[Asterisk-Users] Problem with call originating from Cisco
Mark Raming
Mark at Eraze.nl
Sun Nov 7 23:24:57 MST 2004
> On Sun, 2004-11-07 at 12:06, Mark Raming wrote:
> [snip]
> >
> > ; Incomming calls from Pilmo
> > [212.26.192.155]
> > type=user
> > insecure=yes
> > context=incomming
>
> It is usually spelled as "incoming". Typo or intentional?
Yes, this is a typo. But one I made consistently throughout. More like brain
malfunction...
>
> > ; Outgoing calls to Pilmo
> > [pilmo]
> > type=peer
> > insecure=yes
> > fromdomain=nelson.ritstele.com
> > host=nelson.ritstele.com
> > fromuser=31165570909
> > allow=gsm
> > disallow=ulaw;
> > disallow=alaw;
> > dtmfmode=rfc2833
>
> The order is:
> 1) disallow=all
> 2) allow=...
> 3) allow=... etc.
>
> Remove:
> allow=gsm
> disallow=ulaw; <-- please note the ";" should not be there afaik
> disallow=alaw; <-- please note the ";" should not be there afaik
>
> Try with:
> disallow=all
> allow=gsm
That's what I currently have: disallow=all, allow=gsm, and that is working
but causes Asterisk to do the encoding.
What I want is outgoing alaw. So if I remove all disallow/allow lines from
the [pilmo] section, asterisk does outgoing alaw (ie it doesn't have to do
any transcoding). But then I other party cannot hear me.
Thanks,
Mark
>
> Regards,
> Patrick
>
>
>
> ------------------------------
>
> Message: 10
> Date: Sun, 7 Nov 2004 05:45:35 -0800 (PST)
> From: jafar mohammed <sonztechnology at yahoo.com>
> Subject: [Asterisk-Users] Siemens GSM terminal with Wildcard FXO
> To: asterisk-users at lists.digium.com
> Message-ID: <20041107134535.96323.qmail at web53706.mail.yahoo.com>
> Content-Type: text/plain; charset=us-ascii
>
> Hi,
>
> I would like to implement GSM origination for a VOIP
> system i am developing. I am purchasing a Siemens M20
> Terminal and would like to know if i can plug it into
> my Wildcard FXO device to get incoming GSM calls
> routed to the Asterisk server. If anyone has been able
> or successful in using this terminal please let me
> know. And if any of you have this terminal can you
> hook it up to a telephone headset and see if incoming
> calls will ring the headset.
>
> Thank you.
>
>
>
>
>
> __________________________________
> Do you Yahoo!?
> Check out the new Yahoo! Front Page.
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>
>
>
>
> ------------------------------
>
> Message: 11
> Date: Sun, 7 Nov 2004 09:01:57 -0500
> From: "Steve Totaro" <asterisk at totarotechnologies.com>
> Subject: Re: [Asterisk-Users] press # to execute
> To: "Mike Roberts" <manipura at gmail.com>, "Asterisk Users
> Mailing List
> - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <002601c4c4d2$58602a10$0302a8c0 at 600m>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
> reply-type=original
>
>
> That would be implimented on the phone.
>
> Grandstream is like that but on the snom you press OK.
>
>
> ----- Original Message -----
> From: "Mike Roberts" <manipura at gmail.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Sunday, November 07, 2004 7:08 AM
> Subject: [Asterisk-Users] press # to execute
>
>
> >I have this.
> >
> > exten => 8,1,ANSWER
> > exten => 8,2,DigitTimeout,5
> > exten => 8,3,ResponseTimeout,10
> > exten => 8,4,playback(IVR/en_enter_destination)
> >
> > exten => _1XXXXXXX.,1,dial(SIP/${EXTEN}@146.82.15.241)
> >
> > Basicaly its like pressing 8 for long distance, but more controled.
> > But it has to wait until the timeout before it starts to
> dial. Is there
> > a way to make them press # when they are done dialing the num
> > in order to execute the _1XXXXXXX. I want to turn the timeout up
> > but don't want to have them waiting forever. I also need to have a
> > "exten => _011." in there as well. So it won't have the same
> > amount of digits everytime.
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> ------------------------------
>
> Message: 12
> Date: Sun, 7 Nov 2004 06:09:51 -0800
> From: Mike Roberts <manipura at gmail.com>
> Subject: Re: [Asterisk-Users] press # to execute
> To: Steve Totaro <asterisk at totarotechnologies.com>
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <6e0a8bf404110706097e378616 at mail.gmail.com>
> Content-Type: text/plain; charset=US-ASCII
>
> I'm trying to do this from PSTN -> DID -> *
>
> And yes, please spare me the lecture of security, I already know.
>
>
> On Sun, 7 Nov 2004 09:01:57 -0500, Steve Totaro
> <asterisk at totarotechnologies.com> wrote:
> >
> > That would be implimented on the phone.
> >
> > Grandstream is like that but on the snom you press OK.
> >
> >
> >
> >
> > ----- Original Message -----
> > From: "Mike Roberts" <manipura at gmail.com>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Sunday, November 07, 2004 7:08 AM
> > Subject: [Asterisk-Users] press # to execute
> >
> > >I have this.
> > >
> > > exten => 8,1,ANSWER
> > > exten => 8,2,DigitTimeout,5
> > > exten => 8,3,ResponseTimeout,10
> > > exten => 8,4,playback(IVR/en_enter_destination)
> > >
> > > exten => _1XXXXXXX.,1,dial(SIP/${EXTEN}@146.82.15.241)
> > >
> > > Basicaly its like pressing 8 for long distance, but more
> controled.
> > > But it has to wait until the timeout before it starts to
> dial. Is there
> > > a way to make them press # when they are done dialing the num
> > > in order to execute the _1XXXXXXX. I want to turn the timeout up
> > > but don't want to have them waiting forever. I also need to have a
> > > "exten => _011." in there as well. So it won't have the same
> > > amount of digits everytime.
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
>
>
> ------------------------------
>
> Message: 13
> Date: Sun, 7 Nov 2004 07:04:31 -0800
> From: Mike Roberts <manipura at gmail.com>
> Subject: Re: [Asterisk-Users] press # to execute
> To: Steve Totaro <asterisk at totarotechnologies.com>
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <6e0a8bf40411070704cee38ce at mail.gmail.com>
> Content-Type: text/plain; charset=US-ASCII
>
> I found it, read() does exactly what I need
>
>
> On Sun, 7 Nov 2004 06:09:51 -0800, Mike Roberts
> <manipura at gmail.com> wrote:
> > I'm trying to do this from PSTN -> DID -> *
> >
> > And yes, please spare me the lecture of security, I already know.
> >
> >
> >
> >
> > On Sun, 7 Nov 2004 09:01:57 -0500, Steve Totaro
> > <asterisk at totarotechnologies.com> wrote:
> > >
> > > That would be implimented on the phone.
> > >
> > > Grandstream is like that but on the snom you press OK.
> > >
> > >
> > >
> > >
> > > ----- Original Message -----
> > > From: "Mike Roberts" <manipura at gmail.com>
> > > To: <asterisk-users at lists.digium.com>
> > > Sent: Sunday, November 07, 2004 7:08 AM
> > > Subject: [Asterisk-Users] press # to execute
> > >
> > > >I have this.
> > > >
> > > > exten => 8,1,ANSWER
> > > > exten => 8,2,DigitTimeout,5
> > > > exten => 8,3,ResponseTimeout,10
> > > > exten => 8,4,playback(IVR/en_enter_destination)
> > > >
> > > > exten => _1XXXXXXX.,1,dial(SIP/${EXTEN}@146.82.15.241)
> > > >
> > > > Basicaly its like pressing 8 for long distance, but
> more controled.
> > > > But it has to wait until the timeout before it starts
> to dial. Is there
> > > > a way to make them press # when they are done dialing the num
> > > > in order to execute the _1XXXXXXX. I want to turn the timeout up
> > > > but don't want to have them waiting forever. I also
> need to have a
> > > > "exten => _011." in there as well. So it won't have the same
> > > > amount of digits everytime.
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> >
>
>
> ------------------------------
>
> Message: 14
> Date: Sun, 7 Nov 2004 09:36:31 -0600
> From: "Greg Scasny" <gscasny at golden-tech.com>
> Subject: RE: [Asterisk-Users] Channel Banks
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
>
> <12AC7004F95DBA42889BCA2B64A23F68343256 at highlife.golden-tech.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Do not buy an adtran - no auto impedance match on FXO ports
> (echo, echo,
> echo) and no call disconnect supervision on FXO ports.
>
> Get the ADIT 600 (CAC) (also called a Cactus Lite), they have
> all those
> features and have a 48 port capability, plus callerid works wonderful.
>
> If you still want an adtran, I have 3 I can sell you for cheap :)
>
> Greg
>
> Gregory P. Scasny
>
> Golden Technologies Inc.
>
> http://www.golden-tech.com
>
> 219-462-7200
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jay
> Brussels
> Sent: Friday, November 05, 2004 1:11 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Channel Banks
>
> We have an Asterisk Server with 5 X100's and a 4-port openline card we
> have been using for 6 months.
>
> My only complaint is echo. Some lines sound good, others echo all the
> time, some echo intermittantly or only when conferencing. I
> have tweeked the tx/rx gains and played with echo timing as much as
> possible. It is time to go to a channel bank (a PRI is still
> about $200/month more than POT's lines) .
>
> It appears the favorite channel banks are CAC and adtran. Am
> I missing
> anyone? Does one make or model perform better echo
> cancallation then the others? Are some channel banks still having
> problems with caller-id?
>
> Jay
>
>
>
>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ------------------------------
>
> Message: 15
> Date: Sun, 7 Nov 2004 17:20:12 +0100
> From: "Nicklas Bondesson" <nicklas.bondesson at mindping.com>
> Subject: [Asterisk-Users] No busy-tone
> To: <asterisk-users at lists.digium.com>
> Message-ID: <20041107162012.D97182FDBF3 at lists.digium.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi,
>
> I don't hear a busy-tone when calling an external extension
> that's busy. I
> just get the Busy Here 486 message in the debugging log. Any ideas?
>
> Cheers,
> Nicklas
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>
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