IVR codec issue (I think)
HI Everybody! I had a recent problem where a transferred call's audio was highly degraded. The cause was identified as mismatched codecs - thanks again. Today, I have a VERY VERY similar problem. The difference is that the first step is an IVR, not the receptionist. How do I set the CODEC for an incoming call to 711u when it is routed to an IVR. Thanks, Sean
Try: export data="absolute_codec_string=PCMU" before the answer in the dialplan. After answer, the codec is already established. Ian. On Wed, May 21, 2014 at 5:22 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
HI Everybody!
I had a recent problem where a transferred call’s audio was highly degraded. The cause was identified as mismatched codecs – thanks again.
Today, I have a VERY VERY similar problem. The difference is that the first step is an IVR, not the receptionist. How do I set the CODEC for an incoming call to 711u when it is routed to an IVR.
Thanks,
Sean
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
If possible, try to use RFC-2833 for the DTMF and you shouldn't have any codec problems. 711u is only required if you are doing INBAND DTMF. Guillermo On Wed, May 21, 2014 at 6:35 PM, Ian McMaster <ian.mcmaster@gmail.com>wrote:
Try:
export data="absolute_codec_string=PCMU"
before the answer in the dialplan.
After answer, the codec is already established.
Ian.
On Wed, May 21, 2014 at 5:22 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
HI Everybody!
I had a recent problem where a transferred call’s audio was highly degraded. The cause was identified as mismatched codecs – thanks again.
Today, I have a VERY VERY similar problem. The difference is that the first step is an IVR, not the receptionist. How do I set the CODEC for an incoming call to 711u when it is routed to an IVR.
Thanks,
Sean
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
-- Guillermo Ruiz Camauer
Hi Guillermo, Thank you for your response. I wish understood more, but let me tell what I know and ask you another question. My (voice) ISP only supports “ULaw, ALaw, GSM and G729a” My phones are all Cisco504G’s. They support “G711u, G711a, G729a, G722, G726-16 (-24, -24 and -32). Then there are 2 other items that must be relevant: “DTMF Process AVT: Yes or No” and “DTMF Tx method: Inband, AVT, Info, Auto, Inband + Info, AVT + Info” I think Tx Method defaulted to AVT+Info as I got warnings about receive DTMF in INFO on FS. I am set to AVT and DTMF is working fine. Here is what I think (I would know, but it may not be correct) about codecs: G711u = ULaw, G711a=ALaw, The phones don’t seem to support GSM (?) and my VISP doesn’t support 722 or any 726. So that leaves me with G711a, G711u, G729. I have abundant bandwidth everywhere involved. I know G729 is fairly high quality compression with regard to bandwidth vs audio quality. I also know I have to get a license for each G729 channel on FS for about $10 each. I do not understand the difference in G711u and G711a. What is my best choice for DTMF Tx Method? And codec? Thank you for your time. Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer Sent: Wednesday, May 21, 2014 7:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] IVR codec issue (I think) If possible, try to use RFC-2833 for the DTMF and you shouldn't have any codec problems. 711u is only required if you are doing INBAND DTMF. Guillermo On Wed, May 21, 2014 at 6:35 PM, Ian McMaster <ian.mcmaster@gmail.com<mailto:ian.mcmaster@gmail.com>> wrote: Try: export data="absolute_codec_string=PCMU" before the answer in the dialplan. After answer, the codec is already established. Ian. On Wed, May 21, 2014 at 5:22 PM, Sean Devoy <sdevoy@bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote: [cid:image001.gif@01CF7533.E75C4700] HI Everybody! I had a recent problem where a transferred call’s audio was highly degraded. The cause was identified as mismatched codecs – thanks again. Today, I have a VERY VERY similar problem. The difference is that the first step is an IVR, not the receptionist. How do I set the CODEC for an incoming call to 711u when it is routed to an IVR. Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 -- Guillermo Ruiz Camauer
AVT seems to be another way of referring to RFC2833. This is a way of sending the DTMFs out of band. That is, through a separate data stream than the voice. Just like SIP carried the signaling and RTP carries the voice. RFC 2833 has several advantages over INBand DTMFs. INBand DTMFs must be first extracted from the voice stream and then interpreted, which consumes much more CPU than just reading a number. The DTMF decoders will also get confused some times with people with high pitched voices and generate spurious DTMFs. Also, if the RTP packets follow a different route that the SIP packets, your FS server will not be able to listen for DTMFs once the call has been established (because the RTP packets are not passing through FS). RFC 2833 signaling goes together with SIP, and therefore will always be available to FS, so it will be able to monitor key presses and react accordingly. So if bandwidth is not a problem, use G711u, set Provess AVT to yes, and DTMF Tx method to AVT (only) on the Ciscos, and RFC2833 on the FS side sand you'll be fine. Guillermo Sent from my iPhone
On 21/05/2014, at 21:33, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi Guillermo,
Thank you for your response. I wish understood more, but let me tell what I know and ask you another question.
My (voice) ISP only supports “ULaw, ALaw, GSM and G729a” My phones are all Cisco504G’s. They support “G711u, G711a, G729a, G722, G726-16 (-24, -24 and -32). Then there are 2 other items that must be relevant: “DTMF Process AVT: Yes or No” and “DTMF Tx method: Inband, AVT, Info, Auto, Inband + Info, AVT + Info”
I think Tx Method defaulted to AVT+Info as I got warnings about receive DTMF in INFO on FS. I am set to AVT and DTMF is working fine.
Here is what I think (I would know, but it may not be correct) about codecs: G711u = ULaw, G711a=ALaw, The phones don’t seem to support GSM (?) and my VISP doesn’t support 722 or any 726. So that leaves me with G711a, G711u, G729. I have abundant bandwidth everywhere involved. I know G729 is fairly high quality compression with regard to bandwidth vs audio quality. I also know I have to get a license for each G729 channel on FS for about $10 each.
I do not understand the difference in G711u and G711a.
What is my best choice for DTMF Tx Method? And codec?
Thank you for your time. Sean
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer Sent: Wednesday, May 21, 2014 7:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] IVR codec issue (I think)
If possible, try to use RFC-2833 for the DTMF and you shouldn't have any codec problems. 711u is only required if you are doing INBAND DTMF.
Guillermo
On Wed, May 21, 2014 at 6:35 PM, Ian McMaster <ian.mcmaster@gmail.com> wrote: Try:
export data="absolute_codec_string=PCMU"
before the answer in the dialplan.
After answer, the codec is already established.
Ian.
On Wed, May 21, 2014 at 5:22 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: <image001.gif> HI Everybody!
I had a recent problem where a transferred call’s audio was highly degraded. The cause was identified as mismatched codecs – thanks again.
Today, I have a VERY VERY similar problem. The difference is that the first step is an IVR, not the receptionist. How do I set the CODEC for an incoming call to 711u when it is routed to an IVR.
Thanks, Sean
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
-- Guillermo Ruiz Camauer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Excellent write-up but I have one small correction. RFC2833 is another payload type in the same RTP stream as the audio. It follows RTP, not SIP. On Wednesday, May 21, 2014, Guillermo Ruiz Camauer <grcamauer@gmail.com> wrote:
AVT seems to be another way of referring to RFC2833. This is a way of sending the DTMFs out of band. That is, through a separate data stream than the voice. Just like SIP carried the signaling and RTP carries the voice. RFC 2833 has several advantages over INBand DTMFs. INBand DTMFs must be first extracted from the voice stream and then interpreted, which consumes much more CPU than just reading a number. The DTMF decoders will also get confused some times with people with high pitched voices and generate spurious DTMFs. Also, if the RTP packets follow a different route that the SIP packets, your FS server will not be able to listen for DTMFs once the call has been established (because the RTP packets are not passing through FS). RFC 2833 signaling goes together with SIP, and therefore will always be available to FS, so it will be able to monitor key presses and react accordingly. So if bandwidth is not a problem, use G711u, set Provess AVT to yes, and DTMF Tx method to AVT (only) on the Ciscos, and RFC2833 on the FS side sand you'll be fine.
Guillermo
Sent from my iPhone
On 21/05/2014, at 21:33, Sean Devoy <sdevoy@bizfocused.com<javascript:_e(%7B%7D,'cvml','sdevoy@bizfocused.com');>> wrote:
Hi Guillermo,
Thank you for your response. I wish understood more, but let me tell what I know and ask you another question.
My (voice) ISP only supports “ULaw, ALaw, GSM and G729a”
My phones are all Cisco504G’s. They support “G711u, G711a, G729a, G722, G726-16 (-24, -24 and -32). Then there are 2 other items that must be relevant: “DTMF Process AVT: Yes or No” and “DTMF Tx method: Inband, AVT, Info, Auto, Inband + Info, AVT + Info”
I think Tx Method defaulted to AVT+Info as I got warnings about receive DTMF in INFO on FS. I am set to AVT and DTMF is working fine.
Here is what I think (I would know, but it may not be correct) about codecs:
G711u = ULaw, G711a=ALaw, The phones don’t seem to support GSM (?) and my VISP doesn’t support 722 or any 726. So that leaves me with G711a, G711u, G729. I have abundant bandwidth everywhere involved. I know G729 is fairly high quality compression with regard to bandwidth vs audio quality. I also know I have to get a license for each G729 channel on FS for about $10 each.
I do not understand the difference in G711u and G711a.
What is my best choice for DTMF Tx Method? And codec?
Thank you for your time.
Sean
*From:* freeswitch-users-bounces@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');>[ mailto:freeswitch-users-bounces@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');>] *On Behalf Of *Guillermo Ruiz Camauer *Sent:* Wednesday, May 21, 2014 7:53 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] IVR codec issue (I think)
If possible, try to use RFC-2833 for the DTMF and you shouldn't have any codec problems. 711u is only required if you are doing INBAND DTMF.
Guillermo
On Wed, May 21, 2014 at 6:35 PM, Ian McMaster <ian.mcmaster@gmail.com<javascript:_e(%7B%7D,'cvml','ian.mcmaster@gmail.com');>> wrote:
Try:
export data="absolute_codec_string=PCMU"
before the answer in the dialplan.
After answer, the codec is already established.
Ian.
On Wed, May 21, 2014 at 5:22 PM, Sean Devoy <sdevoy@bizfocused.com<javascript:_e(%7B%7D,'cvml','sdevoy@bizfocused.com');>> wrote:
<image001.gif>
HI Everybody!
I had a recent problem where a transferred call’s audio was highly degraded. The cause was identified as mismatched codecs – thanks again.
Today, I have a VERY VERY similar problem. The difference is that the first step is an IVR, not the receptionist. How do I set the CODEC for an incoming call to 711u when it is routed to an IVR.
Thanks,
Sean
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','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
-- Guillermo Ruiz Camauer
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','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
-- Sent from mobile device
Amen. On May 22, 2014, at 1:46 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Excellent write-up but I have one small correction. RFC2833 is another payload type in the same RTP stream as the audio. It follows RTP, not SIP.
On Wednesday, May 21, 2014, Guillermo Ruiz Camauer <grcamauer@gmail.com> wrote: AVT seems to be another way of referring to RFC2833. This is a way of sending the DTMFs out of band. That is, through a separate data stream than the voice. Just like SIP carried the signaling and RTP carries the voice. RFC 2833 has several advantages over INBand DTMFs. INBand DTMFs must be first extracted from the voice stream and then interpreted, which consumes much more CPU than just reading a number. The DTMF decoders will also get confused some times with people with high pitched voices and generate spurious DTMFs. Also, if the RTP packets follow a different route that the SIP packets, your FS server will not be able to listen for DTMFs once the call has been established (because the RTP packets are not passing through FS). RFC 2833 signaling goes together with SIP, and therefore will always be available to FS, so it will be able to monitor key presses and react accordingly. So if bandwidth is not a problem, use G711u, set Provess AVT to yes, and DTMF Tx method to AVT (only) on the Ciscos, and RFC2833 on the FS side sand you'll be fine.
Guillermo
Sent from my iPhone
On 21/05/2014, at 21:33, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi Guillermo,
Thank you for your response. I wish understood more, but let me tell what I know and ask you another question.
My (voice) ISP only supports “ULaw, ALaw, GSM and G729a”
My phones are all Cisco504G’s. They support “G711u, G711a, G729a, G722, G726-16 (-24, -24 and -32). Then there are 2 other items that must be relevant: “DTMF Process AVT: Yes or No” and “DTMF Tx method: Inband, AVT, Info, Auto, Inband + Info, AVT + Info”
I think Tx Method defaulted to AVT+Info as I got warnings about receive DTMF in INFO on FS. I am set to AVT and DTMF is working fine.
Here is what I think (I would know, but it may not be correct) about codecs:
G711u = ULaw, G711a=ALaw, The phones don’t seem to support GSM (?) and my VISP doesn’t support 722 or any 726. So that leaves me with G711a, G711u, G729. I have abundant bandwidth everywhere involved. I know G729 is fairly high quality compression with regard to bandwidth vs audio quality. I also know I have to get a license for each G729 channel on FS for about $10 each.
I do not understand the difference in G711u and G711a.
What is my best choice for DTMF Tx Method? And codec?
Thank you for your time.
Sean
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer Sent: Wednesday, May 21, 2014 7:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] IVR codec issue (I think)
If possible, try to use RFC-2833 for the DTMF and you shouldn't have any codec problems. 711u is only required if you are doing INBAND DTMF.
Guillermo
On Wed, May 21, 2014 at 6:35 PM, Ian McMaster <ian.mcmaster@gmail.com> wrote:
Try:
export data="absolute_codec_string=PCMU"
before the answer in the dialplan.
After answer, the codec is already established.
Ian.
On Wed, May 21, 2014 at 5:22 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
<image001.gif>
HI Everybody!
I had a recent problem where a transferred call’s audio was highly degraded. The cause was identified as mismatched codecs – thanks again.
Today, I have a VERY VERY similar problem. The difference is that the first step is an IVR, not the receptionist. How do I set the CODEC for an incoming call to 711u when it is routed to an IVR.
Thanks,
Sean
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
-- Guillermo Ruiz Camauer
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
-- Sent from mobile device _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Many thanks Ian. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ian McMaster Sent: Wednesday, May 21, 2014 5:36 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] IVR codec issue (I think) Try: export data="absolute_codec_string=PCMU" before the answer in the dialplan. After answer, the codec is already established. Ian. On Wed, May 21, 2014 at 5:22 PM, Sean Devoy <sdevoy@bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote: [cid:image001.gif@01CF7530.9C1E86C0] HI Everybody! I had a recent problem where a transferred call’s audio was highly degraded. The cause was identified as mismatched codecs – thanks again. Today, I have a VERY VERY similar problem. The difference is that the first step is an IVR, not the receptionist. How do I set the CODEC for an incoming call to 711u when it is routed to an IVR. Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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
participants (5)
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Guillermo Ruiz Camauer
-
Ian McMaster
-
Kristian Kielhofner
-
Sean Devoy
-
servtelar