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: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, August 14, 2004 1:41 PM Subject: Asterisk-Dev digest, Vol 1 #835 - 7 msgs > Send Asterisk-Dev mailing list submissions to > [EMAIL PROTECTED] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-dev > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Dev digest..." > > > 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Date: Fri, 13 Aug 2004 18:01:14 -0400 > Organization: XODATEL > Subject: [Asterisk-Dev] IAX/SIP listener ??? > Reply-To: [EMAIL PROTECTED] > > 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 <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Dev] IAX/SIP listener ??? > Reply-To: [EMAIL PROTECTED] > > 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="[EMAIL PROTECTED]" 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:[EMAIL PROTECTED] > 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]>, > <[EMAIL PROTECTED]> > Subject: [Asterisk-Dev] HELP: BYE-request not sent to SIP-peer > Reply-To: [EMAIL PROTECTED] > > 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:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP > [myIP]:5060;branch=3Dz9hG4bK4246930c..From: "423663098668" <sip: > [EMAIL PROTECTED]>;tag=3Das10b2c259..To: > <sip:[EMAIL PROTECTED]>..Contact: <sip:[EMAIL PROTECTED]>..Call > -ID: [EMAIL PROTECTED]: 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:[EMAIL PROTECTED]>;tag=3Das10b2c25 > 9..To: <sip:[EMAIL PROTECTED]>..CSeq: 102 > INVITE..Call-ID: [EMAIL PROTECTED]: > alfredko > [EMAIL PROTECTED]: 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:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP > [myIP]:5060;branch=3Dz9hG4bK4246930c..From: "423663098668" <sip:alf > [EMAIL PROTECTED]>;tag=3Das10b2c259..To: > <sip:[EMAIL PROTECTED]>..Contact: > <sip:[EMAIL PROTECTED]>..Call-ID > : [EMAIL PROTECTED]: 102 ACK..User-Agent: > Grandstream..Content-Length: 0.... =20 > # > U [myIP]:5060 -> [peerIP]:5060 > INVITE sip:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP > [myIP]:5060;branch=3Dz9hG4bK069df2d9..From: "423663098668" <sip: > [EMAIL PROTECTED]>;tag=3Das10b2c259..To: > <sip:[EMAIL PROTECTED]>..Contact: <sip:[EMAIL PROTECTED]>..Call > -ID: [EMAIL PROTECTED]: 103 > INVITE..User-Agent: Grandstream..Authorization: Digest > username=3D"[EMAIL PROTECTED] > uglastelecom.com", realm=3D"sip.provider.com", algorithm=3DMD5, > uri=3D"[EMAIL PROTECTED]", 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:[EMAIL PROTECTED]>..From: <sip:alfr > [EMAIL PROTECTED]>;tag=3Das10b2c259..CSeq: 103 INVITE..Call-ID: > [EMAIL PROTECTED]: [EMAIL PROTECTED] > 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:[EMAIL PROTECTED]>..From: <sip:alf > [EMAIL PROTECTED]>;tag=3Das10b2c259..CSeq: 103 INVITE..Call-ID: > [EMAIL PROTECTED]: [EMAIL PROTECTED] > .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:[EMAIL PROTECTED]>..From: <sip:alfredko > [EMAIL PROTECTED]>;tag=3Das10b2c259..CSeq: 103 INVITE..Call-ID: > [EMAIL PROTECTED]: [EMAIL PROTECTED] > 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:[EMAIL PROTECTED] > @fog smart partners > > --__--__-- > > Message: 4 > Date: Fri, 13 Aug 2004 23:13:13 -0500 > From: Josh Roberson <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Dev] HELP: BYE-request not sent to SIP-peer > Reply-To: [EMAIL PROTECTED] > > 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:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP > >[myIP]:5060;branch=z9hG4bK4246930c..From: "423663098668" <sip: > > [EMAIL PROTECTED]>;tag=as10b2c259..To: > ><sip:[EMAIL PROTECTED]>..Contact: <sip:[EMAIL PROTECTED]>..Call > > -ID: [EMAIL PROTECTED]: 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:[EMAIL PROTECTED]>;tag=as10b2c25 > > 9..To: <sip:[EMAIL PROTECTED]>..CSeq: 102 > >INVITE..Call-ID: [EMAIL PROTECTED]: > >alfredko > > [EMAIL PROTECTED]: 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:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP > >[myIP]:5060;branch=z9hG4bK4246930c..From: "423663098668" <sip:alf > > [EMAIL PROTECTED]>;tag=as10b2c259..To: > ><sip:[EMAIL PROTECTED]>..Contact: > ><sip:[EMAIL PROTECTED]>..Call-ID > > : [EMAIL PROTECTED]: 102 ACK..User-Agent: > >Grandstream..Content-Length: 0.... > ># > >U [myIP]:5060 -> [peerIP]:5060 > > INVITE sip:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP > >[myIP]:5060;branch=z9hG4bK069df2d9..From: "423663098668" <sip: > > [EMAIL PROTECTED]>;tag=as10b2c259..To: > ><sip:[EMAIL PROTECTED]>..Contact: <sip:[EMAIL PROTECTED]>..Call > > -ID: [EMAIL PROTECTED]: 103 > >INVITE..User-Agent: Grandstream..Authorization: Digest > >username="[EMAIL PROTECTED] > > uglastelecom.com", realm="sip.provider.com", algorithm=MD5, > >uri="[EMAIL PROTECTED]", 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:[EMAIL PROTECTED]>..From: <sip:alfr > > [EMAIL PROTECTED]>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID: > >[EMAIL PROTECTED]: [EMAIL PROTECTED] > > 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:[EMAIL PROTECTED]>..From: <sip:alf > > [EMAIL PROTECTED]>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID: > >[EMAIL PROTECTED]: [EMAIL PROTECTED] > > .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:[EMAIL PROTECTED]>..From: <sip:alfredko > > [EMAIL PROTECTED]>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID: > >[EMAIL PROTECTED]: [EMAIL PROTECTED] > > 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:[EMAIL PROTECTED] > >@fog smart partners > > > > > > > --__--__-- > > Message: 5 > Date: Sat, 14 Aug 2004 01:53:19 -0700 (PDT) > From: chaye wala <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: [Asterisk-Dev] What happened to #asterisk on irc.freenode.net > Reply-To: [EMAIL PROTECTED] > > 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 <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Dev] What happened to #asterisk on irc.freenode.net > Reply-To: [EMAIL PROTECTED] > > 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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: [EMAIL PROTECTED] > > 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,[EMAIL PROTECTED] > 2000 =3D> 4321,rajeev1,[EMAIL PROTECTED] > > 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,[EMAIL PROTECTED]">1234,rajeev,[EMAIL PROTECTED]</A>= > </DIV> > <DIV>2000 =3D> <A=20 > href=3D"mailto:4321,rajeev1,[EMAIL PROTECTED]">4321,rajeev1,[EMAIL PROTECTED] > </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-- > > > > --__--__-- > > _______________________________________________ > Asterisk-Dev mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > End of Asterisk-Dev Digest > _______________________________________________ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
