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...ht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Steve Underwood Gesendet: Donnerstag, 4. Mai 2006 17:00 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] DTMF detection when outgoing call to mobilephones Marc Scheuffler wrote: >Hi all, > >I am trying to detect DTMF keys from a mobile when asterisk make an >outgoing call to the mobile. > >The DTMF detection on incoming calls (also FROM mobiles) is working >very well. >The only problem is if asterisk called the phone... N...

Qestion hvow can i use my SE w800i to reed flac files from a meorycard Best regards Fredrik

Hi there. The title basicly explains it. When we get incomming calls from cellular phones, the callers tend to echo ALOT. They hear their own voice at very high volums. This is a problem only for mobilphone users that calls in to us. I'm using wifi IP-phones. Asteirsk: CVS-Nv1-0-7-04/19/05 Any way to fix this on the asterisk server? Regards, Arne Morten --------------

...39;)),col=black, bty='n',cex=0.9) Luego al tratar de colocar las cusriva utilizando par(font=3) tampoco funciona. ¿Alguna idea de un script para hacer esto? Gracias Eric Flores De Gracia School of Biosciences University of Exeter, Cornwall Campus Penryn, Cornwall TR10 9EZ United Kingdom Mobilephone: +044 07578724705 http://biosciences.exeter.ac.uk/cec/staff/postgradresearch/index.php?web_id=Eric_Flores_de_Gracia&tab=profile

...k)) El factor bloque es random pero no se si esta es la nomenclatura correcta para testar mi hipotesis, o simplemente debo omitir el termino -strata- Gracias,saludos Eric Flores De Gracia School of Biosciences University of Exeter, Cornwall Campus Penryn, Cornwall TR10 9EZ United Kingdom Mobilephone: +044 07578724705 http://biosciences.exeter.ac.uk/cec/staff/postgradresearch/index.php?web_id=Eric_Flores_de_Gracia&tab=profile [[alternative HTML version deleted]]

...the Gluster server, we get the following error: gluster volume status tier1data clients FAILED: Commit failed on master1. Error: Unable to decode brick op response. Could anyone recommend a potential solution for this? Thanks, Anant Anant Saraswat DevOps Lead IT | Technology Blueprint Ltd. [mobilePhone] +44-8450047142 (5020)<tel:+44-8450047142%20(5020)> | +91-9818761614<tel:+91-9818761614> [emailAddress] anant.saraswat at techblue.co.uk<mailto:anant.saraswat at techblue.co.uk> [website] https://www.technologyblueprint.co.uk [address] Salisbury House, Station Road, Cambridge, Cam...

...tier1data clients > FAILED: Commit failed on master1. Error: Unable to decode brick op > response. > > Could anyone recommend a potential solution for this? > > Thanks, > Anant > > Anant Saraswat > > DevOps Lead > > IT | Technology Blueprint Ltd. > [image: mobilePhone] > +44-8450047142 (5020) <+44-8450047142%20(5020)> | +91-9818761614 > > [image: emailAddress] > anant.saraswat at techblue.co.uk > > [image: website] > https://www.technologyblueprint.co.uk > > [image: address] > Salisbury House, Station Road, Cambridge, Cambridg...

Hi all, What i'm going to do is that enable caller sing while playing a background music (likes karaoke). My approach is using Monitor and Meetme apps.Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music, then Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a

...ugh bluetooth. After some sutrggling, I foun chan_mobile. As some one already used this mobile with what result? I'm considering a simple asterisk system (for home use/test purpose) with : - one SIP service provider (with link to PSTN) - one SIP phone (a softphone on the asterisk box) - one mobilephone (ericsson W300i) let's call it A (for Asterisk) Mobile A is supposed to be connected to asterisk. I want to interact with this system with another mobile B. The idea would be to use free calls between mobile A and B and use the SIP provider mobile B => ..... => SIP provider (low cost c...

Hi, I've been learning asterisk for a couple of weeks now - and it worked for me as faar as standard configurations where concerned (sip/iax outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO). Now I recently I'evaluating to use asterisk as a replacemnt for our companies (15 employees) legacy pbx system and I'm experiencing multiple problems with the hfc isdn cards:

Have set up machine accounts as recommended in your docs, using root as authority to change to domain but half of the XP clients complain 'specified user does not exist'. This problem is not consistent and has occurred seemingly at random with a test XP pro workstation on continually attempting to leave & then rejoin the domain - any ideas out there? Please reply to