Hi, <br>My Sip.conf is like this:<br><br>[general]<br>port = 5060<br>bindaddr = <a href="http://0.0.0.0">0.0.0.0</a><br>context = others<br><br>register =><a href="http://outraspace:whatever@voipuser.org/outraspace">outraspace:whatever@voipuser.org/outraspace</a><br>
nat=yes<br>externip=<a href="http://58.251.75.251">58.251.75.251</a><br>localnet=<a href="http://192.168.1.0/255.255.255.0">192.168.1.0/255.255.255.0</a><br>canreinvite=no<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>allow=gsm<br>
qualify=yes<br><br><br><div class="gmail_quote">On Mon, Mar 17, 2008 at 9:57 PM, Steve Totaro <<a href="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Paste the sip.conf for your softphone.<br>
<br>
Thanks,<br>
<font color="#888888">Steve Totaro<br>
</font><div><div></div><div class="Wj3C7c"><br>
On Mon, Mar 17, 2008 at 9:38 AM, Pete Kay <<a href="mailto:petedao@gmail.com">petedao@gmail.com</a>> wrote:<br>
> Hi,<br>
><br>
> Here is the SIP debug output for the playback test. Thank you so much for<br>
> your help.<br>
><br>
> <------------><br>
> [Mar 18 05:33:08] -- Executing [333@my-phones:1]<br>
> Answer("SIP/2000-081e0738", "") in new stack<br>
> [Mar 18 05:33:08] Audio is at <a href="http://192.168.1.101" target="_blank">192.168.1.101</a> port 10028<br>
> [Mar 18 05:33:08] Adding codec 0x4 (ulaw) to SDP<br>
> [Mar 18 05:33:08] Adding codec 0x8 (alaw) to SDP<br>
> [Mar 18 05:33:08] Adding non-codec 0x1 (telephone-event) to SDP<br>
> [Mar 18 05:33:08]<br>
> <--- Reliably Transmitting (NAT) to <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a> ---><br>
> SIP/2.0 200 OK<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a>;branch=z9hG4bK-d87543-f917f17a8205cc03-1--d87543-;received=<a href="http://192.168.1.102" target="_blank">192.168.1.102</a>;rport=8526<br>
> From: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=902ece11<br>
> To: "333"<<a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a>>;tag=as1c53735e<br>
> Call-ID: ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.<br>
> CSeq: 2 INVITE<br>
> User-Agent: Asterisk PBX<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
> Supported: replaces<br>
> Contact: <<a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a>><br>
> Content-Type: application/sdp<br>
> Content-Length: 262<br>
><br>
> v=0<br>
> o=root 616 616 IN IP4 <a href="http://192.168.1.101" target="_blank">192.168.1.101</a><br>
> s=session<br>
> c=IN IP4 <a href="http://192.168.1.101" target="_blank">192.168.1.101</a><br>
> t=0 0<br>
> m=audio 10028 RTP/AVP 0 8 101<br>
> a=rtpmap:0 PCMU/8000<br>
> a=rtpmap:8 PCMA/8000<br>
> a=rtpmap:101 telephone-event/8000<br>
> a=fmtp:101 0-16<br>
> a=silenceSupp:off - - - -<br>
> a=ptime:20<br>
> a=sendrecv<br>
><br>
> <------------><br>
> [Mar 18 05:33:08] -- Executing [333@my-phones:2]<br>
> Playback("SIP/2000-081e0738", "vm-goodbye") in new stack<br>
> [Mar 18 05:33:08] -- <SIP/2000-081e0738> Playing 'vm-goodbye' (language<br>
> 'en')<br>
> [Mar 18 05:33:08]<br>
> <--- SIP read from <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a> ---><br>
> ACK <a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a> SIP/2.0<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a>;branch=z9hG4bK-d87543-52064b41251a4a1c-1--d87543-;rport<br>
> Max-Forwards: 70<br>
> Contact: <sip:2000@192.168.1.102:8526><br>
> To: "333"<<a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a>>;tag=as1c53735e<br>
> From: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=902ece11<br>
> Call-ID: ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.<br>
> CSeq: 2 ACK<br>
> Proxy-Authorization: Digest<br>
> username="2000",realm="asterisk",nonce="387941cf",uri="<a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a>",response="0a44bf3bf1daf39f8d32aac795d6b7c9",algorithm=MD5<br>
> User-Agent: X-Lite release 1011s stamp 41150<br>
> Content-Length: 0<br>
><br>
><br>
> <-------------><br>
> [Mar 18 05:33:08] --- (11 headers 0 lines) ---<br>
> [Mar 18 05:33:12]<br>
> <--- SIP read from <a href="http://192.168.1.102:5060" target="_blank">192.168.1.102:5060</a> ---><br>
> OPTIONS <a href="mailto:sip:ping@192.168.1.101">sip:ping@192.168.1.101</a> SIP/2.0<br>
> Via: SIP/2.0/UDP <a href="http://192.168.1.102:5060" target="_blank">192.168.1.102:5060</a>;rport;branch=z9hG4bK793126083<br>
> From: 2001 <<a href="mailto:sip:2001@192.168.1.101">sip:2001@192.168.1.101</a>>;tag=2612560371<br>
> To: <<a href="mailto:sip:ping@192.168.1.101">sip:ping@192.168.1.101</a>><br>
> Call-ID: <a href="mailto:2808830214@192.168.1.102">2808830214@192.168.1.102</a><br>
> CSeq: 20 OPTIONS<br>
> Max-Forwards: 70<br>
> User-Agent: wengo/v1/wengophoneng/wengo/rev12359/trunk/<br>
> Expires: 120<br>
> Accept: application/sdp<br>
> Content-Length: 0<br>
><br>
><br>
> <-------------><br>
> [Mar 18 05:33:12] --- (11 headers 0 lines) ---<br>
> [Mar 18 05:33:12] Looking for ping in others (domain <a href="http://192.168.1.101" target="_blank">192.168.1.101</a>)<br>
> [Mar 18 05:33:12]<br>
> <--- Transmitting (NAT) to <a href="http://192.168.1.102:5060" target="_blank">192.168.1.102:5060</a> ---><br>
> SIP/2.0 404 Not Found<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:5060" target="_blank">192.168.1.102:5060</a>;branch=z9hG4bK793126083;received=<a href="http://192.168.1.102" target="_blank">192.168.1.102</a>;rport=5060<br>
> From: 2001 <<a href="mailto:sip:2001@192.168.1.101">sip:2001@192.168.1.101</a>>;tag=2612560371<br>
> To: <<a href="mailto:sip:ping@192.168.1.101">sip:ping@192.168.1.101</a>>;tag=as0ca1ddb0<br>
> Call-ID: <a href="mailto:2808830214@192.168.1.102">2808830214@192.168.1.102</a><br>
> CSeq: 20 OPTIONS<br>
> User-Agent: Asterisk PBX<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
> Supported: replaces<br>
> Accept: application/sdp<br>
> Content-Length: 0<br>
><br>
><br>
> <------------><br>
> [Mar 18 05:33:12] Scheduling destruction of SIP dialog<br>
> '<a href="mailto:2808830214@192.168.1.102">2808830214@192.168.1.102</a>' in 32000 ms (Method: OPTIONS)<br>
> [Mar 18 05:33:13]<br>
> <--- SIP read from <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a> ---><br>
> BYE <a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a> SIP/2.0<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a>;branch=z9hG4bK-d87543-f409c54c895d2452-1--d87543-;rport<br>
> Max-Forwards: 70<br>
> Contact: <sip:2000@192.168.1.102:8526><br>
> To: "333"<<a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a>>;tag=as1c53735e<br>
> From: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=902ece11<br>
> Call-ID: ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.<br>
> CSeq: 3 BYE<br>
> Proxy-Authorization: Digest<br>
> username="2000",realm="asterisk",nonce="387941cf",uri="<a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a>",response="c48a3b608e9c1806c3b5f1c6d7fbab01",algorithm=MD5<br>
> User-Agent: X-Lite release 1011s stamp 41150<br>
> Reason: SIP;description="User Hung Up"<br>
> Content-Length: 0<br>
><br>
><br>
> <-------------><br>
> [Mar 18 05:33:13] --- (12 headers 0 lines) ---<br>
> [Mar 18 05:33:13] Sending to <a href="http://192.168.1.102" target="_blank">192.168.1.102</a> : 8526 (NAT)<br>
> [Mar 18 05:33:13]<br>
> <--- Transmitting (NAT) to <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a> ---><br>
> SIP/2.0 200 OK<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a>;branch=z9hG4bK-d87543-f409c54c895d2452-1--d87543-;received=<a href="http://192.168.1.102" target="_blank">192.168.1.102</a>;rport=8526<br>
> From: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=902ece11<br>
> To: "333"<<a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a>>;tag=as1c53735e<br>
> Call-ID: ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.<br>
> CSeq: 3 BYE<br>
> User-Agent: Asterisk PBX<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
> Supported: replaces<br>
> Contact: <<a href="mailto:sip:333@192.168.1.101">sip:333@192.168.1.101</a>><br>
> Content-Length: 0<br>
><br>
><br>
> <------------><br>
> [Mar 18 05:33:13] == Spawn extension (my-phones, 333, 2) exited non-zero<br>
> on 'SIP/2000-081e0738'<br>
> [Mar 18 05:33:14] Really destroying SIP dialog<br>
> 'ZGU0NzM1M2I3ZmM1OGQ4OTViZTlhMDdmNzQ2MTdjMzQ.' Method: BYE<br>
> [Mar 18 05:33:17]<br>
> <--- SIP read from <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a> ---><br>
> SUBSCRIBE <a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a> SIP/2.0<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a>;branch=z9hG4bK-d87543-5a0fd851e47c773d-1--d87543-;rport<br>
> Max-Forwards: 70<br>
> Contact: <sip:2000@192.168.1.102:8526><br>
> To: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>><br>
> From: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=181de57f<br>
> Call-ID: YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.<br>
> CSeq: 1 SUBSCRIBE<br>
> Expires: 300<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,<br>
> INFO<br>
> User-Agent: X-Lite release 1011s stamp 41150<br>
> Event: message-summary<br>
> Content-Length: 0<br>
><br>
><br>
> <-------------><br>
> [Mar 18 05:33:17] --- (13 headers 0 lines) ---<br>
> [Mar 18 05:33:17] Creating new subscription<br>
> [Mar 18 05:33:17] Sending to <a href="http://192.168.1.102" target="_blank">192.168.1.102</a> : 8526 (NAT)<br>
> [Mar 18 05:33:17] Found peer '2000'<br>
> [Mar 18 05:33:17]<br>
> <--- Transmitting (NAT) to <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a> ---><br>
> SIP/2.0 401 Unauthorized<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a>;branch=z9hG4bK-d87543-5a0fd851e47c773d-1--d87543-;received=<a href="http://192.168.1.102" target="_blank">192.168.1.102</a>;rport=8526<br>
> From: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=181de57f<br>
> To: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=as392594ef<br>
> Call-ID: YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.<br>
> CSeq: 1 SUBSCRIBE<br>
> User-Agent: Asterisk PBX<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
> Supported: replaces<br>
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2897b4aa"<br>
> Content-Length: 0<br>
><br>
><br>
> <------------><br>
> [Mar 18 05:33:17] Scheduling destruction of SIP dialog<br>
> 'YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.' in 6976 ms (Method:<br>
> SUBSCRIBE)<br>
> [Mar 18 05:33:17]<br>
> <--- SIP read from <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a> ---><br>
> SUBSCRIBE <a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a> SIP/2.0<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a>;branch=z9hG4bK-d87543-904ff3127e03aa31-1--d87543-;rport<br>
> Max-Forwards: 70<br>
> Contact: <sip:2000@192.168.1.102:8526><br>
> To: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>><br>
> From: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=181de57f<br>
> Call-ID: YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.<br>
> CSeq: 2 SUBSCRIBE<br>
> Expires: 300<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,<br>
> INFO<br>
> User-Agent: X-Lite release 1011s stamp 41150<br>
> Authorization: Digest<br>
> username="2000",realm="asterisk",nonce="2897b4aa",uri="<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>",response="f1bcbbc23e4069ea95962b8c2fbb12b0",algorithm=MD5<br>
> Event: message-summary<br>
> Content-Length: 0<br>
><br>
><br>
> <-------------><br>
> [Mar 18 05:33:17] --- (14 headers 0 lines) ---<br>
> [Mar 18 05:33:17] Creating new subscription<br>
> [Mar 18 05:33:17] Sending to <a href="http://192.168.1.102" target="_blank">192.168.1.102</a> : 8526 (NAT)<br>
> [Mar 18 05:33:17] Found peer '2000'<br>
> [Mar 18 05:33:17] Looking for 2000 in my-phones (domain <a href="http://192.168.1.101" target="_blank">192.168.1.101</a>)<br>
> [Mar 18 05:33:17]<br>
> <--- Transmitting (NAT) to <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a> ---><br>
> SIP/2.0 404 Not Found<br>
> Via: SIP/2.0/UDP<br>
> <a href="http://192.168.1.102:8526" target="_blank">192.168.1.102:8526</a>;branch=z9hG4bK-d87543-904ff3127e03aa31-1--d87543-;received=<a href="http://192.168.1.102" target="_blank">192.168.1.102</a>;rport=8526<br>
> From: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=181de57f<br>
> To: "2000"<<a href="mailto:sip:2000@192.168.1.101">sip:2000@192.168.1.101</a>>;tag=as392594ef<br>
> Call-ID: YjBmMzBlMjY3ZTMyYTA0YTUxYjI0NDExNTgxMjlmMzE.<br>
> CSeq: 2 SUBSCRIBE<br>
> User-Agent: Asterisk PBX<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
> Supported: replaces<br>
> Content-Length: 0<br>
><br>
><br>
><br>
><br>
><br>
><br>
> On Mon, Mar 17, 2008 at 7:26 PM, Steve Totaro<br>
> <<a href="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.com</a>> wrote:<br>
> > SIP debug output please.<br>
> ><br>
> > Thanks,<br>
> > Steve Totaro<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > On Mon, Mar 17, 2008 at 7:17 AM, Pete Kay <<a href="mailto:petedao@gmail.com">petedao@gmail.com</a>> wrote:<br>
> > > Hi,<br>
> > > Thanks for pointing out. I checked the extenip and it is fine. The<br>
> thing<br>
> > > is that I have already configure gsm as one of the codec in the<br>
> sip.conf:<br>
> > ><br>
> > > [general]<br>
> > > port = 5060<br>
> > > bindaddr = <a href="http://0.0.0.0" target="_blank">0.0.0.0</a><br>
> > > context = others<br>
> > ><br>
> > > register =><a href="http://outraspace:whatever@voipuser.org/outraspace" target="_blank">outraspace:whatever@voipuser.org/outraspace</a><br>
> > > nat=yes<br>
> > > externip=58.251.75.333<br>
> > ><br>
> > > localnet=<a href="http://192.168.1.0/255.255.255.0" target="_blank">192.168.1.0/255.255.255.0</a><br>
> > > canreinvite=no<br>
> > > disallow=all<br>
> > > allow=ulaw<br>
> > > allow=alaw<br>
> > > allow=gsm<br>
> > > qualify=yes<br>
> > ><br>
> > > Any other hints?<br>
> > ><br>
> > ><br>
> > ><br>
> > ><br>
> > > On Mon, Mar 17, 2008 at 6:47 PM, Anselm Martin Hoffmeister<br>
> > > <<a href="mailto:anselm@hoffmeister-online.de">anselm@hoffmeister-online.de</a>> wrote:<br>
> > ><br>
> > > > Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay:<br>
> > > ><br>
> > > > > Hi,<br>
> > > > > I am new to Asterisk and I am having a setup problem that I am<br>
> trying<br>
> > > > > to resolved for the last couple days without any success. I am<br>
> pretty<br>
> > > > > much desperated on this issue and I don't know why. Can someone<br>
> > > > > please kindly help me to troubleshoot this? I can't hear any audio<br>
> > > > > from Asterisk when running Playback or VoiceMail tests.<br>
> > > ><br>
> > > > Dear Pete,<br>
> > > ><br>
> > > > my first idea would be that something with your codecs is borken (TM).<br>
> I<br>
> > > > personally use a setup quite similar to yours, with the one visible<br>
> > > > difference that I also allow the "gsm" codec, owing to the fact that<br>
> at<br>
> > > > least my home-recorded prompts are gsm only. I _guess_ asterisk could<br>
> or<br>
> > > > should handle format conversion from audio files automagically, but<br>
> for<br>
> > > > making sure, please try adding "gsm", at least for now.<br>
> > > ><br>
> > > > You might also want to setup the<br>
> > > > [sipclient] stanza in sip.conf such that "nat" is set to "no",<br>
> although<br>
> > > > I do not see why that should break things. Especially as "Echo" works.<br>
> > > ><br>
> > > > The externip is set to your current external IP, right? (Knowing full<br>
> > > > well that some DSL lines get a new IP as often as 6 times a day, or as<br>
> a<br>
> > > > P2P bandwidth countermeasure down to five minute intervals at certain<br>
> > > > restrictive providers once your "fair use" volume is used up). Again<br>
> > > > this should not be the culprit...<br>
> > > ><br>
> > > > Poking with a stick in the swamps, but perhaps hitting the bug :-P<br>
> > > ><br>
> > > > BR<br>
> > > > Anselm<br>
> > > ><br>
> > > ><br>
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> > > ><br>
> > ><br>
> > ><br>
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> > ><br>
> ><br>
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><br>
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