviour:
>
> 2009-04-27 12:11:12 [ERR] sofia_glue.c:2785 sofia_glue_negotiate_sdp()
> a=crypto in RTP/AVP, refer to rfc3711
>
> Can anything be done with this?
>
> Ognjen
>
>
> On Mon, Apr 27, 2009 at 11:22 AM, Hostinsky Miroslav
>  > wrote:
>
> I think SRTP on S60 is not supported in native SIP/IMS client bundled
> with phone.
>
> There is pdf document about RTP/SRTP implementation on the forum-nokia
> website
>
> http://nds2.fds-forum.nokia.com/fdp/interface?fid=A1A1BNIMDIYB&st=vmeOzm
> volzmnda95395bfe31cd07f895a9bd39b45e007033b8900c4f7df925ac9de383011968dc
> 
> 
> d3b65f0b52fa4cb08fab3705157dc42e3af9022012ab2ac111cbeb00d9079d531933758c
> b95b6eab7016953987e9489c30075d9e08c51f2857cec41f60409d58ce806fbae1127e2c
> 87f420efbe962c28eecf16b7c108d6127de0310ea9d5d34c07076ce111ba72f7c91d795e
> dc2adaa57534d08024eb1061a70aa6ba0c12be59ae7a6c354cd177&lid=FN
>
> I didn't find any other information about SRTP and S60...
>
> --
> Miro
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> 
> [mailto:freeswitch-users-boun...@lists.freeswitch.org
> ] On Behalf Of
> Peter P GMX
> Sent: Saturday, April 25, 2009 7:01 PM
> To: freeswitch-users@lists.freeswitch.org
> 
> Subject: Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60
> 3rd) with Pjsip and TLS/SRTP?
>
> Thanks Jason,
>
> I do TLS/SRTP with some other Phones (Snom/X-Lite/Other_Freeswitches).
> This works nad I have those crypto lines.
> But I couldn't get it work with my E71 and I am struggling whether I
> should invest more time into it. I am sure it's on the Symbian side.
>
> This was just to ask if anybody manged to get it to work. I you
> all tell
> me: "Peter I tried, it doesn't work, please do not invest any
> time", or
> "it works", this would be very helpful information for me.
>
> Best regards
> Peter
>
> Jason White schrieb:
> > Peter P GMX  > wrote:
> >
> >> has anybody tried successfully to setup a Nokia E71 (or similar
> symbian
> >> S60 3rd phone) with Pjsip and TLS/SRTP?
> >> TLS seems to work but what about the SRTP part?
> >>
> >
> > Do you have log entries like this?
> >
> > 2009-04-24 11:05:19 [INFO] switch_rtp.c:782
> switch_rtp_add_crypto_key()
> > Activati
> > ng Secure RTP SEND
> > 2009-04-24 11:05:19 [INFO] switch_rtp.c:762
> switch_rtp_add_crypto_key()
> > Activati
> > ng Secure RTP RECV
> >
> > If so, this establishes that SRTP is being used. Of course, you
> could
> also
> > intercept the packets just to be sure, but I think the above log
> entries
> > should be a reliable indicator.
> >
> >
> > ___
> > 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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Re: [Freeswitch-users] any way ring fifo members one by one?

[Anthony Minessale]

keep in mind mod_fifo is not a call center app. it's a simple *fifo* queue
hence the name.


On Tue, Apr 28, 2009 at 8:30 AM, dujinfang  wrote:

> Ah, right, that works. I had thought the purpose of members is for
> sequential hunting. looks I was wrong.
> However, add a | sep'ed dial string is hard to do round robin hunting, as
> we don't want the first agent always busy while others have nothing to do.
> It is possible to add/delete members using another script to set different
> dialstring to emulate a round robin hunt, but why not implement it in the
> queue logic?
>

Other strategies to place calls could be added with a patch to the code.
The goal of the module was to be basic and have
most of the control and logic remain outside the module.



>
> questions:
>
> 1) What's the purpose for members? just for simultaneous ring?
>

They are essentially on-hook agents for a setup where you don't need people
waiting on the phone.


> 2) What's the best use case of agents dial in a fifo to wait callers? They
> just listening to music and waiting if no caller? I guess that would be for
> very busy call centers.
>

Yes its so you can call in an pop all the calls off the fifo in the same one
call.



> 3) In my test, other members keep ring after one answered, some times it
> even ring a long time after the caller hangup.
>

It's not ring-all, there is exactly one outbound call generated for every
one person in the queue who is waiting so sometimes there is collateral
damage.



>
>
> I'm currently using trixbox. when a call comes in, it just play a greeting
> and ring one free agent and fail over to other agents if no answer or
> playing moh if all agents are busy. I just want to implement the same logic
> in FS and replace it as * causing a lot of problems.
>
> I know there are already rules of how to pull a call out from a fifo, and
> guess it would be possible to add some params to do sequential/round robin
> hunting for members, and by using dp tools to dynamically add/delete members
> it would me more powerful.
>

You could post it as a bounty, a change like that is a lot to do as a wish
request.




>
> Thanks.
>
> On Apr 28, 2009, at 8:33 PM, Anthony Minessale wrote:
>
> learn to think 4th dimensionally =D
>
> Add one  with a | sep list in the dial string.
>
>
> On Mon, Apr 27, 2009 at 11:35 PM, seven  wrote:
>
>> Hi,
>>
>> I'm on trunk 13174, and route a call to fifo, but two members ring at
>> the same time. I want it ring one by one in a round robin manner,
>> what's wrong with me?
>>
>> here is fifo.xml
>>
>> 
>>   > lag="20">{fifo_member_wait=wait}user/1...@$${domain}
>>   > lag="20">{fifo_member_wait=wait}user/1...@$${domain}
>> 
>>
>> We want to implement a call center where agents register to waiting
>> customers, when a customer calls in, it will drop in a queue and
>> search one available agent(in round robin manner). Most fifo functions
>> seems implemented for scenarios where agents dial in and waiting
>> callers, which is unnecessary on our condition.
>>
>>
>> Thanks.
>>
>> ___
>> 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
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com 
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org 
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
> ___
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> 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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>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
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Re: [Freeswitch-users] any way ring fifo members one by one?

[Antonio Gallo]

dujinfang ha scritto:
> It is possible to add/delete members using another script
I'm not such expert but i think you can use variables everywhere in FS.
So your member config instead of having a static string 
"us...@domain|us...@domain" probably you can use a variables.
Then you script just push the user in/out fo that database variable string.

Antonio

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[Freeswitch-users] Sofia doubts

[Arsen Chaloyan]

What I want to discuss concerns mostly Sofia-SIP, but not primary FreeSWITCH.
Surely it would be better to use Sofia's list instead and I already did it, but 
have got no response so far.

Overall Sofia-SIP is a good library, which works reliable enough on Linux, but 
Windows port causes me some doubts.
Does anybody use FreeSWITCH/Sofia-SIP in production ready environment on 
Windows?
I know it's working, but how far it has been tested.

Just to be clear, the issues I've observed concerns UniMRCP/Sofia-SIP 1.12.10 
and I just wanted to know how reliable it works on your end.
http://sourceforge.net/mailarchive/forum.php?thread_name=708583.92229.qm%40web111308.mail.gq1.yahoo.com&forum_name=sofia-sip-devel
http://sourceforge.net/mailarchive/forum.php?thread_name=1ae1714d0903290440ree148b5m67fe183da58ee776%40mail.gmail.com&forum_name=sofia-sip-devel

Thanks,
Arsen.
www.unimrcp.org
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Re: [Freeswitch-users] any way ring fifo members one by one?

[dujinfang]

Ah, right, that works. I had thought the purpose of members is for  
sequential hunting. looks I was wrong.


However, add a | sep'ed dial string is hard to do round robin hunting,  
as we don't want the first agent always busy while others have nothing  
to do. It is possible to add/delete members using another script to  
set different dialstring to emulate a round robin hunt, but why not  
implement it in the queue logic?


questions:

1) What's the purpose for members? just for simultaneous ring?
2) What's the best use case of agents dial in a fifo to wait callers?  
They just listening to music and waiting if no caller? I guess that  
would be for very busy call centers.
3) In my test, other members keep ring after one answered, some times  
it even ring a long time after the caller hangup.



I'm currently using trixbox. when a call comes in, it just play a  
greeting and ring one free agent and fail over to other agents if no  
answer or playing moh if all agents are busy. I just want to implement  
the same logic in FS and replace it as * causing a lot of problems.


I know there are already rules of how to pull a call out from a fifo,  
and guess it would be possible to add some params to do sequential/ 
round robin hunting for members, and by using dp tools to dynamically  
add/delete members it would me more powerful.


Thanks.

On Apr 28, 2009, at 8:33 PM, Anthony Minessale wrote:


learn to think 4th dimensionally =D

Add one  with a | sep list in the dial string.


On Mon, Apr 27, 2009 at 11:35 PM, seven  wrote:
Hi,

I'm on trunk 13174, and route a call to fifo, but two members ring at
the same time. I want it ring one by one in a round robin manner,
what's wrong with me?

here is fifo.xml


  {fifo_member_wait=wait}user/1...@$${domain}
  {fifo_member_wait=wait}user/1...@$${domain}


We want to implement a call center where agents register to waiting
customers, when a customer calls in, it will drop in a queue and
search one available agent(in round robin manner). Most fifo functions
seems implemented for scenarios where agents dial in and waiting
callers, which is unnecessary on our condition.


Thanks.

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Re: [Freeswitch-users] [Freeswitch-dev] Compiling the FreeSwitch on Windows XP

[David A. Horner]

Actually, you might have it it doesn't look like the download
scripts are workingcould be file permissions or strange data.
I'd delete the directory and do a fresh checkout build too.

--dave
http://dave.thehorners.com/tech-talk/programming

On Tue, Apr 28, 2009 at 8:52 AM, David A. Horner
 wrote:
> Start with installing the windows sdk.
> error C2143: syntax error : missing ']' before 'constant'       C:\Program
> Files\Microsoft SDKs\Windows\v6.0A\include\ras.h
>
> http://msdn.microsoft.com/en-us/windows/bb980924.aspx
>
> --dave
> http://dave.thehorners.com/tech-talk/programming
>
> On Tue, Apr 28, 2009 at 4:15 AM, Santosh
>  wrote:
>> Hi,
>> I am compiling the FreeSwitch 1.0.3 on Windows machine and getting hundreds
>> of error.I guess this has something to do with Visual studio Settings.Can
>> you please help me with this.I am attaching the Error file also(in which
>> there are some missing files too).Can you send me a working FreeSwitch code
>> for windows?
>>
>> Regards,
>> Santosh
>>
>>
>>
>>
>> Disclaimer: The information contained in this e-mail and attachments if any
>> are privileged and confidential and are intended for the individual(s) or
>> entity(ies) named in this e-mail. If the reader or recipient is not the
>> intended recipient, or employee or agent responsible for delivering to the
>> intended recipient, you are hereby notified that dissemination, distribution
>> or copying of this communication or attachments thereof is strictly
>> prohibited. IF YOU RECEIVE this communication in error, please immediately
>> notify the sender and return the original message.
>>
>>
>>
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>

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Re: [Freeswitch-users] [Freeswitch-dev] Compiling the FreeSwitch on Windows XP

[David A. Horner]

Start with installing the windows sdk.
error C2143: syntax error : missing ']' before 'constant'   C:\Program
Files\Microsoft SDKs\Windows\v6.0A\include\ras.h

http://msdn.microsoft.com/en-us/windows/bb980924.aspx

--dave
http://dave.thehorners.com/tech-talk/programming

On Tue, Apr 28, 2009 at 4:15 AM, Santosh
 wrote:
> Hi,
> I am compiling the FreeSwitch 1.0.3 on Windows machine and getting hundreds
> of error.I guess this has something to do with Visual studio Settings.Can
> you please help me with this.I am attaching the Error file also(in which
> there are some missing files too).Can you send me a working FreeSwitch code
> for windows?
>
> Regards,
> Santosh
>
>
>
>
> Disclaimer: The information contained in this e-mail and attachments if any
> are privileged and confidential and are intended for the individual(s) or
> entity(ies) named in this e-mail. If the reader or recipient is not the
> intended recipient, or employee or agent responsible for delivering to the
> intended recipient, you are hereby notified that dissemination, distribution
> or copying of this communication or attachments thereof is strictly
> prohibited. IF YOU RECEIVE this communication in error, please immediately
> notify the sender and return the original message.
>
>
>
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Re: [Freeswitch-users] Internal.xml using Public context/dialplan?

[Fred-145]

Thanks Brian. I'd like to make sure I finally got how dialplans, contexts,
domains, SIP profiles, and extensions in the directory work together:

"Freeswitch supports different SIP profiles in conf/sip_profiles/
("internal.xml", "external.xml", etc.), which are loaded through
conf/autoload_configs/sofia.conf.xml. Each profile maps to a domain, which
corresponds to the part after @ (eg. "1...@192.168.0.1": The Freeswitch
server is listening on 192.168.0.1, and this is the domain to which
extension 1000 belongs.)

Each profile also maps to a context (dialplan) located under conf/dialplan/.
Note that, for security reasons, by default, all SIP profiles are set to use
the Public context; Internal extensions must map to the Default (private)
context explicitely through the "user_context" variable.

Extensions are located under conf/directory/. Each sub-directory matches the
domain to which an extension belongs. As set in conf/vars.xml, the default
domain matches the server's IP address which itself maps to the Default
domain, so all extensions that belong to this domain are located under
conf/directory/default/."

Is this correct?
-- 
View this message in context: 
http://www.nabble.com/Internal.xml-using-Public-context-dialplan--tp23175441p23276533.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?

[Anthony Minessale]

Are you geting 183+sdp from the nokia?
the media timer only operates once media is established and only
counts against you if the channel is being read from and that does not
happen until you get a 183 or 200 w/sdp

try putting a debug line in switch_rtp.c around 1520
printf("MISSED PACKETS %u/%u\n", rtp_session->missed_count,
rtp_session->max_missed_packets);

but try updating first there was a recent fix that may have prevented a
timer surge at the beginning of calls.


On Tue, Apr 28, 2009 at 6:20 AM, Mikael Aleksander Bjerkeland <
mik...@bjerkeland.com> wrote:

> Hi,
>
> I have been testing inbound calls to a Nokia phone with handover to a
> cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and
> had to set rtp-timeout to a very low 6 seconds in order to get "fast"
> handover. This introduces an interesting side-effect that hangs up calls
> even in the ringing state after 6 seconds. Is this the desired behaviour
> of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should
> only be valid for established calls where the two endpoints have
> exchanged rtp at some point but have stopped exchanging media. As far as
> I know a phone call in ringing state has not shared any RTP with the
> other endpoint until it gets early media or is answered. Should
> rtp-timeout-sec really be valid even when ringing?
>
> It seems to me that setting rtp-timeout-sec to 60 seconds would add an
> absolute time limit on ringing phone calls to 60 seconds, which I
> believe is not the actual purpose of this limit. Could anyone please
> share their thoughts on this matter?
>
>
> Thanks,
> Mikael
>
>
>
>
>
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Re: [Freeswitch-users] any way ring fifo members one by one?

[Anthony Minessale]

learn to think 4th dimensionally =D

Add one  with a | sep list in the dial string.


On Mon, Apr 27, 2009 at 11:35 PM, seven  wrote:

> Hi,
>
> I'm on trunk 13174, and route a call to fifo, but two members ring at
> the same time. I want it ring one by one in a round robin manner,
> what's wrong with me?
>
> here is fifo.xml
>
> 
>lag="20">{fifo_member_wait=wait}user/1...@$${domain}
>lag="20">{fifo_member_wait=wait}user/1...@$${domain}
> 
>
> We want to implement a call center where agents register to waiting
> customers, when a customer calls in, it will drop in a queue and
> search one available agent(in round robin manner). Most fifo functions
> seems implemented for scenarios where agents dial in and waiting
> callers, which is unnecessary on our condition.
>
>
> Thanks.
>
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

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Re: [Freeswitch-users] uuid_displace & FIFO help

[TTNC - Adnan Barakat]

Chris Danielson wrote:
> 
>> Excellent thanks, this is what I was looking for.
>>
>> One last question if you don't mind; is there anyway to pull the caller 
>> out of a fifo after a certain time either from api or by setting a 
>> variable (eg. the destination didn't answer after sometime, so carry on 
>> in the dialplan to eg. voicemail)?
>>   
> There is a "uuid_transfer" that will allow you to route them accordingly.
Thanks Chris


Adnan

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[Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?

[Mikael Aleksander Bjerkeland]

Hi,

I have been testing inbound calls to a Nokia phone with handover to a
cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and
had to set rtp-timeout to a very low 6 seconds in order to get "fast"
handover. This introduces an interesting side-effect that hangs up calls
even in the ringing state after 6 seconds. Is this the desired behaviour
of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should
only be valid for established calls where the two endpoints have
exchanged rtp at some point but have stopped exchanging media. As far as
I know a phone call in ringing state has not shared any RTP with the
other endpoint until it gets early media or is answered. Should
rtp-timeout-sec really be valid even when ringing?

It seems to me that setting rtp-timeout-sec to 60 seconds would add an
absolute time limit on ringing phone calls to 60 seconds, which I
believe is not the actual purpose of this limit. Could anyone please
share their thoughts on this matter?


Thanks,
Mikael





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Re: [Freeswitch-users] Adding Spanish support to say

[Roberto Pereyra]

Hi


I can also help with the translation into Spanish.

Spanish is my native language.

I can be contacted at:

rjpereyra (at) gmail (dot) com


roberto

-- 

The best dedicated servers - LiquidWeb

http://www.liquidweb.com/?RID=contenid

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