[asterisk-users] Fotos 18/08 .
11:09:12 AM Fotos 18/08..: Imagens Anexadas..: DSC_0401.jpg - DSC_0402.jpg - DSC_0403.jpg Videos Hotmail.com..: www.hotmail.com/videos.avi _ Brrr... its getting cold out there Find someone to snuggle up with http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fdating%2Enz%2Emsn%2Ecom%2Fchannel%2Findex%2Easpx%3Ftrackingid%3D1048628&_t=773568480&_r=nzWINDOWSliveMAILemailTAGLINES&_m=EXT___ -- 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
Re: [asterisk-users] Help need to do Lookup from odbc database
Thanks. --- On Thu, 14/5/09, Tilghman Lesher wrote: > From: Tilghman Lesher > Subject: Re: [asterisk-users] Help need to do Lookup from odbc database > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Thursday, 14 May, 2009, 4:20 AM > On Wednesday 13 May 2009 17:55:41 > carl Lougher wrote: > > Howdy, > > How do i perform a lookup from a remote odbc database > in the asterisk > > dialplan? > > > > I can do it with mysql but not sure of commands for > odbc connection. > > See func_odbc.conf for examples. You'll also need to > setup res_odbc.conf, as > this is where func_odbc obtains its handles. > > -- > Tilghman > > ___ > -- 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 > ___ -- 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] Help need to do Lookup from odbc database
Howdy, How do i perform a lookup from a remote odbc database in the asterisk dialplan? I can do it with mysql but not sure of commands for odbc connection. Cheers!!! ___ -- 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] Help with radius
Hi, I'm trying to get my Asterisk 1.4.24.1 server working with radius and aradial. I have radiusclient-ng installed and asterisk radius cdr. My cdr's fail to write to the database and i'm not sure how to authenticate each call. Anyone got this working or can offer any help. I've read all the radius docs and followed them to a tee.. Cheers!!! ___ -- 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] Parked calls for multiple customers
Ok cheers. Any idea when 1.6 goes stable for prod? - Original Message From: Mike To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, 24 April, 2009 0:54:59 Subject: Re: [asterisk-users] Parked calls for multiple customers No, but as I understand it 1.6 would have that possibility. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of carl Lougher > Sent: Thursday, April 23, 2009 4:54 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Parked calls for multiple customers > > > Hi, > > Is there any method of getting call park working on different numbers for > different customers on the same asterisk server? > Currently running asterisk 1.4.23.1 > > Cheers!! > > > > > ___ > -- 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 ___ -- 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 ___ -- 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] Parked calls for multiple customers
Hi, Is there any method of getting call park working on different numbers for different customers on the same asterisk server? Currently running asterisk 1.4.23.1 Cheers!! ___ -- 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] Canreinvite after media connection
Howdy, Is it possible to send a reinvite after the media has connected? Scenario: Inbound call hits asterisk ivr then is sent out to an extension using the dial command. We have to carry the rtp streams in this case as asterisk cant send the reinvite after the ivr has stopped playing the message as we already connected the call. Question: Any way around this or is there a better way we can do it? Cheers, Taff ___ -- 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] Stun clients and canreinvite
Howdy, Scenario: Asterisk server Customer connected over internet using nat Customer phones are Linksys 942 with Stun enabled Issue: Inbound and Outbound calls work fine. But when phones call each other internally we have to carry the voice stream ie using t on dial commands. Question: Is there a better way of doing this or another way to get the media to stream internally on the customer network rather than us carrying it? We have to keep Stun on the phones to get the media to flick off on outbound calls. Cheers, Taff.. ___ -- 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] Call-limit=1 breaks attended transfer
Yeah but doesnt help for extensions that have or require call-limit=1. --- On Tue, 31/3/09, carl Lougher wrote: > From: carl Lougher > Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Tuesday, 31 March, 2009, 2:20 AM > > We use call-limit set to 1 for hints. I guess i'll look > into the dtmf method and debug the linksys phone to see what > it uses for attended transfers. > > Cheers > > --- On Mon, 30/3/09, Mark Michelson > wrote: > > > From: Mark Michelson > > Subject: Re: [asterisk-users] Call-limit=1 breaks > attended transfer > > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > > Date: Monday, 30 March, 2009, 10:50 PM > > carl Lougher wrote: > > > Howdy, > > > Was there ever a fix for this? > > > > > > I have Trix 2.6 running asterisk 1.4 and have to > set > > an extension with call-limit=1. However that user can > no > > longer do attended transfers from Linkys 962 ip > phone. > > > > > > Is there anyway around this? > > > > > > Cheers, > > > Taff.. > > > > > > > Yes, set call-limit to something else :P > > > > Seriously though, there's no "fix" for that since it > is > > behaving exactly as it > > should. When attempting to transfer the call, Asterisk > has > > no way of knowing > > that the new SIP INVITE it receives (in order to call > the > > transfer target) is an > > attempt to transfer the call. It appears that the same > SIP > > peer is attempting to > > make a second call. Since the call-limit is set to 1, > > Asterisk rejects the > > second call attempt. > > > > I haven't tried this yet, but it may actually be > possible > > to use DTMF transfers > > when the call limit is that low since Asterisk is the > one > > that actually > > initiates the new call to the transfer target instead > of > > the transferer's phone. > > To use DTMF transfers, you need to set a DTMF sequence > in > > features.conf and use > > the 't' or 'T' flag (depending on which party should > have > > the ability to > > transfer the call) in your calls to Dial() or > Queue(). > > > > Why do you have the call-limit set to 1, anyway? > > > > Mark Michelson > > > > ___ > > -- 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 > > > > > > > ___ > -- 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 > ___ -- 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] Call-limit=1 breaks attended transfer
We use call-limit set to 1 for hints. I guess i'll look into the dtmf method and debug the linksys phone to see what it uses for attended transfers. Cheers --- On Mon, 30/3/09, Mark Michelson wrote: > From: Mark Michelson > Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Monday, 30 March, 2009, 10:50 PM > carl Lougher wrote: > > Howdy, > > Was there ever a fix for this? > > > > I have Trix 2.6 running asterisk 1.4 and have to set > an extension with call-limit=1. However that user can no > longer do attended transfers from Linkys 962 ip phone. > > > > Is there anyway around this? > > > > Cheers, > > Taff.. > > > > Yes, set call-limit to something else :P > > Seriously though, there's no "fix" for that since it is > behaving exactly as it > should. When attempting to transfer the call, Asterisk has > no way of knowing > that the new SIP INVITE it receives (in order to call the > transfer target) is an > attempt to transfer the call. It appears that the same SIP > peer is attempting to > make a second call. Since the call-limit is set to 1, > Asterisk rejects the > second call attempt. > > I haven't tried this yet, but it may actually be possible > to use DTMF transfers > when the call limit is that low since Asterisk is the one > that actually > initiates the new call to the transfer target instead of > the transferer's phone. > To use DTMF transfers, you need to set a DTMF sequence in > features.conf and use > the 't' or 'T' flag (depending on which party should have > the ability to > transfer the call) in your calls to Dial() or Queue(). > > Why do you have the call-limit set to 1, anyway? > > Mark Michelson > > ___ > -- 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 > ___ -- 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] Call-limit=1 breaks attended transfer
Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff.. ___ -- 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] Stun with hosted asterisk solution???
Howdy, I have the following issue and would like to know if anyone has got around this before. IP Phones - Linksys 942 Sip server - Asterisk 1.4.13 Stun server - Vovida Ok heres the issue. We have multiple client phones on their own network behind a natted connection. We have setup the phones to be natted and also pointing to our stun server. Now when the phones make an outside call to the PSTN stun kicks in and their rtp streams are carried from the phones to the sip provider without any issues. Now when the phones dial each other internally the rtp stream is still carried via stun and therefore fails as its pointing to the same ip on the same router. Now by adding t to the asterisk dial commands for each internal phone the inbound calls work fine but the rtp streams are carried through asterisk rather than between themselves on their network. Also in this scenario when you try conference an outside phone with an inside phone it fails due to stun and outside address problems. So my question is can we set up or change something on the phones or asterisk to allow the phones rtp to go across the local network on internal calls and via stun for outbound pstn calls? 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] Music on hold for sub tenants
This seems to be related to inbound calls. So would this work for music on transfers within that context as well as hitting the hold key on calls? --- On Fri, 26/9/08, Darrick Hartman <[EMAIL PROTECTED]> wrote: > From: Darrick Hartman <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Music on hold for sub tenants > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Friday, 26 September, 2008, 4:52 AM > ...since everyone else top posted. > > Take a look at the application setmusiconhold. > > CLI> core show application SetMusicOnHold > > You can use this in a dialplan as follows: > > [tenant1incoming] > exten => s,1,Wait(1) > exten => s,n,Answer() > exten => s,n,Background(tenant1sounds/welcome) > exten => s,n,SetMusicOnHold(tenant1) > > [tenant2incoming] > exten => s,1,Wait(1) > exten => s,n,Answer() > exten => s,n,Background(tentant2sounds/welcome) > exten => s,n,SetMusicOnHold(tenant2) > > Use that with the previously supplied info. > > Darrick > > carl Lougher wrote: > > Hi, > > I tried this but it still uses the default moh. Is > there some way to define it based on a context in the > sip.conf or extensions.conf??? > > > > Taff... > > > > > > --- On Fri, 26/9/08, Nhadie <[EMAIL PROTECTED]> > wrote: > > > >> From: Nhadie <[EMAIL PROTECTED]> > >> Subject: Re: [asterisk-users] Music on hold for > sub tenants > >> To: "Asterisk Users Mailing List - > Non-Commercial Discussion" > > >> Date: Friday, 26 September, 2008, 4:10 AM > >> Hi, > >> > >> i think you can define it like this: > >> > >> [moh-company-a] > >> mode=files > >> directory=/var/lib/asterisk/moh/companya > >> > >> [moh-company-b] > >> mode=files > >> directory=/var/lib/asterisk/moh/companyb > >> > >> regards, > >> nhadie > >> > >> > >> carl Lougher wrote: > >>> Howdy, > >>> Is there a way to apply a music on hold class > to > >> different context user groups? > >>> I have multiple clients on my asterisk server > and they > >> each want different music on hold. > >>> Company A > >>> Company B > > ___ > -- 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] Music on hold for sub tenants
Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie <[EMAIL PROTECTED]> wrote: > From: Nhadie <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Music on hold for sub tenants > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Date: Friday, 26 September, 2008, 4:10 AM > Hi, > > i think you can define it like this: > > [moh-company-a] > mode=files > directory=/var/lib/asterisk/moh/companya > > [moh-company-b] > mode=files > directory=/var/lib/asterisk/moh/companyb > > regards, > nhadie > > > carl Lougher wrote: > > Howdy, > > Is there a way to apply a music on hold class to > different context user groups? > > > > I have multiple clients on my asterisk server and they > each want different music on hold. > > > > Company A > > Company B > > > > Any help much appreciated.. > > > > Thanks, > > Taff... > > > > > > > > > > ___ > > -- 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
[asterisk-users] Monitoring simul calls
Howdy, Running asterisk 1.4 Is there a way to check the simultaneous sip calls in asterisk and display with mrtg or some graphing app??? Also is there a way to segregate these based on extension or context? Cheers, Taff.. ___ -- 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] Music on hold for sub tenants
Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B Any help much appreciated.. Thanks, Taff... ___ -- 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] Sip reload casuing issues
Howdy, Running asterisk 1.4.13 Sometime when running a sip reload the clients are unable to make and receive calls.. Any pointers? No errors in debug or asterisk console so far.. Cheers, Taff.. ___ -- 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] How to change Internal and external callerid
Howdy, Whats the best way to change the callerid for internal and external calls. At the moment using callerid- Fred <04412345> sends callerid as Fred 04412345 for internal calls when his internal extension is 200. How can i change the callerid for internal calls but also keep the specific external callerid for PSTN calls??? Much appreciated!!! Taff... __ Sent from Yahoo! Mail. More Ways to Keep in Touch. http://uk.docs.yahoo.com/nowyoucan.html ___ -- 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 call quality detection
Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ ___ --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] Clicking Noise on Pure Voip Calls
Setup: Asterisk server in NY. Cisco 7960 IP Phones in NY and London. Dedicated T1 from NY to Ldn. T1: Latency - 100ms Qos applied No errors Default codec on Ldn IP Phones = g711alaw Default codec on NY IP Phones = g711ulaw Both codecs allowed on each phone. Issue: Calls on IP Phones from NY to London hear clicking noise on NY end. Anyone experienced something similar or can offer some assistance? Thanks, Taf.. Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --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] Macro help needed!!!!
Upgrading to ver 1.2.10 fixed it.
--- carl Lougher <[EMAIL PROTECTED]> wrote:
> Hi,
> Need to get the following working:
>
> 1. User calls ext 750.
> 2. If no answer or busy go elsewhere.
> 3. If answered and press 1 accept call.
> 4. If answered and not pressed 1 or timed out then
> send call to be redirected to the busy or no answer
> option.
>
> The issue is that the call gets accepted if any
> number
> is pressed or a timeout. How do i throw the call
> back
> out of the macro???
>
> asterisk ver 1-0-9
>
> [sip-clients]
> exten => 750,1,Dial(SIP/225|60|gM(mobile))
> exten => 750,2,GotoIf($[${DIALSTATUS} = BUSY]?7:3)
> exten => 750,3,GotoIf($[${DIALSTATUS} = NOANSWER]?7)
> exten => 750,4,Hangup()
> exten => 750,7,Dial(SIP/226,15,t)
>
> [macro-mobile]
> exten => s,1,DigitTimeout(4)
> exten => s,2,ResponseTimeout(5)
> exten => s,3,Read(ACCEPT|press one now to accept|1)
> ;
> exten => s,4,GotoIf($[${ACCEPT} = 1]?5:6)
> exten => s,5,SetVar(MACRO_RESULT=CONTINUE)
> exten => s,6,Hangup()
>
>
>
>
>
>
>
>
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>
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Macro help needed!!!!
Hi,
Need to get the following working:
1. User calls ext 750.
2. If no answer or busy go elsewhere.
3. If answered and press 1 accept call.
4. If answered and not pressed 1 or timed out then
send call to be redirected to the busy or no answer
option.
The issue is that the call gets accepted if any number
is pressed or a timeout. How do i throw the call back
out of the macro???
asterisk ver 1-0-9
[sip-clients]
exten => 750,1,Dial(SIP/225|60|gM(mobile))
exten => 750,2,GotoIf($[${DIALSTATUS} = BUSY]?7:3)
exten => 750,3,GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten => 750,4,Hangup()
exten => 750,7,Dial(SIP/226,15,t)
[macro-mobile]
exten => s,1,DigitTimeout(4)
exten => s,2,ResponseTimeout(5)
exten => s,3,Read(ACCEPT|press one now to accept|1) ;
exten => s,4,GotoIf($[${ACCEPT} = 1]?5:6)
exten => s,5,SetVar(MACRO_RESULT=CONTINUE)
exten => s,6,Hangup()
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[asterisk-users] Finding far end echo in Verizon network
This is a weird one. Network: Asterisk ver 1-0-9 on DL360. 10 Cisco 7960g phones with 3.8.2 SIP Load. Gateway - Cisco 2811 router with 4 x verizon bri's. Network - Private vlan with 1ms response times to all devices. Issue: Intermittent echo on outbound/inbound calls. Users hearing their own voice about 0.5sec later. Tried so far: Upgraded firmware on some phones to 3.8.2 Upgraded software on Cisco router. Changed gain and attentuation settings on cisco router Got Verizon to test bris Moved rtp from asterisk direct to phone and router (canreinvite=yes) load tested asterisk None of the above made any difference. They are hearing their own voice so that means the issue is on the far end. But should it be up to me to control the possible delay or slippage in the verizon bri network? Any help much appreciated. Taf. ___ All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html ___ --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] External call press 1
Hi,
Running asterisk ver 1-0-9
Trying to send a call to a mobile phone and playback a
message to the user to press one to accept the call.
If 1 isn't pressed then the call needs to be re-routed
back into the asterisk dialplan.
Tried various macros etc but if one isn't pressed the
call still gets accepted?
Any clues???
exten => 333,1,Macro(test)
exten => 333,2,Hangup
exten => 334,1,Dial(SIP/XXX)
[macro-test]
exten => s,1,Wait(1)
exten => s,2,Read(ACCEPT|press-one |1)
exten => s,3,GotoIf($[${ACCEPT} = 1 ]?4:5)
exten => s,4,NoOp(Caller accepted)
exten => s,5,Goto(client,334,1)
exten => i,1,Set(MACRO_RESULT=CONTINUE)
exten => t,1,Set(MACRO_RESULT=CONTINUE)
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[Asterisk-Users] RE:Poor Voicemail / Ivr announcement quality
Howdy, The first 5 secs of each Voicemail or IVR announcement is stuttered and u can hardly hear the sound. After that its ok. Running TOP showed a high CPU usage on start up of the announcement as running command X?? Is this a PC CPU/RAM issue or something else related to Asterisk OS : Redhat v9 PC : AMD K2 512 Cheers, Carl. _ Store more e-mails with MSN Hotmail Extra Storage 4 plans to choose from! http://click.atdmt.com/AVE/go/onm00200362ave/direct/01/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Qu.
When I call Voicemail I get a very slow underwater sounding voice for the
first few seconds then it corrects itself. Any idea?
Output from Console:
-- Executing VoiceMailMain("SIP/2101-20db", "") in new stack
-- Playing 'vm-login' (language 'en')
Feb 26 14:45:58 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:45:58 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to
schedule in the past?!?!
_
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