Re: [asterisk-users] TCP Trigger on incoming call request
Thank for the hint. I will have a look into it. Daniel -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von isr...@gmail.com Gesendet: Freitag, 6. Mai 2011 15:22 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] TCP Trigger on incoming call request Look at function CURL -Original Message- From: Daniel Isenmann Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 6 May 2011 13:04:09 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] TCP Trigger on incoming call request -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Background music during a call
Will this work: exten=> 123,1,Meetme(1234) exten=> 123,n,Hangup() exten=> 5000,1,Dial(Local/123@bk_music/n,,m()) exten=> 5000,2,Goto(bk_music,123,1) Parties can call 123 to enter a meeting room. and with the help of a callfile ic an dial a local channel to 5000 extension which in return calls a local channel to exten 123 to enter meet me. The dial command with second local channel will use m() option with moh call defind for each caller. will ring indefinately with moh and conf members will listen to it. Not tested it yet. Just sharing, will try it and let you know list. Cheers Mon, May 9, 2011 at 8:47 AM, Rizwan Hisham wrote: > Thanks for the reply. I looked into the G option of Dial applications. No > problem with that but How do I create a ghost call? > > My dial plan will look like this: > > Caller A calls Caller B normally: > > exten=> _XXX,1,SomePreDialApps() > exten=> _XXX,n,Dial(SIP/B) > exten=> _XXX,n,Hangup() > > Caller A calls caller B ith background music > exten=> _*9XXX,1,SomePreDialApps() > exten=> _*9XXX,n,Dial(SIP/B,,G(10)) > exten=> _*9XXX,n,Hangup() > > exten=> _*9XXX,10,Goto(mm,1,1) > exten=> _*9XXX,11,Goto(mm,1,1) > > Waiting for your replies > > Thanks > > On Fri, May 6, 2011 at 10:37 PM, Ioan Indreias wrote: > >> On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham >> wrote: >> >> > I am in desperate need of this feature. I want to play background music >> > during a call while the 2 parties are having some lovely conversation >> (or >> > maybe give them a sort of cursing background if they are cursing each >> > other). >> >> Let's start with your actual dialplan (without the background music) >> and we could start from that point. >> Hint: I am planning to use option G of the Dial application + a meetme >> room where a "ghost" call will play the specified MOH class >> (lovely/cursing). >> >> HTH, >> Ioan. >> >> -- >> _ >> -- 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 >> > > > > -- > Best Ragards > Rizwan Qureshi > VoIP/Asterisk Engineer > Axvoice Inc. > > V: +92 (0) 6767 26 > E: rizwanhas...@gmail.com > W: www.axvoice.com > > -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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
Re: [asterisk-users] Background music during a call
Thanks for the reply. I looked into the G option of Dial applications. No problem with that but How do I create a ghost call? My dial plan will look like this: Caller A calls Caller B normally: exten=> _XXX,1,SomePreDialApps() exten=> _XXX,n,Dial(SIP/B) exten=> _XXX,n,Hangup() Caller A calls caller B ith background music exten=> _*9XXX,1,SomePreDialApps() exten=> _*9XXX,n,Dial(SIP/B,,G(10)) exten=> _*9XXX,n,Hangup() exten=> _*9XXX,10,Goto(mm,1,1) exten=> _*9XXX,11,Goto(mm,1,1) Waiting for your replies Thanks On Fri, May 6, 2011 at 10:37 PM, Ioan Indreias wrote: > On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham > wrote: > > > I am in desperate need of this feature. I want to play background music > > during a call while the 2 parties are having some lovely conversation (or > > maybe give them a sort of cursing background if they are cursing each > > other). > > Let's start with your actual dialplan (without the background music) > and we could start from that point. > Hint: I am planning to use option G of the Dial application + a meetme > room where a "ghost" call will play the specified MOH class > (lovely/cursing). > > HTH, > Ioan. > > -- > _ > -- 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 > -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- 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
Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?
Run more of your systems as diskless. Make your tftp setup indispensable :) On Sun, 2011-05-08 at 22:37 +0100, Sebastian Arcus wrote: > Hi James, > > Thanks for the reply. I'm not concerned about performance. But I've > learned that every extra daemon software on a server comes with its > security caveats. I would feel much better about not having another one > to worry about and keep an eye on. > > Sebastian > > > On 05/08/2011 10:30 PM, James Miller wrote: > > I have my tftp daemon running all the time and it really doesnt affect > > the performance of the machine. Is there a reason why you want to shut > > it down? > > > > “I see blindness more as the ability and sight > > more as the disability, I only see that which > > is within a person.” > > Patrick Henry Hughes - 2009 > > > > > > On 5/8/2011 5:19 PM, Sebastian Arcus wrote: > >> Hi all, > >> > >> Sorry for posting here - but I figured there are many people with > >> Cisco IP phones here - and I use them with Asterisk :-) > >> > >> I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK, > >> loaded the SIP configuration files OK, they work with Asterisk just fine. > >> > >> My question is - will I have to keep on running the tftp server for > >> them for ever and ever? Isn't there any option for them to just use > >> the settings they have already loaded form the tftp server - so that I > >> can kill tftpd on my server machine? I tried doing that, and then the > >> phones stop booting, going in a loop looking for the tftpd server. > >> > >> It seems a bit pointless, having to run the tftpd daemon all the time > >> - although I've already loaded the firmware and configurations I want. > >> > >> Thank you, > >> > >> Sebastian > >> > >> -- > >> _ > >> -- 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 > > > > -- > > _ > > -- 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 > > > > -- > _ > -- 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 signature.asc Description: This is a digitally signed message part -- _ -- 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
Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?
Hi James, Thanks for the reply. I'm not concerned about performance. But I've learned that every extra daemon software on a server comes with its security caveats. I would feel much better about not having another one to worry about and keep an eye on. Sebastian On 05/08/2011 10:30 PM, James Miller wrote: I have my tftp daemon running all the time and it really doesnt affect the performance of the machine. Is there a reason why you want to shut it down? “I see blindness more as the ability and sight more as the disability, I only see that which is within a person.” Patrick Henry Hughes - 2009 On 5/8/2011 5:19 PM, Sebastian Arcus wrote: Hi all, Sorry for posting here - but I figured there are many people with Cisco IP phones here - and I use them with Asterisk :-) I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK, loaded the SIP configuration files OK, they work with Asterisk just fine. My question is - will I have to keep on running the tftp server for them for ever and ever? Isn't there any option for them to just use the settings they have already loaded form the tftp server - so that I can kill tftpd on my server machine? I tried doing that, and then the phones stop booting, going in a loop looking for the tftpd server. It seems a bit pointless, having to run the tftpd daemon all the time - although I've already loaded the firmware and configurations I want. Thank you, Sebastian -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?
I have my tftp daemon running all the time and it really doesnt affect the performance of the machine. Is there a reason why you want to shut it down? “I see blindness more as the ability and sight more as the disability, I only see that which is within a person.” Patrick Henry Hughes - 2009 On 5/8/2011 5:19 PM, Sebastian Arcus wrote: Hi all, Sorry for posting here - but I figured there are many people with Cisco IP phones here - and I use them with Asterisk :-) I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK, loaded the SIP configuration files OK, they work with Asterisk just fine. My question is - will I have to keep on running the tftp server for them for ever and ever? Isn't there any option for them to just use the settings they have already loaded form the tftp server - so that I can kill tftpd on my server machine? I tried doing that, and then the phones stop booting, going in a loop looking for the tftpd server. It seems a bit pointless, having to run the tftpd daemon all the time - although I've already loaded the firmware and configurations I want. Thank you, Sebastian -- _ -- 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 -- _ -- 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] Cisco 7940 phone and tftpd provisioning - for ever?
Hi all, Sorry for posting here - but I figured there are many people with Cisco IP phones here - and I use them with Asterisk :-) I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK, loaded the SIP configuration files OK, they work with Asterisk just fine. My question is - will I have to keep on running the tftp server for them for ever and ever? Isn't there any option for them to just use the settings they have already loaded form the tftp server - so that I can kill tftpd on my server machine? I tried doing that, and then the phones stop booting, going in a loop looking for the tftpd server. It seems a bit pointless, having to run the tftpd daemon all the time - although I've already loaded the firmware and configurations I want. Thank you, Sebastian -- _ -- 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
Re: [asterisk-users] [SOT] Virtualising Asterisk
On Sat, 2011-05-07 at 16:24 +0100, --[ UxBoD ]-- wrote: > I know a lot has changed over the past couple of years, and even > monthly, and that Asterisk running within a virtualised environment is > very happy indeed. If one would only be using SIP/IAX would Xen/KVM be > the best solution ? / or perhaps VServer/LXC maybe advantageous due to > binary hashing. Your thoughts would be very welcome. > -- At work we have a couple of asterisk instances. Some of them are used for doing interfaceing job, like towards isdn-BA or isdn-PRA, skype or GSM. (those do only _that_ and nothing more. Others (dialplan, voicemail, etc etc) are running as an XEN-image. Reason: much more easier to replace Some of the are still 1.4 while others run several different 1.6 versions. Currently i'm experimenting with a bunch of virtual 1.8-systems. from my point of view sles/xen works rock solid. hw -- _ -- 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
Re: [asterisk-users] [SOT] Virtualising Asterisk
On Sat, May 07, 2011 at 04:24:08PM +0100, --[ UxBoD ]-- wrote: > I know a lot has changed over the past couple of years, and even monthly, and > that Asterisk running within a virtualised environment is very happy indeed. > If one would only be using SIP/IAX would Xen/KVM be the best solution ? / or > perhaps VServer/LXC maybe advantageous due to binary hashing. Your thoughts > would be very welcome. VServer / OpenVZ / LXC run the the instances on top of the same kernel. They use kernel-level seperation. So this means that they basically see the subsets of the same file system and such. KVM / Xen provide the view of a different machine. The guest machine run its own kernel. I guess this is better if you don't trust the guest, but you pay more in resource consumption. If you want to run a different kernel, you must use this one. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] txgain no effect
Hi All, I have set following milliwatt and dial in from my mobile phone. and i adjust txgain value in chan_dahdi.conf but no effect on dahdi_monitor value is same. even i change its to 20 and more but monitor still showing same value. But i can hear tone variation on my mobile when i am changing txgain... is there any specific value i can set in txgain or leave it default 0.0 ( # = Audio Level * = Max Audio Hit ) <(RX)> <(TX)> #* 0) Tx: 4705 ( 4719) [milliwatt] ;Asterisk rx/tx gain testing exten => s,1,Answer exten => s,n,PlayTones(1004/1000) exten => s,n,Wait(300) -- _ -- 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
Re: [asterisk-users] no ringback tone on outgoing call PRI line
https://issues.asterisk.org/view.php?id=18868 -Original Message- From: satish patel Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 8 May 2011 11:43:41 To: asterisk-users Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] no ringback tone on outgoing call PRI line -- _ -- 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 -- _ -- 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] no ringback tone on outgoing call PRI line
Hi, I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example SIP->PRI > mobile I have set progress=yes in chan_dahdi.conf but still not working if i call inbound from my mobile to internal extension ringing working please help me -S -- _ -- 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
