Re: [asterisk-users] help with Sipura SPA 3000
On 11 de abr de 2007, at 21:07, James Harper wrote:
A dialplan of '(S0:s)' will get your phone to jump straight into
the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
Just remember that having Asterisk supply the dialtone does add (a
slight) additional load, rather than it just routing calls between
endpoints. Not an issue with one or two ATA's though.
i have just one ATA anyway, this is intended to be used solely at
home... I'll probably give it up in favor of pbxes.org...
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for
you.
From the 'sip_extension_in' context you can make a single '0' or
'*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
A few things to check:
. ${EXTEN:1} will be empty because the extension can only be '0'.
Change
it to 'SIP/LinkSysOut' instead
Done.
. I'm not sure but I think that the SPA3000 can either present a
'false'
dialtone to the SIP call on the PSTN line, take the digits, then send
them to the PSTN then connect the SIP call to it, or it can give the
real PSTN dialtone and connect the call immediately. I think the
latter
is what you want but I can't remember the name of the setting. Maybe
'one stage dialling'?
Done It works! I had to disable one stage dialing and setting the
VOIP DP to none. However, this is giving me one trouble: I use also
my cellphone (E61) to make calls, and it would be nice to do one
stage dialing with it. I don't think it's possible to make it one
stage with the mobile and two stage with the FXS of the Sipura...
Cheers, and thanks a lot!!!
Francis
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Re: [asterisk-users] help with Sipura SPA 3000
Hello Francis,
I also hev asterisk and sipura. Can we chat online on gmail/yahoo. Let's
make some experiments... I hev the same problem like you.
On 4/12/07, Francis Augusto Medeiros [EMAIL PROTECTED] wrote:
On 10 de abr de 2007, at 23:05, James Harper wrote:
2 - How can I gain full control to the FXS? I mean, a simple * dialed
is
not sent for asterisk (the server) interpretation, probably because
it's
used by Sipura's suplementary services, I don't know. Also, is it
possible
to get a dial tone from ASterisk, instead of Sipura's? My goal with
this
is to provide users with direct access to the PSTN line pressing 0,
instead of collecting calls and making the call themselves, or at
least
making ignorepat to work!
A dialplan of '(S0:s)' will get your phone to jump straight into the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for you.
From the 'sip_extension_in' context you can make a single '0' or '*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
Cheers,
Francis
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Re: [asterisk-users] help with Sipura SPA 3000
On 10 de abr de 2007, at 23:05, James Harper wrote: I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of that port because I wasn't sounding loud enough. Get the impedance settings right. An impedance mismatch will cause echo (but may not be the only cause) Thanks a lot for your answer!! But, how do I found out what's the correct impedance of lines here? But there are two things I would like to do with the device, and I'd appreciate if anyone could help me out: 1 - Is there a way to stop cutting other people when I speak through the PSTN? What I mean is that, when sound is captured by my telephone, it dimishes the other peer's voice, and sometimes it makes communication harder, as if the line weren't full duplex. I think the 'echo suppression' setting causes this. It is meant to reduce the incoming audio (and hence the echo) while you are talking, which can be annoying but is supposed to be less annoying than the echo itself. I see... 2 - How can I gain full control to the FXS? I mean, a simple * dialed is not sent for asterisk (the server) interpretation, probably because it's used by Sipura's suplementary services, I don't know. Also, is it possible to get a dial tone from ASterisk, instead of Sipura's? My goal with this is to provide users with direct access to the PSTN line pressing 0, instead of collecting calls and making the call themselves, or at least making ignorepat to work! A dialplan of '(S0:s)' will get your phone to jump straight into the 's' extension in asterisk as soon as someone picks it up. From there you can do something like: [sip_ata_incoming] exten = s,1,Answer exten = s,n,DISA(no-password|sip_extension_in) so Asterisk will give you dialtone and do the dialplan stuff for you. From the 'sip_extension_in' context you can make a single '0' or '*' call the PSTN line. I think if I choose the * to get a dialtone it won't work because it seems that the SPA-3000 will pick up that character and use it as if I was trying to access its own services... By the way, for transfering calls, will asterisk or the SPA the one that will actually do the transfer? Good luck with the echo situation. I have an spa3000 and no matter what I do I get echo coming back to me with almost no reduction in volume!!! Thanks... I don't mind if the echo is small, I actually prefer a small echo than that cutting thing... :( Cheers, Francis James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with Sipura SPA 3000
On 10 de abr de 2007, at 23:05, James Harper wrote:
2 - How can I gain full control to the FXS? I mean, a simple * dialed
is
not sent for asterisk (the server) interpretation, probably because
it's
used by Sipura's suplementary services, I don't know. Also, is it
possible
to get a dial tone from ASterisk, instead of Sipura's? My goal with
this
is to provide users with direct access to the PSTN line pressing 0,
instead of collecting calls and making the call themselves, or at
least
making ignorepat to work!
A dialplan of '(S0:s)' will get your phone to jump straight into the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for you.
From the 'sip_extension_in' context you can make a single '0' or '*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
Cheers,
Francis
___
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asterisk-users mailing list
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RE: [asterisk-users] help with Sipura SPA 3000
A dialplan of '(S0:s)' will get your phone to jump straight into
the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
Just remember that having Asterisk supply the dialtone does add (a
slight) additional load, rather than it just routing calls between
endpoints. Not an issue with one or two ATA's though.
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for
you.
From the 'sip_extension_in' context you can make a single '0' or
'*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
A few things to check:
. ${EXTEN:1} will be empty because the extension can only be '0'. Change
it to 'SIP/LinkSysOut' instead
. I'm not sure but I think that the SPA3000 can either present a 'false'
dialtone to the SIP call on the PSTN line, take the digits, then send
them to the PSTN then connect the SIP call to it, or it can give the
real PSTN dialtone and connect the call immediately. I think the latter
is what you want but I can't remember the name of the setting. Maybe
'one stage dialling'?
. Related to the above, I think you might need to set the dialplan on
the VoIP to PSTN settings to 'none'.
HTH
James
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[asterisk-users] help with Sipura SPA 3000
Hi there everyone! I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of that port because I wasn't sounding loud enough. But there are two things I would like to do with the device, and I'd appreciate if anyone could help me out: 1 - Is there a way to stop cutting other people when I speak through the PSTN? What I mean is that, when sound is captured by my telephone, it dimishes the other peer's voice, and sometimes it makes communication harder, as if the line weren't full duplex. 2 - How can I gain full control to the FXS? I mean, a simple * dialed is not sent for asterisk (the server) interpretation, probably because it's used by Sipura's suplementary services, I don't know. Also, is it possible to get a dial tone from ASterisk, instead of Sipura's? My goal with this is to provide users with direct access to the PSTN line pressing 0, instead of collecting calls and making the call themselves, or at least making ignorepat to work! Cheers, Francis -- Francis Augusto Medeiros ICQ:7825595 Skype: francisaugusto AIM/iChat: francisaugusto Vitória da Conquista - Bahia - Brasil Lâmpada para os meus pés é a Tua palavra, e luz para o meu caminho --- Salmo 119:105 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] help with Sipura SPA 3000
I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of that port because I wasn't sounding loud enough. Get the impedance settings right. An impedance mismatch will cause echo (but may not be the only cause) But there are two things I would like to do with the device, and I'd appreciate if anyone could help me out: 1 - Is there a way to stop cutting other people when I speak through the PSTN? What I mean is that, when sound is captured by my telephone, it dimishes the other peer's voice, and sometimes it makes communication harder, as if the line weren't full duplex. I think the 'echo suppression' setting causes this. It is meant to reduce the incoming audio (and hence the echo) while you are talking, which can be annoying but is supposed to be less annoying than the echo itself. 2 - How can I gain full control to the FXS? I mean, a simple * dialed is not sent for asterisk (the server) interpretation, probably because it's used by Sipura's suplementary services, I don't know. Also, is it possible to get a dial tone from ASterisk, instead of Sipura's? My goal with this is to provide users with direct access to the PSTN line pressing 0, instead of collecting calls and making the call themselves, or at least making ignorepat to work! A dialplan of '(S0:s)' will get your phone to jump straight into the 's' extension in asterisk as soon as someone picks it up. From there you can do something like: [sip_ata_incoming] exten = s,1,Answer exten = s,n,DISA(no-password|sip_extension_in) so Asterisk will give you dialtone and do the dialplan stuff for you. From the 'sip_extension_in' context you can make a single '0' or '*' call the PSTN line. Good luck with the echo situation. I have an spa3000 and no matter what I do I get echo coming back to me with almost no reduction in volume!!! James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
