[Freeswitch-users] Question regardin freeswitch startup info
Hareem Haque
hareem.haque at gmail.com
Mon Feb 21 06:55:09 MSK 2011
i start freeswitch it says max sessions 1000
and session rate 30.. what exact do these numbers mean and how can i
increase them
Your help is greatly appreciated.
Best Regards
Hareem. Haque
On Sun, Feb 20, 2011 at 12:11 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:
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> Today's Topics:
>
> 1. Re: Codec negotiation, again (Steven Ayre)
> 2. Re: Any experience with DTMF from FreeSwitch to Sonus with
> Vega ATAs? (Yehavi Bourvine)
> 3. Need help with CDR setup and call routing patterns (Hareem Haque)
> 4. Re: Need help with CDR setup and call routing patterns
> (Steven Ayre)
> 5. Re: Need help with CDR setup and call routing patterns
> (Steven Ayre)
>
>
> ---------- Forwarded message ----------
> From: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 20 Feb 2011 12:36:45 +0000
> Subject: Re: [Freeswitch-users] Codec negotiation, again
> In default behaviour, FS will negotiate the codec for the aleg, then do the
> dialplan, then do the bleg. As a result it can have already picked the 729
> codec by the time it hits the dialplan. With transcoding that's fine but
> with 729 not.
>
> The workaround is the late-negotiation profile parameter. It delays picking
> the aleg codec until the bleg picks one, then uses that if possible for the
> aleg.
>
> Steve on iPhone
>
>
> On 20 Feb 2011, at 01:33, Serge Yuriev <me at nevian.org> wrote:
>
> > Hello,
> >
> > A offers g729, g711a, g711u
> > FS allows all of them
> > B offers g711a, g711u
> >
> > Call fails with INCOMPATIBLE DESTINATION.
> >
> > Cant understand why FS offers only ONE codec (g729) to B - wiki says it
> should offer ordered LIST of..
> > GIT few days old, default config
> >
> > --
> > wbr,
> > Serge
> >
> > _______________________________________________
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> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> > http://www.freeswitch.org
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>
>
>
> ---------- Forwarded message ----------
> From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 20 Feb 2011 14:39:18 +0200
> Subject: Re: [Freeswitch-users] Any experience with DTMF from FreeSwitch to
> Sonus with Vega ATAs?
> Hello Steve,
>
> Thanks for clarifying it.
>
> __Yehavi:
>
> 2011/2/20 Steve Underwood <steveu at coppice.org>
>
>> Hi Yehavi,
>>
>> Section 3.2 of RFC2833 is poorly worded, and causes a lot of issues in
>> the real world. The first paragraph says you can choose to send both
>> audio and events, but doesn't clarify how. Presumably RFC2198 should be
>> used. Is your system using RFC2198 encoding? I doubt it. The second
>> paragraph talks about tone onset getting through as audio. Valid DTMF
>> can be as short as 45ms, so 60ms is far more than anything which might
>> be termed onset. Because they didn't specify anything about what is
>> tolerable as tone onset passing through the channel as audio we have
>> chaos today.
>>
>> RFC2833 is an obsolete spec, and we should be talking about RFC4733
>> today. Section 2.5.1.3.1 of RFC4733 does clearly call for combined
>> payloads to be sent as RFC2198 packets. RFC4733 is even vaguer about the
>> onset issue, though.
>>
>> Steve
>>
>>
>> On 02/20/2011 06:33 PM, Yehavi Bourvine wrote:
>> > Hello Steve,
>> > I uderstand from paragraph 3.2 in RFC-2833 that the sender may send
>> > some audio of the DTMF while sending the events, until it revognises
>> > that this is a DTMF tone. The 60mSec is what I've been told by Vega
>> > engineers.
>> > Thanks! __Yehavi:
>> >
>> > 2011/2/20 Steve Underwood <steveu at coppice.org <mailto:
>> steveu at coppice.org>>
>> >
>> > On 02/20/2011 03:31 PM, Yehavi Bourvine wrote:
>> > > Hello,
>> > > Anyone has an experience with the above configuration? All
>> > sides are
>> > > marked as RFC-2833 and the Sonus recognises most of the DTMF's
>> > twice.
>> > > The Vega sends both RFC-2833 events and about 60msec of the DTMF
>> > tone
>> > > (according to Vega's engineers it takes it about 60 msec to
>> > detect the
>> > > DTMF tones).
>> > > According to the RFC this is acceptable. The service provider I am
>> > > working with says that it shouldn't be like that. Before I waste
>> > time
>> > > in wars, has anybody had this issue and knows whether it is a
>> matter
>> > > of configuration at the Sonus side?
>> > > Thanks! __Yehavi:
>> > Are you saying the Vega sends 60ms of DTMF as audio, then sends
>> > RFC2833
>> > DTMF packets and mutes the audio? In what reading of the RFC is that
>> > considered appropriate action? In addition, a DTMF detector should
>> > detect is more like 40ms. 60ms is rather long.
>> >
>> > Steve
>> >
>> >
>> > _______________________________________________
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>> > FreeSWITCH-users at lists.freeswitch.org
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>> >
>> >
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>
>
> ---------- Forwarded message ----------
> From: Hareem Haque <hareem.haque at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sun, 20 Feb 2011 11:08:08 -0500
> Subject: [Freeswitch-users] Need help with CDR setup and call routing
> patterns
> Thank you very much for helping me out with the initial switch setup. I
> really appreciate it. However, I am stuck with the cdr and call routing
> patterns setup.
>
>
> A. I have attached my cdr_csv file and its cdr log file. As you may see in
> the conf xml that i have 22 items that i need posted onto my cdr logs.
> However, the csv file only shows 15 items. How can i make this work
>
> B. I have setup a few sip trunks. Now how do i set them up so that if i
> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
> goes to gateway B
>
> Many thanks for all the help. I really appreciate it
>
> Best Regards
> Hareem. Haque
>
>
>
>
> ---------- Forwarded message ----------
> From: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 20 Feb 2011 17:01:47 +0000
> Subject: Re: [Freeswitch-users] Need help with CDR setup and call routing
> patterns
> After changing the config, did you reload the config?
>
> reloadxml
> reload mod_cdr_csv
>
> Steve on iPhone
>
>
>
> On 20 Feb 2011, at 16:08, Hareem Haque <hareem.haque at gmail.com> wrote:
>
> > Thank you very much for helping me out with the initial switch setup. I
> really appreciate it. However, I am stuck with the cdr and call routing
> patterns setup.
> >
> >
> > A. I have attached my cdr_csv file and its cdr log file. As you may see
> in the conf xml that i have 22 items that i need posted onto my cdr logs.
> However, the csv file only shows 15 items. How can i make this work
> >
> > B. I have setup a few sip trunks. Now how do i set them up so that if i
> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
> goes to gateway B
> >
> > Many thanks for all the help. I really appreciate it
> >
> > Best Regards
> > Hareem. Haque
> >
> >
> > <cdr_csv.conf.xml>
> > <hareem.csv>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
>
> ---------- Forwarded message ----------
> From: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 20 Feb 2011 17:10:34 +0000
> Subject: Re: [Freeswitch-users] Need help with CDR setup and call routing
> patterns
>
>> B. I have setup a few sip trunks. Now how do i set them up so that if i
>> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
>> goes to gateway B
>>
>
> Handle each in a different extension in the dialplan:
>
> <extension name="route1">
> <condition field="destination_number" expression="^(01192)$">
> <action application="bridge" data="sofia/gateway/gatewaya/$1"/>
> </condition>
> </extension>
> <extension name="route2">
> <condition field="destination_number" expression="^(1\d\d\d)$">
> <action application="bridge" data="sofia/gateway/gatewayb/$1"/>
> </condition>
> </extension>
>
> Adjust the expressions to suit the numbers you want to handle better. The
> contents of the () brackets are placed in variable $1, which is being used
> in the bridge to dial that number via the gateway. You can add prefixes etc
> there if you need them. ^ and $ match the start and end of the string, not
> actual characters. They're called regular expressions if you want to read up
> on them, and are more flexible than just matching absolute strings because
> you match patterns instead.
>
> An example of changing a prefix, should your gateway need that (for example
> converting a local number to an international prefix):
> <condition field="destination_number" expression="^0(1192)$">
> <action application="bridge" data="sofia/gateway/gatewaya/44$1"/>
> </condition>
> 01192 would be dialed as 441192 through gateway a.
>
> -Steve
>
>
>
>
> On 20 February 2011 16:08, Hareem Haque <hareem.haque at gmail.com> wrote:
>
>> Thank you very much for helping me out with the initial switch setup. I
>> really appreciate it. However, I am stuck with the cdr and call routing
>> patterns setup.
>>
>>
>> A. I have attached my cdr_csv file and its cdr log file. As you may see in
>> the conf xml that i have 22 items that i need posted onto my cdr logs.
>> However, the csv file only shows 15 items. How can i make this work
>>
>> B. I have setup a few sip trunks. Now how do i set them up so that if i
>> dial 01192 that call goes to gateway A and if i dial 1XXX then that call
>> goes to gateway B
>>
>> Many thanks for all the help. I really appreciate it
>>
>> Best Regards
>> Hareem. Haque
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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