Re: [Asterisk-Users] SIP VoIP to PSTN provider who allows multiple outbound calls?
Have you looked at www.packet8.cnet Here is some talk about it. http://www.dslreports.com/forum/remark,5845308~root=voip~mode=flat They claim you have to use at least one of their DTA-310 It is made by their parent company www.8x8.com, it uses SIP. - Original Message - From: "William X Walsh" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, February 26, 2003 5:50 PM Subject: [Asterisk-Users] SIP VoIP to PSTN provider who allows multiple outbound calls? > > IConnecthere is great, but I'm looking for something that allows > multiple outbound calls (not a plethora, but a few would be nice). > > It makes no sense to me to restrict number of simultaneous calls, > because you are paying for the minutes you use, but I guess that is > company policy there. > > Anyone aware of another solution? > > -- > William Walsh <[EMAIL PROTECTED]> > Jabber: [EMAIL PROTECTED] > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message waiting light on Cisco 7960
Can I get the voicemail application turn on / off the MWI (message waiting indicator) on the Cisco 7960? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice-mail App
Indeed rtp.c: static int dtmftimeout = 400; /* was 300 samples */ Did seem to fix the problemThat was almost too easy. All hell will probable brake out tomorrow when real users get on the system. Mark Spencer wrote: I have played with the timeouts :int timeout, int ftimeout for ast_readstring(chan, password, sizeof(password) - 1, 2000, 1, "#") To no effect, Could you give me a pointer to where I can start looking next to track down this strangeness. Maybe something in the SIP driver? Getting in and out of band DTMF? etc... Best place to look is probably the RTP code, where the digits are generated. Specifically look at the rfc2833 routines, assuming that's how they're being sent. I just got a 7960 on loan, so I'm going to set it up so that I can try to duplicate any problems you're having. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF with IConnectHere fix
Is anyone working on a fix to send / receive DTMF with IConnectHere? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directory() application, answers but doesn't dial...
I have the following in my config files: extensions.conf: exten => ,1,Directory(employees) [employees] exten => ,1,Playback(transfer,skip) exten => ,2,Macro(stdexten,,SIP/lenny) ;; exten => 5556,1,Playback(transfer,skip) exten => 5556,2,Macro(stdexten,5556,SIP/brian) voicemail.conf: => ,Lenny Tropiano,[EMAIL PROTECTED] 5556 => 5556,Brian Sinclair,[EMAIL PROTECTED] If I dial , the Directory application answer, I type TRO or SIN and wait... nothing, after a few seconds later it goes onto the "Bye, thank you for calling asterisk" Any ideas? Lenny irc: InetNomad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
My small Norstar / Asterisk config: NORSTAR MICS 5.1 0x32 KSU NT7B53FA 4 port LS Caller ID Trunk Cartridge NT5B41GA 3 T7316 Norstar phones 1 S100U and 1 ATA186s connected to the norstar trunk cartridge. Each telephone has its own Asterisk line appearance in addition to the intercom line appearance. This works well for me since asterisk can handle the incoming calls and then extend them to one or more of the phones with caller id, personal greeting voicemail, and message waiting indicators working perfectly. I would like to use the norstar's PRI card to integrate with asterisk, but I am not sure how to do this. On Thursday, February 27, 2003, at 06:20 PM, Brian Johnson wrote: What model? I'm looking at integrating a Nortel Norstar Compact ICS with asterisk and digium dev lite kit hardware to achieve a hydrid analog/VoIP solution This particular model has a ISDN interface as well as a 4 port analog PSTN connection. The handsets are a mixture of Nortel/Meridian models (mostly M7208) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Davis Sent: Thursday, February 27, 2003 19:06 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intercom and Paging I would also like to know what could I do to use asterisk in front of our current Norstar PBX. Could I pass calls to/from the norstar and the asterisk box(with the 400p)? David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Clark Sent: Thursday, February 27, 2003 2:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intercom and Paging These are really nice phones for Asterisk. I am interested in where they can be found refurbished for $65. Currently I use Asterisk behind a Norstar MICS system so I can still have the nice line appearances, message waiting, busy lamps, paging and voice call (intercom) calling, but I would like to add more telephones and these nortel phones look like a cheaper alternative to adding more norstar phones and trunk cards. Does anyone know if Asterisk can be connected directly to the T1 service card on Norstar MICS? On Thursday, February 27, 2003, at 12:25 PM, Jeff Noxon wrote: The ones I ordered are M9417CW's. Several other nortel models support the paging feature too. I paid $65 ea for refurbs. On Thu, Feb 27, 2003 at 05:17:04PM +, Brian Johnson wrote: What models? Jeff Noxon ([EMAIL PROTECTED]) wrote*: I just purchased a bunch of Nortel Meridian POTS phones that support intercom on the 3rd pair. I intend to get it working with Asterisk. The phones support MWI, have a 3-line display, callerID, call waiting callerID, 2 lines...very nice. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Collect Digits for CO Blind Transfer
You can also do that using Background application:
[transfer]
exten => s,1,Background,some-file ;it can be silence
exten => _XXX,1,Flash ;collecting the digits
exten => _XXX,2,SendDTMF,${EXTEN}
exten => _XXX,3,Hangup
[called_context]
exten => 1000,1,Goto,transfer|s|1 ;magical number
regards
Martin
On Thu, 27 Feb 2003, Steven Critchfield wrote:
> Not to take away from your prize collection, but for now wouldn't that
> be trivial to do either as a patternmatch with assignment to a variable,
> or slightly less trivialy as a agi app that already has access to the
> getdata function?
>
> On Thu, 2003-02-27 at 17:42, Mark Spencer wrote:
> > Sounds like we need an app_getdata (front end to the "getdata" function).
> >
> > I'll do it for any item in my thinkgeek wishlist (you can search by
> > [EMAIL PROTECTED] or by my name).
> >
> > Mark
> >
> > On Thu, 27 Feb 2003, Ben Clark wrote:
> >
> > > I have a blind transfer feature available to me from my telephone
> > > provider and was wondering if asterisk can take advantage of this so
> > > that when a certain extension is called the user is asked for the 11
> > > digit pstn number they wish to call then asterisk flashes the line,
> > > dials the transfer codes and hangs up. I have figured out how to do
> > > everything except collecting the digits from the user and then using
> > > them in the transfer codes. Is it possible to do this with the
> > > asterisk dial plan?
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Collect Digits for CO Blind Transfer
Yup, but it's really not that tough in C even :) Maybe if i get bored i'll do it anyway heh. It sounds like a nifty app! Mark On Thu, 27 Feb 2003, Steven Critchfield wrote: > Not to take away from your prize collection, but for now wouldn't that > be trivial to do either as a patternmatch with assignment to a variable, > or slightly less trivialy as a agi app that already has access to the > getdata function? > > On Thu, 2003-02-27 at 17:42, Mark Spencer wrote: > > Sounds like we need an app_getdata (front end to the "getdata" function). > > > > I'll do it for any item in my thinkgeek wishlist (you can search by > > [EMAIL PROTECTED] or by my name). > > > > Mark > > > > On Thu, 27 Feb 2003, Ben Clark wrote: > > > > > I have a blind transfer feature available to me from my telephone > > > provider and was wondering if asterisk can take advantage of this so > > > that when a certain extension is called the user is asked for the 11 > > > digit pstn number they wish to call then asterisk flashes the line, > > > dials the transfer codes and hangs up. I have figured out how to do > > > everything except collecting the digits from the user and then using > > > them in the transfer codes. Is it possible to do this with the > > > asterisk dial plan? > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Collect Digits for CO Blind Transfer
Not to take away from your prize collection, but for now wouldn't that be trivial to do either as a patternmatch with assignment to a variable, or slightly less trivialy as a agi app that already has access to the getdata function? On Thu, 2003-02-27 at 17:42, Mark Spencer wrote: > Sounds like we need an app_getdata (front end to the "getdata" function). > > I'll do it for any item in my thinkgeek wishlist (you can search by > [EMAIL PROTECTED] or by my name). > > Mark > > On Thu, 27 Feb 2003, Ben Clark wrote: > > > I have a blind transfer feature available to me from my telephone > > provider and was wondering if asterisk can take advantage of this so > > that when a certain extension is called the user is asked for the 11 > > digit pstn number they wish to call then asterisk flashes the line, > > dials the transfer codes and hangs up. I have figured out how to do > > everything except collecting the digits from the user and then using > > them in the transfer codes. Is it possible to do this with the > > asterisk dial plan? > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intercom and Paging
What model? I'm looking at integrating a Nortel Norstar Compact ICS with asterisk and digium dev lite kit hardware to achieve a hydrid analog/VoIP solution This particular model has a ISDN interface as well as a 4 port analog PSTN connection. The handsets are a mixture of Nortel/Meridian models (mostly M7208) > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of David Davis > Sent: Thursday, February 27, 2003 19:06 > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Intercom and Paging > > > I would also like to know what could I do to use asterisk in front of > our current Norstar PBX. > Could I pass calls to/from the norstar and the asterisk box(with the > 400p)? > > David > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Ben Clark > Sent: Thursday, February 27, 2003 2:26 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Intercom and Paging > > > These are really nice phones for Asterisk. I am interested in where > they can be found refurbished for $65. Currently I use Asterisk behind > a Norstar MICS system so I can still have the nice line appearances, > message waiting, busy lamps, paging and voice call (intercom) calling, > but I would like to add more telephones and these nortel phones look > like a cheaper alternative to adding more norstar phones and trunk > cards. > > Does anyone know if Asterisk can be connected directly to the T1 > service card on Norstar MICS? > > On Thursday, February 27, 2003, at 12:25 PM, Jeff Noxon wrote: > > > The ones I ordered are M9417CW's. Several other nortel models support > > > the paging feature too. I paid $65 ea for refurbs. > > > > On Thu, Feb 27, 2003 at 05:17:04PM +, Brian Johnson wrote: > >> What models? > >> > >> Jeff Noxon ([EMAIL PROTECTED]) wrote*: > >>> > >>> I just purchased a bunch of Nortel Meridian POTS phones that support > > >>> intercom on the 3rd pair. I intend to get it working with Asterisk. > > >>> The phones support MWI, have a 3-line display, callerID, call > >>> waiting callerID, 2 lines...very nice. > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intercom and Paging
I would also like to know what could I do to use asterisk in front of our current Norstar PBX. Could I pass calls to/from the norstar and the asterisk box(with the 400p)? David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Clark Sent: Thursday, February 27, 2003 2:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intercom and Paging These are really nice phones for Asterisk. I am interested in where they can be found refurbished for $65. Currently I use Asterisk behind a Norstar MICS system so I can still have the nice line appearances, message waiting, busy lamps, paging and voice call (intercom) calling, but I would like to add more telephones and these nortel phones look like a cheaper alternative to adding more norstar phones and trunk cards. Does anyone know if Asterisk can be connected directly to the T1 service card on Norstar MICS? On Thursday, February 27, 2003, at 12:25 PM, Jeff Noxon wrote: > The ones I ordered are M9417CW's. Several other nortel models support > the paging feature too. I paid $65 ea for refurbs. > > On Thu, Feb 27, 2003 at 05:17:04PM +, Brian Johnson wrote: >> What models? >> >> Jeff Noxon ([EMAIL PROTECTED]) wrote*: >>> >>> I just purchased a bunch of Nortel Meridian POTS phones that support >>> intercom on the 3rd pair. I intend to get it working with Asterisk. >>> The phones support MWI, have a 3-line display, callerID, call >>> waiting callerID, 2 lines...very nice. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Collect Digits for CO Blind Transfer
Sounds like we need an app_getdata (front end to the "getdata" function). I'll do it for any item in my thinkgeek wishlist (you can search by [EMAIL PROTECTED] or by my name). Mark On Thu, 27 Feb 2003, Ben Clark wrote: > I have a blind transfer feature available to me from my telephone > provider and was wondering if asterisk can take advantage of this so > that when a certain extension is called the user is asked for the 11 > digit pstn number they wish to call then asterisk flashes the line, > dials the transfer codes and hangs up. I have figured out how to do > everything except collecting the digits from the user and then using > them in the transfer codes. Is it possible to do this with the > asterisk dial plan? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interest in E1 channel banks?
>It seems a bit strange to me to manufacture devices in 2003 allowing >only connection of analog phones --- unless I did not understand your > channel bank spec. The channel bank can be used to retain existing (Analog) PBX infrastructure in areas where Telcos are dropping analogue trunk support. We already manufacture ISDN BRI & PRI protocol converters and could possibly make a generic BRI module for the channel bank if demand justified it. However we haven't had any real demand for such a module to date. Brendan. -Original Message- From: Emmanuel Michon [mailto:[EMAIL PROTECTED] Sent: Thursday, February 27, 2003 11:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Interest in E1 channel banks? "Dooley, Brendan" <[EMAIL PROTECTED]> writes: > Our company manufactures an E1 channel bank that is approved for use in > Australia (it should also be compatible with Euro standards). It is modular > and available in 10, 20 or 30 analog port configurations. Signal monitoring > and configuration is via Ethernet. It seems a bit strange to me to manufacture devices in 2003 allowing only connection of analog phones --- unless I did not understand your channel bank spec. Connecting plain digital ISDN phones (not channel bank specific) is a requirement for me. Sincerely yours, > > These units are manufactured in low quantities for specific telco > requirements. However if there was enough interest, we would be able to > manufacture and sell the units at pricing levels under US $2000. > > So how much interest is out there? > > Brendan > - > Brendan Dooley > General Manager > > Open Access Pty Ltd > PO Box 301 > Crows Nest NSW 1585 > > Phone +612 9978 7009 > Fax +612 9978 7099 > Email [EMAIL PROTECTED] > - > This email is intended only for the use of the individual or entity named > above and may contain information that is confidential and privileged. If > you are not the intended recipient, you are hereby notified that any > dissemination, distribution or copying of this email is strictly prohibited. > If you have received this email in error, please notify us immediately by > return email or telephone +612 9978 7009 and destroy the original message. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Emmanuel Michon Chef de projet REALmagic France SAS Mobile: 0614372733 GPGkeyID: D2997E42 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Unfortunately, no. It would be nice if SNOM added NBS support while they're adding IAX :) On Thu, Feb 27, 2003 at 12:41:31PM -1000, James H. Thompson wrote: > Do any of the SIP phones support intercom/paging? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Collect Digits for CO Blind Transfer
I have a blind transfer feature available to me from my telephone provider and was wondering if asterisk can take advantage of this so that when a certain extension is called the user is asked for the 11 digit pstn number they wish to call then asterisk flashes the line, dials the transfer codes and hangs up. I have figured out how to do everything except collecting the digits from the user and then using them in the transfer codes. Is it possible to do this with the asterisk dial plan? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Do any of the SIP phones support intercom/paging? Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, February 27, 2003 11:59 AM Subject: Re: [Asterisk-Users] Intercom and Paging > Then use 2 lines per phone. > > You should connect 2 phones together ( the way they want specify ) and hook an > oscilloscope to the > 3rd pair. It probably uses a simple signal to indicate page-intercom, maybe even > DTMF. > > You then need to rig a voltage convertor from telco batt voltage to whatever they > use ( probably 12v > or less ) > > > Jeff Noxon wrote: > > Yes but I want * to be able to page me, announce queued calls, etc. > > > > Basically glorified talking caller ID. ;) > > > > On Thu, Feb 27, 2003 at 04:03:09PM -0500, [EMAIL PROTECTED] wrote: > > > >>Jeff - If you make special phone cords that break out the 3rd pair at the > >>wiring closet you can just join those all together in parallel and your > >>paging will be * independant. > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
These are really nice phones for Asterisk. I am interested in where they can be found refurbished for $65. Currently I use Asterisk behind a Norstar MICS system so I can still have the nice line appearances, message waiting, busy lamps, paging and voice call (intercom) calling, but I would like to add more telephones and these nortel phones look like a cheaper alternative to adding more norstar phones and trunk cards. Does anyone know if Asterisk can be connected directly to the T1 service card on Norstar MICS? On Thursday, February 27, 2003, at 12:25 PM, Jeff Noxon wrote: The ones I ordered are M9417CW's. Several other nortel models support the paging feature too. I paid $65 ea for refurbs. On Thu, Feb 27, 2003 at 05:17:04PM +, Brian Johnson wrote: What models? Jeff Noxon ([EMAIL PROTECTED]) wrote*: I just purchased a bunch of Nortel Meridian POTS phones that support intercom on the 3rd pair. I intend to get it working with Asterisk. The phones support MWI, have a 3-line display, callerID, call waiting callerID, 2 lines...very nice. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3-Way call problems
For the first time last night, I tried placing a 3-way call, and had problems... First, here is my setup. 2 X100P - Both going to telco 1 T100P - Going to a CAC channel bank (all internal phones) When I flash-hooked, and dial the second number, everything seemed fine. After I heard the call start to ring, I flash-hooked again to connect the 3-way. At that point there was just this terrible noise. I could hear static that sounded like it was the other call still ringing. The caller that was connected could not hear me, and I could not hear them, but they heard the noises. I flash-hooked again, and was able to continue on with the first caller just fine. Any ideas? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom phones with sip, at asterisk.
The snom200 phone is considerably better than the 100. Make sure you have a few extra phones to test. The latest image for the snom200 works great, but they have released some broken images in the past. Be cautious about upgrading the firware just because there is a new release. Otherwise, it is a very configureable phone. Wade Weppler wrote: The SNOM 100's work ok with Asterisk, but the sound quality is poor on the handset, and the speakerphone is unusable. Apparently the latest revision of the SNOM 100 is better, but I haven't used it. The SNOM 200 is supposed to be considerably better, and is only around $50 more. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Tello Abrego Sent: Thursday, February 27, 2003 3:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] snom phones with sip, at asterisk. How is the behaivor of the snom phones with asterisk? Im thinking of getting 40 snom phones, to manage a voip solution... How's the sound quality? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom phones with sip, at asterisk.
What other IP phone, can I use with *, cheap and comercially available, since I in mexico, I'm unable to get many phones, I even think about importing this snome phones... On Thu, 27 Feb 2003, Wade Weppler wrote: > The SNOM 100's work ok with Asterisk, but the sound quality is poor on the > handset, and the speakerphone is unusable. Apparently the latest revision > of the SNOM 100 is better, but I haven't used it. The SNOM 200 is supposed > to be considerably better, and is only around $50 more. > > -wade > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Andres Tello Abrego > > Sent: Thursday, February 27, 2003 3:35 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] snom phones with sip, at asterisk. > > > > > > How is the behaivor of the snom phones with asterisk? > > > > Im thinking of getting 40 snom phones, to manage a voip solution... > > > > How's the sound quality? > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Then use 2 lines per phone. You should connect 2 phones together ( the way they want specify ) and hook an oscilloscope to the 3rd pair. It probably uses a simple signal to indicate page-intercom, maybe even DTMF. You then need to rig a voltage convertor from telco batt voltage to whatever they use ( probably 12v or less ) Jeff Noxon wrote: Yes but I want * to be able to page me, announce queued calls, etc. Basically glorified talking caller ID. ;) On Thu, Feb 27, 2003 at 04:03:09PM -0500, [EMAIL PROTECTED] wrote: Jeff - If you make special phone cords that break out the 3rd pair at the wiring closet you can just join those all together in parallel and your paging will be * independant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom phones with sip, at asterisk.
The SNOM 100's work ok with Asterisk, but the sound quality is poor on the handset, and the speakerphone is unusable. Apparently the latest revision of the SNOM 100 is better, but I haven't used it. The SNOM 200 is supposed to be considerably better, and is only around $50 more. -wade > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andres Tello Abrego > Sent: Thursday, February 27, 2003 3:35 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] snom phones with sip, at asterisk. > > > How is the behaivor of the snom phones with asterisk? > > Im thinking of getting 40 snom phones, to manage a voip solution... > > How's the sound quality? > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Yes but I want * to be able to page me, announce queued calls, etc. Basically glorified talking caller ID. ;) On Thu, Feb 27, 2003 at 04:03:09PM -0500, [EMAIL PROTECTED] wrote: > Jeff - If you make special phone cords that break out the 3rd pair at the > wiring closet you can just join those all together in parallel and your > paging will be * independant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Jeff - If you make special phone cords that break out the 3rd pair at the wiring closet you can just join those all together in parallel and your paging will be * independant. Bill Jeff Noxon wrote: I forsee one of the following: 1) All phones on same paging system. Uses 1 FXS port or a sound card on the asterisk box. 2) Paging system addressable by groups. Uses 1 FXS port per paging group on asterisk. (Plus 1 or 2 FXS for a 1/2 line phone) 3) Paging system addressable by phone. Uses 1 FXS port per phone dedicated to paging. (Up to 3 FXS per phone, somewhat wasteful.) Since this is my home system and I just want simple paging, I'm going to go for option #1. Obviously a digital phone would make a better solution, but that isn't an option right now. Regards, Jeff On Thu, Feb 27, 2003 at 09:22:06AM -1000, James H. Thompson wrote: Are you going to need two ports on asterisk for each phone or does the intercom connect outside of asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
On Thu, Feb 27, 2003 at 10:27:17AM -1000, James H. Thompson wrote: > Do the phones support talkback after a page? I don't believe they do. I'm still looking for the perfect Asterisk phone. Regards, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom phones with sip, at asterisk.
How is the behaivor of the snom phones with asterisk? Im thinking of getting 40 snom phones, to manage a voip solution... How's the sound quality? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
You can do overhead paging with chan_nbs and nbs out of CVS (as well as play MP3's throughout your house) Mark On Thu, 27 Feb 2003, Jeff Noxon wrote: > I forsee one of the following: > > 1) All phones on same paging system. Uses 1 FXS port or a sound card > on the asterisk box. > > 2) Paging system addressable by groups. Uses 1 FXS port per paging > group on asterisk. (Plus 1 or 2 FXS for a 1/2 line phone) > > 3) Paging system addressable by phone. Uses 1 FXS port per phone > dedicated to paging. (Up to 3 FXS per phone, somewhat wasteful.) > > Since this is my home system and I just want simple paging, I'm going > to go for option #1. > > Obviously a digital phone would make a better solution, but that isn't > an option right now. > > Regards, > > Jeff > > On Thu, Feb 27, 2003 at 09:22:06AM -1000, James H. Thompson wrote: > > Are you going to need two ports on asterisk for each phone or does the intercom > > connect outside of > > asterisk? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Do the phones support talkback after a page? Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: "Jeff Noxon" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, February 27, 2003 9:46 AM Subject: Re: [Asterisk-Users] Intercom and Paging > I forsee one of the following: > > 1) All phones on same paging system. Uses 1 FXS port or a sound card > on the asterisk box. > > 2) Paging system addressable by groups. Uses 1 FXS port per paging > group on asterisk. (Plus 1 or 2 FXS for a 1/2 line phone) > > 3) Paging system addressable by phone. Uses 1 FXS port per phone > dedicated to paging. (Up to 3 FXS per phone, somewhat wasteful.) > > Since this is my home system and I just want simple paging, I'm going > to go for option #1. > > Obviously a digital phone would make a better solution, but that isn't > an option right now. > > Regards, > > Jeff > > On Thu, Feb 27, 2003 at 09:22:06AM -1000, James H. Thompson wrote: > > Are you going to need two ports on asterisk for each phone or does the intercom > > connect outside of > > asterisk? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
I got excited about my M7208 models ... but I don't think they're the same Jeff Noxon ([EMAIL PROTECTED]) wrote*: > >The ones I ordered are M9417CW's. Several other nortel models support >the paging feature too. I paid $65 ea for refurbs. > >On Thu, Feb 27, 2003 at 05:17:04PM +, Brian Johnson wrote: >> What models? >> >> Jeff Noxon ([EMAIL PROTECTED]) wrote*: >> > >> >I just purchased a bunch of Nortel Meridian POTS phones that support >> >intercom on the 3rd pair. I intend to get it working with Asterisk. >> >The phones support MWI, have a 3-line display, callerID, call waiting >> >callerID, 2 lines...very nice. >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
I forsee one of the following: 1) All phones on same paging system. Uses 1 FXS port or a sound card on the asterisk box. 2) Paging system addressable by groups. Uses 1 FXS port per paging group on asterisk. (Plus 1 or 2 FXS for a 1/2 line phone) 3) Paging system addressable by phone. Uses 1 FXS port per phone dedicated to paging. (Up to 3 FXS per phone, somewhat wasteful.) Since this is my home system and I just want simple paging, I'm going to go for option #1. Obviously a digital phone would make a better solution, but that isn't an option right now. Regards, Jeff On Thu, Feb 27, 2003 at 09:22:06AM -1000, James H. Thompson wrote: > Are you going to need two ports on asterisk for each phone or does the intercom > connect outside of > asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
Are you going to need two ports on asterisk for each phone or does the intercom connect outside of asterisk? Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: "Jeff Noxon" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, February 27, 2003 8:25 AM Subject: Re: [Asterisk-Users] Intercom and Paging > The ones I ordered are M9417CW's. Several other nortel models support > the paging feature too. I paid $65 ea for refurbs. > > On Thu, Feb 27, 2003 at 05:17:04PM +, Brian Johnson wrote: > > What models? > > > > Jeff Noxon ([EMAIL PROTECTED]) wrote*: > > > > > >I just purchased a bunch of Nortel Meridian POTS phones that support > > >intercom on the 3rd pair. I intend to get it working with Asterisk. > > >The phones support MWI, have a 3-line display, callerID, call waiting > > >callerID, 2 lines...very nice. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HARDWARE
Please: I´m looking for FXS (one port ) harware and It´s a bit difficult for me to import the ones listed in asterisk.org to Argentina, so I need the full list of supported hardware to check in mi local providers. In the mailing list I read about snom and actiontec. Can anyone tell me if them are supported?? Thanks in advance. Ariel P. AichinoCisbCorrientes 314 Of. 18 y 19S2000CTP Rosario ArgentinaTel. Fax. +54 341 4484004http://cisb.mine.nuemail: [EMAIL PROTECTED]
Re: [Asterisk-Users] snom phones and redirect
> We don't do any sort of 3XX processing at the moment. We could modify > Asterisk to relocate. now i know - reading source code of chan_sip.c. I thing if there is no other way to solve this, maybe if you can and wish, you can change code to relocate sip calls. regards Marian > > Mark > -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inbound isdn call [SOLVED]
Thanks Iain to guide me through to solved this and all others for patches, all is working now fine > Iain > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will the Bells crush Net calling?
On Thu, 2003-02-27 at 07:28, TC wrote: > Heads up anybodt doing USA VoIP termination ... > > http://news.com.com/2010-1069-985856.html?tag=fd_nc_1 Listers, I think we shall see the Telcoes line up their contacts at the Federal, State and Local governments by pointing out the tax revenue issues to those governments. Never mind the competition, technology or logic. Howard, things could get messy, White ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
The ones I ordered are M9417CW's. Several other nortel models support the paging feature too. I paid $65 ea for refurbs. On Thu, Feb 27, 2003 at 05:17:04PM +, Brian Johnson wrote: > What models? > > Jeff Noxon ([EMAIL PROTECTED]) wrote*: > > > >I just purchased a bunch of Nortel Meridian POTS phones that support > >intercom on the 3rd pair. I intend to get it working with Asterisk. > >The phones support MWI, have a 3-line display, callerID, call waiting > >callerID, 2 lines...very nice. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup problems...
Hi Guys, Its been almost a week now and I thought I would mention that this seems to have fixed my problem completely. Heh.. of course, the boss is smacking his head against the wall over the fact that he didn't need the T1 card or the channel bank, but.. we would have needed to upgrade in a year or so anyway, so its no big loss. Thanks again for your help! Jim - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, February 24, 2003 2:42 PM Subject: Re: [Asterisk-Users] Hangup problems... > It should detect if the party that called in your > X100P card hanged up. Then if asterisk hears fast busy > it's going to assume that it's the proper time > to hang up the channel. But in your case it's not working > properly and it's hanging up when there is no fast busy. > > regards > Martin > > On Mon, 24 Feb 2003, Jim Howarth wrote: > > > Yes, it was in there, I've commented it out and we shall see what happens. > > Thanks :) > > > > Just curious.. do you know what that is for? > > > > Jim > > > > - Original Message - > > From: "Martin Pycko" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Monday, February 24, 2003 1:57 PM > > Subject: Re: [Asterisk-Users] Hangup problems... > > > > > > > Do you have busydetect=yes in /etc/asterisk/zapata.conf ? > > > If you do then comment it out and see if that helps. > > > > > > regards > > > Martin > > > > > > On Mon, 24 Feb 2003, Jim Howarth wrote: > > > > > > > Howdy, > > > > > > > > We've been trying to get Asterisk to work properly for us and we aren't > > > > having much luck when it comes to simply speaking on the phone. We are > > > > experiencing hangups don't occur at any specific time. The equipment we > > use > > > > are two X101P's and one T100P. We originally tried with with the USB > > S100U > > > > as well (so channel bank isn't an issue). Sometimes the caller will be > > > > completely hung up on and sometimes they will be spit into voicemail. > > My > > > > headset (if it matters) is a Plantronics S10 and the phone is a Bell > > Venture > > > > 3 line phone. > > > > > > > > Has anyone experienced this? If so, how did you fix it. This is > > connected > > > > to our tech support lines and as you can imagine, customers have a > > problem > > > > with being hung up on - on a regular basis. :D > > > > > > > > Thanks! > > > > > > > > Jim > > > > > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aggressive Suppression and Comfort Noise
Voip channel detection by the zap channel would be more helpful. The Aggressive Suppressor would only operate when a channel is patched to voip. All other times it could either be set to lower suppression thresholds or revert to one of the other (or no) echo_cans. I would suspect that adding noise is low on the echo_can priority list until the basic functionality becomes more ideal. John >>We can all agree that the Aggressive Suppression code has gone a long way to solving the echo problems of Asterisk. I, and others, have noticed that its methods do seem a little harsh at time because it seems all sound cuts out during suppression. Would it be possible to add comfort noise to suppression periods so instead of all sound cutting out during suppression, you would at least hear what sounds like normal line noise. I think this would greatly round out the overall sound and make the suppression sound softer and less harsh. This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
What models? Jeff Noxon ([EMAIL PROTECTED]) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson wrote: >> Anyway to get paging/intercom functions using asterisk with SIP or ADSI phones? >> On my current phone, I can intercom another extension (or all extensions) and make an announcement, >> the person at the other end can talk back without touching their phone (if its set to handsfree >> intercom answer). >> Can any of the SIP or ADSI based phones do this? >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Execute asterisk command in shell
asterisk -r -x "this should work" On Thu, 27 Feb 2003, Rattana BIV wrote: > it doesn't work with me =( > > > - Message d'origine - > De : "Martin Pycko" <[EMAIL PROTECTED]> > À : <[EMAIL PROTECTED]> > Envoyé : jeudi 27 février 2003 17:10 > Objet : Re: [Asterisk-Users] Execute asterisk command in shell > > > > asterisk -r -x 'put the command here' > > > > regards > > Martin > > > > On Thu, 27 Feb 2003, Rattana BIV wrote: > > > > > Hi, > > > > > > Does anyone know how to execute an asterisk command in shell ? > > > > > > I wanted to make a script who put extension in asterisk. > > > > > > Regards > > > > > > Rattana > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Execute asterisk command in shell
it doesn't work with me =( - Message d'origine - De : "Martin Pycko" <[EMAIL PROTECTED]> À : <[EMAIL PROTECTED]> Envoyé : jeudi 27 février 2003 17:10 Objet : Re: [Asterisk-Users] Execute asterisk command in shell > asterisk -r -x 'put the command here' > > regards > Martin > > On Thu, 27 Feb 2003, Rattana BIV wrote: > > > Hi, > > > > Does anyone know how to execute an asterisk command in shell ? > > > > I wanted to make a script who put extension in asterisk. > > > > Regards > > > > Rattana > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom and Paging
I just purchased a bunch of Nortel Meridian POTS phones that support intercom on the 3rd pair. I intend to get it working with Asterisk. The phones support MWI, have a 3-line display, callerID, call waiting callerID, 2 lines...very nice. On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson wrote: > Anyway to get paging/intercom functions using asterisk with SIP or ADSI phones? > On my current phone, I can intercom another extension (or all extensions) and make > an announcement, > the person at the other end can talk back without touching their phone (if its set > to handsfree > intercom answer). > Can any of the SIP or ADSI based phones do this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Execute asterisk command in shell
asterisk -r -x 'put the command here' regards Martin On Thu, 27 Feb 2003, Rattana BIV wrote: > Hi, > > Does anyone know how to execute an asterisk command in shell ? > > I wanted to make a script who put extension in asterisk. > > Regards > > Rattana > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] logging all console output?
Yes, you can: asterisk -vvvgcn|tee /tmp/log regards Martin On Thu, 27 Feb 2003, Roy Sigurd Karlsbakk wrote: > hi > > can I log all console output while having console access as with > > asterisk -vvvgc > > ? > -- > Roy Sigurd Karlsbakk, Datavaktmester > ProntoTV AS - http://www.pronto.tv/ > Tel: +47 9801 3356 > > Computers are like air conditioners. > They stop working when you open Windows. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom phones and redirect
> does anybody suceesfully setup snom phones with sip firmware with > asterisk to redirect call when phone is set to redirect if busy/ or > allways redirect ? > My console says : > > chan_sip.c Line 3000 (handle response) > Dunno anything about a 302 Moved Temporarily from SIP... We don't do any sort of 3XX processing at the moment. We could modify Asterisk to relocate. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Execute asterisk command in shell
Hi, Does anyone know how to execute an asterisk command in shell ? I wanted to make a script who put extension in asterisk. Regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BUG: chan_oss outgoing calls autoanswer=no
Hi, today i setup my desktop box as an asterisk server to use chan_oss for my headset (gnophone and my soundsystem give me huge delays). in oss.conf i set autoanswer=no because i dont want incoming callers to eavesdrop on me when i am singing loud to my ABBA mp3s ;-) incoming calls work fine, i type "answer" on the console and the call is connected. but outgoing calls are somewhat odd: - i do a "dial 17001578573" - oss answers the channel i hear the "pling" - app_dial starts - and continues ringing indication forever although the other party can hear me already - when i type "answer" (remember it's an outgoing call) i get connected and can hear them, too. Mark, is this the wanted behaviour of chan_oss? or a small bug? regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705390 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will the Bells crush Net calling?
Hi, I looked around at the pricing for a few VoIP services, like Vonage, this week and was rather puzzled. The IDD rates they all quote for many destinations (eg the US) seem about twice what I pay when I pick up my phone and use the local Telco to make an IDD call. I live on Hong Kong, and I was looking up prices based on being a Hong Kong subscriber to these VoIP services. It strikes me they aren't likely to get too many subscribers here. It looks like the Bells don't have to tax VoIP out of business :-) Regards, Steve TC wrote: Heads up anybodt doing USA VoIP termination ... http://news.com.com/2010-1069-985856.html?tag=fd_nc_1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intercom and Paging
James: The Snom phones have an auto answer mode setting that answers in the speaker phone mode. However, once it's set, it automaticly answers every call. Also, the audio on the speaker phone mode in the Snom 200 is much, much, much better than the Snom 100. Pingtel also has a java app you can load into their phone which allows it to do the same thing, but only for a pre-programmed (in the phone) set of numbers. If the number is not on the list, it just rings normally. Hope that helps, Jerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Thompson Sent: Thursday, February 27, 2003 6:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Intercom and Paging Anyway to get paging/intercom functions using asterisk with SIP or ADSI phones? On my current phone, I can intercom another extension (or all extensions) and make an announcement, the person at the other end can talk back without touching their phone (if its set to handsfree intercom answer). Can any of the SIP or ADSI based phones do this? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inbound isdn call
--On Thursday, February 27, 2003 1:54 pm +0100 Marian Danisek <[EMAIL PROTECTED]> wrote: this mean that i need 2 different patches ? I already found isdn_audio.c and isdn_audio.h patch... this is for i4l. You meat that i need another patch for asterisk ? If you want asterisk to handle dtmf then you need Pauline Middelink's dsp patch - isdn-dsp.txt - which was posted to the list in January. This patch allows asterisk's dsp routines - the same ones used for the zaptel interfaces - to provide dtmf support for the ISDN line. Without it, you will have no dtmf support if you apply the isdn-audio.c patch. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aggressive Suppression and Comfort Noise
We can all agree that the Aggressive Suppression code has gone a long way to solving the echo problems of Asterisk. I, and others, have noticed that its methods do seem a little harsh at time because it seems all sound cuts out during suppression. Would it be possible to add comfort noise to suppression periods so instead of all sound cutting out during suppression, you would at least hear what sounds like normal line noise. I think this would greatly round out the overall sound and make the suppression sound softer and less harsh. Raymond McKay Partner Support Services [EMAIL PROTECTED]
Re: [Asterisk-Users] Intercom and Paging
> Anyway to get paging/intercom functions using asterisk with SIP or ADSI phones? > On my current phone, I can intercom another extension (or all extensions) and make an announcement, > the person at the other end can talk back without touching their phone (if its set to handsfree > intercom answer). > Can any of the SIP or ADSI based phones do this? I have numerous clients that would switch to Asterisk if this were possible. So far, all the responses I've seen have been to the effect of setting up seperate individual paging speakers and the like. Unless something has changed to make this possible, could somebody please comment on what would need to be added on both the client and server side to make this possible. I am currently in the process of researching developing my own low cost IP phone and that is one of the key features that would need to be added to it. Raymond McKay Partner Support Services [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Will the Bells crush Net calling?
Heads up anybodt doing USA VoIP termination ... http://news.com.com/2010-1069-985856.html?tag=fd_nc_1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inbound isdn call
> > Sounds like the i4l dtmf problem. Assuming you are using i4l, the kernel > dtmf detection routines are poor and quite frequently misinterpret speech > as dtmf tones. You need to patch asterisk to handle dtmf and i4l not to > detect dtmf (or silence). There are a few posts on this list about fixing > this issue. > > Iain this mean that i need 2 different patches ? I already found isdn_audio.c and isdn_audio.h patch... this is for i4l. You meat that i need another patch for asterisk ? regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interest in E1 channel banks?
"Dooley, Brendan" <[EMAIL PROTECTED]> writes: > Our company manufactures an E1 channel bank that is approved for use in > Australia (it should also be compatible with Euro standards). It is modular > and available in 10, 20 or 30 analog port configurations. Signal monitoring > and configuration is via Ethernet. It seems a bit strange to me to manufacture devices in 2003 allowing only connection of analog phones --- unless I did not understand your channel bank spec. Connecting plain digital ISDN phones (not channel bank specific) is a requirement for me. Sincerely yours, > > These units are manufactured in low quantities for specific telco > requirements. However if there was enough interest, we would be able to > manufacture and sell the units at pricing levels under US $2000. > > So how much interest is out there? > > Brendan > - > Brendan Dooley > General Manager > > Open Access Pty Ltd > PO Box 301 > Crows Nest NSW 1585 > > Phone +612 9978 7009 > Fax +612 9978 7099 > Email [EMAIL PROTECTED] > - > This email is intended only for the use of the individual or entity named > above and may contain information that is confidential and privileged. If > you are not the intended recipient, you are hereby notified that any > dissemination, distribution or copying of this email is strictly prohibited. > If you have received this email in error, please notify us immediately by > return email or telephone +612 9978 7009 and destroy the original message. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Emmanuel Michon Chef de projet REALmagic France SAS Mobile: 0614372733 GPGkeyID: D2997E42 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inbound isdn call
Sounds like the i4l dtmf problem. Assuming you are using i4l, the kernel dtmf detection routines are poor and quite frequently misinterpret speech as dtmf tones. You need to patch asterisk to handle dtmf and i4l not to detect dtmf (or silence). There are a few posts on this list about fixing this issue. Iain --On Thursday, February 27, 2003 12:06 pm +0100 Marian Danisek <[EMAIL PROTECTED]> wrote: hello, when call is made via asterisk from isdn line to the snom sip phones and caller on isdn line is speaking loudly to the microfone, people on the sip phones didnt hear voice but tones, like dtmf. how can i firuge out this problem ? can echo cancel algorithm ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] logging all console output?
hi can I log all console output while having console access as with asterisk -vvvgc ? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inbound isdn call
hello, when call is made via asterisk from isdn line to the snom sip phones and caller on isdn line is speaking loudly to the microfone, people on the sip phones didnt hear voice but tones, like dtmf. how can i firuge out this problem ? can echo cancel algorithm ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intercom and Paging
Anyway to get paging/intercom functions using asterisk with SIP or ADSI phones? On my current phone, I can intercom another extension (or all extensions) and make an announcement, the person at the other end can talk back without touching their phone (if its set to handsfree intercom answer). Can any of the SIP or ADSI based phones do this? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AGI and fast-entered DTMF codes
> Thats fine, didn't expect you to be able to measure it. The no pauses > raises a bit of concern. There is supposed to be a set time between > keypresses. It must be the phone I was using - which is an internal phone system phone, rather than your average PSTN phone one would have at home. I think my concerns were that if I could cause a problem like that with that particular phone, random users could also cause this. > If they are typing fast due to a contest, chances are they already know > that they can type faster than speed dial, but they can outrun the telco > and have problems. This is something I did back when I tried to win > things from the radio stations. At one point I started using a modem > that I could tune the pause and tone length till I was getting only > about 10% failures. The time to dial was significantly faster. Sorry, I think I explained that a bit wrong. People will receive a code. Then they must call in and enter the code. Once that is successfully verified in the database, they proceed to enter details (for example, date of birth) and then record a short message. The contest is based on the message they enter. Therefore, they will not be in a rush to type in the code faster than others, but my fear was that they might be quick typers and cause a dead line. Again, I used the same (obviously incorrectly configured) phone to call another IVR service that seemed to be able to handle these quick-fire DTMF tones, so I thought that it could be an Asterisk/Zaptel problem, rather than a problem with that particular phone. > If it isn't recognised as DTMF, it is sent as audio. There wouldn't be any garbage. I see. Unfortunately, my telecoms experience is very limited (I only really started using Asterisk about 4 weeks ago!), so forgive my understanding of it! Would that audio cause Asterisk problems in that case, if Asterisk were listening out for X DTMF digits, causing it to bomb out of the "GET DATA" routine? > Good, you didn't see that as the begining of a language holy war. For 2 > lines you are correct it should be plenty enough. As for the idea of > building a multi threaded java server, do you want to tie the system to > another single process that could fail and take all lines down with it? > We have a app we have been working with, and rather like the idea of the > app starting and dieing on a per channel/per call basis so as to make > sure the system is always respawning a fresh copy for a call. Our > current software has threads that hang occasionally and tie up the phone > line till we interactively reset it. This is why I spend so much time > with asterisk since it will eliminate that interactive reseting. Excellent point. My background is in the server-side J2EE arena and I recently built a multi-threaded Java server to handle high volume SMS messages routed through 3 network providers. This is a little bit of a paradigm shift from what I was doing, as the calls through Asterisk are interactive and the user knows if something is wrong and can redial. For SMS messages, the user "fires and forgets" and if no reply is forthcoming, does not know whether that is meant to happen, or the message was lost. This would suggest that, with AGI, the most effective way to go would be a single process that is created and destroyed per call. It does require a trade-off between language functionality and speed of development against the performance profile of that language. Therefore Java might be a bit heavy handed in a large scale environment on that basis. As for the stability of a single server process, monitoring threads can be put in place to ensure a process is in place. However, my preferred way would be to allow the lightweight process that Asterisk calls first to start or restart the server if communication fails. You might have some extra delays, but for high volumes, the extra efficiency might be worth the trade off. Cheers for the help and sorry for going slightly off topic! Alan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interest in E1 channel banks?
Very interested! Are you planning for european certification?(expensive!) If so even more interested! Florian Overkamp said: > At 18:45 27-2-2003 +1100, you wrote: >>Our company manufactures an E1 channel bank that is approved for use in >>Australia (it should also be compatible with Euro standards). It is >> modular >>and available in 10, 20 or 30 analog port configurations. Signal >> monitoring >>and configuration is via Ethernet. >> >>These units are manufactured in low quantities for specific telco >>requirements. However if there was enough interest, we would be able to >>manufacture and sell the units at pricing levels under US $2000. >> >>So how much interest is out there? > > /me raises hand (well, we've done our current infra, but it may be a > consideration none the less - at these levels of pricing) > > Florian > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interest in E1 channel banks?
At 18:45 27-2-2003 +1100, you wrote: Our company manufactures an E1 channel bank that is approved for use in Australia (it should also be compatible with Euro standards). It is modular and available in 10, 20 or 30 analog port configurations. Signal monitoring and configuration is via Ethernet. These units are manufactured in low quantities for specific telco requirements. However if there was enough interest, we would be able to manufacture and sell the units at pricing levels under US $2000. So how much interest is out there? /me raises hand (well, we've done our current infra, but it may be a consideration none the less - at these levels of pricing) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom phones and redirect
Hello, does anybody suceesfully setup snom phones with sip firmware with asterisk to redirect call when phone is set to redirect if busy/ or allways redirect ? My console says : chan_sip.c Line 3000 (handle response) Dunno anything about a 302 Moved Temporarily from SIP... regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interest in E1 channel banks?
Hello, We would be ineterested to get such a device. Most E1 channel banks are too expensive compared to T1 channel banks (I have no idea why). So, any CB at a cost lower than 2000$ is ceratinly interesting. Best regards, Vlasis. "Dooley, Brendan" wrote: > Our company manufactures an E1 channel bank that is approved for use in > Australia (it should also be compatible with Euro standards). It is modular > and available in 10, 20 or 30 analog port configurations. Signal monitoring > and configuration is via Ethernet. > > These units are manufactured in low quantities for specific telco > requirements. However if there was enough interest, we would be able to > manufacture and sell the units at pricing levels under US $2000. > > So how much interest is out there? > > Brendan > - > Brendan Dooley > General Manager > > Open Access Pty Ltd > PO Box 301 > Crows Nest NSW 1585 > > Phone +612 9978 7009 > Fax +612 9978 7099 > Email [EMAIL PROTECTED] > - > This email is intended only for the use of the individual or entity named > above and may contain information that is confidential and privileged. If > you are not the intended recipient, you are hereby notified that any > dissemination, distribution or copying of this email is strictly prohibited. > If you have received this email in error, please notify us immediately by > return email or telephone +612 9978 7009 and destroy the original message. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interest in E1 channel banks?
Our company manufactures an E1 channel bank that is approved for use in Australia (it should also be compatible with Euro standards). It is modular and available in 10, 20 or 30 analog port configurations. Signal monitoring and configuration is via Ethernet. These units are manufactured in low quantities for specific telco requirements. However if there was enough interest, we would be able to manufacture and sell the units at pricing levels under US $2000. So how much interest is out there? Brendan - Brendan Dooley General Manager Open Access Pty Ltd PO Box 301 Crows Nest NSW 1585 Phone +612 9978 7009 Fax +612 9978 7099 Email [EMAIL PROTECTED] - This email is intended only for the use of the individual or entity named above and may contain information that is confidential and privileged. If you are not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this email is strictly prohibited. If you have received this email in error, please notify us immediately by return email or telephone +612 9978 7009 and destroy the original message. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
