[asterisk-users] Desperately need help with Asterisk setup
Pete Kay
petedao at gmail.com
Mon Mar 17 09:27:25 CDT 2008
Hi,
My Sip.conf is like this:
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
register =>outraspace:whatever at voipuser.org/outraspace
nat=yes
externip=58.251.75.251
localnet=192.168.1.0/255.255.255.0
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=yes
On Mon, Mar 17, 2008 at 9:57 PM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:
> Paste the sip.conf for your softphone.
>
> Thanks,
> Steve Totaro
>
> On Mon, Mar 17, 2008 at 9:38 AM, Pete Kay <petedao at gmail.com> wrote:
> > Hi,
> >
> > Here is the SIP debug output for the playback test. Thank you so much
> for
> > your help.
> >
> > <------------>
> > [Mar 18 05:33:08] -- Executing [333 at my-phones:1]
> > Answer("SIP/2000-081e0738", "") in new stack
> > [Mar 18 05:33:08] Audio is at 192.168.1.101 port 10028
> > [Mar 18 05:33:08] Adding codec 0x4 (ulaw) to SDP
> > [Mar 18 05:33:08] Adding codec 0x8 (alaw) to SDP
> > [Mar 18 05:33:08] Adding non-codec 0x1 (telephone-event) to SDP
> > [Mar 18 05:33:08]
> > <--- Reliably Transmitting (NAT) to 192.168.1.102:8526 --->
> > SIP/2.0 200 OK
> > Via: SIP/2.0/UDP
> > 192.168.1.102:8526
> ;branch=z9hG4bK-d87543-f917f17a8205cc03-1--d87543-;received=192.168.1.102
> ;rport=8526
> > From: "2000"<sip:2000 at 192.168.1.101>;tag=902ece11
> > To: "333"<sip:333 at 192.168.1.101>;tag=as1c53735e
> > Call-ID: ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.
> > CSeq: 2 INVITE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Contact: <sip:333 at 192.168.1.101>
> > Content-Type: application/sdp
> > Content-Length: 262
> >
> > v=0
> > o=root 616 616 IN IP4 192.168.1.101
> > s=session
> > c=IN IP4 192.168.1.101
> > t=0 0
> > m=audio 10028 RTP/AVP 0 8 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > a=sendrecv
> >
> > <------------>
> > [Mar 18 05:33:08] -- Executing [333 at my-phones:2]
> > Playback("SIP/2000-081e0738", "vm-goodbye") in new stack
> > [Mar 18 05:33:08] -- <SIP/2000-081e0738> Playing 'vm-goodbye'
> (language
> > 'en')
> > [Mar 18 05:33:08]
> > <--- SIP read from 192.168.1.102:8526 --->
> > ACK sip:333 at 192.168.1.101 SIP/2.0
> > Via: SIP/2.0/UDP
> > 192.168.1.102:8526
> ;branch=z9hG4bK-d87543-52064b41251a4a1c-1--d87543-;rport
> > Max-Forwards: 70
> > Contact: <sip:2000 at 192.168.1.102:8526>
> > To: "333"<sip:333 at 192.168.1.101>;tag=as1c53735e
> > From: "2000"<sip:2000 at 192.168.1.101>;tag=902ece11
> > Call-ID: ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.
> > CSeq: 2 ACK
> > Proxy-Authorization: Digest
> > username="2000",realm="asterisk",nonce="387941cf",uri="
> sip:333 at 192.168.1.101
> ",response="0a44bf3bf1daf39f8d32aac795d6b7c9",algorithm=MD5
> > User-Agent: X-Lite release 1011s stamp 41150
> > Content-Length: 0
> >
> >
> > <------------->
> > [Mar 18 05:33:08] --- (11 headers 0 lines) ---
> > [Mar 18 05:33:12]
> > <--- SIP read from 192.168.1.102:5060 --->
> > OPTIONS sip:ping at 192.168.1.101 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bK793126083
> > From: 2001 <sip:2001 at 192.168.1.101>;tag=2612560371
> > To: <sip:ping at 192.168.1.101>
> > Call-ID: 2808830214 at 192.168.1.102
> > CSeq: 20 OPTIONS
> > Max-Forwards: 70
> > User-Agent: wengo/v1/wengophoneng/wengo/rev12359/trunk/
> > Expires: 120
> > Accept: application/sdp
> > Content-Length: 0
> >
> >
> > <------------->
> > [Mar 18 05:33:12] --- (11 headers 0 lines) ---
> > [Mar 18 05:33:12] Looking for ping in others (domain 192.168.1.101)
> > [Mar 18 05:33:12]
> > <--- Transmitting (NAT) to 192.168.1.102:5060 --->
> > SIP/2.0 404 Not Found
> > Via: SIP/2.0/UDP
> > 192.168.1.102:5060;branch=z9hG4bK793126083;received=192.168.1.102
> ;rport=5060
> > From: 2001 <sip:2001 at 192.168.1.101>;tag=2612560371
> > To: <sip:ping at 192.168.1.101>;tag=as0ca1ddb0
> > Call-ID: 2808830214 at 192.168.1.102
> > CSeq: 20 OPTIONS
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Accept: application/sdp
> > Content-Length: 0
> >
> >
> > <------------>
> > [Mar 18 05:33:12] Scheduling destruction of SIP dialog
> > '2808830214 at 192.168.1.102' in 32000 ms (Method: OPTIONS)
> > [Mar 18 05:33:13]
> > <--- SIP read from 192.168.1.102:8526 --->
> > BYE sip:333 at 192.168.1.101 SIP/2.0
> > Via: SIP/2.0/UDP
> > 192.168.1.102:8526
> ;branch=z9hG4bK-d87543-f409c54c895d2452-1--d87543-;rport
> > Max-Forwards: 70
> > Contact: <sip:2000 at 192.168.1.102:8526>
> > To: "333"<sip:333 at 192.168.1.101>;tag=as1c53735e
> > From: "2000"<sip:2000 at 192.168.1.101>;tag=902ece11
> > Call-ID: ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.
> > CSeq: 3 BYE
> > Proxy-Authorization: Digest
> > username="2000",realm="asterisk",nonce="387941cf",uri="
> sip:333 at 192.168.1.101
> ",response="c48a3b608e9c1806c3b5f1c6d7fbab01",algorithm=MD5
> > User-Agent: X-Lite release 1011s stamp 41150
> > Reason: SIP;description="User Hung Up"
> > Content-Length: 0
> >
> >
> > <------------->
> > [Mar 18 05:33:13] --- (12 headers 0 lines) ---
> > [Mar 18 05:33:13] Sending to 192.168.1.102 : 8526 (NAT)
> > [Mar 18 05:33:13]
> > <--- Transmitting (NAT) to 192.168.1.102:8526 --->
> > SIP/2.0 200 OK
> > Via: SIP/2.0/UDP
> > 192.168.1.102:8526
> ;branch=z9hG4bK-d87543-f409c54c895d2452-1--d87543-;received=192.168.1.102
> ;rport=8526
> > From: "2000"<sip:2000 at 192.168.1.101>;tag=902ece11
> > To: "333"<sip:333 at 192.168.1.101>;tag=as1c53735e
> > Call-ID: ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.
> > CSeq: 3 BYE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Contact: <sip:333 at 192.168.1.101>
> > Content-Length: 0
> >
> >
> > <------------>
> > [Mar 18 05:33:13] == Spawn extension (my-phones, 333, 2) exited
> non-zero
> > on 'SIP/2000-081e0738'
> > [Mar 18 05:33:14] Really destroying SIP dialog
> > 'ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.' Method: BYE
> > [Mar 18 05:33:17]
> > <--- SIP read from 192.168.1.102:8526 --->
> > SUBSCRIBE sip:2000 at 192.168.1.101 SIP/2.0
> > Via: SIP/2.0/UDP
> > 192.168.1.102:8526
> ;branch=z9hG4bK-d87543-5a0fd851e47c773d-1--d87543-;rport
> > Max-Forwards: 70
> > Contact: <sip:2000 at 192.168.1.102:8526>
> > To: "2000"<sip:2000 at 192.168.1.101>
> > From: "2000"<sip:2000 at 192.168.1.101>;tag=181de57f
> > Call-ID: YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.
> > CSeq: 1 SUBSCRIBE
> > Expires: 300
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE,
> > INFO
> > User-Agent: X-Lite release 1011s stamp 41150
> > Event: message-summary
> > Content-Length: 0
> >
> >
> > <------------->
> > [Mar 18 05:33:17] --- (13 headers 0 lines) ---
> > [Mar 18 05:33:17] Creating new subscription
> > [Mar 18 05:33:17] Sending to 192.168.1.102 : 8526 (NAT)
> > [Mar 18 05:33:17] Found peer '2000'
> > [Mar 18 05:33:17]
> > <--- Transmitting (NAT) to 192.168.1.102:8526 --->
> > SIP/2.0 401 Unauthorized
> > Via: SIP/2.0/UDP
> > 192.168.1.102:8526
> ;branch=z9hG4bK-d87543-5a0fd851e47c773d-1--d87543-;received=192.168.1.102
> ;rport=8526
> > From: "2000"<sip:2000 at 192.168.1.101>;tag=181de57f
> > To: "2000"<sip:2000 at 192.168.1.101>;tag=as392594ef
> > Call-ID: YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.
> > CSeq: 1 SUBSCRIBE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
> nonce="2897b4aa"
> > Content-Length: 0
> >
> >
> > <------------>
> > [Mar 18 05:33:17] Scheduling destruction of SIP dialog
> > 'YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.' in 6976 ms (Method:
> > SUBSCRIBE)
> > [Mar 18 05:33:17]
> > <--- SIP read from 192.168.1.102:8526 --->
> > SUBSCRIBE sip:2000 at 192.168.1.101 SIP/2.0
> > Via: SIP/2.0/UDP
> > 192.168.1.102:8526
> ;branch=z9hG4bK-d87543-904ff3127e03aa31-1--d87543-;rport
> > Max-Forwards: 70
> > Contact: <sip:2000 at 192.168.1.102:8526>
> > To: "2000"<sip:2000 at 192.168.1.101>
> > From: "2000"<sip:2000 at 192.168.1.101>;tag=181de57f
> > Call-ID: YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.
> > CSeq: 2 SUBSCRIBE
> > Expires: 300
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE,
> > INFO
> > User-Agent: X-Lite release 1011s stamp 41150
> > Authorization: Digest
> > username="2000",realm="asterisk",nonce="2897b4aa",uri="
> sip:2000 at 192.168.1.101
> ",response="f1bcbbc23e4069ea95962b8c2fbb12b0",algorithm=MD5
> > Event: message-summary
> > Content-Length: 0
> >
> >
> > <------------->
> > [Mar 18 05:33:17] --- (14 headers 0 lines) ---
> > [Mar 18 05:33:17] Creating new subscription
> > [Mar 18 05:33:17] Sending to 192.168.1.102 : 8526 (NAT)
> > [Mar 18 05:33:17] Found peer '2000'
> > [Mar 18 05:33:17] Looking for 2000 in my-phones (domain 192.168.1.101)
> > [Mar 18 05:33:17]
> > <--- Transmitting (NAT) to 192.168.1.102:8526 --->
> > SIP/2.0 404 Not Found
> > Via: SIP/2.0/UDP
> > 192.168.1.102:8526
> ;branch=z9hG4bK-d87543-904ff3127e03aa31-1--d87543-;received=192.168.1.102
> ;rport=8526
> > From: "2000"<sip:2000 at 192.168.1.101>;tag=181de57f
> > To: "2000"<sip:2000 at 192.168.1.101>;tag=as392594ef
> > Call-ID: YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.
> > CSeq: 2 SUBSCRIBE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Content-Length: 0
> >
> >
> >
> >
> >
> >
> > On Mon, Mar 17, 2008 at 7:26 PM, Steve Totaro
> > <stotaro at totarotechnologies.com> wrote:
> > > SIP debug output please.
> > >
> > > Thanks,
> > > Steve Totaro
> > >
> > >
> > >
> > >
> > > On Mon, Mar 17, 2008 at 7:17 AM, Pete Kay <petedao at gmail.com> wrote:
> > > > Hi,
> > > > Thanks for pointing out. I checked the extenip and it is fine. The
> > thing
> > > > is that I have already configure gsm as one of the codec in the
> > sip.conf:
> > > >
> > > > [general]
> > > > port = 5060
> > > > bindaddr = 0.0.0.0
> > > > context = others
> > > >
> > > > register =>outraspace:whatever at voipuser.org/outraspace
> > > > nat=yes
> > > > externip=58.251.75.333
> > > >
> > > > localnet=192.168.1.0/255.255.255.0
> > > > canreinvite=no
> > > > disallow=all
> > > > allow=ulaw
> > > > allow=alaw
> > > > allow=gsm
> > > > qualify=yes
> > > >
> > > > Any other hints?
> > > >
> > > >
> > > >
> > > >
> > > > On Mon, Mar 17, 2008 at 6:47 PM, Anselm Martin Hoffmeister
> > > > <anselm at hoffmeister-online.de> wrote:
> > > >
> > > > > Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay:
> > > > >
> > > > > > Hi,
> > > > > > I am new to Asterisk and I am having a setup problem that I am
> > trying
> > > > > > to resolved for the last couple days without any success. I am
> > pretty
> > > > > > much desperated on this issue and I don't know why. Can someone
> > > > > > please kindly help me to troubleshoot this? I can't hear any
> audio
> > > > > > from Asterisk when running Playback or VoiceMail tests.
> > > > >
> > > > > Dear Pete,
> > > > >
> > > > > my first idea would be that something with your codecs is borken
> (TM).
> > I
> > > > > personally use a setup quite similar to yours, with the one
> visible
> > > > > difference that I also allow the "gsm" codec, owing to the fact
> that
> > at
> > > > > least my home-recorded prompts are gsm only. I _guess_ asterisk
> could
> > or
> > > > > should handle format conversion from audio files automagically,
> but
> > for
> > > > > making sure, please try adding "gsm", at least for now.
> > > > >
> > > > > You might also want to setup the
> > > > > [sipclient] stanza in sip.conf such that "nat" is set to "no",
> > although
> > > > > I do not see why that should break things. Especially as "Echo"
> works.
> > > > >
> > > > > The externip is set to your current external IP, right? (Knowing
> full
> > > > > well that some DSL lines get a new IP as often as 6 times a day,
> or as
> > a
> > > > > P2P bandwidth countermeasure down to five minute intervals at
> certain
> > > > > restrictive providers once your "fair use" volume is used up).
> Again
> > > > > this should not be the culprit...
> > > > >
> > > > > Poking with a stick in the swamps, but perhaps hitting the bug :-P
> > > > >
> > > > > BR
> > > > > Anselm
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
> > > > >
> > > > > asterisk-users mailing list
> > > > > To UNSUBSCRIBE or update options visit:
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> > > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
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> > > >
> > >
> > > _______________________________________________
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > > asterisk-users mailing list
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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