Hello, I¹ve posted on the question on this topic before but I¹ll try again with more detail. I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com I¹m assuming that DTMF rc2833 is the default, but I can¹t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A. The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn¹t sure where to set it up. I added this to my dialplan <extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1"/> </condition> </extension> Should I add it to my sip trunk? Not sure how to troubleshoot this issue Thanks Sean
The start_dtmf app simply turns on listening in-band for DTMFs, so this should be just fine. Have bw.com enable in-band DTMFs and give it a test drive. -MC On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean@obscuradigital.com> wrote:
Hello,
I’ve posted on the question on this topic before but I’ll try again with more detail.
I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com
I’m assuming that DTMF rc2833 is the default, but I can’t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A.
The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn’t sure where to set it up. I added this to my dialplan
<extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1 "/> </condition> </extension>
Should I add it to my sip trunk?
Not sure how to troubleshoot this issue
Thanks Sean
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Yea still getting no love. Had bandwidth turn on inband but still no success. On 3/30/10 12:10 PM, "Michael Collins" <msc@freeswitch.org> wrote:
The start_dtmf app simply turns on listening in-band for DTMFs, so this should be just fine. Have bw.com <http://bw.com> enable in-band DTMFs and give it a test drive. -MC
On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean@obscuradigital.com> wrote:
Hello,
I¹ve posted on the question on this topic before but I¹ll try again with more detail.
I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com
I¹m assuming that DTMF rc2833 is the default, but I can¹t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A.
The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn¹t sure where to set it up. I added this to my dialplan
<extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1"/> </condition> </extension>
Should I add it to my sip trunk?
Not sure how to troubleshoot this issue
Thanks Sean
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Use wireshark to see (and playback) the DTMF coming down the line. For reliable inband I believe you need to be using 711 - so that is worth checking too. On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean@obscuradigital.com> wrote:
Yea still getting no love. Had bandwidth turn on inband but still no success.
On 3/30/10 12:10 PM, "Michael Collins" <msc@freeswitch.org> wrote:
The start_dtmf app simply turns on listening in-band for DTMFs, so this should be just fine. Have bw.com <http://bw.com> enable in-band DTMFs and give it a test drive.
-MC
On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean@obscuradigital.com> wrote:
Hello,
I’ve posted on the question on this topic before but I’ll try again with more detail.
I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com
I’m assuming that DTMF rc2833 is the default, but I can’t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A.
The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn’t sure where to set it up. I added this to my dialplan
<extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1 "/> </condition> </extension>
Should I add it to my sip trunk?
Not sure how to troubleshoot this issue
Thanks Sean
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Ok I¹ve run wireshark, but not sure what to look for. It¹s using codec 711 in the rtp stream, but not sure what to look for in regards to DTMF? Thanks Sean On 3/30/10 12:50 PM, "Phillip Jones" <pjintheusa@gmail.com> wrote:
Use wireshark to see (and playback) the DTMF coming down the line.
For reliable inband I believe you need to be using 711 - so that is worth checking too.
On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean@obscuradigital.com> wrote:
Yea still getting no love. Had bandwidth turn on inband but still no success.
On 3/30/10 12:10 PM, "Michael Collins" <msc@freeswitch.org <http://msc@freeswitch.org> > wrote:
The start_dtmf app simply turns on listening in-band for DTMFs, so this should be just fine. Have bw.com <http://bw.com> <http://bw.com> enable in-band DTMFs and give it a test drive.
-MC
On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean@obscuradigital.com <http://sean@obscuradigital.com> > wrote:
Hello,
I¹ve posted on the question on this topic before but I¹ll try again with more detail.
I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com
I¹m assuming that DTMF rc2833 is the default, but I can¹t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A.
The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn¹t sure where to set it up. I added this to my dialplan
<extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1"/> </condition> </extension>
Should I add it to my sip trunk?
Not sure how to troubleshoot this issue
Thanks Sean
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Wireshark lets you play back the audio of the call. It also shows any RFC2833 DTMF sequences. I'd start by listening to the audio. Do you hear any DTMFs? -MC On Tue, Mar 30, 2010 at 2:40 PM, Sean Holt <sean@obscuradigital.com> wrote:
Ok I’ve run wireshark, but not sure what to look for. It’s using codec 711 in the rtp stream, but not sure what to look for in regards to DTMF?
Thanks Sean
On 3/30/10 12:50 PM, "Phillip Jones" <pjintheusa@gmail.com> wrote:
Use wireshark to see (and playback) the DTMF coming down the line.
For reliable inband I believe you need to be using 711 - so that is worth checking too.
On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean@obscuradigital.com> wrote:
Yea still getting no love. Had bandwidth turn on inband but still no success.
On 3/30/10 12:10 PM, "Michael Collins" <msc@freeswitch.org < http://msc@freeswitch.org> > wrote:
The start_dtmf app simply turns on listening in-band for DTMFs, so this should be just fine. Have bw.com <http://bw.com> <http://bw.com> enable in-band DTMFs and give it a test drive.
-MC
On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean@obscuradigital.com < http://sean@obscuradigital.com> > wrote:
Hello,
I’ve posted on the question on this topic before but I’ll try again with more detail.
I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com
I’m assuming that DTMF rc2833 is the default, but I can’t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A.
The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn’t sure where to set it up. I added this to my dialplan
<extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1 "/> </condition> </extension>
Should I add it to my sip trunk?
Not sure how to troubleshoot this issue
Thanks Sean
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Unfortunately I don¹t see any reference to DTMF coming from outside source. It shows it¹s using 711 Thanks Sean On 3/30/10 12:50 PM, "Phillip Jones" <pjintheusa@gmail.com> wrote:
Use wireshark to see (and playback) the DTMF coming down the line.
For reliable inband I believe you need to be using 711 - so that is worth checking too.
On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean@obscuradigital.com> wrote:
Yea still getting no love. Had bandwidth turn on inband but still no success.
On 3/30/10 12:10 PM, "Michael Collins" <msc@freeswitch.org <http://msc@freeswitch.org> > wrote:
The start_dtmf app simply turns on listening in-band for DTMFs, so this should be just fine. Have bw.com <http://bw.com> <http://bw.com> enable in-band DTMFs and give it a test drive.
-MC
On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean@obscuradigital.com <http://sean@obscuradigital.com> > wrote:
Hello,
I¹ve posted on the question on this topic before but I¹ll try again with more detail.
I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com
I¹m assuming that DTMF rc2833 is the default, but I can¹t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A.
The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn¹t sure where to set it up. I added this to my dialplan
<extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1"/> </condition> </extension>
Should I add it to my sip trunk?
Not sure how to troubleshoot this issue
Thanks Sean
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On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean@obscuradigital.com> wrote:
Unfortunately I don’t see any reference to DTMF coming from outside source.
It shows it’s using 711
Thanks Sean
What about listening to the call? Do you hear any DTMFs in the audio stream? If not then it looks like the far end isn't sending DTMFs which means you will need to talk to your provider to continue troubleshooting. -MC
Thanks man... Looking closer at the wireshark capture I¹m only seeing RTP going out and nothing coming. I¹ll contact Bandwidth again Sean On 3/30/10 4:11 PM, "Michael Collins" <msc@freeswitch.org> wrote:
On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean@obscuradigital.com> wrote:
Unfortunately I don¹t see any reference to DTMF coming from outside source.
It shows it¹s using 711
Thanks Sean
What about listening to the call? Do you hear any DTMFs in the audio stream? If not then it looks like the far end isn't sending DTMFs which means you will need to talk to your provider to continue troubleshooting.
-MC
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So it appears that Bandwidth requires the audio to be routed on the external public IP address, but since my freeswitch pbx box is behind a NAT the audio is trying to route to a private IP. The tech at bandwidth says that won¹t work. Not sure how to correct this? Thanks Sean On 3/30/10 4:11 PM, "Michael Collins" <msc@freeswitch.org> wrote:
On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean@obscuradigital.com> wrote:
Unfortunately I don¹t see any reference to DTMF coming from outside source.
It shows it¹s using 711
Thanks Sean
What about listening to the call? Do you hear any DTMFs in the audio stream? If not then it looks like the far end isn't sending DTMFs which means you will need to talk to your provider to continue troubleshooting.
-MC
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On Tue, Mar 30, 2010 at 5:09 PM, Sean Holt <sean@obscuradigital.com> wrote:
So it appears that Bandwidth requires the audio to be routed on the external public IP address, but since my freeswitch pbx box is behind a NAT the audio is trying to route to a private IP. The tech at bandwidth says that won’t work.
Not sure how to correct this?
Thanks Sean
I strongly recommend you read up on NAT: http://wiki.freeswitch.org/wiki/Nat
and http://wiki.freeswitch.org/wiki/NAT_Traversal and http://wiki.freeswitch.org/wiki/Auto_NAT You need to set up your external profile properly, which means you need to understand your network and your NAT device. Did you turn off any SIP ALG's in your device? Do you have a static public IP address? Those are important questions to answer before you continue. -MC
On 3/30/10 4:11 PM, "Michael Collins" <msc@freeswitch.org> wrote:
On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean@obscuradigital.com> wrote:
Unfortunately I don’t see any reference to DTMF coming from outside source.
It shows it’s using 711
Thanks Sean
What about listening to the call? Do you hear any DTMFs in the audio stream? If not then it looks like the far end isn't sending DTMFs which means you will need to talk to your provider to continue troubleshooting.
-MC
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Thanks michael for all the help. I¹ve seem to have fixed this by adding the public IP address in my sip profile file under the ext-rtp-ip and ext-sip-ip param. Now I can see the DTMF tones come in while capturing. Sean On 3/30/10 5:29 PM, "Michael Collins" <msc@freeswitch.org> wrote:
On Tue, Mar 30, 2010 at 5:09 PM, Sean Holt <sean@obscuradigital.com> wrote:
So it appears that Bandwidth requires the audio to be routed on the external public IP address, but since my freeswitch pbx box is behind a NAT the audio is trying to route to a private IP. The tech at bandwidth says that won¹t work.
Not sure how to correct this?
Thanks Sean
I strongly recommend you read up on NAT: http://wiki.freeswitch.org/wiki/Nat
and
http://wiki.freeswitch.org/wiki/NAT_Traversal
and
http://wiki.freeswitch.org/wiki/Auto_NAT
You need to set up your external profile properly, which means you need to understand your network and your NAT device. Did you turn off any SIP ALG's in your device? Do you have a static public IP address? Those are important questions to answer before you continue.
-MC
On 3/30/10 4:11 PM, "Michael Collins" <msc@freeswitch.org <http://msc@freeswitch.org> > wrote:
On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean@obscuradigital.com <http://sean@obscuradigital.com> > wrote:
Unfortunately I don¹t see any reference to DTMF coming from outside source.
It shows it¹s using 711
Thanks Sean
What about listening to the call? Do you hear any DTMFs in the audio stream? If not then it looks like the far end isn't sending DTMFs which means you will need to talk to your provider to continue troubleshooting.
-MC
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participants (3)
-
Michael Collins
-
Phillip Jones
-
Sean Holt