<DIV>Hi all,</DIV> <DIV>i am still challanged to get the SIP phones to work. this is the output of debug for one of the SIP phones. I called the other sip phone and collected the output. if you look through the debug post (below), you'll notice the bolded text "address incomplete", what is the cause of this sort of error? and how can one eliminate this problem?</DIV> <DIV>my * server: 10.47.200.136</DIV> <DIV>phone1: 10.47.200.137 (2173 is the number to dial to reach this phone)</DIV> <DIV>phone2: 10.47.200.141 (2172 is the number to dial to reach this phone)</DIV> <DIV> </DIV> <DIV>thank you in advance for any help.</DIV> <DIV>Ama</DIV> <DIV> </DIV> <DIV>---------------------------------------------------------------------------------------------------------------------</DIV> <DIV>--- (16 headers 24 lines)---<BR>Using INVITE request as basis
request - <A href="mailto:29df48c8e354007d15c53467f311ecf4@10.47.200.137">29df48c8e354007d15c53467f311ecf4@10.47.200.137</A><BR>Sending to 10.47.200.137 : 5060 (non-NAT)<BR>Found user 'stargate3'<BR>Found RTP audio format 0<BR>Found RTP audio format 18<BR>Found RTP audio format 96<BR>Found RTP audio format 102<BR>Found RTP audio format 107<BR>Found RTP audio format 104<BR>Found RTP audio format 105<BR>Found RTP audio format 106<BR>Found RTP audio format 4<BR>Found RTP audio format 97<BR>Found RTP audio format 98<BR>Found RTP audio format 2<BR>Found RTP audio format 99<BR>Found RTP audio format 8<BR>Found RTP audio format 101<BR>Peer audio RTP is at port 10.47.200.137:3000<BR>Found description format PCMU<BR>Found description format G729<BR>Found description format BV16<BR>Found description format BV32<BR>Found description format L16<BR>Found description format PCMU<BR>Found description format PCMA<BR>Found description format L16<BR>Found description format G723<BR>Found description
format G726-16<BR>Found description format G726-24<BR>Found description format G726-32<BR>Found description format G726-40<BR>Found description format PCMA<BR>Found description format telephone-event<BR>Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x55d (g723|ulaw|alaw|g726|slin|g729|ilbc)/video=0x0 (nothing), combined - 0x40c (ulaw|alaw|ilbc)<BR>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<BR>Looking for 2172 in default (domain 10.47.200.136)<BR>Reliably Transmitting (no NAT) to 10.47.200.137:5060:<BR><STRONG>SIP/2.0 484 Address Incomplete<BR></STRONG>Via: SIP/2.0/UDP 10.47.200.137;branch=z9hG4bKaa5252205;received=10.47.200.137<BR>From: stargate3 <sip:stargate3@10.47.200.136:5060>;tag=95f353525026b9d<BR>To: 2172 <sip:2172@10.47.200.136:5060>;tag=as529cb545<BR>Call-ID: <A
href="mailto:29df48c8e354007d15c53467f311ecf4@10.47.200.137">29df48c8e354007d15c53467f311ecf4@10.47.200.137</A><BR>CSeq: 775968348 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>Max-Forwards: 70<BR>Contact: <sip:2172@10.47.200.136><BR>Content-Length: 0</DIV> <DIV><BR>---<BR>VoIP-*CLI> <BR><-- SIP read from 10.47.200.137:5060: <BR>ACK sip:2172@10.47.200.136:5060 SIP/2.0<BR>Via: SIP/2.0/UDP 10.47.200.137;branch=z9hG4bKaa5252205<BR>Max-Forwards: 70<BR>Content-Length: 0<BR>To: 2172 <sip:2172@10.47.200.136:5060>;tag=as529cb545<BR>From: stargate3 <sip:stargate3@10.47.200.136:5060>;tag=95f353525026b9d<BR>Call-ID: <A href="mailto:29df48c8e354007d15c53467f311ecf4@10.47.200.137">29df48c8e354007d15c53467f311ecf4@10.47.200.137</A><BR>CSeq: 775968348 ACK<BR>Proxy-Authorization:Digest
response="a94c49745a7cde16ebf111a12426493c",username="stargate3",realm="asterisk",nonce="366925e5",uri="sip:2172@10.47.200.136:5060"<BR>User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26</DIV> <DIV>---------------------------------------------------------------------------------------------------------------------------<BR><BR><B><I>abc def <xisterisk@yahoo.com></I></B> wrote:</DIV> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"> <DIV>not sure but this is the output from the pbx:</DIV> <DIV> </DIV> <DIV>>sip show registry<BR>Host Username Refresh State
<BR>local_sip:5060 stargate3 105 Registered <BR>local_sip:5060 stargate2 105 Registered <BR>local_sip:5060 stargate1 105 Registered <BR></DIV> <DIV>from sip phone I can any other phone (cisco with sccp or iax protocol) but I can't call any other sip phone, or receive phone calls.</DIV> <DIV><BR><B><I>Facundo Ameal <fameal@gmail.com></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">are you sure your sip phone is registering ok?<BR><BR>2006/2/1, abc def <XISTERISK@YAHOO.COM>:<BR>> Thanks Facundo for instruction but it didn't work. there is nothing new in<BR>> your suggestion compare to my conf files nevertheless I tried it but it<BR>> didn't work. I can make call from my sip phone but can't receive any phone<BR>> call. I am sure some one had had the same problem an d solved it.<BR>> as always I appreciate for your suggestion, advice and/or correction to my<BR>> config files.<BR>> if you know how to solve this problem please give me some hint.<BR>><BR>> thank you<BR>><BR>> Facundo Ameal <FAMEAL@GMAIL.COM>wrote:<BR>> i've tested it with this config files and i worked:<BR>><BR>> extensions.conf<BR>><BR>> exten => 55,1,Dial(SIP/2271,20)<BR>><BR>><BR>> sip.conf<BR>><BR>> [2271]<BR>>
type=friend<BR>> host=dynamic<BR>> secret=sip<BR>> allow=all<BR>> qualify=200<BR>> nat=no<BR>><BR>><BR>> Instead of 2271 you can put whatever you want.<BR>><BR>> good luck.<BR>><BR>><BR>><BR>> 2006/1/31, Facundo Ameal :<BR>> > Are you using a SIP Softphone or an ATA?<BR>> ><BR>> > 2006/1/31, Facundo Ameal :<BR>> > > does it registers well?<BR>> > > although i think you have to add "context=default" to the stargate1<BR>> section.<BR>> ; > ><BR>> > > try that and see what happens.<BR>> > ><BR>> > > 2006/1/31, abc def :<BR>> > > > Hi all, I am resending this message, so far no one has helped me with<BR>> this<BR>> > > > incoming call issue. there is no problem with outbound call but there<BR>> is no<BR>> > > > inbound call to my sip phone. the only message I get when I call from<BR>> pstn<BR>> > > > is "unable to
create local channel for call forward to<BR>> > > > 'Local/sipphone@default' (case =0)". my configuration files are<BR>> attached<BR>> > > > below. any help would be greatly appreciated. many thanks in advance.<BR>> > > > ABC<BR>> > > ><BR>> > > > abc def wrote:<BR>> > > ><BR>> > > > there is no error message coming up on the pbx for in-bound calls<BR>> (there is<BR>> > > > only debugging messages for outbound calls).<BR>> > > ><BR>> > > > thanks in advance for any hint or suggestion.<BR>> > > > Ama<BR>> > > ><BR>> > > > I just post my configuration file here for sip phone:<BR>> > > > extensions.conf<BR>> > > ><BR>> -------------------------------------------------------------------------<BR>> > > > [globals]<BR>> > > > [default]<BR>> > > > include =>
incoming<BR>> > > > include => outgoing<BR>> > > > include => iax<BR>> > > > inculde => sip<BR>> > > > include => sccp<BR>> > > > [sip]<BR>> > > > exten => 2171,1,Dial(SIP/stargate1,20)<BR>> > > > ;exten => 2171,1,Dial(SIP/2171,20)<BR>> > > > exten => 2171,2,Hangup<BR>> & gt; > > exten => 2172,1,Dial(SIP/stargate2,20)<BR>><BR>> > > > ;exten => 2172,1,Dial(SIP/2172,20)<BR>> > > > exten => 2172,2,Hangup<BR>> > > > exten => 2173,1,Dial(SIP/stargate3,20)<BR>> > > > ;exten => 2173,1,Dial(SIP/2173,20)<BR>> > > > exten => 2173,2,Hangup<BR>> > > > [sccp]<BR>> > > > [skinny]<BR>> > > > [incoming]<BR>> > > > exten => ; _214943[5-9]6,1,Dial(SIP/stargate3)<BR>> > > > exten => _214943[5-9]6,2,Hangup<BR>> > > >
[outgoing]<BR>> > > > exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN})<BR>> > > > exten => _XXXXXXXX,2,Hangup<BR>> > > ><BR>> -------------------------------------------------------------------------<BR>> > > > sip.conf<BR>> > > ><BR>> -------------------------------------------------------------------------<BR>> > > > [general]<BR>> > > > context=default ; Default context for incoming calls<BR>&g t; > > > ; Set this to your host name or domain name<BR>> > > > bindport=5060 ; UDP Port to bind to (SIP standard port is<BR>> > > > 5060)<BR>> > > > bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to<BR>> > > > all)<BR>> > > > srvlookup=yes ; Enable DNS SRV lookups on outbound calls<BR>> > > ><BR>> > > > register => stargate1:1stargate@local_sip/2171<BR>> > > > register =>
stargate2:2stargate@local_sip/2172<BR>> > > > register => stargate3:3stargate@local_sip/2173<BR>> > > > ;---------------------------------------------- NAT<BR>> SUPPORT<BR>> > > > ------------------------<BR>> > > > nat=no ; Global NAT settings (Affects all peers and<BR>> > > > users)<BR>> > > ><BR>> > > ><BR>> > > > [local_sip]<BR>> > > > type=friend<BR>> > > > host=10.47.200.136<BR>> > > > context=default<BR>> > > > [stargate1] ;cisco 9760<BR>> > > > ;[2171]<BR>> > > > ; type=friend<BR>><BR>> > > > host=dynamic ;10.47.200.140 ;dynamic<BR>> > > > defaultip=10.47.200.140<BR>> > > > username=stargate1<BR>> > > > secret=xxx<BR>> > > > callerid="21495071" <2171><BR>> > > > allow=all<BR>> > > > qualify=200<BR>> > >
> nat=no<BR>> > > > defaultip=10.47.200.140<BR>> > > ><BR>> > > > [stargate2] ;Polycom 601<BR>> > > > ;[2172]<BR>> > > > type=friend<BR>> > > > host=dynamic ;10.47.200.141 ;dynamic<BR>> > > > defaultip=10.47.200.141<BR>> > > > username=xxx<BR>> > > > secret=2stargate<BR>> > > > callerid="21495072" <2172><BR>> > > > allow=all<BR>> > &g t; > qualify=200<BR>> > > > nat=no<BR>> > > > defaultip=10.47.200.141<BR>> > > > [stargate3] ;Aastra 480i<BR>> > > > ;[2173]<BR>> > > > type=friend<BR>> > > > host=dynamic ;10.47.200.137 ;dynamic<BR>> > > > defaultip=10.47.200.137<BR>> > > > username=stargate3<BR>> > > > callerid="starg ate3" <2173><BR>> > > > secret=xxx<BR>> > > > allow=all<BR>> > > >
qualify=200<BR>> > > > nat=no<BR>> > > > defaultip=10.47.200.137<BR>> > > ><BR>> ----------------------------------------------------------------------------<BR>> > > ><BR>> > > ><BR>> > > > pdhales@optusnet.com.au wrote:<BR>> > > ><BR>> > > > What error do you get when trying to call the SIP phones?<BR>> > > ><BR>> > > > PaulH<BR>> > > ><BR>> > > ><BR>> > > > ----- Original Message -----<BR>> > > > From: abc def<BR>> > > > To: asterisk-users@lists.digium.com<BR>> > > > Sent: Wednesday, January 25, 2006 11:58 PM<BR>> > > > Subject: [Asterisk-Users] Help with sip setup because can't receive<BR>> calls<BR>> > > ><BR>> > > ><BR>> > > ><BR>> > > > Hi all,<BR>> > > > I read many posts on asterisk mail site and been trying
many<BR>> different<BR>> > > > things but still I can't get my sip phones to work with asterisk.<BR>> > > > I have a full blown-up voip netwok with two asterisk servers connected<BR>> > > > to pstn network with iax phones and cisco sccp phones which all work<BR>> fine.<BR>> > > > however, I have been struggeling to configure my sip phones (polycom<BR>> 601,<BR>> > > > Aastra 480i and cisco 9760) to work with aster isk. I can call out from<BR>> sip<BR>> > > > phones to anywhere else but not receive phone calls. I can see the<BR>> phones on<BR>> > > > "sip show registry" and "sip show peers" but no track phone calls for<BR>> sip.<BR>> > > ><BR>> > > > can you please shed some light on me how to go about solving this<BR>> > > > problem?<BR>> > > ><BR>> > > > thank you and best regards,<BR>> > > > Ama<BR>> >
> ><BR>> > > > < HR SIZE=1> Do you Yahoo!?<BR>> > > > With a free 1 GB, there's more in store with Yahoo! Mail.<BR>> > > > ________________________________<BR>> > > ><BR>> > > > _______________________________________________<BR>> > > > --Bandwidth and Colocation provided by Easynews.com --<BR>> > > ><BR>> > > > Asterisk-Users mailing list<BR>> > > > To UNSUBSCRIBE or update options visit:<BR>> > > ><BR>> > > ><BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > > > _______________________________________________<BR>> > > > --Bandwidth and Colocation provided by Easynews.com --<BR>> > > ><BR>> > > > Asterisk-Users mailing list<BR>> > > > To UNSUBSCRIBE or update options visit:<BR>> > > ><BR>>
http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > > ><BR>> > > ><BR>> > > ><BR>> > > > ________________________________<BR>> > > > Bring words and photos to gether (easily) with<BR>> > > > PhotoMail - it's free and works with Yahoo!<BR>> > > > Mail._______________________________________________<BR>> > > > --Bandwidth and Colocation provided by Easynews.com --<BR>> > > ><BR>> > > > Asterisk-Users mailing list<BR>> > > > To UNSUBSCRIBE or update options visit:<BR>> > > ><BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > > ><BR>> > > ><BR>> > > ><BR>> > > ><BR>> > > > ________________________________<BR>> > > > Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more<BR>> on new<BR>> > > > and used
cars.<BR>> > > ><BR>> > > ><BR>> > > > _______________________________________________<BR>> > > > --Bandwidth and Colocation provided by Easynews.com --<BR>> > > ><BR>> > > > Asterisk-Users mailing list<BR>> > > > To UNSUBSCRIBE or update options visit:<BR>> > > ><BR>> & gt; > ><BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> > > ><BR>> > > ><BR>> > > ><BR>> > ><BR>> > ><BR>> > > --<BR>> > > Facundo Ameal.<BR>> > > famealgmailcom<BR>> > > Linux User #395088<BR>> > ><BR>> > > FWD: 741664<BR>> > > MSN: asadolamorcillacomar<BR>> > > ICQ: 74005793<BR>> > ><BR>> > ><BR>> > > Open your mind, use open source.<BR>> > ><BR>> ><BR>> ><BR>> > --<BR>> > Facundo Ameal.<BR>> >
famealgmailcom<BR>> > Linux User #395088<BR>> ><BR>> > FWD: 741664<BR>> > MSN: asadolamorcillacomar<BR>> > ICQ: 74005793<BR>> ><BR>> ><BR>> > Open your mind, use open source.<BR>> ><BR>><BR>><BR>> --<BR>> Facundo Ameal.<BR>> famealgmailcom<BR>> Linux User #395088<BR>><BR>> FWD: 741664<BR>> MSN: asadolamorcillacomar<BR>> ICQ: 74005793<BR>><BR>><BR>> Open your mind, use open source.<BR>> _______________________________________________<BR>> --Bandwidth and Colocation provided by Easynews.com --<BR>><BR>> Asterisk-Users mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>><BR>><BR>><BR>><BR>> ________________________________<BR>><BR>> What are the most popular cars? Find out at Yahoo! Autos<BR>><BR>><BR>> _______________________________________________<BR>> --Bandwidth and
Colocation provided by Easynews.com --<BR>><BR>> Asterisk-Users mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>><BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR>><BR>><BR>><BR><BR><BR>--<BR>Facundo Ameal.<BR>fameal<AT>gmail<DOT>com<BR>Linux User #395088<BR><BR>FWD: 741664<BR>MSN: asado<AT>lamorcilla<DOT>com<DOT>ar<BR>ICQ: 74005793<BR><BR><BR>Open your mind, use open source.<BR>_______________________________________________<BR>--Bandwidth and Colocation provided by Easynews.com --<BR><BR>Asterisk-Users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE> <DIV><BR></DIV> <div> <HR SIZE=1> <A href="http://us.rd.yahoo.com/evt=38381/%20ylc=X3oDMTEzcGlrdGY5BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDMWF1dG9z/*http://autos.yahoo.com/index.html">Yahoo! Autos</A>. Looking for a sweet ride? Get pricing, reviews, & more on new and used
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<hr size=1> <a href="http://us.rd.yahoo.com/evt=38381/ ylc=X3oDMTEzcGlrdGY5BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDMWF1dG9z/*http://autos.yahoo.com/index.html ">Yahoo! Autos</a>. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.<p>
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