Re: [Asterisk-Users] spandsp
Steve, ? Daniel thanks for reply posts the location i download from is as per technote on * installation; export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot prior to the last download i had to manually install the rxfax / txfax applications from opencall.org after latest download rxFAX / txfax are loaded ?? assuming this is latest version of spandsp applications do you have any views on how to proceed with the debug of the failed fax receipt. ?? thanks for your help GT - Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Sunday, September 26, 2004 3:22 PM Subject: Re: [Asterisk-Users] spandsp > Graham Turner wrote: > > >have posted a while ago on issues of receiving faxes by an Asterisk host > >using an x100p fxo interface attached to BT pstn > > > >the asterisk installation is the cvs download as of 23/09/04 > > > >is anyone able to confirm that the rxfax / txfax application that seems to > >be 'bundled' in thecvs download is the latest as per the www.opencall.org > >site which i think is at 0.0.1k ?? > > > >TIA > > > >GT > > > > > Which CVS download are you refering to? rxfax and txfax aren't in > Digium's CVS as far as I know. > > Regards, > Steve > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp
have posted a while ago on issues of receiving faxes by an Asterisk host using an x100p fxo interface attached to BT pstn the asterisk installation is the cvs download as of 23/09/04 is anyone able to confirm that the rxfax / txfax application that seems to be 'bundled' in thecvs download is the latest as per the www.opencall.org site which i think is at 0.0.1k ?? TIA GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: latest cvs / spandsp
apologies as i forget to mention to the receiving device connected to PSTN is x100p fxo i/f - Original Message - From: "Graham Turner" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, September 24, 2004 1:12 PM Subject: latest cvs / spandsp > i am experiencing errors with the rxfax application when receiving faxes > from a 'brother' fax device. > > the rxfax application picks up the incoming fax but the subseqeuent > 'negotiation' process seems to fail with the messages logged to the asterisk > console as below; > > fast carrier up > > coarse carrier frequency 1697.01(8) > > training error 666.008746 > > traininig failed (convergence failed) > > fast carrier down / up > > coarse carrier frequency 1692.62(8) > > training error 661.340515 > > traininig failed (convergence failed) > > fast carrier down / up > > coarse carrier frequency 1694.14(8) > > training error 667.475496 > > traininig failed (convergence failed) > > fast carrier up / down couple of times > > frequency 1426.08 > > carrier down > > hungup / zap > > > > have seen a few similalry related posts but no conclusive answers it would > seem > > the asterisk installation is built from the cvs source as of last night > (23/09/04) > > TIA > > GT > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] latest cvs / spandsp
i am experiencing errors with the rxfax application when receiving faxes from a 'brother' fax device. the rxfax application picks up the incoming fax but the subseqeuent 'negotiation' process seems to fail with the messages logged to the asterisk console as below; fast carrier up coarse carrier frequency 1697.01(8) training error 666.008746 traininig failed (convergence failed) fast carrier down / up coarse carrier frequency 1692.62(8) training error 661.340515 traininig failed (convergence failed) fast carrier down / up coarse carrier frequency 1694.14(8) training error 667.475496 traininig failed (convergence failed) fast carrier up / down couple of times frequency 1426.08 carrier down hungup / zap have seen a few similalry related posts but no conclusive answers it would seem the asterisk installation is built from the cvs source as of last night (23/09/04) TIA GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] uk caller id
kevin, i will give the latest cvs of asterisk (and libpri / zaptel ) as seems good practice would you be happy to share with me (off topic if necessary) your zapata.conf (for X100P ??) GT - Original Message - From: "Kevin Walsh" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Tuesday, September 21, 2004 10:41 AM Subject: RE: [Asterisk-Users] uk caller id > Graham Turner [EMAIL PROTECTED] lazily top-posted: > > i have installed asterisk / zaptel from cvs distribution as of 17/09/04 > > so i assume this does it > > > If you have a TDM/FXO then you'll need the latest CVS code. If you > have a X100P then you'll need any CVS (the latest is usually a good > choice) and some patches. I have the X100P running with today's CVS > version and with UK (BT) Caller*ID support. > > > > > have configured zapata.conf as per instruction but i would have expected > > to have seen the callerid on the asterisk console as it receives the call > > but then may be not ?? > > > > the relevant my extensions.conf is ; > > > > exten => s,1,answer > > exten => s,2,Dial(SIP/1001|10) > > > > it is quite possible that callerid is being seen by * but i would have > > expected it to have been echoed to the console or at least written to the > > CDR entries ??? > > > There's no need to answer before dialling, btw. The SIP phone's > answer will filter through the system. > > Aside from that, the (UK) Caller*ID will only be available if you have > one of the setups described above (TDM with latest CVS code or X100P > with patches). > > > > > going a bit further on, the whole point of this exercise is to allow this > > CALLERID to be displayed on the console of a SIP peer (7940 ip phone) that > > is defined as an asterisk extension > > > That will happen once you're set up, yes. > > -- >_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ > _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h > _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] > _/ _/ _/_/_/_/ _/_/_/_/ _/_/ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp / compilation errors
much better - thanks very much !! now have my asterisk back to its former 'glory' and am getting something out of rxfax - not immediate success fax call was clearly 'answered' but a whole load of messages along the lines of 'carrier down' flashed past any quick way of capturing them ?? had a look in /var/log/asterisk/messages but not in there ?? - Original Message - From: "Mike Machado" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Monday, September 20, 2004 5:51 PM Subject: Re: [Asterisk-Users] spandsp / compilation errors > Add '/usr/local/lib' to /etc/ld.so.conf if not already, and run > 'ldconfig' as root. Then start asterisk. > > On Mon, 2004-09-20 at 09:45, Graham Turner wrote: > > Daniel, thanks for mail back - this has got me much further through spandsp > > installation process > > > > i have progressed through your technote by applying patches to rxfac.c > > /txfax.c and applying the Makefile patch > > > > i assume by rebuild of Asterisk this is make clean; make install in the > > /usr/src/asterisk directory ?? - which is as i have done > > > > do i need to do the same with zaptel / librpi as per the asterisk install > > guide ?? > > > > however i am now in the unfortunate position where the asterisk does not now > > start correctly - the console logs the message; > > > > libspandsp.so - cannot open shared object file - no such file or directory > > .. > > loading module app_rxfax.so failed ! > > > > clues on this will be VERY gladly received > > > > GT > > > > > > - Original Message - > > From: "administrator tootai" <[EMAIL PROTECTED]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <[EMAIL PROTECTED]> > > Sent: Monday, September 20, 2004 3:54 PM > > Subject: Re: [Asterisk-Users] spandsp / compilation errors > > > > > > > Graham Turner a écrit : > > > > > > >I am attempting installation of spandsp on to an Asterisk installation > > on > > > >Linux RH9 > > > > > > > >the distribution i am using is that are URL http://ftp2.tootai.net - the > > > >README for which i have followed verbatim - > > > > > > > > > > > It's not a special distribution, it's the original one. It's just here > > > as in august the opencall website was down a long time ;-) > > > > > > >my only issue on this was the target for the port.h / tif_dir.h / > > tiffiop.h > > > >files in the 'headers' folder of the distribtion > > > > > > > >i put these in the /usr/include folder based simply on the fact that > > there > > > >is nothing in the /usr/local/include > > > > > > > >the tiffio.h / tiffvers.h files are not in here so i am beginning to > > suspect > > > >the installation of libtiff on the system - however i checked 'rpm -qa' > > and > > > >it does confirm libtiff 3.5.7 as being installed > > > > > > > > > > > You have libtiff-3.5.7 but what about libtiff-devel-3.5.7 which provide > > > tiffio.f and consors? > > > > > > >any clues on the debug of failed compilation will be gladly received > > > > > > > >GT > > > > > > > > > > > -- > > > Daniel > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp / compilation errors
Daniel, thanks for mail back - this has got me much further through spandsp installation process i have progressed through your technote by applying patches to rxfac.c /txfax.c and applying the Makefile patch i assume by rebuild of Asterisk this is make clean; make install in the /usr/src/asterisk directory ?? - which is as i have done do i need to do the same with zaptel / librpi as per the asterisk install guide ?? however i am now in the unfortunate position where the asterisk does not now start correctly - the console logs the message; libspandsp.so - cannot open shared object file - no such file or directory .. loading module app_rxfax.so failed ! clues on this will be VERY gladly received GT - Original Message - From: "administrator tootai" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Monday, September 20, 2004 3:54 PM Subject: Re: [Asterisk-Users] spandsp / compilation errors > Graham Turner a écrit : > > >I am attempting installation of spandsp on to an Asterisk installation on > >Linux RH9 > > > >the distribution i am using is that are URL http://ftp2.tootai.net - the > >README for which i have followed verbatim - > > > > > It's not a special distribution, it's the original one. It's just here > as in august the opencall website was down a long time ;-) > > >my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h > >files in the 'headers' folder of the distribtion > > > >i put these in the /usr/include folder based simply on the fact that there > >is nothing in the /usr/local/include > > > >the tiffio.h / tiffvers.h files are not in here so i am beginning to suspect > >the installation of libtiff on the system - however i checked 'rpm -qa' and > >it does confirm libtiff 3.5.7 as being installed > > > > > You have libtiff-3.5.7 but what about libtiff-devel-3.5.7 which provide > tiffio.f and consors? > > >any clues on the debug of failed compilation will be gladly received > > > >GT > > > > > -- > Daniel > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp / compilation errors
I am attempting installation of spandsp on to an Asterisk installation on Linux RH9 the distribution i am using is that are URL http://ftp2.tootai.net - the README for which i have followed verbatim - my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h files in the 'headers' folder of the distribtion i put these in the /usr/include folder based simply on the fact that there is nothing in the /usr/local/include the tiffio.h / tiffvers.h files are not in here so i am beginning to suspect the installation of libtiff on the system - however i checked 'rpm -qa' and it does confirm libtiff 3.5.7 as being installed any clues on the debug of failed compilation will be gladly received GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] uk caller id
Kevin, thanks for post reply . i have installed asterisk / zaptel from cvs distribution as of 17/09/04 so i assume this does it have configured zapata.conf as per instruction but i would have expected to have seen the callerid on the asterisk console as it receives the call but then may be not ?? the relevant my extensions.conf is ; exten => s,1,answer exten => s,2,Dial(SIP/1001|10) it is quite possible that callerid is being seen by * but i would have expected it to have been echoed to the console or at least written to the CDR entries ??? would you have any suggestions as to how to confirm this going a bit further on, the whole point of this exercise is to allow this CALLERID to be displayed on the console of a SIP peer (7940 ip phone) that is defined as an asterisk extension thanks 4 yr help GT - Original Message - From: "Kevin Walsh" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Sunday, September 19, 2004 1:48 PM Subject: RE: [Asterisk-Users] uk caller id > Graham Turner [EMAIL PROTECTED] wrote: > > dear all, i am looking to enable CALLERID on an Asterisk system > > comprising a X101P FXO interface connecting to BT PSTN in the uk > > > > seems this is supported by the interface but there seems to be varying > > information on how to enable it in zapata.conf > > > > 1. usecallerid=uk > > > > 2. ukcallerid=yes > > > > being two of the configuration statements offered > > > The current method is "usecallerid = uk". Of course, you need to > patch Zaptel and Asterisk first. The "ukcallerid = yes" was used in > an earlier version of the patch. > > -- >_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ > _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h > _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] > _/ _/ _/_/_/_/ _/_/_/_/ _/_/ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uk caller id
dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered TIA GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco reinvite
it seems that the use of renivite in sip peer configuration is very much dependent on sip endpoint have read of what seems defnite "no no" when using the cisco ata 186 it seems eminently preferable from a networking / performance view for the media data transfer to be between the two endpoints and not using the proxy, especially when all hosts are behind a NAT was wondering if the list could provide me with general view on the use of reinvite given endpoints of cisco 7940 ip phones (latest 7.1 sip image) and cisco 1760 router GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] changing the ip address of an asterisk pbx
looking to move an asterisk pbx server to a different vlan and as such looking to check the impact of this change on the asterisk application obviously we have the linux interface reconfiguration to complete are there any application level settings that need to be changed to reflect the changed ip address of the host ? GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk console messages
was wondering if someone could give any indication of the messages that are appearing on the console of an Asterisk PBX WARNING[1116941120]: chan_sip.c:532 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (non-critical request) 192.168.90.1 is a 7940 ip phone configured as a SIP dial peer on asterisk pbx i mght added that the call seems to take place ok but this message appears every time - was hoping to some 'heads-up' on the severity of this message as it does seem to indicate some sort of failiure / misconfiguration ?? Thanks GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] using asterisk with iptel addreses
was wondering if anyone could give us a run through an explanation of the
wiki and other examples of connecting to iptel's sip express router using
asterisk pbx so i can understand better the call processing ..
given the example i work from on john todd's www.loligo.com site ;
exten => _3.,1,SetCallerID(${IPTELUSERID})
exten => _3.,2,SetCIDname(${IPTELUSERNAME})
exten => _3.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _3.,4,Playback(invalid)
exten => _3.,5,Hangup
are the first two statements mandatory to connection to iptel.org ??
i understand the dial plan of any numbers prefixed by 3 are interpreted as
iptel extensions, and that the third extension priority strips off the
prefix
presumably the call is then 'processed' by the [iptel] section of sip.conf
and generates sip call with the credentials in this section as those passed
to the iptel.org server ??
how (iassuming they do) these relate in any way to the IPTELUSERID /
IPTELUSERNAME variables defined in extensions.conf ??
presumably (and i am using a 7940 sip device) i can dial an asterisk
extension of targetiptelusername prefixed with 3 to call the sip user
registered with
qu. does the target iptel username need the iptel.org domain appended to it
or is it somehow 'implied' by the above ??
another wiki example suggests the use of a 'fromdomain' statement in an
[iptel] section of sip.conf ??
TIA
GT
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[Asterisk-Users] (no subject)
was wondering if anyone could give us a run through
an explanation of the wiki and other examples of connecting to iptel's sip
express router using asterisk pbx so i can understand better the call
processing ..
given the example i work from on john todd's
www.loligo.com site ;
exten =>
_3.,1,SetCallerID(${IPTELUSERID})
exten =>
_3.,2,SetCIDname(${IPTELUSERNAME})
exten =>
_3.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _3.,4,Playback(invalid)
exten => _3.,5,Hangup
are the first two statements mandatory to
connection to iptel.org ??
i understand the dial plan of any numbers prefixed
by 3 are interpreted as iptel extensions, and that the
third extension priority strips off the prefix
presumably the call is then 'processed' by the
[iptel] section of sip.conf and generates sip call with the credentials in
this section as those passed to the iptel.org server ??
how (iassuming they do) these relate in any
way to the IPTELUSERID / IPTELUSERNAME variables defined in extensions.conf
??
presumably (and i am using a 7940 sip device) i can
dial an asterisk extension of targetiptelusername prefixed with 3
to call the sip user registered with
qu. does the target iptel username need the
iptel.org domain appended to it or is it somehow 'implied' by the above
??
another wiki example suggests the use of a
'fromdomain' statement in an [iptel] section of sip.conf
??
TIA
GT
[Asterisk-Users] extension pattern matching
dear all, was hoping someone could give me instruction on the syntax of extension pattern matching for letters the proposed 'dial plan' is one where any letter in the dialled digits causes the pbx to assume we are dilaling a sip url and as such forward to the appropraite sip service provider was hoping to avoid the plan in john todd's example that assumes anything prefixed with 3 is a sip address gt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail customization
have managed to establish voicemail functionality using voicemail / voicemailmain applications the documentation on these applications from digium.com suggests that voicemail greetings are customizable (as one would be expect), but am not able to find any supporting documentation can anyone refer me to said documentation or provide assistance on how to proceed GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940
Brian, thanks for your post reply .
2 further qu's if i may
in yr exten statement you use voicemailmain as the application.
i have got exten => 1001,2,Voicemail(u1001)
i know there has been recent developements to the voicemail application but
is this correct given a cvs download of early this month ??
2nd qu - where do i configure the 'voicemail uri' - have been through the
phone / line settings - or do i have to configure the SIP or
sipdefault.cnf files ??
GT
- Original Message -
From: "Brian Cuthie" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 18, 2004 2:47 PM
Subject: Re: [Asterisk-Users] asterisk voicemail retrieval using a cisco
7940
> Graham,
>
> You need to configure something in extensions.conf to access voicemail.
> I usually use something like this:
>
> exten => 8500,1,VoiceMailMain(s${CALLERIDNUM})
> exten => 8500,2,Congestion
>
> Then you'll want to configure the voicemail URI on the 7940 so that it
> calls extension 8500.
>
> One nice thing about the Cisco phone is that they will keep track of WMI
> separately for each configured line.
>
> -brian
>
> Graham Turner wrote:
>
> >can anyone give me a reference to the retrieval of voicemail from the
> >Asterisk PBX using a cisco 7940 phine running sip image.
> >
> >i have configured a single voicemail box using the script, the
corresponding
> >entry in voicemail.conf and configured the extension to use the voicemail
> >box .
> >
> >i can see from the asterisk console the message being passed to the voice
> >mailbox, and correspondingly the sip phone indicates voicemail by the
> >flashing red on the handset and the envelope on its console
> >
> >it would seem further configuration work is required to access the voice
> >mailbox
> >
> >TIA
> >
> >GT
> >
> >___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
> ___
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>
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[Asterisk-Users] asterisk voicemail retrieval using a cisco 7940
can anyone give me a reference to the retrieval of voicemail from the Asterisk PBX using a cisco 7940 phine running sip image. i have configured a single voicemail box using the script, the corresponding entry in voicemail.conf and configured the extension to use the voicemail box . i can see from the asterisk console the message being passed to the voice mailbox, and correspondingly the sip phone indicates voicemail by the flashing red on the handset and the envelope on its console it would seem further configuration work is required to access the voice mailbox TIA GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mailbox numbers
hopefully a quick and not too daft a question but just wanted to check to see if the extension numbers as defined in extensions.conf and the sections of sip.conf needed to have a numbering plan that was exclusive of the numbers that are allocated to the mailbox numbers that are established for voicemail. ie can we have an extension number and a mailbox number of the same number ??? this would seem to afford the simplest mapping of an extension to an identically numbered mailbox. or as i perhaps suspect the mailbox numbers are in fact "extensions" and as such need to have a numbering scheme that is exclusive from the extension numbers the wiki example seem to suggest we can have an identical numbering scheme for the two ??? GT ie can ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
