Skypiax- how to know which card to use?
Hey guys, I am attempting to setup Skypiax and Freeswitch on Ubuntu Hardy. I've had a couple of previous problems which updating to SVN trunk release seemed to resolve, but I've got one lingering one to ask about. When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS picks up and rings the desired extension. No problems there. What happens though is, the person that is on the extension in the office can hear the calling party just fine, but the calling party hears stuttered choppy sound. I've already tested my internet connection, and through QOS have assigned my test box 512K up and 512K down, so on one test call bandwidth should not be an issue. What do you think causes the stuttering on my audio for the calling party? Do you think it might be a codec? I have everything setup right now to use g711u. I'd like to use something smaller and higher quality, but I'm not sure what is a better option. I had made some phone calls as listed above and they tested out just fine, but in that same time frame I also recompiled Alsa drivers with the Skypiax dummy file, so do you think there might be something in that causing the stutter? Looking for some advice. Thanks, Doc
On Fri, Feb 19, 2010 at 10:30 PM, Doc <ledoktre@meanie.us> wrote:
I am attempting to setup Skypiax and Freeswitch on Ubuntu Hardy. I've had a couple of previous problems which updating to SVN trunk release seemed to resolve, but I've got one lingering one to ask about.
Hi Doc, before to delve in the troubleshooting, I have to say that I'm modifying the audio skypiax code in svn, so maybe it's just my fault ;). please be patient for a little while, I hope to have done with it in a couple days. I'll announce to the mailing list when done. In the mean time, at least one good news for you user of SkypeIn service: a new feature of mod_skypiax is intended to recognize the DTMFs coming from SkypeIn, so the incoming calls will be able to use ivr, voicemail, etc.
When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS picks up and rings the desired extension. No problems there. What happens though is, the person that is on the extension in the office can hear the calling party just fine, but the calling party hears stuttered choppy sound. What do you think causes the stuttering on my audio for the calling party?
Do you think it might be a codec? I have everything setup right now to use g711u. I'd like to use something smaller and higher quality, but I'm not sure what is a better option.
I would exclude is a codec problem
I had made some phone calls as listed above and they tested out just fine, but in that same time frame I also recompiled Alsa drivers with the Skypiax dummy file, so do you think there might be something in that causing the stutter?
more probably is some change I made to the code, please be patient for a while, couple days. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Giovanni, Awesome to hear from the creator of Skypiax. You might be right, maybe it is the code in SVN. I had recently updated SVN (at the time that I re-compiled with the skypiax dummy file). I will look forward to an update on this. I am also excited to hear that you have some new things coming that will allow for DTMF and IVR's, etc. Yay! Thanks for your reply, Doc
Hi Doc,
before to delve in the troubleshooting, I have to say that I'm modifying the audio skypiax code in svn, so maybe it's just my fault ;).
please be patient for a little while, I hope to have done with it in a couple days.
I'll announce to the mailing list when done.
In the mean time, at least one good news for you user of SkypeIn service: a new feature of mod_skypiax is intended to recognize the DTMFs coming from SkypeIn, so the incoming calls will be able to use ivr, voicemail, etc.
When a call comes in to my SkypeIn number (PSTN that rings my Skype), FS picks up and rings the desired extension. No problems there. What happens though is, the person that is on the extension in the office can hear the calling party just fine, but the calling party hears stuttered choppy sound. What do you think causes the stuttering on my audio for the calling party?
Do you think it might be a codec? I have everything setup right now to use g711u. I'd like to use something smaller and higher quality, but I'm not sure what is a better option.
I would exclude is a codec problem
I had made some phone calls as listed above and they tested out just fine, but in that same time frame I also recompiled Alsa drivers with the Skypiax dummy file, so do you think there might be something in that causing the stutter?
more probably is some change I made to the code, please be patient for a while, couple days.
-giovanni
Hello, I was writing to inquire how this skypiax update was coming along. I didn't see it in the mailing list, but also since it had been nearly 1 week, I wanted to be sure if I didn't miss the announcement. I am very anxious to try this new update of the module.. It should be awesome! Thanks, Tim Giovanni Maruzzelli wrote:
before to delve in the troubleshooting, I have to say that I'm modifying the audio skypiax code in svn, so maybe it's just my fault ;).
please be patient for a little while, I hope to have done with it in a couple days.
I'll announce to the mailing list when done.
In the mean time, at least one good news for you user of SkypeIn service: a new feature of mod_skypiax is intended to recognize the DTMFs coming from SkypeIn, so the incoming calls will be able to use ivr, voicemail, etc.
On Wed, Feb 24, 2010 at 7:02 PM, Tim Streit <ledoktre@meanie.us> wrote:
Hello,
I was writing to inquire how this skypiax update was coming along. I didn't see it in the mailing list, but also since it had been nearly 1 week, I wanted to be sure if I didn't miss the announcement. I am very anxious to try this new update of the module.. It should be awesome!
Hehehe, no, you've not missed the announcement. Is taking me some more time than I was expecting. But's arriving... I'll post here the announcement ;) -giovanni
Thanks,
Tim
Giovanni Maruzzelli wrote:
before to delve in the troubleshooting, I have to say that I'm modifying the audio skypiax code in svn, so maybe it's just my fault ;).
please be patient for a little while, I hope to have done with it in a couple days.
I'll announce to the mailing list when done.
In the mean time, at least one good news for you user of SkypeIn service: a new feature of mod_skypiax is intended to recognize the DTMFs coming from SkypeIn, so the incoming calls will be able to use ivr, voicemail, etc.
_______________________________________________ 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, Giovanni Maruzzelli Cell : +39-347-2665618
Giovanni, Any progress updates on the skypiax updates for sound quality and also DMTF detection? With this update coming, I am pretty excited to be trying it out. I have been trying very, very hard to be patient... :-D Thanks, Giovanni Maruzzelli wrote:
On Wed, Feb 24, 2010 at 7:02 PM, Tim Streit <ledoktre@meanie.us> wrote:
Hello,
I was writing to inquire how this skypiax update was coming along. I didn't see it in the mailing list, but also since it had been nearly 1 week, I wanted to be sure if I didn't miss the announcement. I am very anxious to try this new update of the module.. It should be awesome!
Hehehe, no, you've not missed the announcement.
Is taking me some more time than I was expecting. But's arriving... I'll post here the announcement ;)
-giovanni
Thanks,
Tim
Giovanni Maruzzelli wrote:
before to delve in the troubleshooting, I have to say that I'm modifying the audio skypiax code in svn, so maybe it's just my fault ;).
please be patient for a little while, I hope to have done with it in a couple days.
I'll announce to the mailing list when done.
In the mean time, at least one good news for you user of SkypeIn service: a new feature of mod_skypiax is intended to recognize the DTMFs coming from SkypeIn, so the incoming calls will be able to use ivr, voicemail, etc.
_______________________________________________ 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
On Wed, Mar 10, 2010 at 10:26 PM, Tim Streit <ledoktre@meanie.us> wrote:
With this update coming, I am pretty excited to be trying it out. I have been trying very, very hard to be patient... :-D
Ciao Tim hehehe, you've missed the announce, so! http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/026987.htm... -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Giovanni, Thanks for your reply. I wondered if I somehow missed it !! Giovanni Maruzzelli wrote:
On Wed, Mar 10, 2010 at 10:26 PM, Tim Streit <ledoktre@meanie.us> wrote:
Ciao Tim
hehehe, you've missed the announce, so!
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/026987.htm...
participants (3)
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Doc
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Giovanni Maruzzelli
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Tim Streit