[Asterisk-Dev] ATA 186 connected phone is not ringing (Rajeev) - Solved
Rajeev
rajeev at cefib.com
Mon Aug 16 09:37:08 MST 2004
Hi,
The above mentioned problem is solved by editing the sip.conf file.
allow=all is changed to diallow=all and added allow=ulaw.
Thanks
Rajeevk
----- Original Message -----
From: <asterisk-dev-request at lists.digium.com>
To: <asterisk-dev at lists.digium.com>
Sent: Saturday, August 14, 2004 1:41 PM
Subject: Asterisk-Dev digest, Vol 1 #835 - 7 msgs
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> Today's Topics:
>
> 1. IAX/SIP listener ??? (Jan Hulala)
> 2. Re: IAX/SIP listener ??? (David Pollak)
> 3. HELP: BYE-request not sent to SIP-peer (Roland Zagler)
> 4. Re: HELP: BYE-request not sent to SIP-peer (Josh Roberson)
> 5. What happened to #asterisk on irc.freenode.net (chaye wala)
> 6. Re: What happened to #asterisk on irc.freenode.net (Jeremy McNamara)
> 7. ATA 186 connected phone is not ringing (Rajeev)
>
> --__--__--
>
> Message: 1
> From: "Jan Hulala" <hulala at xodatel.com>
> To: <asterisk-dev at lists.digium.com>
> Date: Fri, 13 Aug 2004 18:01:14 -0400
> Organization: XODATEL
> Subject: [Asterisk-Dev] IAX/SIP listener ???
> Reply-To: asterisk-dev at lists.digium.com
>
> This is a multi-part message in MIME format.
>
> ------=_NextPart_000_019C_01C4815F.85FCA5E0
> Content-Type: text/plain;
> charset="Windows-1252"
> Content-Transfer-Encoding: quoted-printable
>
> Hi folks,
>
> I need simple application (the best in perl :), which will just listen =
> for incoming calls, read callerID information. It should just ring or to =
> answer as busy, depend on callerID number. How can I do this? Do I need =
> to use Asterisk for this?
>
> (I want to use a VoicePulse Connect for this)
>
> Any help will be very appreciate.
>
> Thank you!
>
> Jan
>
> ------=_NextPart_000_019C_01C4815F.85FCA5E0
> Content-Type: text/html;
> charset="Windows-1252"
> Content-Transfer-Encoding: quoted-printable
>
> <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
> <HTML><HEAD>
> <META http-equiv=3DContent-Type content=3D"text/html; =
> charset=3Dwindows-1252">
> <META content=3D"MSHTML 6.00.2737.800" name=3DGENERATOR>
> <STYLE></STYLE>
> </HEAD>
> <BODY bgColor=3D#ffffff>
> <DIV><FONT size=3D2>Hi folks,</FONT></DIV>
> <DIV><FONT size=3D2></FONT> </DIV>
> <DIV><FONT size=3D2>I need simple application (the best in perl :), =
> which will=20
> just listen for incoming calls, read callerID information. It should =
> just ring=20
> or to answer as busy, depend on callerID number. How can I do this? Do I =
> need to=20
> use Asterisk for this?</FONT></DIV>
> <DIV><FONT size=3D2></FONT> </DIV>
> <DIV><FONT size=3D2>(I want to use a VoicePulse Connect for =
> this)</FONT></DIV>
> <DIV><FONT size=3D2></FONT> </DIV>
> <DIV><FONT size=3D2>Any help will be very appreciate.</FONT></DIV>
> <DIV><FONT size=3D2></FONT> </DIV>
> <DIV><FONT size=3D2>Thank you!</FONT></DIV>
> <DIV><FONT size=3D2></FONT> </DIV>
> <DIV><FONT size=3D2>Jan</FONT></DIV></BODY></HTML>
>
> ------=_NextPart_000_019C_01C4815F.85FCA5E0--
>
>
> --__--__--
>
> Message: 2
> Date: Fri, 13 Aug 2004 15:28:37 -0700
> From: David Pollak <dpp-asterisk at projectsinmotion.com>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] IAX/SIP listener ???
> Reply-To: asterisk-dev at lists.digium.com
>
> This is a multi-part message in MIME format.
> --------------030108030000000100020600
> Content-Type: multipart/alternative;
> boundary="------------090206070803090601060004"
>
>
> --------------090206070803090601060004
> Content-Type: text/plain; charset=windows-1252; format=flowed
> Content-Transfer-Encoding: 7bit
>
> Jan,
>
> Using Asterisk for this is a lot like swatting flies with a Buick.
> Asterisk will do the job, shine your shoes, get you a latte (or coffee
> if you don't live on one of the coasts), and walk your dog.
>
> If you want wanted to build such an app in Perl, you could use the
> Asterisk AGI:
> http://www.voip-info.org/wiki-Asterisk+AGI
>
> You could probably even write a Dial Plan that would do the job.
>
> Thanks,
>
> David
>
> Jan Hulala wrote:
>
> > Hi folks,
> >
> > I need simple application (the best in perl :), which will just listen
> > for incoming calls, read callerID information. It should just ring or
> > to answer as busy, depend on callerID number. How can I do this? Do I
> > need to use Asterisk for this?
> >
> > (I want to use a VoicePulse Connect for this)
> >
> > Any help will be very appreciate.
> >
> > Thank you!
> >
> > Jan
>
>
> --------------090206070803090601060004
> Content-Type: text/html; charset=windows-1252
> Content-Transfer-Encoding: 8bit
>
> <!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
> <html>
> <head>
> <meta content="text/html;charset=windows-1252"
> http-equiv="Content-Type">
> <title></title>
> </head>
> <body bgcolor="#ffffff" text="#000000">
> Jan,<br>
> <br>
> Using Asterisk for this is a lot like swatting flies with a Buick.
> Asterisk will do the job, shine your shoes, get you a latte (or coffee
> if you don't live on one of the coasts), and walk your dog.<br>
> <br>
> If you want wanted to build such an app in Perl, you could use the
> Asterisk AGI:<br>
> <a class="moz-txt-link-freetext"
href="http://www.voip-info.org/wiki-Asterisk+AGI">http://www.voip-info.org/w
iki-Asterisk+AGI</a><br>
> <br>
> You could probably even write a Dial Plan that would do the job.<br>
> <br>
> Thanks,<br>
> <br>
> David<br>
> <br>
> Jan Hulala wrote:
> <blockquote cite="mid019f01c48181$0e05ec50$6701a8c0 at HULALA" type="cite">
> <meta http-equiv="Content-Type" content="text/html; ">
> <meta content="MSHTML 6.00.2737.800" name="GENERATOR">
> <style></style>
> <div><font size="2">Hi folks,</font></div>
> <div> </div>
> <div><font size="2">I need simple application (the best in perl :),
> which will just listen for incoming calls, read callerID information.
> It should just ring or to answer as busy, depend on callerID number.
> How can I do this? Do I need to use Asterisk for this?</font></div>
> <div> </div>
> <div><font size="2">(I want to use a VoicePulse Connect for
this)</font></div>
> <div> </div>
> <div><font size="2">Any help will be very appreciate.</font></div>
> <div> </div>
> <div><font size="2">Thank you!</font></div>
> <div> </div>
> <div><font size="2">Jan</font></div>
> </blockquote>
> </body>
> </html>
>
> --------------090206070803090601060004--
>
> --------------030108030000000100020600
> Content-Type: text/x-vcard; charset=utf8;
> name="dpp-asterisk.vcf"
> Content-Transfer-Encoding: 7bit
> Content-Disposition: attachment;
> filename="dpp-asterisk.vcf"
>
> begin:vcard
> fn:David Pollak
> n:Pollak;David
> org:Projects In Motion
> adr:#204;;1032 Irving St;San Francisco;CA;94122;USA
> email;internet:dpp at projectsinmotion.com
> title:CEO
> tel;work:415-462-1504
> tel;fax:415-680-2437
> tel;cell:415-812-2394
> x-mozilla-html:TRUE
> url:http://www.projectsinmotion.com
> version:2.1
> end:vcard
>
>
> --------------030108030000000100020600--
>
>
> --__--__--
>
> Message: 3
> Date: Sat, 14 Aug 2004 00:53:02 +0200
> From: "Roland Zagler" <laureen at laureen.at>
> To: <asterisk-users at lists.digium.com>,
> <asterisk-dev at lists.digium.com>
> Subject: [Asterisk-Dev] HELP: BYE-request not sent to SIP-peer
> Reply-To: asterisk-dev at lists.digium.com
>
> Hello,
>
> When i have a "Hangup" in my dialplan (extensions.conf) the RFC says to
> terminate the session is to send a BYE request to UA. After tracing the
> traffic on port 5060 UDP i recognized that my asterisk is NOT sending a
> BYE request to it's peer, so the peer doen't know to end the session and
> continues to send RTP packages to me. Does anyone know how to fix this?
>
> Here's the complete trace from ngrep(make call, speak for 5 seconds,
> hangup): UDP port 5060 in all directions
>
>
> U [myIP]:5060 -> [peerIP]:5060
> INVITE sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
> [myIP]:5060;branch=3Dz9hG4bK4246930c..From: "423663098668" <sip:
> user at sip.provider.com>;tag=3Das10b2c259..To:
> <sip:011423663900828 at sip.provider.com>..Contact: <sip:user@[myIP]>..Call
> -ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 102
> INVITE..User-Agent: Grandstream..Date: Fri, 13 Aug 2004 21:57:57
> GMT..Allow: INVI
> TE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type:
> application/sdp..Content-Length: 184....v=3D0..o=3Droot 3608 3608 IN IP4
> [myIP]..s=3Dsessio
> n..c=3DIN IP4 [myIP]..t=3D0 0..m=3Daudio 19430 RTP/AVP 8 =
> 0..a=3Drtpmap:8
> PCMA/8000..a=3Drtpmap:0 PCMU/8000..a=3DsilenceSupp:off - - - -.. =
> =20
> #
> U [peerIP]:5060 -> [myIP]:5060
> SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
> [myIP]:5060;branch=3Dz9hG4bK4246930c..From:
> <sip:user at sip.provider.com>;tag=3Das10b2c25
> 9..To: <sip:011423663900828 at sip.provider.com>..CSeq: 102
> INVITE..Call-ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact:
> alfredko
> hl@[myIP]..WWW-Authenticate: Digest
> realm=3D"sip.provider.com",algorithm=3D"MD5",qop=3D"auth",nonce=3D"573FEF=
> AFEF25C
> B48",opaque=3D"901158A266D
> 481F7"..Max-Forwards: 70..Content-Length: 0....
>
> #
> U [myIP]:5060 -> [peerIP]:5060
> ACK sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
> [myIP]:5060;branch=3Dz9hG4bK4246930c..From: "423663098668" <sip:alf
> redkohl at sip.provider.com>;tag=3Das10b2c259..To:
> <sip:011423663900828 at sip.provider.com>..Contact:
> <sip:user@[myIP]>..Call-ID
> : 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 102 ACK..User-Agent:
> Grandstream..Content-Length: 0.... =20
> #
> U [myIP]:5060 -> [peerIP]:5060
> INVITE sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
> [myIP]:5060;branch=3Dz9hG4bK069df2d9..From: "423663098668" <sip:
> user at sip.provider.com>;tag=3Das10b2c259..To:
> <sip:011423663900828 at sip.provider.com>..Contact: <sip:user@[myIP]>..Call
> -ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 103
> INVITE..User-Agent: Grandstream..Authorization: Digest
> username=3D"user at s1.do
> uglastelecom.com", realm=3D"sip.provider.com", algorithm=3DMD5,
> uri=3D"user@[myIP]", nonce=3D"573FEFAFEF25CB48", response=3D"00e118ce8d2
> 72181311a762c91ea6cdc", opaque=3D"901158A266D481F7", qop=3D"auth",
> cnonce=3D"7de950c3", nc=3D00000001..Date: Fri, 13 Aug 2004 21:57:57
> GMT..Allow: INVIT
> E, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type:
> application/sdp..Content-Length: 184....v=3D0..o=3Droot 3608 3609 IN IP4
> [myIP]..s=3Dsession
> ..c=3DIN IP4 [myIP]..t=3D0 0..m=3Daudio 19430 RTP/AVP 8 =
> 0..a=3Drtpmap:8
> PCMA/8000..a=3Drtpmap:0 PCMU/8000..a=3DsilenceSupp:off - - - -..
>
> #
> U [peerIP]:5060 -> [myIP]:5060
> SIP/2.0 100 trying..Via: SIP/2.0/UDP
> [myIP]:5060;branch=3Dz9hG4bK069df2d9..To:
> <sip:011423663900828 at sip.provider.com>..From: <sip:alfr
> edkohl at sip.provider.com>;tag=3Das10b2c259..CSeq: 103 INVITE..Call-ID:
> 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62.
> 99.190.238..Max-Forwards: 70..Content-Length: 0....
>
> #
> U [peerIP]:5060 -> [myIP]:5060
> SIP/2.0 180 ringing..Via: SIP/2.0/UDP
> [myIP]:5060;branch=3Dz9hG4bK069df2d9..To:
> <sip:011423663900828 at sip.provider.com>..From: <sip:alf
> redkohl at sip.provider.com>;tag=3Das10b2c259..CSeq: 103 INVITE..Call-ID:
> 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62
> .99.190.238..Max-Forwards: 70..Content-Length: 0....
>
> #
> U [peerIP]:5060 -> [myIP]:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
> [myIP]:5060;branch=3Dz9hG4bK069df2d9..To:
> <sip:011423663900828 at sip.provider.com>..From: <sip:alfredko
> hl at sip.provider.com>;tag=3Das10b2c259..CSeq: 103 INVITE..Call-ID:
> 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62.99.1
> 90.238..Content-type: application/sdp..Max-Forwards:
> 70..Content-Length: 133....v=3D0..o=3Dnone 0 0 IN IP4 =
> [peerIP]..s=3D-..c=3DIN
> IP4 198.31.231.1
> 7..t=3D0 0..m=3Daudio 18691 RTP/AVP 8..a=3Drtpmap:8 =
> PCMA/8000..a=3Dptime:30..
>
>
>
> Thanxxxx
>
> Roland Zagler
> mailto:laureen at laureen.at
> @fog smart partners
>
> --__--__--
>
> Message: 4
> Date: Fri, 13 Aug 2004 23:13:13 -0500
> From: Josh Roberson <twisted at indigent-networks.com>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] HELP: BYE-request not sent to SIP-peer
> Reply-To: asterisk-dev at lists.digium.com
>
> Roland, It would have been nice to post a followup :P
>
> A few of us took a crack at this on IRC, and have decided that the real
> problem here are the contact headers being set by the provider, and * is
> not at fault at all, since we honored the contact headers. Many thanks
> to pfn, as he pointed out the providers headers were wrong, and asked
> Roland to set nat=yes, which solved the issue temproarily.
>
> -Josh (twisted)
>
> Roland Zagler wrote:
>
> >Hello,
> >
> >When i have a "Hangup" in my dialplan (extensions.conf) the RFC says to
> >terminate the session is to send a BYE request to UA. After tracing the
> >traffic on port 5060 UDP i recognized that my asterisk is NOT sending a
> >BYE request to it's peer, so the peer doen't know to end the session and
> >continues to send RTP packages to me. Does anyone know how to fix this?
> >
> >Here's the complete trace from ngrep(make call, speak for 5 seconds,
> >hangup): UDP port 5060 in all directions
> >
> >
> >U [myIP]:5060 -> [peerIP]:5060
> > INVITE sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
> >[myIP]:5060;branch=z9hG4bK4246930c..From: "423663098668" <sip:
> > user at sip.provider.com>;tag=as10b2c259..To:
> ><sip:011423663900828 at sip.provider.com>..Contact: <sip:user@[myIP]>..Call
> > -ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 102
> >INVITE..User-Agent: Grandstream..Date: Fri, 13 Aug 2004 21:57:57
> >GMT..Allow: INVI
> > TE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type:
> >application/sdp..Content-Length: 184....v=0..o=root 3608 3608 IN IP4
> >[myIP]..s=sessio
> > n..c=IN IP4 [myIP]..t=0 0..m=audio 19430 RTP/AVP 8 0..a=rtpmap:8
> >PCMA/8000..a=rtpmap:0 PCMU/8000..a=silenceSupp:off - - - -..
> >#
> >U [peerIP]:5060 -> [myIP]:5060
> > SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
> >[myIP]:5060;branch=z9hG4bK4246930c..From:
> ><sip:user at sip.provider.com>;tag=as10b2c25
> > 9..To: <sip:011423663900828 at sip.provider.com>..CSeq: 102
> >INVITE..Call-ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact:
> >alfredko
> > hl@[myIP]..WWW-Authenticate: Digest
> >realm="sip.provider.com",algorithm="MD5",qop="auth",nonce="573FEFAFEF25C
> >B48",opaque="901158A266D
> > 481F7"..Max-Forwards: 70..Content-Length: 0....
> >
> >#
> >U [myIP]:5060 -> [peerIP]:5060
> > ACK sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
> >[myIP]:5060;branch=z9hG4bK4246930c..From: "423663098668" <sip:alf
> > redkohl at sip.provider.com>;tag=as10b2c259..To:
> ><sip:011423663900828 at sip.provider.com>..Contact:
> ><sip:user@[myIP]>..Call-ID
> > : 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 102 ACK..User-Agent:
> >Grandstream..Content-Length: 0....
> >#
> >U [myIP]:5060 -> [peerIP]:5060
> > INVITE sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
> >[myIP]:5060;branch=z9hG4bK069df2d9..From: "423663098668" <sip:
> > user at sip.provider.com>;tag=as10b2c259..To:
> ><sip:011423663900828 at sip.provider.com>..Contact: <sip:user@[myIP]>..Call
> > -ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 103
> >INVITE..User-Agent: Grandstream..Authorization: Digest
> >username="user at s1.do
> > uglastelecom.com", realm="sip.provider.com", algorithm=MD5,
> >uri="user@[myIP]", nonce="573FEFAFEF25CB48", response="00e118ce8d2
> > 72181311a762c91ea6cdc", opaque="901158A266D481F7", qop="auth",
> >cnonce="7de950c3", nc=00000001..Date: Fri, 13 Aug 2004 21:57:57
> >GMT..Allow: INVIT
> > E, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type:
> >application/sdp..Content-Length: 184....v=0..o=root 3608 3609 IN IP4
> >[myIP]..s=session
> > ..c=IN IP4 [myIP]..t=0 0..m=audio 19430 RTP/AVP 8 0..a=rtpmap:8
> >PCMA/8000..a=rtpmap:0 PCMU/8000..a=silenceSupp:off - - - -..
> >
> >#
> >U [peerIP]:5060 -> [myIP]:5060
> > SIP/2.0 100 trying..Via: SIP/2.0/UDP
> >[myIP]:5060;branch=z9hG4bK069df2d9..To:
> ><sip:011423663900828 at sip.provider.com>..From: <sip:alfr
> > edkohl at sip.provider.com>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID:
> >6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62.
> > 99.190.238..Max-Forwards: 70..Content-Length: 0....
> >
> >#
> >U [peerIP]:5060 -> [myIP]:5060
> > SIP/2.0 180 ringing..Via: SIP/2.0/UDP
> >[myIP]:5060;branch=z9hG4bK069df2d9..To:
> ><sip:011423663900828 at sip.provider.com>..From: <sip:alf
> > redkohl at sip.provider.com>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID:
> >6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62
> > .99.190.238..Max-Forwards: 70..Content-Length: 0....
> >
> >#
> >U [peerIP]:5060 -> [myIP]:5060
> > SIP/2.0 200 OK..Via: SIP/2.0/UDP
> >[myIP]:5060;branch=z9hG4bK069df2d9..To:
> ><sip:011423663900828 at sip.provider.com>..From: <sip:alfredko
> > hl at sip.provider.com>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID:
> >6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62.99.1
> > 90.238..Content-type: application/sdp..Max-Forwards:
> >70..Content-Length: 133....v=0..o=none 0 0 IN IP4 [peerIP]..s=-..c=IN
> >IP4 198.31.231.1
> > 7..t=0 0..m=audio 18691 RTP/AVP 8..a=rtpmap:8 PCMA/8000..a=ptime:30..
> >
> >
> >
> >Thanxxxx
> >
> >Roland Zagler
> >mailto:laureen at laureen.at
> >@fog smart partners
> >
> >
>
>
> --__--__--
>
> Message: 5
> Date: Sat, 14 Aug 2004 01:53:19 -0700 (PDT)
> From: chaye wala <chayewala at yahoo.com>
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] What happened to #asterisk on irc.freenode.net
> Reply-To: asterisk-dev at lists.digium.com
>
> I am unable to get to #asterisk on irc.freenode.net
> for quite sometime now. Is it still available?
> Thanks.
>
>
>
> __________________________________
> Do you Yahoo!?
> Yahoo! Mail - 50x more storage than other providers!
> http://promotions.yahoo.com/new_mail
>
> --__--__--
>
> Message: 6
> Date: Sat, 14 Aug 2004 04:56:50 -0400
> From: Jeremy McNamara <jj at nufone.net>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] What happened to #asterisk on irc.freenode.net
> Reply-To: asterisk-dev at lists.digium.com
>
> chaye wala wrote:
>
> > I am unable to get to #asterisk on irc.freenode.net
> > for quite sometime now. Is it still available?
>
>
> Register your nick
>
>
> Jeremy McNamara
>
> --__--__--
>
> Message: 7
> From: "Rajeev" <rajeev at cefib.com>
> To: <asterisk-dev at lists.digium.com>
> Date: Sat, 14 Aug 2004 12:23:22 +0100
> Organization: CEFIB Internet (Outgoing)
> Subject: [Asterisk-Dev] ATA 186 connected phone is not ringing
> Reply-To: asterisk-dev at lists.digium.com
>
> This is a multi-part message in MIME format.
>
> ------=_NextPart_000_0384_01C481F9.7D4BDCA0
> Content-Type: text/plain;
> charset="iso-8859-1"
> Content-Transfer-Encoding: quoted-printable
>
> Hi,
>
> I am new to Linux and Asterisk. I have installed latest version of =
> Asterisk on a system running Redhat Linux 8.0. I have followed the =
> instructions from Andy Powell's getting started guide. The installation =
> of Asterisk was successful.
>
> I have installed X-lite softphone (Xten Networks) in Windows 2000 =
> professional and configured Cisco ATA 186 as the second extension. I =
> have disabled silence suppression in X-Lite. When dial from X-lite, =
> Asterisk is directly going to voice mail. But when i call from ATA186 =
> the softphone is ringing and upon attending the call it is getting =
> disconnected.(hung up)
>
> my configuration is as follows.
>
> sip.conf
>
> [general]
>
> port=3D5060
> bindaddr =3D 0.0.0.0
> context =3D bogon-calls
> allow=3Dall
>
> [2000]
> type=3Dfriend
> username=3D2000
> secret=3D2000
> host=3Ddynamic
> context=3Dfrom-sip
> mailbox=3D100
>
> [2001]
> type=3Dfriend
> username=3D2001
> secret=3D2001
> host=3Ddynamic
> context=3Dfrom-sip
> mailbox=3D101
>
>
> extension.conf
>
> static=3Dyes
> writeprotect=3Dyes
>
> [bogon-calls]
>
> exten=3D> _.,1,Congestion
>
> [from-sip]
>
> exten =3D> 2000,1,Dial(SIP/2000,20)
> exten =3D> 2000,2,Voicemail(u2000)
> exten =3D> 2000,102,Voicemail(b2000)
> exten =3D> 2000,103,Hangup
>
> exten =3D> 2001,1,Dial(SIP/2001,20)
> exten =3D> 2000,2,Voicemail(u2001)
> exten =3D> 2000,102,Voicemail(b2001)
> exten =3D> 2000,103,Hangup
>
> exten=3D> 2999,1,VoicemailMain(${CALLERIDNUM})
>
> voicemail.conf
>
> [general]
>
> format=3Dwav
>
> [local]
>
> 2000 =3D> 1234,rajeev,rajeev at xyz.com
> 2000 =3D> 4321,rajeev1,rajeev1 at xyz.com
>
> These configurations i have copied from ONLamp.com website. I am not =
> using any additional hardware for testing this setup.=20
> I have followed the instructions of Andy Powell in configuring the ATA. =
> If the host is mentioned as "dynamic " the ATA is not getting =
> registered. I am giving IP address now.=20
>
> The error messages are=20
>
> When calling from X-Lite
>
> rtp.c:275 process_rfc3389: RFC3389 support incomplete. Turn off on =
> client if possible.
> spawn extension (from-sip, 2000, 102) exited non-zero on 'SIP/2001-3c02'
>
> When calling from Cisco ATA
>
> spawn extension (from-sip, 2001, 1) exited non-zero on 'SIP/2000-bfed'
> Got SIP response 481 " Call Leg/Transaction Does Not Exist" back from =
> 192.168.1.136
>
>
> Thanks a lot for your time.
>
> Rajeevk
>
>
>
>
> ------=_NextPart_000_0384_01C481F9.7D4BDCA0
> Content-Type: text/html;
> charset="iso-8859-1"
> Content-Transfer-Encoding: quoted-printable
>
> <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
> <HTML><HEAD>
> <META content=3D"text/html; charset=3Diso-8859-1" =
> http-equiv=3DContent-Type>
> <META content=3D"MSHTML 5.00.3819.300" name=3DGENERATOR>
> <STYLE></STYLE>
> </HEAD>
> <BODY bgColor=3D#ffffff>
> <DIV><FONT face=3DArial size=3D2>
> <DIV><FONT face=3DArial size=3D2>
> <DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial size=3D2>I am new to Linux and Asterisk. I have =
> installed=20
> latest version of Asterisk on a system running Redhat Linux 8.0. I have =
> followed=20
> the instructions from Andy Powell's getting started guide. The =
> installation of=20
> Asterisk was successful.</FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial size=3D2>I have installed X-lite softphone (Xten =
> Networks)=20
> in Windows 2000 professional and configured Cisco ATA 186 as the second=20
> extension. I have disabled silence suppression in X-Lite. =
> When dial=20
> from X-lite, Asterisk is directly going to voice mail. But when i call =
> from=20
> ATA186 the softphone is ringing and upon attending the call it is =
> getting=20
> disconnected.(hung up)</FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial size=3D2>my configuration is as =
> follows.</FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial =
> size=3D2><U><STRONG>sip.conf</STRONG></U></FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial size=3D2>[general]</FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial size=3D2>port=3D5060</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>bindaddr =3D 0.0.0.0</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>context =3D =
> bogon-calls</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>allow=3Dall</FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial size=3D2>[2000]</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>type=3Dfriend</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>username=3D2000</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>secret=3D2000</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>host=3Ddynamic</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>context=3Dfrom-sip</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>mailbox=3D100</FONT></DIV>
> <DIV> </DIV>
> <DIV><FONT face=3DArial size=3D2>[2001]</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>
> <DIV><FONT face=3DArial size=3D2>type=3Dfriend</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>username=3D2001</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>secret=3D2001</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>host=3Ddynamic</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>context=3Dfrom-sip</FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>mailbox=3D101</FONT></DIV>
> <DIV> </DIV>
> <DIV> </DIV>
> <DIV><STRONG><U>extension.conf</U></STRONG></DIV>
> <DIV> </DIV>
> <DIV>static=3Dyes</DIV>
> <DIV>writeprotect=3Dyes</DIV>
> <DIV> </DIV>
> <DIV>[bogon-calls]</DIV>
> <DIV> </DIV>
> <DIV>exten=3D> _.,1,Congestion</DIV>
> <DIV> </DIV>
> <DIV>[from-sip]</DIV>
> <DIV> </DIV>
> <DIV>exten =3D> 2000,1,Dial(SIP/2000,20)</DIV>
> <DIV>exten =3D> 2000,2,Voicemail(u2000)</DIV>
> <DIV>exten =3D> 2000,102,Voicemail(b2000)</DIV>
> <DIV>exten =3D> 2000,103,Hangup</DIV>
> <DIV> </DIV>
> <DIV>
> <DIV>exten =3D> 2001,1,Dial(SIP/2001,20)</DIV>
> <DIV>exten =3D> 2000,2,Voicemail(u2001)</DIV>
> <DIV>exten =3D> 2000,102,Voicemail(b2001)</DIV>
> <DIV>exten =3D> 2000,103,Hangup</DIV></DIV>
> <DIV> </DIV>
> <DIV>exten=3D> 2999,1,VoicemailMain(${CALLERIDNUM})</DIV>
> <DIV> </DIV>
> <DIV><STRONG><U>voicemail.conf</U></STRONG></DIV>
> <DIV> </DIV>
> <DIV>[general]</DIV>
> <DIV> </DIV>
> <DIV>format=3Dwav</DIV>
> <DIV> </DIV>
> <DIV>[local]</DIV>
> <DIV> </DIV>
> <DIV>2000 =3D> <A=20
> href=3D"mailto:1234,rajeev,rajeev at xyz.com">1234,rajeev,rajeev at xyz.com</A>=
> </DIV>
> <DIV>2000 =3D> <A=20
> href=3D"mailto:4321,rajeev1,rajeev1 at xyz.com">4321,rajeev1,rajeev1 at xyz.com=
> </A></DIV>
> <DIV> </DIV>
> <DIV>These configurations i have copied from ONLamp.com website. I am =
> not using=20
> any additional hardware for testing this setup. </DIV>
> <DIV>I have followed the instructions of Andy Powell in configuring the =
> ATA. If=20
> the host is mentioned as "dynamic " the ATA is not getting registered. I =
> am=20
> giving IP address now. </DIV>
> <DIV> </DIV>
> <DIV>The error messages are </DIV>
> <DIV> </DIV>
> <DIV>When calling from X-Lite</DIV>
> <DIV> </DIV>
> <DIV>rtp.c:275 process_rfc3389: RFC3389 support incomplete. Turn off on =
> client=20
> if possible.</DIV>
> <DIV>spawn extension (from-sip, 2000, 102) exited non-zero on=20
> 'SIP/2001-3c02'</DIV>
> <DIV> </DIV>
> <DIV>When calling from Cisco ATA</DIV>
> <DIV> </DIV>
> <DIV>
> <DIV>spawn extension (from-sip, 2001, 1) exited non-zero on=20
> 'SIP/2000-bfed'</DIV>
> <DIV>Got SIP response 481 " Call Leg/Transaction Does Not Exist" back =
> from=20
> 192.168.1.136</DIV>
> <DIV> </DIV>
> <DIV> </DIV>
> <DIV>Thanks a lot for your time.</DIV>
> <DIV> </DIV>
> <DIV>Rajeevk</DIV></DIV>
> <DIV> </DIV>
> <DIV> </DIV>
> <DIV> </DIV></FONT></DIV></FONT></DIV></FONT></DIV></BODY></HTML>
>
> ------=_NextPart_000_0384_01C481F9.7D4BDCA0--
>
>
>
> --__--__--
>
> _______________________________________________
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> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
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