[asterisk-users] Desperately need help with Asterisk setup
Steve Totaro
stotaro at totarotechnologies.com
Mon Mar 17 08:57:24 CDT 2008
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
> > > >
> > > >
> > > > _______________________________________________
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> > > >
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> > > >
> > >
> > >
> > > _______________________________________________
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> > >
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> > >
> >
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>
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