httapi - dial/bridge only working when it comes first
I'm using httapi and everything is working well for the most part. The one thing I noticed is that whenever I do anything before trying to "Dial" or bridge a call to the gateway: 1. The first thing never happens, even though the console says it did. 2. The outbound call "Dial" won't work, it hangs for about 60 seconds and then returns a NORMAL_TEMPORARY_FAILURE every time. Things to note: - If I start with the dial request (B below) then it connects the call fine, it just can't come after anything else. - If I never return a request to dial (pretend B below starts a conference instead) then everything works fine, the TTS works before and the conference works after. Here is an example: A. When FS makes it's first request to my URL I return this: <document type="text/freeswitch-httapi"> <work> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document> B: When FS makes it's next request to my action URL (same URL with different parameters) I return the dial command and attempt to connect to the outbound gateway. <document type="text/freeswitch-httapi"> <work> <!-- Hold music URL --> <execute application="set" data="ringback=http://blabla.com/beat.wav "></execute> <!-- Set the caller ID information --> <execute application="set" data="effective_caller_id_name=My Name"></execute> <execute application="set" data="effective_caller_id_number=12144635555"></execute> <!-- The actual call or calls, entrie string separated by commas --> <execute application="export" data="nolocal:codec_string=G711"></execute> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> <execute application="answer"/> </execute> </work> </document> What happens? It never speaks the text (although the console says it did) and when dialing out I get "NORMAL_TEMPORARY_FAILURE" and the call never connects. Keep in mind, if the second thing I return (B above) is NOT dial then everything works fine, it will speak the text, play a file, start a conference, or any other application. This is the weirdest thing to me and of course almost everything is useless if you can't connect a call at some point, if anyone has any advice I'd be highly appreciative.
Anyone? On Wed, Jul 27, 2016 at 7:54 PM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
I'm using httapi and everything is working well for the most part. The one thing I noticed is that whenever I do anything before trying to "Dial" or bridge a call to the gateway:
1. The first thing never happens, even though the console says it did. 2. The outbound call "Dial" won't work, it hangs for about 60 seconds and then returns a NORMAL_TEMPORARY_FAILURE every time.
Things to note:
- If I start with the dial request (B below) then it connects the call fine, it just can't come after anything else. - If I never return a request to dial (pretend B below starts a conference instead) then everything works fine, the TTS works before and the conference works after.
Here is an example:
A. When FS makes it's first request to my URL I return this:
<document type="text/freeswitch-httapi"> <work> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document>
B: When FS makes it's next request to my action URL (same URL with different parameters) I return the dial command and attempt to connect to the outbound gateway.
<document type="text/freeswitch-httapi"> <work> <!-- Hold music URL --> <execute application="set" data="ringback=http://blabla.com/beat.wav "></execute> <!-- Set the caller ID information --> <execute application="set" data="effective_caller_id_name=My Name"></execute> <execute application="set" data="effective_caller_id_number=12144635555 "></execute>
<!-- The actual call or calls, entrie string separated by commas --> <execute application="export" data="nolocal:codec_string=G711"></execute> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> <execute application="answer"/> </execute> </work> </document>
What happens? It never speaks the text (although the console says it did) and when dialing out I get "NORMAL_TEMPORARY_FAILURE" and the call never connects. Keep in mind, if the second thing I return (B above) is NOT dial then everything works fine, it will speak the text, play a file, start a conference, or any other application.
This is the weirdest thing to me and of course almost everything is useless if you can't connect a call at some point, if anyone has any advice I'd be highly appreciative.
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
I have made some progress here. I added the following lines to the dial plan: <action application="pre_answer"/> <action application="sleep" data="2000"/> (this one was the trick) Now, I can speak TTS, or play a file, or any other application before bridging the call to a gateway. However, when bridging the call I'm still getting "NORMAL_TEMPORARY_FAILURE". And of course, if I call bridge the call first (before any other application) then it connects the call with no problem. Any ideas?
Don- maybe try: 1: <document type='xml/freeswitch-httapi'> <params/> <variables/> <work> <execute application="answer"/> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document> 2: <document type='xml/freeswitch-httapi'> <params/> <variables> <ringback>http://blabla.com/beat.wav</ringback> <effective_caller_id_name>My Name</effective_caller_id_name> <effective_caller_id_number>2144635555</effective_caller_id_number> <ringback>http://blabla.com/beat.wav</ringback> </variables> <work> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> </work> </document> On Thu, Jul 28, 2016 at 11:38 AM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
Anyone?
On Wed, Jul 27, 2016 at 7:54 PM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
I'm using httapi and everything is working well for the most part. The one thing I noticed is that whenever I do anything before trying to "Dial" or bridge a call to the gateway:
1. The first thing never happens, even though the console says it did. 2. The outbound call "Dial" won't work, it hangs for about 60 seconds and then returns a NORMAL_TEMPORARY_FAILURE every time.
Things to note:
- If I start with the dial request (B below) then it connects the call fine, it just can't come after anything else. - If I never return a request to dial (pretend B below starts a conference instead) then everything works fine, the TTS works before and the conference works after.
Here is an example:
A. When FS makes it's first request to my URL I return this:
<document type="text/freeswitch-httapi"> <work> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document>
B: When FS makes it's next request to my action URL (same URL with different parameters) I return the dial command and attempt to connect to the outbound gateway.
<document type="text/freeswitch-httapi"> <work> <!-- Hold music URL --> <execute application="set" data="ringback=http://blabla.com/beat.wav "></execute> <!-- Set the caller ID information --> <execute application="set" data="effective_caller_id_name=My Name"></execute> <execute application="set" data="effective_caller_id_number=12144635555 "></execute>
<!-- The actual call or calls, entrie string separated by commas --> <execute application="export" data="nolocal:codec_string=G711"></execute> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> <execute application="answer"/> </execute> </work> </document>
What happens? It never speaks the text (although the console says it did) and when dialing out I get "NORMAL_TEMPORARY_FAILURE" and the call never connects. Keep in mind, if the second thing I return (B above) is NOT dial then everything works fine, it will speak the text, play a file, start a conference, or any other application.
This is the weirdest thing to me and of course almost everything is useless if you can't connect a call at some point, if anyone has any advice I'd be highly appreciative.
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Tony Bourdeaux *Intelecenter, LLC* ph: 805-428-3031 Skype: tony.bourdeaux "This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
I think this is actually a completely different problem than I though. I tried to connect a call with X-Lite Softphone on my Windows PC and FAILED (I've been testing with this). I tried to connect a call with ZoiPer on my Android Softphone and it was a SUCCESS. So, there seems to be some kind of codec issue going on here. Any ideas? On Fri, Jul 29, 2016 at 11:10 AM, Tony Bourdeaux <tony@intelecenter.com> wrote:
Don-
maybe try:
1:
<document type='xml/freeswitch-httapi'> <params/> <variables/> <work> <execute application="answer"/> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document>
2: <document type='xml/freeswitch-httapi'> <params/> <variables> <ringback>http://blabla.com/beat.wav</ringback> <effective_caller_id_name>My Name</effective_caller_id_name> <effective_caller_id_number>2144635555</effective_caller_id_number> <ringback>http://blabla.com/beat.wav</ringback> </variables> <work> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> </work> </document>
On Thu, Jul 28, 2016 at 11:38 AM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
Anyone?
On Wed, Jul 27, 2016 at 7:54 PM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
I'm using httapi and everything is working well for the most part. The one thing I noticed is that whenever I do anything before trying to "Dial" or bridge a call to the gateway:
1. The first thing never happens, even though the console says it did. 2. The outbound call "Dial" won't work, it hangs for about 60 seconds and then returns a NORMAL_TEMPORARY_FAILURE every time.
Things to note:
- If I start with the dial request (B below) then it connects the call fine, it just can't come after anything else. - If I never return a request to dial (pretend B below starts a conference instead) then everything works fine, the TTS works before and the conference works after.
Here is an example:
A. When FS makes it's first request to my URL I return this:
<document type="text/freeswitch-httapi"> <work> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document>
B: When FS makes it's next request to my action URL (same URL with different parameters) I return the dial command and attempt to connect to the outbound gateway.
<document type="text/freeswitch-httapi"> <work> <!-- Hold music URL --> <execute application="set" data="ringback=http://blabla.com/beat.wav "></execute> <!-- Set the caller ID information --> <execute application="set" data="effective_caller_id_name=My Name"></execute> <execute application="set" data="effective_caller_id_number=12144635555 "></execute>
<!-- The actual call or calls, entrie string separated by commas --> <execute application="export" data="nolocal:codec_string=G711"></execute> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> <execute application="answer"/> </execute> </work> </document>
What happens? It never speaks the text (although the console says it did) and when dialing out I get "NORMAL_TEMPORARY_FAILURE" and the call never connects. Keep in mind, if the second thing I return (B above) is NOT dial then everything works fine, it will speak the text, play a file, start a conference, or any other application.
This is the weirdest thing to me and of course almost everything is useless if you can't connect a call at some point, if anyone has any advice I'd be highly appreciative.
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
--
Tony Bourdeaux
*Intelecenter, LLC*
ph: 805-428-3031
Skype: tony.bourdeaux
"This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
This issue is now resolved, with the beautiful little line of XML below: <execute application="export" data="nolocal:absolute_codec_string=PCMU"></execute> It seems as if this provider wanted only PCMU, although I did look this up before, I had PCMU in the list along with quite a few other codec's they "said" they supported. On Tue, Aug 2, 2016 at 11:32 PM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
I think this is actually a completely different problem than I though.
I tried to connect a call with X-Lite Softphone on my Windows PC and FAILED (I've been testing with this). I tried to connect a call with ZoiPer on my Android Softphone and it was a SUCCESS.
So, there seems to be some kind of codec issue going on here. Any ideas?
On Fri, Jul 29, 2016 at 11:10 AM, Tony Bourdeaux <tony@intelecenter.com> wrote:
Don-
maybe try:
1:
<document type='xml/freeswitch-httapi'> <params/> <variables/> <work> <execute application="answer"/> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document>
2: <document type='xml/freeswitch-httapi'> <params/> <variables> <ringback>http://blabla.com/beat.wav</ringback> <effective_caller_id_name>My Name</effective_caller_id_name> <effective_caller_id_number>2144635555</effective_caller_id_number> <ringback>http://blabla.com/beat.wav</ringback> </variables> <work> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> </work> </document>
On Thu, Jul 28, 2016 at 11:38 AM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
Anyone?
On Wed, Jul 27, 2016 at 7:54 PM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
I'm using httapi and everything is working well for the most part. The one thing I noticed is that whenever I do anything before trying to "Dial" or bridge a call to the gateway:
1. The first thing never happens, even though the console says it did. 2. The outbound call "Dial" won't work, it hangs for about 60 seconds and then returns a NORMAL_TEMPORARY_FAILURE every time.
Things to note:
- If I start with the dial request (B below) then it connects the call fine, it just can't come after anything else. - If I never return a request to dial (pretend B below starts a conference instead) then everything works fine, the TTS works before and the conference works after.
Here is an example:
A. When FS makes it's first request to my URL I return this:
<document type="text/freeswitch-httapi"> <work> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document>
B: When FS makes it's next request to my action URL (same URL with different parameters) I return the dial command and attempt to connect to the outbound gateway.
<document type="text/freeswitch-httapi"> <work> <!-- Hold music URL --> <execute application="set" data="ringback=http://blabla.com/beat.wav "></execute> <!-- Set the caller ID information --> <execute application="set" data="effective_caller_id_name=My Name"></execute> <execute application="set" data="effective_caller_id_number=12144635555 "></execute>
<!-- The actual call or calls, entrie string separated by commas --> <execute application="export" data="nolocal:codec_string=G711"></execute> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> <execute application="answer"/> </execute> </work> </document>
What happens? It never speaks the text (although the console says it did) and when dialing out I get "NORMAL_TEMPORARY_FAILURE" and the call never connects. Keep in mind, if the second thing I return (B above) is NOT dial then everything works fine, it will speak the text, play a file, start a conference, or any other application.
This is the weirdest thing to me and of course almost everything is useless if you can't connect a call at some point, if anyone has any advice I'd be highly appreciative.
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
--
Tony Bourdeaux
*Intelecenter, LLC*
ph: 805-428-3031
Skype: tony.bourdeaux
"This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
you can turn on SIP trace to see your SIP messages in fs cli (and the CODEC's used): sofia profile <profilename> siptrace on|off OR sofia global siptrace [on|off] On Tue, Aug 2, 2016 at 9:32 PM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
I think this is actually a completely different problem than I though.
I tried to connect a call with X-Lite Softphone on my Windows PC and FAILED (I've been testing with this). I tried to connect a call with ZoiPer on my Android Softphone and it was a SUCCESS.
So, there seems to be some kind of codec issue going on here. Any ideas?
On Fri, Jul 29, 2016 at 11:10 AM, Tony Bourdeaux <tony@intelecenter.com> wrote:
Don-
maybe try:
1:
<document type='xml/freeswitch-httapi'> <params/> <variables/> <work> <execute application="answer"/> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document>
2: <document type='xml/freeswitch-httapi'> <params/> <variables> <ringback>http://blabla.com/beat.wav</ringback> <effective_caller_id_name>My Name</effective_caller_id_name> <effective_caller_id_number>2144635555</effective_caller_id_number> <ringback>http://blabla.com/beat.wav</ringback> </variables> <work> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> </work> </document>
On Thu, Jul 28, 2016 at 11:38 AM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
Anyone?
On Wed, Jul 27, 2016 at 7:54 PM, Don Hawkins <hawkins@hawkinsegroup.com> wrote:
I'm using httapi and everything is working well for the most part. The one thing I noticed is that whenever I do anything before trying to "Dial" or bridge a call to the gateway:
1. The first thing never happens, even though the console says it did. 2. The outbound call "Dial" won't work, it hangs for about 60 seconds and then returns a NORMAL_TEMPORARY_FAILURE every time.
Things to note:
- If I start with the dial request (B below) then it connects the call fine, it just can't come after anything else. - If I never return a request to dial (pretend B below starts a conference instead) then everything works fine, the TTS works before and the conference works after.
Here is an example:
A. When FS makes it's first request to my URL I return this:
<document type="text/freeswitch-httapi"> <work> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla..." name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document>
B: When FS makes it's next request to my action URL (same URL with different parameters) I return the dial command and attempt to connect to the outbound gateway.
<document type="text/freeswitch-httapi"> <work> <!-- Hold music URL --> <execute application="set" data="ringback=http://blabla.com/beat.wav "></execute> <!-- Set the caller ID information --> <execute application="set" data="effective_caller_id_name=My Name"></execute> <execute application="set" data="effective_caller_id_number=12144635555 "></execute>
<!-- The actual call or calls, entrie string separated by commas --> <execute application="export" data="nolocal:codec_string=G711"></execute> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabla... " application="bridge" data="sofia/gatewayAlcazar/12144635555"> <execute application="answer"/> </execute> </work> </document>
What happens? It never speaks the text (although the console says it did) and when dialing out I get "NORMAL_TEMPORARY_FAILURE" and the call never connects. Keep in mind, if the second thing I return (B above) is NOT dial then everything works fine, it will speak the text, play a file, start a conference, or any other application.
This is the weirdest thing to me and of course almost everything is useless if you can't connect a call at some point, if anyone has any advice I'd be highly appreciative.
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
--
Tony Bourdeaux
*Intelecenter, LLC*
ph: 805-428-3031
Skype: tony.bourdeaux
"This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Tony Bourdeaux *Intelecenter, LLC* ph: 805-428-3031 Skype: tony.bourdeaux "This message and any attachments are solely for the intended recipient and may contain confidential or privileged information. If you are not the intended recipient, any disclosure, copying, use, or distribution of the information included in this message and any attachments is prohibited. If you have received this communication in error, please notify me by reply e-mail and immediately and permanently delete this message and any attachments."
participants (2)
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Don Hawkins
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Tony Bourdeaux