[asterisk-users] Problem with making outgoing calls

VoIP Street
Wed, 12 Jul 2006 07:57:39 -0000

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Crazy Boy wrote:
<blockquote
 cite="mid20060712132227.36480.qmail@web37114.mail.mud.yahoo.com"
 type="cite">
<font
color="#000000">Hi,
<span
class="312502706-12072006">
<font
color="#000000">We could
make&nbsp;calls to USA using Teliax service upto 11, July 2006 with
Asterisk. But, since 11, July 2006 evening, we are unable to make calls
sometimes and could not connect to Teliax server sometimes. I have
realized that Teliax server was down for few hours.

<span
class="312502706-12072006">
<font
color="#000000">Currently
our Asterisk server is connecting with Teliax. But, When I am trying to
make call to USA, Its giving me one ring and being disconnected. I
could not understand what could be the problem? Is&nbsp;there any problem
with my connection to Teliax server?
<span
class="312502706-12072006">

<font
color="#000000">With the
configuration mentioned below, we could successfully make calls to USA.
But, from this morning we are unable to make calls to USA.


<font
color="#000000">IAX.CONF
file contents:
<span
class="312502706-12072006">
<font
color="#000000">disallow=all
<font
color="#000000">allow=ulaw
<span
class="312502706-12072006">
<span
class="312502706-12072006">SIP.CONF file
contents:
<span
class="312502706-12072006">
<font
color="#000000">[101]
<span
class="312502706-12072006">
<font
color="#000000">type=friend
<font
color="#000000">username=101
<font
color="#000000">secret=abcd
<font
color="#000000">callerid="Ani"
<font
color="#000000">host=dynamic
<font
color="#000000">context=tutorial
<span
class="312502706-12072006">
<font
color="#000000">[general]
register => <a class="moz-txt-link-abbreviated"
href="mailto:ab.cd:xxxxx@voip-co1.teliax.com">ab.cd:xxxxx@voip-co1.teliax.com
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;

[authentication]
auth = <a class="moz-txt-link-abbreviated"
href="mailto:ab.cd:xxxxxx@voip-co1.teliax.com">ab.cd:xxxxxx@voip-co1.teliax.com

[teliax]
context=default
<font
color="#000000">type=friend
username=ab.cd
user=ab.cd
host=voip-co1.teliax.com
secret=xxxxxx
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
<span
class="312502706-12072006">
<span
class="312502706-12072006">
<font
color="#000000">EXTENSIONS.CONF
file contents:
<span
class="312502706-12072006">
<font
color="#000000">exten
=> 101,1,Dial(SIP/101,15)
<font
color="#000000">
<font
color="#000000">exten
=> 101,2,Voicemail(u101)

<font
color="#000000">exten
=> 101,3,Voicemail(b101)

<font
color="#000000">exten
=> 101,4,Hangup
<span
class="312502706-12072006">
exten =>
_1XXXXXXXXXX,1,DIAL(SIP/teliax/${EXTEN},30,tr)

exten => 3031234567,1,Answer()
exten => 3031234567,2,DIAL(SIP/user,20)

<span
class="312502706-12072006">
<span
class="312502706-12072006">VOICEMAIL.CONF file
contents:
<span
class="312502706-12072006">
101 => ani,
Ani, <a
href="mailto:ani@olivetech.com">ani@abc.com, <a
href="mailto:ani@olivetech.com">ani@abc.com



<span
class="312502706-12072006">
<font
color="#000000">Please
let me know the problem ASAP. Looking forward to your response.

<span
class="312502706-12072006">
<font
color="#000000">Thank you.

Regards,
Chandra.


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Do you get some sort of error if you look at the console (CLI) while
attempting the call? If not, give it a try.. If you try and don't see
anything useful, try turning on debug by typing "sip debug" if using
sip or "iax2 debug" if using IAX.

When you are done use "sip no debug" or "iax2 no debug" to
turn it off
:)

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