Re: [asterisk-users] call forwarding number from outside.

That`s the normal behavior of assisted transfers. Try a blind/non-assisted transfer, that should show the original callerid. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio Sent: Friday, July 29, 2011 2:56 AM To: Aste

[asterisk-users] call forwarding number from outside.

Hi! I need help regarding the following problem: when I receive a phone call to the PBX from the number 01234567890 rings the number 100, get up the phone, I transfer (assisted) to the number 100. When the 100 number rings, the display shows the number of those who transferred the call and not t

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that you will receive in that , also read this for better implementation. http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause regards Dhaval On Fri, Jul 29, 2011 at 11:58 AM, Nikhil wrote: > ** > find the inl

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

find the inline comment... On 07/29/2011 12:11 AM, Ishwar Sridharan wrote: The dialplan is very simple. When the call comes in, we hand the call over to adhearsion. This is how the dialplan looks: ;group 0 will be used for incoming calls EXOIN = DAHDI/g0 ;group 11 for outgoing EXOOUT = DAHDI/

Re: [asterisk-users] Dialplan required for recording

Hi Vinod, You Need to look in MIxmonitor application on asterisk. http://www.voip-info.org/wiki/view/MixMonitor http://www.the-asterisk-book.com/unstable/applikationen-mixmonitor.html Where you can find easy dialplan On Fri, Jul 29, 2011 at 4:35 AM, Vinod Dharashive wrote: > Hi team, > > Can

[asterisk-users] Dialplan required for recording

Hi team, Can any one help me to implement dialplan in which conversation between a-party and b-party (call patch) needs to be recorded. Thanks Vinod Dharashive Sent from BlackBerry® on Airtel -- _ -- Bandwidth and Colocation Pro

Re: [asterisk-users] Voicemail not acting as documented.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I found what I believe to be a bug and have submitted it: https://issues.asterisk.org/jira/browse/ASTERISK-18207 Please correct me if I'm wrong. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFOMdzTCFu3bIiwtTARAimrAJ92+

Re: [asterisk-users] Securing Asterisk

On 7/28/2011 11:31 AM, Bruce B wrote: Hmmm, if alwaysauthreject is already breaking RFC rules then why not break another rule for the greater good? It would only add another layer of security. Maybe: *alwaysregreject=yes* * * *To drop SIP packets for both unauthorized registers and anonymous cal

[asterisk-users] Questions about FMFM with linked servers

All; In a linked server environment, running Asterisk 1.6 I am noticing that when a call is placed from server A to server B (via 4 digit extension) and server B ext has a FMFM to call their mobile, the mobile phone shows the default caller ID setting on server B instead of the actual caller id of

Re: [asterisk-users] Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions

> > > I'm looking to disable rejecting calls from my call center employees. > > They are using Polycom phones. Is there a way to either disable the > > reject/DND features on the Polycom phones (don`t think so) or have the > > Asterisk PBX ignore "Got SIP response 486 "Busy Here" back from > > 12

Re: [asterisk-users] Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions

> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Mike > Sent: Thursday, July 28, 2011 3:47 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Disabling Polycom

Re: [asterisk-users] Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions

On 07/28/2011 03:47 PM, Mike wrote: I’m looking to disable rejecting calls from my call center employees. They are using Polycom phones. Is there a way to either disable the reject/DND features on the Polycom phones (don`t think so) or have the Asterisk PBX ignore “Got SIP response 486 "Busy Her

Re: [asterisk-users] MoH - conversion command

In order to get the proper encoding for Asterisk, you must provide the correct values for each of these characteristics. In your case, they are as follows: rate = 8000 data size = 8-bit (byte) data encoding = gsm channels = 1 (mono) Therefore, the command you would use to creat

Re: [asterisk-users] Voicemail not acting as documented.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 07/28/2011 02:42 PM, Barry L. Kline wrote: > Is there some other parameter required to get this to fire or am I > reading more into that sentence from the CHANGES document than is > actually there? Sorry for replying to my own post, but I've don

[asterisk-users] Disabling Polycom "reject" and "DND" or disable Asterisk 486 "Busy Here" actions

Hi, I'm looking to disable rejecting calls from my call center employees. They are using Polycom phones. Is there a way to either disable the reject/DND features on the Polycom phones (don`t think so) or have the Asterisk PBX ignore "Got SIP response 486 "Busy Here" back from 12.23.34.45" res

Re: [asterisk-users] MoH - conversion command

From: "Mike" Sent: Thursday, July 28, 2011 1:29 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] MoH - conversion command > > > I should have said I am trying this both from a landline using ulaw,

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

1) You have to have channels configured for your PRI SOMEWHERE in the Asterisk DAHDI configs. 2) Can't troubleshoot when everything important is masked by an AGI script. Reproduce the problem using standard dialplan stuff. > -Original Message- > From: asterisk-users-boun...@lists.digium

Re: [asterisk-users] MoH - conversion command

From: "Mike" Sent: Thursday, July 28, 2011 1:29 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] MoH - conversion command > > > I should have said I am trying this both from a landline using ulaw, >

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

Hi Eric, There weren't any lines with "PRI channel =>" in the chan_dahdi.conf However, I added the lines you'd mentioned, near the top of the file. Still, no difference in either the behaviour or the asterisk output. Please note that as soon as the call lands on asterisk, we pass the control ove

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

Hi AJS, Our dialplan doesn't have a Dial() statement as that's taken care of by adhearsion. However, I added "exten => y, n, NoOp(Hangup cause was ${HANGUPCAUSE})" at the end of our context, restarted asterisk. The log doesn't have anything new. -- Thanks, Ishwar. On Thu, Jul 28, 2011 at 6

[asterisk-users] Voicemail not acting as documented.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 In http://svnview.digium.com/svn/asterisk/branches/1.8/CHANGES?view=markup, line 180 states: > Voicemail now runs the externnotify script when pollmailboxes is > activated and notices a change. My voicemail.conf configuration for my LDAP vm storage i

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

The dialplan is very simple. When the call comes in, we hand the call over to adhearsion. This is how the dialplan looks: ;group 0 will be used for incoming calls EXOIN = DAHDI/g0 ;group 11 for outgoing EXOOUT = DAHDI/G11 ;This will be used by adhearsion EXOCID= [general] autofallthroug

Re: [asterisk-users] MoH - conversion command

you have to select musical compositions that are 'less incompatible' > with G.729's compression methods. When we chose the current MOH selections > included with Asterisk, our initial list was much larger, but we had to > remove some because they sound terrible when compressed with G.729. > Thank

Re: [asterisk-users] Why no traction for Windows version?

On Thu, 28 Jul 2011 13:08:33 -0500, "Danny Nicholas" wrote: >If they have, it would probably be on www.nerdvittles.com It looks like The Incredible PBX runs on CentOS www.nerdvittles.com/index.php?p=740 -- _ -- Bandwidth and

Re: [asterisk-users] Why no traction for Windows version?

- Original Message - > On Thu, 28 Jul 2011 12:46:03 -0500, "Danny Nicholas" > wrote: > >Interrupting - you have to not use DAHDI (SIP Only) and make sure you > >have > >the necessary libs downloaded in your Cygwin install. > > It's OK, I don't mind using a VoIP gateway instead of a PCI ca

Re: [asterisk-users] Why no traction for Windows version?

If they have, it would probably be on www.nerdvittles.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, July 28, 2011 1:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk

Re: [asterisk-users] Why no traction for Windows version?

On Thu, 28 Jul 2011 12:46:03 -0500, "Danny Nicholas" wrote: >Interrupting - you have to not use DAHDI (SIP Only) and make sure you have >the necessary libs downloaded in your Cygwin install. It's OK, I don't mind using a VoIP gateway instead of a PCI card. Has someone written an HOWTO to compile

Re: [asterisk-users] Why no traction for Windows version?

Interrupting - you have to not use DAHDI (SIP Only) and make sure you have the necessary libs downloaded in your Cygwin install. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, July 28, 20

Re: [asterisk-users] Why no traction for Windows version?

On Thu, 28 Jul 2011 12:04:38 +0500, "Faisal Hanif" wrote: >I have tried asterisk on windows XP using Cygwin and it worked fine. Would you mind explaining how to do this? Thank you. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] MoH - conversion command

On 07/28/2011 01:29 PM, Mike wrote: I should have said I am trying this both from a landline using ulaw, and from a Polycom phone using g729 codec. G729 is noticeablty worst, as you`d expect, maybe this is what is reported by my customers. Is there any way to have a "decent" g729 file, or sho

Re: [asterisk-users] MoH - conversion command

> > > I should have said I am trying this both from a landline using ulaw, > > and from a Polycom phone using g729 codec. G729 is noticeablty worst, > > as you`d expect, maybe this is what is reported by my customers. > > > > Is there any way to have a "decent" g729 file, or should I just give >

Re: [asterisk-users] MoH - conversion command

On 07/28/2011 11:57 AM, Mike wrote: I should have said I am trying this both from a landline using ulaw, and from a Polycom phone using g729 codec. G729 is noticeablty worst, as you`d expect, maybe this is what is reported by my customers. Is there any way to have a "decent" g729 file, or shou

Re: [asterisk-users] hide google voice number

Yes, they used to allow it. Like CallWithUs and Voip.ms (and I'm sure other VTSP's) do. From: A.H. Jos Sent: Thu 7/28/2011 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] hide google voice number Do you mean that was possible to set the CID

Re: [asterisk-users] hide google voice number

Do you mean that was possible to set the CID in the early days of GVoice? On Thu, Jul 28, 2011 at 4:33 PM, Terry Brummell wrote: > Google Voice will show your number no matter what, there was a problem > with abuse when they let you send the CID in the early days. Pretty sure > there is nothin

Re: [asterisk-users] MoH - conversion command

> On 07/28/2011 10:53 AM, Mike wrote: > > Hi, > > > > I've been trying to get MoH files to sound decent. I've got a hold of > > Royalty-free Classical music (a safe choice for most of my customers) > > and I`ve been trying to convert them to the normal telephony/Asterisk > > format using sox. Unfor

Re: [asterisk-users] MoH - conversion command

Personally I like to just hook up an old ghetto blaster / boombox to the line in port on my sound card :) Kidding aside - I think audio quality for MoH is not always going to sound as nice as you might want. I mostly stream online radio over my MoH and the quality is not the greatest. Maybe it's

Re: [asterisk-users] Securing Asterisk

Hmmm, if alwaysauthreject is already breaking RFC rules then why not break another rule for the greater good? It would only add another layer of security. Maybe: *alwaysregreject=yes* * * *To drop SIP packets for both unauthorized registers and anonymous calls. Keep it off by default and then allo

Re: [asterisk-users] MoH - conversion command

On Thu, 28 Jul 2011, Mike wrote: I’ve got a hold of Royalty-free Classical music (a safe choice for most of my customers) and I`ve been trying to convert them to the normal telephony/Asterisk format using sox.  Unfortunately, it sounds really bad. I convert files using: sox "${INPUT

Re: [asterisk-users] MoH - conversion command

On 07/28/2011 10:53 AM, Mike wrote: Hi, I’ve been trying to get MoH files to sound decent. I’ve got a hold of Royalty-free Classical music (a safe choice for most of my customers) and I`ve been trying to convert them to the normal telephony/Asterisk format using sox. Unfortunately, it sounds rea

[asterisk-users] MoH - conversion command

Hi, I've been trying to get MoH files to sound decent. I've got a hold of Royalty-free Classical music (a safe choice for most of my customers) and I`ve been trying to convert them to the normal telephony/Asterisk format using sox. Unfortunately, it sounds really bad. I don't expect concert h

Re: [asterisk-users] hide google voice number

Google Voice will show your number no matter what, there was a problem with abuse when they let you send the CID in the early days. Pretty sure there is nothing you can do about it. From: A.H. Jos Sent: Thu 7/28/2011 9:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] hide g

Re: [asterisk-users] Problem H323 asterisk 1.6.2.19

2011/7/27 Vladimir Mikhelson : > Do you have any network devices or VPN tunnels in between the Asterisk > and Avaya? > Hi , the server does not have connections vpn I have and it in the same LAN that avaya > The reason I am asking it looks like a potential networking issue. ok, but I do ping t

Re: [asterisk-users] Strange network issue

On Thu, Jul 28, 2011 at 4:46 AM, Duncan Turnbull wrote: > On 28/07/2011, at 8:41 PM, Paul Hayes wrote: > > > On 28/07/11 02:58, Mike Diehl wrote: > >> > >> Any ideas? > >> > >> Mike. > > > > I'd go on site if possible and see what actually happens at 19:00. Set > up a wireshark trace capturing a

Re: [asterisk-users] Connect asterisk to normal telephone PBX

Thanks for the reply. I am using an analog phone in normal PBX. I have an extension called 199 in asterisk and an extension 264 in analog PBX. So how do i create an inbound or outbound routes for call between these two extentions ? On Thu, Jul 28, 2011 at 1:39 PM, Carlos M Cruz wrote: > Hi, > >

Re: [asterisk-users] Why no traction for Windows version?

On 11-07-26 03:21 AM, Gilles wrote: Are there just not enough interest and too many, deep, Linux-specific assumptions in the code, that would explain why Asterisk was never officially ported to Windows? Cost? For me to run Asterisk under Windows requires me to purchase a Microsoft license for

Re: [asterisk-users] hide google voice number

I am new to asterisk but I think that with gtalk things are different!!! In my extensions.conf I have: exten => 2103,2,Dial(SIP/sip.generic,20); twinkle client exten => _33X,1,Set(CALLERID(num)=01215063743); that doesn't work exten => _33X,2,Dial(gtalk/asterisk/+${EXTEN}@voice.googl

Re: [asterisk-users] hide google voice number

On 07/28/2011 09:22 AM, A.H. Jos wrote: Hi list, I have Asterisk speaking with google talk, is there any way to set or at least hide my google voice number when I call others? Set a different 'callerid' on either your outgoing sip.conf peer? -- Alex Balashov - Principal Evariste Systems LLC 26

[asterisk-users] hide google voice number

Hi list, I have Asterisk speaking with google talk, is there any way to set or at least hide my google voice number when I call others? thanks for your help, AHJos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Nikhil > Sent: Thursday, July 28, 2011 9:03 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Capturing call Reject/Decline events o

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

On Thursday 28 Jul 2011, Ishwar Sridharan wrote: > Is there a reason why asterisk doesn't recognise the "call decline", and > does it need any configuration changes to enable this? What are you seeing for ${HANGUPCAUSE} when this happens ? (Put a line such as exten => y, n, NoOp(Hangup cau

Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

Can you share the dialplan ,where SIP call is dialing... Thanks Nikhil On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party

[asterisk-users] Capturing call Reject/Decline events on a PRI line

Hello everybody, We have an asterisk 1.8.4.1 setup, connected to a PRI line. We're currently facing an issue where asterisk does not recognise the event when the called party declines/cuts the call. This happens specifically over calls on a PRI line. For calls over SIP, call decline event is capt

Re: [asterisk-users] Connect asterisk to normal telephone PBX

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, July 28, 2011 1:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Connect asterisk to normal telephone PBX Hello All,

Re: [asterisk-users] Lightning and thunder (Claude Hayn

On Thu, 2011-07-28 at 09:57 +0100, Paul Hayes wrote: > On 27/07/11 19:41, Claude Hayn wrote: > > > > The office manager freaks out each time and starts randomly rebooting > > devices in no particular order including the UPS, PBX, Asterisk Gateway, > > firewall and router. > > > > Ahh that old ches

[asterisk-users] [chan_mobile addons] DTMF transfer from calling mobile to Asterisk through called mobile FAILED

Dear All I've setup in lab a model include a *handse*t (nokia 6021 in supported list) connect to an *bluetooth dongle* (Cambridge Silicon Radio, Ltd Bluetooth Dongle (HCI mode)) attached to an *PC *(install Asterisk 1.6.2.19 and Bluez 3.7 with Asteri

Re: [asterisk-users] Lightning and thunder (Claude Hayn

On 27/07/11 19:41, Claude Hayn wrote: The office manager freaks out each time and starts randomly rebooting devices in no particular order including the UPS, PBX, Asterisk Gateway, firewall and router. Ahh that old chestnut. That's never a good thing, try to tell them not to do this, althou

Re: [asterisk-users] Strange network issue

On 28/07/2011, at 8:41 PM, Paul Hayes wrote: > On 28/07/11 02:58, Mike Diehl wrote: >> >> Any ideas? >> >> Mike. > > I'd go on site if possible and see what actually happens at 19:00. Set up a > wireshark trace capturing all traffic through their router. > > -- I am picking a cleaner pluggi

Re: [asterisk-users] Strange network issue

On 28/07/11 02:58, Mike Diehl wrote: Any ideas? Mike. I'd go on site if possible and see what actually happens at 19:00. Set up a wireshark trace capturing all traffic through their router. -- _ -- Bandwidth and Colocatio

Re: [asterisk-users] Connect asterisk to normal telephone PBX

Hi, Did you created your normal Inbound and Outbound routes in freepbx? For use with your zap channels? You'll problably have to change your routes on your pbx too... Regards, Carlos M Cruz 2011/7/28 michael k > Hello All, > > I don't even know the relevancy of my question. Please answer me

[asterisk-users] Radius billing for asterisk

Hi Any company proving radius based billing for asterisk only for accounting ,not authenication and atherization.Please provide some links Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.c

Re: [asterisk-users] Why no traction for Windows version?

I have tried asterisk on windows XP using Cygwin and it worked fine. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antonio Modesto Sent: Thursday, July 28, 2011 1:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Sub