if (first_session.ready()) { console_log("notice","Session state=[" + first_session.state + "] \n"); consoleLog("NOTICE", "ready: Start DTMF\n"); first_session.execute("start_dtmf"); first_session.answer( ); Disposition = "ANS"; first_session.sleep(1500); console_log("notice", "Playing message: " + recording + "\n"); first_session.streamFile(recording, on_event); if (first_session.ready()) { consoleLog("err", "ready: Waiting for input\n"); first_session.streamFile("4.wav",on_event, "dtmf"); consoleLog("err", "ready: Timeout on input\n"); first_session.execute("stop_tone_detect"); //disp_call() first_session.hangup() first_session.execute("sleep", "2000"); consoleLog("NOTICE", "EXITING\n"); exit(); } } ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 18:34 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files we would need to see your script. On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote: Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!! Example 3 calls and the matching playbacks 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav That's 22 seconds for the first one!! Anyone any ideas as to what's going on here? Regards _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400

Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

if (first_session.ready()) {

console_log("notice","Session state=[" + first_session.state + "] \n");

consoleLog("NOTICE", "ready: Start DTMF\n");

first_session.execute("start_dtmf");

first_session.answer( );

Disposition = "ANS";

first_session.sleep(1500);

console_log("notice", "Playing message: " + recording + "\n");

first_session.streamFile(recording, on_event);

if (first_session.ready()) {

consoleLog("err", "ready: Waiting for input\n");

first_session.streamFile("4.wav",on_event, "dtmf");

consoleLog("err", "ready: Timeout on input\n");

first_session.execute("stop_tone_detect");

//disp_call()

first_session.hangup()

first_session.execute("sleep", "2000");

consoleLog("NOTICE", "EXITING\n");

exit();

}

}

________________________________

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 18:34 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files

we would need to see your script.

On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!!

Example 3 calls and the matching playbacks

2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

That's 22 seconds for the first one!!

Anyone any ideas as to what's going on here?

Regards

_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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http://www.freeswitch.org

-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400

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I'm talking of the time when I hit the dial button to the phone 3 ft away starting to ring Regards, -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 20:28 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files On Tue, Feb 17, 2009 at 12:11 PM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

Pretty much

I haven't included the on-event hooks as it never gets to the point where they're called.

Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal?

By "ring" do you mean caller hears ringback tone or target phone audibly rings? I haven't noticed a long delay when dialing but I haven't been looking for one either.

//build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' "

var first_session = new Session(dial_string);

// Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; }

else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; }

disp_call() exit(); }

//set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup");

Can I just say how much I hate JavaScript for stuff like this? :) I can't test this right now but I will lab it up as soon as I can. In the meantime I would have you turn on debugging and capture the results from start to finish for a successful call. Save that for future reference. Then try to recreate the symptoms, also with debug turned on, capturing output. It will be A LOT of output, so you might want to consider rolling log files. (see the "Reporting Bugs" wiki page for hints on how to do that and more) Has anybody else out there used js for something like this, or otherwise have any input on why js seems to be acting up in this case? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

okay, can you do the usual stuff and report a bug on jira? Not sure if it's really bug but having you collect all of the data and submit a bug report will assist us greatly. -MC On Tue, Feb 17, 2009 at 12:30 PM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my office number (using latest trunk)

2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 handle_fatality() Caught signal 11 for unmapped thread!Aborted (core dumped)

Also then I had tone detect on, I'd often get this

freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: Assertion `read_codec != ((void *)0)' failed.

Hardware, HP DL360 G4. Centos 5.2, 4 GB ram.

-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 17 February 2009 20:11 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files

Pretty much

I haven't included the on-event hooks as it never gets to the point where they're called.

Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal?

//build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' "

var first_session = new Session(dial_string);

// Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; }

else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; }

disp_call() exit(); }

//set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup");

-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files

Is this the entire script?! -MC

On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

...

if (first_session.ready()) {

console_log("notice","Session state=[" + first_session.state + "] \n");

consoleLog("NOTICE", "ready: Start DTMF\n");

first_session.execute("start_dtmf");

first_session.answer( );

Disposition = "ANS";

first_session.sleep(1500);

console_log("notice", "Playing message: " + recording + "\n");

first_session.streamFile(recording, on_event);

if (first_session.ready()) {

consoleLog("err", "ready: Waiting for input\n");

first_session.streamFile("4.wav",on_event, "dtmf");

consoleLog("err", "ready: Timeout on input\n");

first_session.execute("stop_tone_detect");

//disp_call()

first_session.hangup()

first_session.execute("sleep", "2000");

consoleLog("NOTICE", "EXITING\n");

exit();

}

}

________________________________

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 18:34 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files

we would need to see your script.

On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!!

Example 3 calls and the matching playbacks

2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

That's 22 seconds for the first one!!

Anyone any ideas as to what's going on here?

Regards

_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

...

http://www.freeswitch.org

-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/

AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400

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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

...

http://www.freeswitch.org

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_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

1, OK, 2. Right now I have a php script calling bgapi via and event socket with the call parameters. Is that what you mean? If not, can you give me a pointer? I had assumed that every time I called bgapi it with the script in it, it would get it's own copy. Regards, ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 20:38 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files 1) turn off crash protection. 2) you cant manipulate more that one call per script, design the script to be run from the application interface so you originate the call with the api interface and transfer the call to the script so each one has it's own copy of the script. On Tue, Feb 17, 2009 at 2:30 PM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote: I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my office number (using latest trunk) 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 handle_fatality() Caught signal 11 for unmapped thread!Aborted (core dumped) Also then I had tone detect on, I'd often get this freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: Assertion `read_codec != ((void *)0)' failed. Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 17 February 2009 20:11 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

if (first_session.ready()) {

console_log("notice","Session state=[" + first_session.state + "] \n");

consoleLog("NOTICE", "ready: Start DTMF\n");

first_session.execute("start_dtmf");

first_session.answer( );

Disposition = "ANS";

first_session.sleep(1500);

console_log("notice", "Playing message: " + recording + "\n");

first_session.streamFile(recording, on_event);

if (first_session.ready()) {

consoleLog("err", "ready: Waiting for input\n");

first_session.streamFile("4.wav",on_event, "dtmf");

consoleLog("err", "ready: Timeout on input\n");

first_session.execute("stop_tone_detect");

//disp_call()

first_session.hangup()

first_session.execute("sleep", "2000");

consoleLog("NOTICE", "EXITING\n");

exit();

}

}

________________________________

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 18:34 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files

we would need to see your script.

On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!!

Example 3 calls and the matching playbacks

2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav

That's 22 seconds for the first one!!

Anyone any ideas as to what's going on here?

Regards

_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org

-- Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/

AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch

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_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400

yes, bgapi counts as an API call. Ken Rice thinks this might be related to a spidermonky concurrency issue...

Well that kinda fits, as I see the audio files stacking up, seems like they're being queued. Question is, what's my alternative, lua? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 20:57 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files On Tue, Feb 17, 2009 at 12:48 PM, Nik Middleton <nik.middleton@noblesolutions.co.uk> wrote:

1, OK,

2. Right now I have a php script calling bgapi via and event socket with the call parameters. Is that what you mean? If not, can you give me a pointer? I had assumed that every time I called bgapi it with the script in it, it would get it's own copy.

Hello, folks - I hope that I can reach someone who knows the answer to this one: I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are locked by Webnet global Communications. From what I can tell, this company went bankrupt, and the ebay seller bought the phones from a bankruptcy auction. He does not know the admin username or password. Nowhere on the linksys site is there a solution to how to unlock these phones. Is there a way, or did I buy 2 interesting looking doorstops? Other than the password thing, they function fine. Thanks - Library Mark

Have you tried resetting the phone via the built-in IVR menu? Pick up the handset and dial ****73738# This should reset the phone to factory defaults, assuming that company didn't lock this feature out. Gabe Mark wrote:

Hello, folks - I hope that I can reach someone who knows the answer to this one:

I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are locked by Webnet global Communications. From what I can tell, this company went bankrupt, and the ebay seller bought the phones from a bankruptcy auction. He does not know the admin username or password. Nowhere on the linksys site is there a solution to how to unlock these phones.

Is there a way, or did I buy 2 interesting looking doorstops? Other than the password thing, they function fine.

Thanks -

Library Mark

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I believe you need to make sure the Ethernet cable is unplugged from the phone when trying to dial that string. Now I've never tried this, but it should theoretically be possible ... Sniff the traffic of the phone and see where it's attempting to pickup the config file. Then setup a local network with your own DNS server, and re-direct the phone (via DNS) to your own web server (assuming it's picking up the config via http) and have a config file on the web server with a username and password you specify to reset the config and get into the phone. Let's hope they didn't setup the phone to provision via https, otherwise you're really SOL If you need help generating a config for the phone, with Linksys' special config tool, contact me offlist. Gabe Mark wrote:

Sadly, ****73738# does not work.

Is there a jumper on the board or some other hardware fix for this?

Quoting "Gabriel Kuri" <gkuri@ieee.org>:

...

Have you tried resetting the phone via the built-in IVR menu?

Pick up the handset and dial ****73738#

This should reset the phone to factory defaults, assuming that company didn't lock this feature out.

Gabe

Mark wrote:

...

Hello, folks - I hope that I can reach someone who knows the answer to this one:

I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are locked by Webnet global Communications. From what I can tell, this company went bankrupt, and the ebay seller bought the phones from a bankruptcy auction. He does not know the admin username or password. Nowhere on the linksys site is there a solution to how to unlock these phones.

Is there a way, or did I buy 2 interesting looking doorstops? Other than the password thing, they function fine.

Thanks -

Library Mark

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Brian West wrote:

Don't they cryptographically sign the config also?

it's an option in the device... some providers do, some don't. but it shouldn't matter too much if they're using https or not, as long as the ata doesn't authenticate via certificate or something. -Ray

It depends on the whether you pass the option to the Linksys/Cisco Profile Compiler to generate the config file. In any case, that shouldn't be an issue. Gabe Brian West wrote:

Don't they cryptographically sign the config also?

/b

On Feb 17, 2009, at 2:58 PM, Gabriel Kuri wrote:

...

If you need help generating a config for the phone, with Linksys' special config tool, contact me offlist.

Gabe

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On the slight chance they're not doing remote provisioning and the phone is just simply locked with a username/password, you'll need to feed the phone a TFTP server via DHCP Option 66 and setup a config file on that tftp server with the name spa922.cfg. Contact me off list about generating a config file for the phone. Gabe Mark wrote:

I did unplug the ethernet cable. I have never been able to make the IVR work on any of the Linksys phones that I have. I must be doing something wrong.

I will try to sniff the traffic on the phone when I start it up. I will report back when I do.

Thanks so much -

Library Mark

Quoting "Gabriel Kuri" <gkuri@ieee.org>:

...

I believe you need to make sure the Ethernet cable is unplugged from the phone when trying to dial that string.

Now I've never tried this, but it should theoretically be possible ...

Sniff the traffic of the phone and see where it's attempting to pickup the config file. Then setup a local network with your own DNS server, and re-direct the phone (via DNS) to your own web server (assuming it's picking up the config via http) and have a config file on the web server with a username and password you specify to reset the config and get into the phone. Let's hope they didn't setup the phone to provision via https, otherwise you're really SOL

If you need help generating a config for the phone, with Linksys' special config tool, contact me offlist.

Gabe

Mark wrote:

...

Sadly, ****73738# does not work.

Is there a jumper on the board or some other hardware fix for this?

Quoting "Gabriel Kuri" <gkuri@ieee.org>:

...

Have you tried resetting the phone via the built-in IVR menu?

Pick up the handset and dial ****73738#

This should reset the phone to factory defaults, assuming that company didn't lock this feature out.

Gabe

Mark wrote:

...

Hello, folks - I hope that I can reach someone who knows the answer to this one:

I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are locked by Webnet global Communications. From what I can tell, this company went bankrupt, and the ebay seller bought the phones from a bankruptcy auction. He does not know the admin username or password. Nowhere on the linksys site is there a solution to how to unlock these phones.

Is there a way, or did I buy 2 interesting looking doorstops? Other than the password thing, they function fine.

Thanks -

Library Mark

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