Hi,

It can be codec negotiation error or else plese try to print hangupcause
sent from telco

On Wed, Apr 18, 2012 at 4:27 PM, Tech wrote:

> Hi,****
>
> ** **
>
> I have a problem where calling “out” of asterisk when the call is answered
> dahdi hangs up immediately.****
>
> For example: Sip Client A calls external number. Route: SIP -> FXO -> GSM
> Gateway ->External Landline.****
>
> However when that external landline answers the call dahdi hangs up
> immediately .****
>
> ** **
>
> Going the other way is fine (External Landline -> GSM Gateway -> FXO ->
> SIP).****
>
> ** **
>
> I’ve tried multiple different internet searches and can’t seem to find any
> information on this problem.****
>
> ** **
>
> Below are my config files.****
>
> ** **
>
> *Sip.conf*
>
> [office-phone](!) ****
>
> type=friend ****
>
> context=sipofficephone ****
>
> host=dynamic ****
>
> nat=yes ****
>
> #secret=xxxx ****
>
> dtmfmode=auto ****
>
> disallow=all ****
>
> ;allow=ulaw ****
>
> allow=alaw ****
>
> allow=GSM****
>
> ** **
>
> [lewisphone](office-phone);lewis mobile****
>
> secret=xxxx****
>
> ** **
>
> *Chan_dahdi.conf*
>
> [channels]****
>
> signalling=fxs_ks ****
>
> context=pstnincomming****
>
> group=0****
>
> channel => 1****
>
> ** **
>
> ** **
>
> *Extensions.conf*
>
> [sipofficephone]****
>
> exten => _X.,1,Verbose(2,Call from VoIP network to ${EXTEN})****
>
> same => n,Dial(DAHDI/1/${EXTEN})****
>
> same => n,Hangup()****
>
> ** **
>
> [pstnincomming]Diamon****
>
> exten => s,1,Answer()****
>
> same => n,Dial(SIP/lewisphone)****
>
> same => n,Hangup()****
>
> ** **
>
> ** **
>
> *Asterisk CLI Output (Verbose 3)*
>
> My comments bold.****
>
> ** **
>
> == Using SIP RTP CoS mark 5****
>
> — Executing [xxxx@sipofficephone:1]
> Verbose(“SIP/lewisphone-0000000a”, “2,Call from VoIP network to xxxx”) in
> new stack****
>
> == Call from VoIP network to xxxx****
>
> — Executing [xxxx@sipofficephone:2] Dial(“SIP/lewisphone-0000000a”,
> “DAHDI/1/xxxx”) in new stack****
>
> — Called DAHDI/1/xxxx****
>
> — DAHDI/1-1 answered SIP/lewisphone-0000000a *GSM Gateway Answering
> Call then Sending it out.*
>
> — Hanging up on ‘DAHDI/1-1′ *Dest answering call to which DAHDI
> hangs up*
>
> — Hungup ‘DAHDI/1-1′****
>
> == Spawn extension (sipofficephone, xxxx, 2) exited non-zero on
> ‘SIP/lewisphone-0000000a’****
>
> ** **
>
> ** **
>
> ** **
>
> Best Regards****
>
> *
>
> *
>
> Lewis ****
>
> [image: digitalselect-e]****
>
> www.Digital-Select.com http://www.digital-select.com/****
>
> *
>
> *****
>
> ** **
>
> –
> _____________________________________________________________________
> — Bandwidth and Colocation Provided by http://www.api-digital.com –
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hi,

I’ve read here and there how Asterisk could send SMS but I didn’t find
much about how to receive SMS and forward them to an email box.

1. First of all, I don’t think my telco would let me receive any SMS
my landline.

2. Maybe I could find providers selling this service for a monthly fee;

3. I could build and operate my own infrastructure.

Given this asterisk-users mailing-list purpose, and for curiosity’s
sake, how could I build my own SMS reception service with Asterisk
(1.6.1 or later) ?
Which channel (chan_mobile, chan_datacard, …) and hardware would be
appropriate ?
Suggestions ?

Regards

Hi,

We noticed a very sharp drop in voice quality when using digium g729a
codec. The problem seems to happen if the A channel (caller’s channel)
is a landline/mobile number contacted using the same outgoing provider
(as a local channel). It sounds like listening to a mono speaker on
low volume.

If I use a softphone that is directly registered to our asterisk box
the audio quality improves, the words come out more clearer and
louder.

I also asked my provider to test call me using their Cisco as5300
system and g729 codec and compared it with ulaw. The difference is
unnoticable.

Hello

I’d like to build a compact, affordable, fanless x86 solution to
handle my home landline.

I know about the following two platforms:

1. www.pcengines.ch/alix.htm
alix1d + case 100€

Does “Availability >500″ mean that it’s just not possible to buy just
one item?

2. www.soekris.com/products.html?limit=all
net4501-30 Board and Case $175.00

Is the net4501 powerful enough to run Asterisk, considering that I’ll
use an external VoIP gateway to connect it to my landline?

Are there other manufacturers I should know about?

Thank you.

HI,

I have installed asterisk 1.6.2.18 with konference 1.7, All things are
working fine but when we start taking DTMF then key 3 not get my asterisk.
When we use landline number(dedicated number) than all DTMF is capture and
asterisk work fine. In case of mobile only key 3 don’t work. Strange when I
use my touch screen number then most of the DTMF digits don’t get’s my
asterisk….

But the same module work with asterisk 1.4.41 without any issue.

So is it asterisk issue or konference module issue please guide me…..
Or all problem is created my mobile divices ?

Thanks in advance ………