[asterisk-users] Sox and bad quality when converting to 8 kHz
Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help best regards Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and PlayBack
Hi, when I audio studio should produce an sound file to play back with Asterisk. Whats the best format they should deliver the audio file? Sample Size: 16-bit (2 bytes) Sample Encoding: signed (2's complement) Channels : 1 Sample Rate: 8000 thanks Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple devices wants to call through single peer (trunking)
Hi list, how can I set up an peer, so that behind one IP (NAT) multiple devices can access to this single peer to make outbound calls. Some of these multiple devices will be SIP phones and these SIP phones are trying to make registrations to this peer. best regards Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to avoid AGI script is canceled if caller HangUp
Hi, is there any way to avoid cancel the AGI script if caller is hanging up. That gives me sometimes data mismatch and it is deffcault to clean up in the h extension. I would like that the PHP script called by AGI will run to end.. Some thing can happend with an Macro if caller hang up exactly when call is answered. An Macro called byi the DIAL command will be stoped and data mismatch can occur.. best regards Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.26.2 died after 80 days uptime
Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? best regards Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] change L(x[:y][:z]) parameter of DIAL command after call is bridged
Hi, is there any way from outside change x,y an z after a call is bridged? maybe with AMI interface? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIAL IAX2 vs. SIP
Hi, documentation shows me: Dial(Tech/User:passw...@host/Extension,Timeout,Optionen) This is working for IAX2. If Iam using DIAL(SIP/u...@secret@sip.domian.tls/123456) Asterisk shoes no host with name "sip.domian.tls/123456" How to put in extension if using the DIAL command with userid and secret? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime difference sipusers sippeers
Hi, I would have expected that peers of type friend ( for example an SIP-phone) registring at Asterisk will be searched in sipusers. But the peers will be searched in sippeers. May be sombody can explain the difference? Asterisk 1.4 thanks Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GROUP() decrement
Hi,
how I can decrement the value of GROUP_COUNT() by one after I have before used
GROUP(), so that other channel will get the correct value of GROUP_COUNT().
for examaple
exten => _X!,n,Set(GROUP()=${Provider})
exten => _X!,n,DIAL(SIP/${ext...@${provider})
When Dialstatus is CONGESTION I want to dial again with another provider but I
have to decrement the GROUP of the unused provider.
If there is no function, woud it be possible to call GROUP() in the Macro
called by the DIAL command if DIALSTATUS is ANSWERED?
Or do I have to progamm it outside with AGI?
best regards
Thomas
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[asterisk-users] line disconnected after 20 seconds no reply to our critical packet
Dear all, I have from time to time problems with disconnect after exact 20 seconds. I have these problems from time to time in LAN after using PickUP() with 1.2 I have these problems from time to time in WAN when the internet connection is not reliable with 1.4 Is there any way to fix it? best regards Thomas [Jan 10 13:18:36] WARNING[4102] chan_sip.c: Maximum retries exceeded on transmission 5e608496-e9a1c...@123.123.123.123 for seqno 102 (C ritical Response) [Jan 10 13:18:36] WARNING[4102] chan_sip.c: Hanging up call 5e608496-e9a1c...@80.80.173.155 - no reply to our critical packet. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec problems when using G.723
On Monday 10 November 2008 16:52, Eric "ManxPower" Wieling wrote:
> Thomas Winter wrote:
> > On Sunday 09 November 2008 20:14, Eric "ManxPower" Wieling wrote:
> >> The best (and maybe only way) is to set your client and your service
> >> provider to only do G.723.
> >
> > Really, thats not the way it should work.
> >
> > How I can find out the codec of an incomming call?
> >
> > Is there any way to use ${SIP_CODEC} to try to change to G.723 and then
> > check success?
> > If OK use provider with only allowed G.723 and if not use provider with
> > allowed alaw and ulaw?
>
> I didn't say that is how it should work. I said that is how it does
> work. No, you cannot change the codec of the incoming call in the
> dialplan. SIP_CODEC only sets the codec for the outgoing leg of the call.
With SIP_CODEC you can change the codec for an incomming call.
[Nov 10 17:21:33] NOTICE[12954]: chan_sip.c:3659 try_suggested_sip_codec:
Changing codec to 'ulaw' for this call because of ${SIP_CODEC} variable
How about:
grep with AGI and AMI sip show channels and find the used codec. (Is there an
better way?)
If found G.723.1 use provider with G.723, if not use provider with alaw and
ulaw
Should work, I will give it a try...
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Re: [asterisk-users] Codec problems when using G.723
On Sunday 09 November 2008 20:14, Eric "ManxPower" Wieling wrote:
> The best (and maybe only way) is to set your client and your service
> provider to only do G.723.
Really, thats not the way it should work.
How I can find out the codec of an incomming call?
Is there any way to use ${SIP_CODEC} to try to change to G.723 and then check
success?
If OK use provider with only allowed G.723 and if not use provider with
allowed alaw and ulaw?
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[asterisk-users] Codec problems when using G.723
Hi, I have a problem with codecs. I have an provider with allowed codec alaw, ulaw, g.723 I have SIP clients with codec allowed alaw, ulaw, g.723 If a SIP clients wants call through with g.723 Asterisk is using alaw to connect to the provider, so its not working because only passthrough would work. How I can prevent this? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue reload
On Tuesday 09 September 2008 12:30, Atis Lezdins wrote: > On Tue, Sep 9, 2008 at 1:22 PM, Thomas Winter <[EMAIL PROTECTED]> wrote: > > On Monday 08 September 2008 14:44, Atis Lezdins wrote: > >> On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter <[EMAIL PROTECTED]> > > > > wrote: > >> > I dont have problem to make a reload by AMI. > >> > My questions was if module reload app_queue.so is the right way to do > >> > this, because whis "reload" I reload everything. > >> > > >> > Its fact that I have to do reload queue otherwise Asterisk did not > >> > load realtime database with new settings. > >> > >> Definitely not. > >> > >> Realtime should reload settings on every new call, and this is working > >> for me on periodic_announce and periodic_announce_frequency. However > >> this will work only for new calls, existing calls will have settings > >> as loaded at their enter queue. > >> > >> My Asterisk version is 1.4.19, addons 1.4.6 > >> > >> If you enable debug 1 you should see in your full log: > > > > Hi, > > I have 1.4.21.2 and addons 1.4.7 > > > > If I do reload I have this: > > > > [Sep 9 12:10:12] DEBUG[4709] res_config_mysql.c: MySQL RealTime: Static > > SQL: SELECT category, var_name, var_val, cat_metr > > ic FROM fileconf WHERE filename='queues.conf' and commented=0 ORDER BY > > filename, cat_metric desc, var_metric asc, category > > , var_name, var_val, id > > > > And I have found only this in debug file: > > > > [Sep 9 12:12:02] DEBUG[20173] app_queue.c: Queue test has no realtime > > members defined. No need for update > > > > Might be this is the reason, I do add agents with AMI QueueAdd. > > > > So changes in realtime queues.conf will not be read, I have to do reload. > > Oooh, so you have "Static realtime". I think it isn't supposed to > reload automatically. Go for "Real Realtime" - > http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue or live > with "module reload" :) I see, I use module reload app_queue.so and it works fine. thanks best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue reload
On Monday 08 September 2008 14:44, Atis Lezdins wrote: > On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter <[EMAIL PROTECTED]> wrote: > > I dont have problem to make a reload by AMI. > > My questions was if module reload app_queue.so is the right way to do > > this, because whis "reload" I reload everything. > > > > Its fact that I have to do reload queue otherwise Asterisk did not load > > realtime database with new settings. > > Definitely not. > > Realtime should reload settings on every new call, and this is working > for me on periodic_announce and periodic_announce_frequency. However > this will work only for new calls, existing calls will have settings > as loaded at their enter queue. > > My Asterisk version is 1.4.19, addons 1.4.6 > > If you enable debug 1 you should see in your full log: Hi, I have 1.4.21.2 and addons 1.4.7 If I do reload I have this: [Sep 9 12:10:12] DEBUG[4709] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metr ic FROM fileconf WHERE filename='queues.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category , var_name, var_val, id And I have found only this in debug file: [Sep 9 12:12:02] DEBUG[20173] app_queue.c: Queue test has no realtime members defined. No need for update Might be this is the reason, I do add agents with AMI QueueAdd. So changes in realtime queues.conf will not be read, I have to do reload. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue reload
On Sunday 07 September 2008 21:49, Atis Lezdins wrote: > On Sun, Sep 7, 2008 at 4:56 PM, Thomas Winter <[EMAIL PROTECTED]> wrote: > > is not work for periodic-announce-frequency and periodic-announce. > > An reload is necessary. > > Asterisk is 1.4.21.1 > > It shouldn't be necessary. However you can try "queue show > " from CLI, that would trigger reloading queue's settings. > If it doesn't work, enable "core set debug 1" and post output when > executing reload. > > As for executing CLI commands, see manager action Command: > http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command I dont have problem to make a reload by AMI. My questions was if module reload app_queue.so is the right way to do this, because whis "reload" I reload everything. Its fact that I have to do reload queue otherwise Asterisk did not load realtime database with new settings. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue reload
On Saturday 06 September 2008 21:47, Brian wrote: > Hi Thomas, > > The queue definitions and its member list will be reloaded each time a > caller joins the queue. So you don't need to reload it manually. Hi, is not work for periodic-announce-frequency and periodic-announce. An reload is necessary. Asterisk is 1.4.21.1 best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime queue reload
Hi, Iam using queues through realtime, works fine. After making changes how do I make Asterisk aware? Is "module reload app_queue.so" through AMI the correct way to do this? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tuesday 05 August 2008 18:50, Al Baker wrote:
> Err - Ok - let me ask this in MUCH simpler way
>
> 1 - In dialplan , you set a Variable called "MYVAR", to "Apple"
>
> 2 - You go into MACRO and NOOP the VALUE of "MYVAR" -What SHOULD it BE ?
You should better use
M(x[^arg]) - Execute the Macro for the *called* channel before connecting
to the calling channel. Arguments can be specified to the Macro
using '^' as a delimeter.
Its not a problem to get vars in the MACRO
> 3 - While IN MACRO you set VALUE of "MYVAR" = "Pear"
>
> 4 - You leave MACRO and get back to DIALPLAN and NOOP the value of MYVAR
> - What should it be
regardless if you set MYVAR, _MYVAR or __MYVAR in the MACRO, it is not
working.
> === 2nd Question ===
> CAN the DIAL command call a SUBROUTINE instead of a MACRO ?
>
> If so WOULD that help him out ?
>
> Any clarification much apprecatted
>
> Tilghman Lesher wrote:
> > On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
> >> Hi all,
> >>
> >> Iam using an DIAL Command wird Macro if callee is answer the call.
> >>
> >> exten => 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
> >> exten => 123,n,NoOp( ${var_from_macro})
> >>
> >>
> >> In Macro test_connect Iam generating an new variable var_from_macro and
> >> would like to use this var in the original dialplan.
> >> I tried also __var_from_macro but didnt work. How can I set vars in
> >> macros called by DIAL so that I can use these vars in the Dialplan or in
> >> the h extention.
> >
> > There isn't any good way to do that, period. When it comes to
> > inheritance, variables are only inherited from a master channel to a
> > slave channel. In the case of the Macro operating within the Dial, that
> > Macro is occurring exclusively on the slave channel. You cannot directly
> > set variables on other channels (for obvious race-condition reasons).
> >
> > However, you could do this in a roundabout way, either by using a
> > database or by using shared variables in trunk. You'd need to first set
> > (in the master channel, before the Dial) an inherited variable containing
> > the name of the master channel, i.e. Set(_masterchan=${CHANNEL(name)}),
> > then use that inherited variable to set the shared variable in the master
> > channel from the slave channel, i.e. Set(SHARED(foo,${masterchan})=...).
> > Finally, you would be able to access the shared variable in the master
> > channel with ${SHARED(foo)}. Again, the SHARED function is only
> > available in trunk at this time, although you could probably backport it
> > to 1.4 with minimal trouble.
>
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Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tuesday 05 August 2008 18:02, Ruddy Gbaguidi wrote: > I don't think you can do that because, asterisk, in the caller thread > will only read MACRO_RESULT to know if he has to connect the call or not. > A workaround will be to : > 1. before the dial, add a row in a database table and retrieve an id > 2. pass the id to test_connect and test_connect will then write his > variable value into the database > 3. after the dial,. use the id to retrieve the needed variable. > thanks, but Iam using additional redirect though AMI, so it could be that the channel is redirected to another context and never see exten after DIAL or h extension in that context. It seemed to be that I have to add additional programming outside the dialplan if doing redirect. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tuesday 05 August 2008 18:04, Tilghman Lesher wrote:
> On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
> > Hi all,
> >
> > Iam using an DIAL Command wird Macro if callee is answer the call.
> >
> > exten => 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
> > exten => 123,n,NoOp( ${var_from_macro})
> >
> >
> > In Macro test_connect Iam generating an new variable var_from_macro and
> > would like to use this var in the original dialplan.
> > I tried also __var_from_macro but didnt work. How can I set vars in
> > macros called by DIAL so that I can use these vars in the Dialplan or in
> > the h extention.
>
> There isn't any good way to do that, period. When it comes to inheritance,
> variables are only inherited from a master channel to a slave channel. In
> the case of the Macro operating within the Dial, that Macro is occurring
> exclusively on the slave channel. You cannot directly set variables on
> other channels (for obvious race-condition reasons).
I see..
> However, you could do this in a roundabout way, either by using a database
> or by using shared variables in trunk. You'd need to first set (in the
> master channel, before the Dial) an inherited variable containing the name
> of the master channel, i.e. Set(_masterchan=${CHANNEL(name)}), then use
> that inherited variable to set the shared variable in the master channel
> from the slave channel, i.e. Set(SHARED(foo,${masterchan})=...). Finally,
> you would be able to access the shared variable in the master channel with
> ${SHARED(foo)}. Again, the SHARED function is only available in trunk at
> this time, although you could probably backport it to 1.4 with minimal
> trouble.
I dont want to use trunk, but thanks for info...
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[asterisk-users] vars in Macros called by DIAL with option M
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten => 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten => 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and would
like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
best regards
Thomas
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[asterisk-users] queues and MEMBERINTERFACE for AGI script
Hi, iam using and queue and starting an AGI script after caller connected to agent. How to find out in the script the connected agent, MEMBERINTERFACE seemed to be not work, either as variable in the queue command and also not in the AGI script. How to found out which agent is connected to calling channel? I try to avoid to using LOCAL channels, because I like the function ringinuse. regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue with different music for each caller
Hi, I tried this before I ask here on the list. In 1.2 SetMusicOnHold did not work. The Moh class defined in queues.conf is overwriting any SetMusicOnHold values of the caller channel. You can see this if you use periodic announce, the Moh call is printed in the CLI and is allways the class defines in queues.conf. I have now the choice to switch to 1.4 or implement for every music an single queue. best regards Thomas On Wednesday 25 June 2008 06:55, Martin Schrott - thinking:systems wrote: > Hello Thomas, > > no problem. > In asterisk <1.6 use > SetMusicOnHold(musiconholdname) > > then it will work in older Asterisk versions! > > br, > Martin > > - Original Message - > From: "Thomas Winter" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, June 24, 2008 5:50 PM > Subject: Re: [asterisk-users] Queue with different music for each caller > > On Tuesday 24 June 2008 15:22, Martin Schrott - thinking:systems wrote: > > Hello Thomas > > > > > > you can use different music for each caller if you like. > > > > in extensions.conf you can set the music class. > > > > exten => s,n,Set(CHANNEL(musicclass)=yourmusicforthiscaller) > > Hi Martin, > > thanks for your suggestion, I forgot to notice that Iam still using 1.2.X > > Jun 24 17:45:31 ERROR[17784]: pbx.c:1437 ast_func_write: Function CHANNEL > not > registered > > So, this didnt work for me. > > best regards > Thomas > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue with different music for each caller
On Tuesday 24 June 2008 15:22, Martin Schrott - thinking:systems wrote: > Hello Thomas > > > you can use different music for each caller if you like. > > in extensions.conf you can set the music class. > > exten => s,n,Set(CHANNEL(musicclass)=yourmusicforthiscaller) Hi Martin, thanks for your suggestion, I forgot to notice that Iam still using 1.2.X Jun 24 17:45:31 ERROR[17784]: pbx.c:1437 ast_func_write: Function CHANNEL not registered So, this didnt work for me. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue with different music for each caller
Hi, is there an possibilty to have for each caller different music when queued. I see there only the global musiconhold = default in queues.conf, what menas same musci for all waiting callers. Any other idea to realize this? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] custom functions is voicemail
Hi, I want to add some custom functions in voicemail. For example user can switch SMS on/off or the voicemail global on/off. Whats best way to do this? modify app_voicemail.c or or do everything in dialplan? or any other solutions (Asterisk 1.2.X please) best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue delay between calls to agents
On Thursday 05 June 2008 01:09, Tariq .. wrote: > you can reduce the 5 seconds to any number you wish.. but from a personal > experience .. if you put the retry to zero.. nothing will change.. i > suggest to use "1" as your minimum aiting number Tarek Sawah thanks, retry = 1 is working retry = 0 looks like default (5s) best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue delay between calls to agents
Hi, I want to reduce the dead time before the queue is calling the next agent. I see there 5 seconds delay. It is possible to reduce this time, or what is Asterisk doing within this timeframe. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue is sending calls to Agents even when they are in use
On Tuesday 03 June 2008 23:22, Atis Lezdins wrote: > chan_agent with AgentCallbackLogin was working but not completely > stable for my dialplan which was quite heavy when I was on 1.2, > however you may try that out. Or just upgrade to 1.4 (or even 1.6 and > try state_interface) Iam using API Action QueueAdd through an WebGUI and quite happy with this. I hope I can jump from 1.2 to 1.6 without touching 1.4:) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] applicationmap in features.conf Asterisk 1.2 is ignoring DIAL tT options
Hi, I found out that GoTo in applicationmap is not working. OK, LOCAL is working. but I expected that applicationmap is using the DIAL option tT. But it doesnt, it works without tT Option, so also callee can use internal functions if callee knows the code. Any workaround avaiable? best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] overlap calls from NT-BRI timeout problem
Hi, Iam getting calls from an POTS system on an NT port. Multiport BRI card running bristuff 0.3. >From time to time the recognized number is incomplete and dial failed. Is there any way to increase timeout waiting for called numbers? Because dialed numbers can be from 3 to 13 digits there is no way to regocnize the completeness of the number. Other option switch to en-block dialing is not possible because of bad documentation of the old POTS system. I hear the old telephone provider can validate numbers and so they can avoid such problems. I guess I do not get access to this POTS club information. best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mpg123 on Thecus N2100
On Sunday 11 November 2007 01:38, Tzafrir Cohen wrote: > On Sat, Nov 10, 2007 at 01:40:23PM -0600, Eric ManxPower Wieling wrote: > > Thomas Winter wrote: > > > Hi, > > > Iam running debian etch on thecus n2100 (Xscale 80219) > > > I do not have MoH because standard mpg123 gives only loud noise. > > > > > > I can not compile mpg123 from asterisk because of option -m486. > > > > > > Any way to get MoH running on this board. > > > > What version of Asterisk? > > > > Asterisk 1.0.x MoH requires mpg123 v0.59r. > > To be more specific, if you do use mpg123, just use any recent version. > For a long time the development of mpg123 has stalled and a number of > buggy versions were around. The version of mpg123 in Etch should owrk well. I have systems with high CPU load with mpg123 from etch, no problems with mpg123 v0.59r at all. I see, on this board I can only use wav for playing MoH thanks... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mpg123 on Thecus N2100
On Saturday 10 November 2007 20:40, Eric "ManxPower" Wieling wrote: > Thomas Winter wrote: > > Hi, > > Iam running debian etch on thecus n2100 (Xscale 80219) > > I do not have MoH because standard mpg123 gives only loud noise. > > > > I can not compile mpg123 from asterisk because of option -m486. > > > > Any way to get MoH running on this board. > > What version of Asterisk? V 1.2 can not compile mpg123 v0.59r. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mpg123 on Thecus N2100
Hi, Iam running debian etch on thecus n2100 (Xscale 80219) I do not have MoH because standard mpg123 gives only loud noise. I can not compile mpg123 from asterisk because of option -m486. Any way to get MoH running on this board. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bristuff: music on hold but no dialoptions tT defined.
Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983] logger.c: -- Started music on hold, class 'default', on channel 'Zap/8-1' Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Stopped music on hold on Zap/8-1 Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Started music on hold, class 'default', on channel 'Zap/8-1' Oct 22 11:20:55 VERBOSE[911] logger.c: == Spawn extension (macro-call, s, 2) exited non-zero on 'Zap /8-1' in macro 'tmp_call' ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] German SIP and/or IAX providers?
On Friday 12 October 2007 04:38, Ken D'Ambrosio wrote: > Hi, all. My company is setting up a branch office in Germany, and I'm > very interested in a VoIP provider over thataway. However, I'd need a few > things: > > - Reliability. Can't have my branch office's DID's just going down. A then you have to chosse for incomming calls ISDN and route from there to other destinations if needed. For outgoing use an sip provider, ISDN for special numbers not supported by SIP provider. > And that's about -it-. I'm even willing to pay a reasonable premium, so > long as it gets me a VoIP provider with the above restrictions. save your money and spend it for ISDN lines as long as the calls have more then 80% final destination in your local office. Consider also that your DSL line can go down, thats not the responsibilty of the SIP provider. ISDN is in Germany extremly reliable. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bristuff for hfc card on Xscale 80219
On Thursday 19 July 2007 04:27, Tzafrir Cohen wrote: > On Wed, Jul 18, 2007 at 12:44:29AM +0200, Thomas Winter wrote: > > Hi, > Frozen or crashed? Do you see the console of the system? serial console is dead. kernel is 2.6.18-4 debian Etch. bristuff is latest zaptel-1.2.19 and asterisk-1.2.22 I tried older bristuff before but same result. > Can you load zaptel and zaphfc those modules with debug=1 yes, I assume the board is frozen because of to much load on the pci bus. if making tail of the syslog I can see tons of lines before board is frozen. I have seen similar messages before if using two HFC cards on old PC, so I assume not enough horse power for cheap HFC card and bristuff. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bristuff for hfc card on Xscale 80219
Hi, compile and load of modules works fine. After ztcfg I can see . . Changing signalling on channel 1 from Unused to Clear channel Changing signalling on channel 2 from Unused to Clear channel Changing signalling on channel 3 from Unused to HDLC with FCS check and then the board is frozen. Any ideas? regards Thomas ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I improve call quality?
On Friday 20 April 2007 20:01, Adrian Marsh wrote: > Hi All, > > I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used > for PSTN calls via IAX2. > Our 'net link is a dedicated 2Mb fibre connection (of which we have ever > used 50% max bandwidth). We've no E1/T1 links, everything is IP based. > > My boss complains that many of the calls he holds with others has a bad > quality. He also says that its not just him. > Iam using Gradwell in UK with SIP with several lines and I do not have these problems. Try SIP and if the problem is gone check your IAX configuration. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hints howto reset if wrong subscription Asterisk 1.2
Hi, sometimes Asterisk told me in the subscription: status confirmed so LED is on if the softphone is disconnected or the registration has expired. So the whole weekend LEDs have the wrong status. Manager Command Extensionstate is working correct, only the subscription is wrong. How can I reset this by hand? SIP clients are in relatime, dialplan is txt file. I tried to delete astdb and fields in the mySQL database, but without success. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play audio and continue to next priority before audio ends...
Am Monday 09 April 2007 23:20 schrieb Alejandro Mejía: > Hello list members. > > I would like to know how to playback an audio file to the caller, and while > it's played asterisk to continue executing the next priorities on > extensions.conf > That's not the case when using "playback" command, because the next > priority is executed until the audio file ends playing. I want to evaluate > some variables while caller hears the audio file. > > Any ideas? Use StartMusicOnHold() and StopMusicOnHold() If use use MoH type=files MoH play from start of file, so ist similar to Playback ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager Originate and Var to long
Hi, I use Originate to make a call. I have problems to bring my vars into the channel. Are there restrictions more then only 24 vars at mentioned at www.voip-info.org? Any workaround to get this running? WARNING[4641]: manager.c:1365 get_input: Dumping long line with no return from 127.0.0.1: regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_musiconhold.c:1243 load_module: No music on hold classes configured
Hi, I am using relatime for musiconhold.conf. After starting Asterisk I have to do an reload, otherwise no MoH is avaiable. Bug or do I have to change loading of modules in modules.con? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP provider did not send BYE if callee is an queue
Hi, I have an strange problem and did not understand what happened here. Iam using an SIP provider to call an analog POTS phone. I orginate the call to the analog phone and send the call in an queue. If the analog phone hangs up, the SIP provider did not send BYE and Asterisk think the line is still up. I can repeat this with version 1.2.16 without any problems. If I make call through from Asterisk to an anlog phone and hang up the phone the Sip provider sends Asterisk an BYE and the call is hang up. So whats the reason why the SIP provider did not send an BYE if the callee is an queue? I did find an workaround to avoid this situation of endless calls but whats the detailed reason for this strange endless call. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play file and action only stop if one defined key has been pressed
Am Friday 09 March 2007 23:51 schrieb Steve Murphy:
> On Fri, 2007-03-09 at 23:01 +0100, Thomas Winter wrote:
> > I didnt see the option.
> >
> > The number can be different and is stored in mySQL
> >
> > exten => ${tmp_var},1,NoOp(INFO key pressed)
> > exten => ${tmp_var},n,GoTo(s,restart)
>
> Woa! can you really do that? I would have to check the code, but I have
> the strong impression that you cannot use a variable in the extension
> name field, they are not evaluated, nor are they really evaluatable. All
> the extensions in a context are compared when looking for a match to a
> target location, but
> I know that goto's etc, can use a variable in a reference, but not in a
> definition like this.
I can do this, but it is not working as I wrote before.
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Re: [asterisk-users] play file and action only stop if one defined key has been pressed
Am Friday 09 March 2007 22:27 schrieb Time Bandit:
> > I would like that user cann press 3 and then actions can be taken.
> > Problem ist if the pressed key not 3 the user jumps to extension i and
> > then the file will be played from start again.
> >
> > I would like that the play of file is only stopped if the user has
> > pressed the key 3.
> >
> > What for an command can i use to make this happened?
>
> check http://www.voip-info.org/wiki-Asterisk+cmd+Background
>
> I think the m option is what you are looking for
thanks,
I didnt see the option.
The number can be different and is stored in mySQL
exten => ${tmp_var},1,NoOp(INFO key pressed)
exten => ${tmp_var},n,GoTo(s,restart)
is not working because when Asterisk reads the dialplan ${tmp_var} is EMPTY.
Iam now using an workaround. I have done 9 different context and depends on
the key Iam using Background with option m in each context
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[asterisk-users] play file and action only stop if one defined key has been pressed
Hi, I would like that user cann press 3 and then actions can be taken. Problem ist if the pressed key not 3 the user jumps to extension i and then the file will be played from start again. I would like that the play of file is only stopped if the user has pressed the key 3. What for an command can i use to make this happened? exten => i,1,GoTo(restart) exten => 3,1,NoOp(action) exten => s,n(restart),Background(file) best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue information in mySQL
Hi, is it possible to have the information stored in /var/log/asterisk/queue_log realtime in mySQL? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: musiconhold.conf in realtime
If i do an asterisk -rx "moh reload" MoH stops and restarts on exsisting
channels.
If I do an moh reload through the Manager Interface Sound is dead on exsisting
channels.
Any other idea for an workaround?
Hi,
I have problems with 1.2.14 and musiconhold.conf and realtime.
I have to do moh reload at CLI to use the classes stored in mysql.
Otherwise nothing is found if using SetMusicOnHold(${v_moh}) or DIAL with
MoH.
Are there known problems or how to get this running?
thanks
Thomas
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[asterisk-users] musiconhold.conf in realtime
Hi,
I have problems with 1.2.14 and musiconhold.conf and realtime.
I have to do moh reload at CLI to use the classes stored in mysql.
Otherwise nothing is found if using SetMusicOnHold(${v_moh}) or DIAL with MoH.
Are there known problems or how to get this running?
thanks
Thomas
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Re: [asterisk-users] queues and LOCAL for members
Am Friday 02 February 2007 23:48 schrieb BJ Weschke: > On 2/2/07, Thomas Winter <[EMAIL PROTECTED]> wrote: > > Hi, > > > > I have an queue stored in relatime and defined members called through > > LOCAL/ > > > > I found out that if I call the members through the LOCAL think the queue > > statistics is not updated. > > > > Any idea, or isnt possible to call members with LOCAL channel. > > There's been some efforts to have Local channels as viable queue > members. I'm not quite sure that I understand your issue. Can you post > some more details possibly in a bug on bugs.digium.com ? Thanks, I found out LOCAL is working. I have been confused because an change of queue-members and an reload has reset the queue statistic. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queues and LOCAL for members
Hi, I have an queue stored in relatime and defined members called through LOCAL/ I found out that if I call the members through the LOCAL think the queue statistics is not updated. Any idea, or isnt possible to call members with LOCAL channel. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] licence quick question
Hi, If I develope an dialplan, some AGI and AMI functions for Asterisk and ship it as an complete product to an coustomer, do I have to put my developed code or the complete product under the GPL? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HFC-card and TDM400 with bristuff
Hi, is it possible to run an HFC-card with bristuff and an TDM400 in one PC? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: mySQL and to many connections with SQL statement UPDATE
Am Sunday 24 December 2006 16:53 schrieb JR Richardson:
> > If Iam doing UPDATE SQL statements I got an overload for connection.
> > am doing everytime an Disconnect ${connid}) but this is ignored.
> >
> > any idea?
>
> You must clear the resut ID and also issue a disconnect to the
> connection ID, see priority 4 and 5.
> Or a simpler method is to setup realtime access to the database and
> use the realtime cmd, this will control the mysql access and you won't
> have to worry about the connections.
>
> http://www.voip-info.org/wiki/view/Asterisk+cmd+RealTime
Iam using lots of SELECT statements and do not have problems.
The problem occurs if using the SQL statement UPDATE.
I tried with and without MYSQL(Clear ${resultid}.) If using I got an working
that there is nothing to clear. Usage seemed to be onyl for SELECT
statements.
MYSQL(Disconnect ${connid}) is obviously not working.
The connection is not closed and mySQL is running shortly out of avaiable
connections.
WARNING[12189] app_addon_sql_mysql.c: aMYSQL_query: mysql_store_result()
failed on query UPDATE ...;
best regards
Thomas
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[asterisk-users] mySQL and to many connections with SQL statement UPDATE
Hi,
If Iam doing UPDATE SQL statements I got an overload for connection.
am doing everytime an Disconnect ${connid}) but this is ignored.
any idea?
best regards
Thomas
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Re: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp
Am Sunday 29 October 2006 01:31 schrieb Dovid B: > Half asleep. Sorry for my last post. I believe you still need port > forwarding for IAX. Time to keep to my bed time. If works as long as you have notransfer=no at both ends. Iam concerned that with SIP Asterisk is bridging up and I do not receive the audio stream. Asterisk should Hangup the line if Audio stream is announced to com from another IP. Iam wonderung that there is no setting for this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk behind NAT and without portforwarding for rtp
Hi, I have an Asterisk behind NAT. NAT=yes and canreinvite=no in globals and for the peer. I call an peer. The peer advice to use another IP for the audio and my Asterisk is sending audio stream to the Audio server. Because of missing port forwarding I will not receive the audio stream and hear nothing. I would expect that Asterisk will cancel the connection, but this didnt happened. Asterisk will follow the reinvite from the peer. Any solution except portforwarding? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and ISDN and Hylafax
Am Thursday 26 October 2006 23:35 schrieben Sie: > On Thu, 26 Oct 2006, Thomas Winter wrote: > I would recommend the Eicon DIVA Server 4BRI cards. They have a > capi interface which is used by chan-capi (chan-capi.org) and > onboards DSPs for the faxing. > You can use this for send and receive faxes and/or use capi4hylafax > in parallel with asterisk/chan-capi. sounds good, you think it will run reliable? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and ISDN and Hylafax
Hi, I have to set up an Asterisk with an 4-port BRI card. Hylafax should send and receive fax. Will this work reliable? Any recommandations for an 4-port BRI card? Other alternatives except analog fax units? thanks for your help best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanIsAvail() in 1.2.12.1
Am Friday 06 October 2006 23:03 schrieb Douglas Garstang:
> *CLI> -- Executing NoOp("SIP/3254101-0817a220", "*** Originated call
> "Chocolate Chip" <3254101> -> 3254103") in new stack -- Executing
> NoOp("SIP/3254101-0817a220", "FOO1") in new stack -- Executing
> ChanIsAvail("SIP/3254101-0817a220", "SIP/3254103") in new stack
>
> It never makes it past the call to ChanIsAvail(). Dialplan processing just
> completely stops at this point. What's up with that???
Asterisk SVN-tag-1.2.12.1
Its working fine. (Iam using Realtime)
best regards
Thomas
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[asterisk-users] Server for Asterisk PCI
Hi, can anybody recommend HP Proliant ML110 for Asterisk and ISDN interface cards? This Server has only two PCI 32Bit/33 MHz 3,3 Volt. Is this OK for PRI cards? thanks.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP user deny and permit for calls through Asterisk
Hi, i tried deny and permit in the peer definition. It works fine for registration purpose. But if the peer is dialing through Asterisk these settings are ignored. Only username and password are used for authentification. Is there anythink additional what I can use to prevent that the phone is making calls from another IP. I tried also host=IP and defaulip, but it did not help when the phone is placing calls through Asterisk. thanks Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sipbuddies realtime fields and latest documentation
Hi, I used voip-info.org for setup my realtime users. The mySQL table did not include for example the option call-limit. Where I can find information whats the correct field name to adjust my mySQL table? thanks Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)
Am Tuesday 25 April 2006 11:24 schrieb Olle E Johansson: > 25 apr 2006 kl. 00.24 skrev Thomas Winter: > > Am Monday 24 April 2006 18:39 schrieb Doug Lytle: > >> Thomas Winter wrote: > >>> Hi, > >>> > >>> I dont want to have in the SIP HEADER the CALLERID(name) (the > >>> Display > >>> Name) for the initial INVITE to an SIP proxy. > >>> > >>> If I use SET(CALLERID(name)=) the display-name is "asterisk". > >> > >> Just a guess, try: > >> > >> > >> SET(CALLERID(name)=" ") > > > > Hi, > > > > Asterisk will use this space. > > -> FROM: " " > We do insert "asterisk" when we have no caller ID name. In what > situations don't you > want a caller ID name at all? I am a bit curious here, in order to > understand. > > Regards > /Olle Hi, I have an gateway provider. He accepts mynumber in the displayname: FROM: "mynumber" or not using the displayname and number at all: FROM: if I do not want to show mynumber on the called POTS phone. With * I have allways an displayname in the FROM field and can not disable showing my CALLERID to the called phone. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)
Am Monday 24 April 2006 18:39 schrieb Doug Lytle: > Thomas Winter wrote: > > Hi, > > > > I dont want to have in the SIP HEADER the CALLERID(name) (the Display > > Name) for the initial INVITE to an SIP proxy. > > > > If I use SET(CALLERID(name)=) the display-name is "asterisk". > > Just a guess, try: > > > SET(CALLERID(name)=" ") > Hi, Asterisk will use this space. -> FROM: " " http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent
Am Wednesday 19 April 2006 16:37 schrieb Marco Mouta: > How do I report a Bug to Digium? or asterisk project? > Did you report this bug? I checked and have seen only an timeout in the channel will kill the dead channels. Iam using GROUP_COUNT, so it is easy to kill my Asterisk if somebody is make some calls and disconnect the SIP-client every time. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP HEADER FROM: without CALLERID(name)
Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP proxy. If I use SET(CALLERID(name)=) the display-name is "asterisk". I want to have the SIP HEADER like this: FROM: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension
Am Thursday 20 April 2006 01:21 schrieb tom: > Thomas Winter wrote: > > I have done additional tests, because the documentation sample was not > > 100 % identical to my register command. > > > > OK: > > register => 44198:[EMAIL PROTECTED]/200 > > This jumps to 200, s is also working > > > > NOT OK: > > user:[EMAIL PROTECTED]/200 > > It looks for extension user and is ignoring 200 or anythink else > > > > I think the non numeric username is the problem. > > > > Yes, I have done an restart of Asteriks after changing the sip.conf. > > Excuse me for sounding silly, but isn't the extension you mark at the > end sent to your provider as the extension that they should use when > calling you (ie. in the authentication statement, the remote server > tries to connect as 200@ ) which is why some providers with > broken sip implementations require you to have a specific extension > after the /. > > ie. the extension they call on, is not neccesarily what is stated in the > register statement, that's just the extension you've told them to call > you on. > > Do a sip debug provider.com in the asterisk CLI to see what happens when > the call comes in. I have used Ethereal. The initial call comes in: SIP/SDP Request Invite sip:[EMAIL PROTECTED] user is allways the username from the register , there is no information regards the user extension from the register command in. (or I didnt see them) If * is register at the sip provider there are contact bindings send to the SIP-proxy with the extension from the register command and some messages with bindings are comming back. I dont know how this is related to the call INVITE. Anyway, I have tested 4 different provider, two numeric and two alphanumeric username. numeric is working and alphanumeric is not working. If the provider software is broken it would be also good to know. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension
I have done additional tests, because the documentation sample was not 100 % identical to my register command. OK: register => 44198:[EMAIL PROTECTED]/200 This jumps to 200, s is also working NOT OK: user:[EMAIL PROTECTED]/200 It looks for extension user and is ignoring 200 or anythink else I think the non numeric username is the problem. Yes, I have done an restart of Asteriks after changing the sip.conf. Am Wednesday 19 April 2006 23:03 schrieb Aaron Daniel: > Have you tried sending it to a different extension number? I've got the > registrations working on my home server where I register with one number > and have it drop in on a totally different number in the context. > > register => 44198:[EMAIL PROTECTED]/200 > > Aaron > > On Wed, 19 Apr 2006, Thomas Winter wrote: > > Hi, > > > > [general] > > context=Sip_in > > register => 1234:[EMAIL PROTECTED]/s > > > > s is the same, it still looks for an extension 1234 in the context Sip_in > > and did not use /s > > > > Asterisk is 1.2.7 > > > > Am Wednesday 19 April 2006 22:48 schrieb Aaron Daniel: > >> I'm not gonna say much for the documentation, but I would suggest if you > >> want to bypass that problem, add /s (or whatever extension) to the > >> register statement so you know for absolute sure that incoming calls on > >> the registration will go to the extension that you expect. > >> > >> Aaron > >> > >> On Wed, 19 Apr 2006, Thomas Winter wrote: > >>> Hi, > >>> > >>> the documentation of sip.conf is telling me this: > >>> > >>> ;register => 1234:[EMAIL PROTECTED] > >>> ; > >>> ; This will pass incoming calls to the 's' extension > >>> > >>> > >>> In reality it jumps to the extension 1234 in the context and not to s > >>> So it is much more complicate to write an proper dialplan. > >>> > >>> Is this an bug or is the documentation not up to date? > >>> > >>> best regards > >>> > >>> Thomas > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: sip.conf and jump from register to the extension
Hi, [general] context=Sip_in register => 1234:[EMAIL PROTECTED]/s s is the same, it still looks for an extension 1234 in the context Sip_in and did not use /s Asterisk is 1.2.7 Am Wednesday 19 April 2006 22:48 schrieb Aaron Daniel: > I'm not gonna say much for the documentation, but I would suggest if you > want to bypass that problem, add /s (or whatever extension) to the > register statement so you know for absolute sure that incoming calls on > the registration will go to the extension that you expect. > > Aaron > > On Wed, 19 Apr 2006, Thomas Winter wrote: > > Hi, > > > > the documentation of sip.conf is telling me this: > > > > ;register => 1234:[EMAIL PROTECTED] > > ; > > ; This will pass incoming calls to the 's' extension > > > > > > In reality it jumps to the extension 1234 in the context and not to s > > So it is much more complicate to write an proper dialplan. > > > > Is this an bug or is the documentation not up to date? > > > > best regards > > > > Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: sip.conf and jump from register to the extension
Hi, the documentation of sip.conf is telling me this: ;register => 1234:[EMAIL PROTECTED] ; ; This will pass incoming calls to the 's' extension In reality it jumps to the extension 1234 in the context and not to s So it is much more complicate to write an proper dialplan. Is this an bug or is the documentation not up to date? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g.726 codec not working in one direction
Hi, Iam using Asterisk Asterisk 1.2.5 Iam calling: NOT OK: phone A ->ulaw -> Asterik-A -> gsm -> Asterisk-B -> g.726 -> POTS phone B NO sound from from phone A to phone B, phone B to phone A works If iam using ulaw to connect from Asterisk-B to POTS phone B everythink is OK: OK: phone A ->ulaw -> Asterik-A -> gsm -> Asterisk-B -> ulaw ->POTS phone B Any idea how this can happened? If additional information required please ask.. best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Group funcations not functioning
On Monday 10 April 2006 22:04, Raymond Chen wrote:
> Dear all,
>
> we have try to limit the outgoing channel by using GROUP() and
> GROUP_COUNT() to limit number of calls to a channel/trunk. but lately
> we upgraded to 1.2.5, 1.2.6 or SVN 1.2 , both functions not work at
> all. Is this a bug or just a misconfiguration on our part?
>
> exten => s,1,Set(GROUP()=${count})
> exten => s,n,GotoIf($[${GROUP_COUNT(${count})} > 1]?IncCount)
It works.
May be this helps:
exten => _9501,n,Set(GROUP(BlueSip9501)=add)
exten => _9501,n,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])} >
1]?Block:Call)
best regards
Thomas
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Re: [Asterisk-Users] Call me for testing my system
On Monday 10 April 2006 13:49, [EMAIL PROTECTED] wrote:
> Could you try again please?
>
> --- Thomas Winter <[EMAIL PROTECTED]> a écrit :
> > On Monday 10 April 2006 11:59, [EMAIL PROTECTED]
> >
> > wrote:
> > > Dear User,
> > >
> > > Anybody could dial these sip uri :
> > >
> > > sip:[EMAIL PROTECTED] (french)
> > > sip:[EMAIL PROTECTED] (music 60s)
> > > sip:[EMAIL PROTECTED] (french)
> >
> > Hi,
> >
> > sip:[EMAIL PROTECTED] (french)
> > sip:[EMAIL PROTECTED] (music 60s)
> >
> >
> > No Sound or voice!
> >
> > -- Executing Dial("SIP/210-f032",
> > "SIP/[EMAIL PROTECTED]|121|@dialout|") in
> > new stack
> > -- parse_srv: SRV mapped to host nxs.yi.org,
> > port 5060
> > -- Called [EMAIL PROTECTED]
> > -- SIP/nxs.yi.org-4a0f answered SIP/210-f032
> > -- Attempting native bridge of SIP/210-f032 and
> > SIP/nxs.yi.org-4a0f
sip:[EMAIL PROTECTED] (music 60s)
I can hear the great Asterisk MoH music!
best regards
Thomas
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Re: [Asterisk-Users] Call me for testing my system
On Monday 10 April 2006 11:59, [EMAIL PROTECTED] wrote:
> Dear User,
>
> Anybody could dial these sip uri :
>
> sip:[EMAIL PROTECTED] (french)
> sip:[EMAIL PROTECTED] (music 60s)
> sip:[EMAIL PROTECTED] (french)
>
Hi,
sip:[EMAIL PROTECTED] (french)
sip:[EMAIL PROTECTED] (music 60s)
No Sound or voice!
-- Executing Dial("SIP/210-f032", "SIP/[EMAIL PROTECTED]|121|@dialout|") in
new stack
-- parse_srv: SRV mapped to host nxs.yi.org, port 5060
-- Called [EMAIL PROTECTED]
-- SIP/nxs.yi.org-4a0f answered SIP/210-f032
-- Attempting native bridge of SIP/210-f032 and SIP/nxs.yi.org-4a0f
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Re: [Asterisk-Users] How to set busy
On Sunday 09 April 2006 08:46, Benoit Panizzon wrote: > On Sunday 09 April 2006 06:02, Miles Scruggs wrote: > > For multiline phones how do you set SIP channels to busy. For instance > > if SIP/101 is on a call then dial would return busy. Right now it just > > starts ringing on line X, and stacks up from there. > > I suppose incominglinit=1 in the sip.conf of that phone works exactly the > wrong way round? incominglimit and outgoinglimit is replaced by call_limit in Asterisk 1.2. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to set busy
On Sunday 09 April 2006 06:02, Miles Scruggs wrote:
> For multiline phones how do you set SIP channels to busy. For instance
> if SIP/101 is on a call then dial would return busy. Right now it just
> starts ringing on line X, and stacks up from there.
${DIALSTATUS} BUSY comes from the phone.
You can limit the possible lines to an phone with call_limit.
If you have call_limit=1 you will never get an BUSY from the phone.
If call_limit smaller or equal the maximum call to an phone, you will never
get an BUSY.
So use groupcount and make your own logic. Then you know whats going on.
best regards
Thomas
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