[Asterisk-Users] Help with sip setup because can't receive calls!!!!!!

abc def xisterisk at yahoo.com
Thu Feb 2 05:00:08 MST 2006


Hi all,
  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?
  my * server:  10.47.200.136
  phone1:        10.47.200.137   (2173 is the number to dial to reach this phone)
  phone2:        10.47.200.141   (2172 is the number to dial to reach this phone)
   
  thank you in advance for any help.
  Ama
   
  ---------------------------------------------------------------------------------------------------------------------
  --- (16 headers 24 lines)---
Using INVITE request as basis request - 29df48c8e354007d15c53467f311ecf4 at 10.47.200.137
Sending to 10.47.200.137 : 5060 (non-NAT)
Found user 'stargate3'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 102
Found RTP audio format 107
Found RTP audio format 104
Found RTP audio format 105
Found RTP audio format 106
Found RTP audio format 4
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 2
Found RTP audio format 99
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.47.200.137:3000
Found description format PCMU
Found description format G729
Found description format BV16
Found description format BV32
Found description format L16
Found description format PCMU
Found description format PCMA
Found description format L16
Found description format G723
Found description format G726-16
Found description format G726-24
Found description format G726-32
Found description format G726-40
Found description format PCMA
Found description format telephone-event
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)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 2172 in default (domain 10.47.200.136)
Reliably Transmitting (no NAT) to 10.47.200.137:5060:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.47.200.137;branch=z9hG4bKaa5252205;received=10.47.200.137
From: stargate3 <sip:stargate3 at 10.47.200.136:5060>;tag=95f353525026b9d
To: 2172 <sip:2172 at 10.47.200.136:5060>;tag=as529cb545
Call-ID: 29df48c8e354007d15c53467f311ecf4 at 10.47.200.137
CSeq: 775968348 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:2172 at 10.47.200.136>
Content-Length: 0
  
---
VoIP-*CLI> 
<-- SIP read from 10.47.200.137:5060: 
ACK sip:2172 at 10.47.200.136:5060 SIP/2.0
Via: SIP/2.0/UDP 10.47.200.137;branch=z9hG4bKaa5252205
Max-Forwards: 70
Content-Length: 0
To: 2172 <sip:2172 at 10.47.200.136:5060>;tag=as529cb545
From: stargate3 <sip:stargate3 at 10.47.200.136:5060>;tag=95f353525026b9d
Call-ID: 29df48c8e354007d15c53467f311ecf4 at 10.47.200.137
CSeq: 775968348 ACK
Proxy-Authorization:Digest response="a94c49745a7cde16ebf111a12426493c",username="stargate3",realm="asterisk",nonce="366925e5",uri="sip:2172 at 10.47.200.136:5060"
User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
  ---------------------------------------------------------------------------------------------------------------------------

abc def <xisterisk at yahoo.com> wrote:
    not sure but this is the output from the pbx:
   
  >sip show registry
Host                            Username       Refresh State               
local_sip:5060                  stargate3          105 Registered          
local_sip:5060                  stargate2          105 Registered          
local_sip:5060                  stargate1          105 Registered          

  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.
  
Facundo Ameal <fameal at gmail.com> wrote:
  are you sure your sip phone is registering ok?

2006/2/1, abc def :
> Thanks Facundo for instruction but it didn't work. there is nothing new in
> your suggestion compare to my conf files nevertheless I tried it but it
> didn't work. I can make call from my sip phone but can't receive any phone
> call. I am sure some one had had the same problem an d solved it.
> as always I appreciate for your suggestion, advice and/or correction to my
> config files.
> if you know how to solve this problem please give me some hint.
>
> thank you
>
> Facundo Ameal wrote:
> i've tested it with this config files and i worked:
>
> extensions.conf
>
> exten => 55,1,Dial(SIP/2271,20)
>
>
> sip.conf
>
> [2271]
> type=friend
> host=dynamic
> secret=sip
> allow=all
> qualify=200
> nat=no
>
>
> Instead of 2271 you can put whatever you want.
>
> good luck.
>
>
>
> 2006/1/31, Facundo Ameal :
> > Are you using a SIP Softphone or an ATA?
> >
> > 2006/1/31, Facundo Ameal :
> > > does it registers well?
> > > although i think you have to add "context=default" to the stargate1
> section.
> ; > >
> > > try that and see what happens.
> > >
> > > 2006/1/31, abc def :
> > > > Hi all, I am resending this message, so far no one has helped me with
> this
> > > > incoming call issue. there is no problem with outbound call but there
> is no
> > > > inbound call to my sip phone. the only message I get when I call from
> pstn
> > > > is "unable to create local channel for call forward to
> > > > 'Local/sipphone at default' (case =0)". my configuration files are
> attached
> > > > below. any help would be greatly appreciated. many thanks in advance.
> > > > ABC
> > > >
> > > > abc def wrote:
> > > >
> > > > there is no error message coming up on the pbx for in-bound calls
> (there is
> > > > only debugging messages for outbound calls).
> > > >
> > > > thanks in advance for any hint or suggestion.
> > > > Ama
> > > >
> > > > I just post my configuration file here for sip phone:
> > > > extensions.conf
> > > >
> -------------------------------------------------------------------------
> > > > [globals]
> > > > [default]
> > > > include => incoming
> > > > include => outgoing
> > > > include => iax
> > > > inculde => sip
> > > > include => sccp
> > > > [sip]
> > > > exten => 2171,1,Dial(SIP/stargate1,20)
> > > > ;exten => 2171,1,Dial(SIP/2171,20)
> > > > exten => 2171,2,Hangup
> & gt; > > exten => 2172,1,Dial(SIP/stargate2,20)
>
> > > > ;exten => 2172,1,Dial(SIP/2172,20)
> > > > exten => 2172,2,Hangup
> > > > exten => 2173,1,Dial(SIP/stargate3,20)
> > > > ;exten => 2173,1,Dial(SIP/2173,20)
> > > > exten => 2173,2,Hangup
> > > > [sccp]
> > > > [skinny]
> > > > [incoming]
> > > > exten => ; _214943[5-9]6,1,Dial(SIP/stargate3)
> > > > exten => _214943[5-9]6,2,Hangup
> > > > [outgoing]
> > > > exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN})
> > > > exten => _XXXXXXXX,2,Hangup
> > > >
> -------------------------------------------------------------------------
> > > > sip.conf
> > > >
> -------------------------------------------------------------------------
> > > > [general]
> > > > context=default ; Default context for incoming calls
&g t; > > > ; Set this to your host name or domain name
> > > > bindport=5060 ; UDP Port to bind to (SIP standard port is
> > > > 5060)
> > > > bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to
> > > > all)
> > > > srvlookup=yes ; Enable DNS SRV lookups on outbound calls
> > > >
> > > > register => stargate1:1stargate at local_sip/2171
> > > > register => stargate2:2stargate at local_sip/2172
> > > > register => stargate3:3stargate at local_sip/2173
> > > > ;---------------------------------------------- NAT
> SUPPORT
> > > > ------------------------
> > > > nat=no ; Global NAT settings (Affects all peers and
> > > > users)
> > > >
> > > >
> > > > [local_sip]
> > > > type=friend
> > > > host=10.47.200.136
> > > > context=default
> > > > [stargate1] ;cisco 9760
> > > > ;[2171]
> > > > ; type=friend
>
> > > > host=dynamic ;10.47.200.140 ;dynamic
> > > > defaultip=10.47.200.140
> > > > username=stargate1
> > > > secret=xxx
> > > > callerid="21495071" <2171>
> > > > allow=all
> > > > qualify=200
> > > > nat=no
> > > > defaultip=10.47.200.140
> > > >
> > > > [stargate2] ;Polycom 601
> > > > ;[2172]
> > > > type=friend
> > > > host=dynamic ;10.47.200.141 ;dynamic
> > > > defaultip=10.47.200.141
> > > > username=xxx
> > > > secret=2stargate
> > > > callerid="21495072" <2172>
> > > > allow=all
> > &g t; > qualify=200
> > > > nat=no
> > > > defaultip=10.47.200.141
> > > > [stargate3] ;Aastra 480i
> > > > ;[2173]
> > > > type=friend
> > > > host=dynamic ;10.47.200.137 ;dynamic
> > > > defaultip=10.47.200.137
> > > > username=stargate3
> > > > callerid="starg ate3" <2173>
> > > > secret=xxx
> > > > allow=all
> > > > qualify=200
> > > > nat=no
> > > > defaultip=10.47.200.137
> > > >
> ----------------------------------------------------------------------------
> > > >
> > > >
> > > > pdhales at optusnet.com.au wrote:
> > > >
> > > > What error do you get when trying to call the SIP phones?
> > > >
> > > > PaulH
> > > >
> > > >
> > > > ----- Original Message -----
> > > > From: abc def
> > > > To: asterisk-users at lists.digium.com
> > > > Sent: Wednesday, January 25, 2006 11:58 PM
> > > > Subject: [Asterisk-Users] Help with sip setup because can't receive
> calls
> > > >
> > > >
> > > >
> > > > Hi all,
> > > > I read many posts on asterisk mail site and been trying many
> different
> > > > things but still I can't get my sip phones to work with asterisk.
> > > > I have a full blown-up voip netwok with two asterisk servers connected
> > > > to pstn network with iax phones and cisco sccp phones which all work
> fine.
> > > > however, I have been struggeling to configure my sip phones (polycom
> 601,
> > > > Aastra 480i and cisco 9760) to work with aster isk. I can call out from
> sip
> > > > phones to anywhere else but not receive phone calls. I can see the
> phones on
> > > > "sip show registry" and "sip show peers" but no track phone calls for
> sip.
> > > >
> > > > can you please shed some light on me how to go about solving this
> > > > problem?
> > > >
> > > > thank you and best regards,
> > > > Ama
> > > >
> > > > < HR SIZE=1> Do you Yahoo!?
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> > > >
> > > >
> > >
> > >
> > > --
> > > Facundo Ameal.
> > > famealgmailcom
> > > Linux User #395088
> > >
> > > FWD: 741664
> > > MSN: asadolamorcillacomar
> > > ICQ: 74005793
> > >
> > >
> > > Open your mind, use open source.
> > >
> >
> >
> > --
> > Facundo Ameal.
> > famealgmailcom
> > Linux User #395088
> >
> > FWD: 741664
> > MSN: asadolamorcillacomar
> > ICQ: 74005793
> >
> >
> > Open your mind, use open source.
> >
>
>
> --
> Facundo Ameal.
> famealgmailcom
> Linux User #395088
>
> FWD: 741664
> MSN: asadolamorcillacomar
> ICQ: 74005793
>
>
> Open your mind, use open source.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
>
>
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>
>


--
Facundo Ameal.
famealgmailcom
Linux User #395088

FWD: 741664
MSN: asadolamorcillacomar
ICQ: 74005793


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