[asterisk-users] Desperately need help with Asterisk setup

Mon Mar 17 08:38:26 CDT 2008

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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