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[Freeswitch-users] Audio routing problem between FS and Voxeo


 
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mszlazak at aol.com
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PostPosted: Wed Dec 10, 2008 1:49 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

Normal 0 MicrosoftInternetExplorer4 <![endif]--> st1\:*{behavior:url(#ieooui) } <![endif]--> /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-parent:""; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-para-margin:0in; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Times New Roman";} <![endif]--> I’m trying to route calls from X-lite <--> FS (Nov. 6 2008 svn) <--> Voceo (Prophecy) to use Voxeo’s ASR instead of FS’s built in PocketSphinx/ASR.
All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi=2 0router that connects a laptop to my desktop in case that matters but I’m not using the laptop for any of this.

I’ve set up an extension to bridge calls to Voxeo. Here is the entry in file conf\dialplan\default.xml:

<extension name="Doctors Office">
<condition field="destination_number" expression="^2007$">
<action application="info"/>
<action application="bridge" data="sofia/internal/sip:Doctorsoffice@127.0.0.1:5068"/>
</condition>
</extension>

I hear one ring then a hang up. No errors in the FS console and Voxeo’s logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeo’s Prophecy ASR called “Doctorsoffice.”



Any ideas?

Thanks.



Mark.


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brian at freeswitch.org
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PostPosted: Wed Dec 10, 2008 1:54 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

You're a month behind.. I highly recommend you update. Chances are this has already been fixed.


/b

On Dec 10, 2008, at 12:39 PM, mszlazak@aol.com (mszlazak@aol.com) wrote:
Quote:
Im trying to route calls from X-lite <--> FS (Nov. 6 2008 svn) <--> Voceo (Prophecy) to use Voxeos ASR instead of FSs built in PocketSphinx/ASR.
All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi=2 0router that connects a laptop to my desktop in case that matters but Im not using the laptop for any of this.

Ive set up an extension to bridge calls to Voxeo. Here is the entry in file conf\dialplan\default.xml:

<extension name="Doctors Office">
<condition field="destination_number" expression="^2007$">
<action application="info"/>
<action application="bridge" data="sofia/internal/[url=sip:Doctorsoffice@127.0.0.1:5068]sip:Doctorsoffice@127.0.0.1:5068[/url]"/>
</condition>
</extension>

I hear one ring then a hang up. No errors in the FS console and Voxeos logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeos Prophecy ASR called Doctorsoffice.



Any ideas?

Thanks.



Mark.


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mszlazak at aol.com
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PostPosted: Wed Dec 10, 2008 2:14 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

Yes but someone else I'm in contact with set up FS a couple days ago and is having the same problems.
Brian should I still update today?





-----Original Message-----
From: Brian West <brian@freeswitch.org>
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 10 Dec 2008 10:45 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo

You're a month behind.. I highly recommend you update. Chances are this has already been fixed.


/b

On Dec 10, 2008, at 12:39 PM, mszlazak@aol.com (mszlazak@aol.com) wrote:
Quote:
I’m trying to route calls from X-lite&nbs p;<--> FS (Nov. 6 2008 svn) <--> Voceo (Prophecy) to use Voxeo’s ASR instead of FS’s built in PocketSphinx/ASR.
All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi=2 0router that connects a laptop to my desktop in case that matters but I’m not using the laptop for any of this.

I’ve set up an extension to bridge calls to=2 0Voxeo. Here is the entry in file conf\dialplan\default.xml:

<extension name="Doctors Office">
<condition field="destination_number" expression="^2007$">
<action application="info"/>
<action application="bridge" data="sofia/internal/sip:Doctorsoffice@127.0.0.1:5068"/>
</condition>
</extension>

I hear one ring then a hang up. No errors in the FS console and Voxeo’s logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeo’s Prophecy ASR called “Doctorsoffice.”



Any ideas?

Thanks.



Mark.


0A
Listen to 350+ music, sports, & news radio stations – including songs for the holidays – FREE while you browse. Start Listening Now!
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brian at freeswitch.org
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PostPosted: Wed Dec 10, 2008 2:25 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

If you're not trying this on the latest code then yes I would update if possible. Do you recall who you were talking too?

/b

On Dec 10, 2008, at 12:55 PM, mszlazak@aol.com (mszlazak@aol.com) wrote:
Quote:
Yes but someone else I'm in contact with set up FS a couple days ago and is having the same problems.
Brian should I still update today?

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mszlazak at aol.com
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PostPosted: Wed Dec 10, 2008 3:05 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

It was someone from Voxeo support. I think John was the main person helping me with this.

I updated but things got worse all over.
I now can't run other extensions Gino's pizza or some db stuff

Got these errors:

2008-12-10 11:40:23 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll
**The specified module could not be found. **
2008-12-10 11:40:23 [CONSOLE] mod_spidermonkey.c:944 sm_load_file() Successfully Loaded [C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_core_db.dll]
2008-12-10 11:40:23 [CONSOLE] mod_spidermonkey.c:944 sm_load_file() Successfully Loaded [C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_socket.dll]
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_spidermonkey]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'javascript'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'jsrun'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'jsapi'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_lua]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'lua'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'luarun'
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'lua'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_say_en]
2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:371 switch_loadable_module_process() Adding Say interface 'en'
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:118 switch_loadable_module_runtime() Starting runtime thread for CORE_SOFTTIMER_MODULE
2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:118 switch_loadable_module_runtime() Starting runtime thread for mod_event_socket
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list dl-candidates default (allow)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 10.0.0.0/8 (deny) to list dl-candidates
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 172.16.0.0/12 (deny) to list dl-candidates
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.0.0/16 (deny) to list dl-candidates
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list rfc1918 default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 10.0.0.0/8 (allow) to list rfc1918
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 172.16.0.0/12 (allow) to list rfc1918
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.0.0/16 (allow) to list rfc1918
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list lan default (allow)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.42.0/24 (deny) to list lan
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.42.42/32 (allow) to list lan
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list strict default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 208.102.123.124/32 (allow) to list strict
2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list domains default (deny)
2008-12-10 11:40:23 [NOTICE] switch_core.c:907 switch_load_network_lists() Adding 1.2.3.4/24 (allow) [brian@10.0.0.2] to list domains
2008-12-10 11:40:23 [CONSOLE] switch_core.c:1258 switch_core_init_and_modload()
FreeSWITCH Version 1.0.trunk (10171M) Started.
Crash Protection [Disabled]
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]

freeswitch@mark-e40edd7b44> 2008-12-10 11:40:24 [ERR] sofia.c:543 sofia_profile_thread_run() Error Creating SIP UA for profile: internal-ipv6
2008-12-10 11:40:45 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/1000@10.0.0.2 [88d1aaf9-a625-444e-883d-c5ac6eeac30e]
2008-12-10 11:40:45 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch->2006 in context default
2008-12-10 11:40:45 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000@10.0.0.2] has been answered
2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
2008-12-10 11:40:48 [NOTICE] switch_ivr_async.c:1845 switch_ivr_detect_speech() Hangup sofia/internal/1000@10.0.0.2 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
2008-12-10 11:40:48 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 1 (sofia/internal/1000@10.0.0.2) Ended
2008-12-10 11:40:48 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000@10.0.0.2 [CS_HANGUP]
2008-12-10 11:41:23 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/1000@10.0.0.2 [67882be4-4c10-d248-8fe1-ac02b9b8fc5c]
2008-12-10 11:41:23 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch->8337 in context default
2008-12-10 11:41:23 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/1000@10.0.0.2] has been answered
2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!
2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey() Cannot allocate speech engine!
2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000@10.0.0.2 [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000@10.0.0.2) Ended
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000@10.0.0.2 [CS_HANGUP]






-----Original Message-----
From: Brian West <brian@freeswitch.org>
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 10 Dec 2008 11:02 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo

If you're not trying this on the latest code then yes I would update if possible. Do you recall who you were talking too?

/b

On Dec 10, 2008, at 12:55 PM, mszlazak@aol.com (mszlazak@aol.com) wrote:
Quote:
Yes but someone else I'm in contact with set up FS a couple days ago and is having the same problems.
Brian should I still update today?




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Quote:
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brian at freeswitch.org
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PostPosted: Wed Dec 10, 2008 3:07 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.

/b

On Dec 10, 2008, at 1:51 PM, mszlazak@aol.com (mszlazak@aol.com) wrote:
Quote:
It was someone from Voxeo support. I think John was the main person helping me with this.

I updated but things got worse all over.
I now can't run other extensions Gino's pizza or some db stuff


2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!
2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey() Cannot allocate speech engine!
2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email]) [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email])) Ended
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email]) [CS_HANGUP]




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mszlazak at aol.com
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PostPosted: Wed Dec 10, 2008 4:34 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

Yup my bad.

But I'm still getting this error:

2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll
**The specified module could not be found. **

It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then?

I'm still having the same problem with no audio from Voxeo.

Mark.





-----Original Message-----
From: Brian West <brian@freeswitch.org>
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 10 Dec 2008 11:59 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo

Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.

/b

On Dec 10, 2008, at 1:51 PM, mszlazak@aol.com (mszlazak@aol.com) wrote:
Quote:
It was someone from Voxeo support. I think John was the main person helping me with this.

I updated but things got worse all over.
I now can't run other extensions Gino's pizza or some db stuff


2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!
2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey() Cannot allocate speech engine!
2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email]) [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email])) Ended
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email]) [CS_HANGUP]






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carlos.talbot at gmail...
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PostPosted: Wed Dec 10, 2008 6:00 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

There was a typecast warning that prevented spidermoneky from compiling in a recent svn. Did you check to see if it compiled?

Sent from my iPhone

On Dec 10, 2008, at 3:24 PM, mszlazak@aol.com (mszlazak@aol.com) wrote:



Quote:
Yup my bad.

But I'm still getting this error:

2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll
**The specified module could not be found. **

It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then?

I'm still having the same problem with no audio from Voxeo.

Mark.





-----Original Message-----
From: Brian West <brian@freeswitch.org (brian@freeswitch.org)>
To: [url=mailto:freeswitch-users@lists.freeswitch.org]freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)[/url]
Sent: Wed, 10 Dec 2008 11:59 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo

Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.

/b

On Dec 10, 2008, at 1:51 PM, [url=mailto:mszlazak@aol.com]mszlazak@aol.com (mszlazak@aol.com)[/url] wrote:
Quote:
It was someone from Voxeo support. I think John was the main person helping me with this.

I updated but things got worse all over.
I now can't run other extensions Gino's pizza or some db stuff


2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]!
2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey() Cannot allocate speech engine!
2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/[url=mailto:1000@10.0.0.2]1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email])[/url] [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/[url=mailto:1000@10.0.0.2]1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email])[/url]) Ended
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/[url=mailto:1000@10.0.0.2]1000@10.0.0.2 ([email]sofia/internal/1000@10.0.0.2[/email])[/url] [CS_HANGUP]






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PostPosted: Wed Dec 10, 2008 6:50 pm    Post subject: [Freeswitch-users] Audio routing problem between FS and Voxe Reply with quote

I only glanced at the compilers output when it ended and it reported no no errors but I did not look at the warnings along the way. Have things been modified since so spider monkey compiles?

More importantly, do you have any ideas as what is going on with my audio problem. I can't attach a wireshark .pcap file since it's to big for your list and my email gets rejected.




-----Original Message-----
From: Carlos Talbot <carlos.talbot@gmail.com>
To: freeswitch-users@lists.freeswitch.org <freeswitch-users@lists.freeswitch.org>
Cc: freeswitch-users@lists.freeswitch.org <freeswitch-users@lists.freeswitch.org>
Sent: Wed, 10 Dec 2008 2:57 pm
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo

There was a typecast warning that prevented spidermoneky from compiling in a recent svn. Did you check to see if it compiled?

Sent from my iPhone

On Dec 10, 2008, at 3:24 PM, mszlazak@aol.com (mszlazak@aol.com) wrote:



Quote:
Yup my bad.

But I'm still getting this error:

2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll
**The specified module could not be found. **

It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then?

I'm still having the same problem with no audio from Voxeo.

Mark.





-----Original Message-----
From: Brian West <brian@freeswitch.org (brian@freeswitch.org)>
To: (freeswitch-users@lists.freeswitch.org)freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Sent: Wed, 10 Dec 2008 11:59 am
Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo

Looks like mod_pocketsphinx and mod_openmrcp isn't loaded.

/b

On Dec 10, 2008, at 1:51 PM, (mszlazak@aol.com)mszlazak@aol.com (mszlazak@aol.com) wrote:
Quote:
It was someone from Voxeo support. I think John was the main person helping me with this.

I updated but things got worse all over.
I now can't run other extensions Gino's pizza or some db stuff


2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]!
2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech=2 0module [openmrcp]!
2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module!
2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey() Cannot allocate speech engine!
2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/ ([email]sofia/internal/1000@10.0.0.2[/email])1000@10.0.0.2 (1000@10.0.0.2) [CS_EXECUTE] [NORMAL_CLEARING]
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/ ([email]sofia/internal/1000@10.0.0.2[/email])1000@10.0.0.2 (1000@10.0.0.2)) Ended
2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/ ([email]sofia/internal/1000@10.0.0.2[/email])1000@10.0.0.2 (1000@10.0.0.2) [CS_HANGUP]






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