Re: [asterisk-users] SPA-941 (and others ) Transmit Sound Quality
Hi guys, I got also problems with SPA 941 and 942, pour sound (a kind of click noise) that when i set volume sound lower almost can't notice, but still exists. I also notice on SIP to SIP calls , echo that could only be justified by Handsets hardware quality. When i make calls using Xlite with Plantronics DSP 400 USB micro and headset everything works like a charm. I've been told, by some one with longer experience with CISCO phones 7960 that if some one try to just replace in the handset the microphone inside with one form a cheaper traditional phone will get this VoIp hardphone working perfect. But in my case i didn't try that. If someone has a SPA942 on their own lab and can try this without damaging the phone would be nice info to share, I believe! Best regards, Marco Mouta On 1/17/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: I too seem to have the same problem, dont know about poor quality but its certainly not loud enough, I have to put my mouth to the microphone, otherwise the other end reports they cannot hear me. This does however seem to do a good job to cancel out the background noise In the SIPura setup change the packet size from .3 to .2. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-941 (and others ) Transmit Sound Quality
ron, i recently fixed the poor quality with our spa942 using canreinvite=yes have you found out the problem with your spa941? i can't get around the problem of poor quality audio when the rtp pass through * [EMAIL PROTECTED] wrote: Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the box to provide hardware timing. The use case is calling from the SIP phones (which are extensions registered with the * 1.2 box) to a VoIP termination service which routes the call to a PSTN number. Everything sounds great on the SIP phone, but the sound on the other end of the line is distant and missing bass, most especially so on the SPA-941 (which is the phone we really want to use). If I use the default handset mic gain value of 0db, the sound is so loud for the other person they have to hold the phone away from their ear. If I set it to -6db, it is still too quiet. The Aastra 9113i sounds a little better, and the Uniden 5.4 GHz Cordless sounds actually very good, so I'm pretty sure my network setup is capable of transmitting good sound. Using the speaker-phone on the SPA-941 sounds significantly better than using the handset. But we need the handset to also sound good. I've tried different providers etc. and always come back to the phone. I'm using G711u codec in all cases and silence suppresion is off. I saw a previous thread that mentioned changing the RTP from .03 to .02, however the post was regarding a MeetMe issue. I tried anyway, and it introduced an echo on the line. I've seen many rave reviews regarding the sound quality on the SPA-941, so I'm wondering if maybe I got a bum handset? Would anyone be willing to receive/place a call to tell me if it sounds the way its supposed to or if there is indeed a problem? All suggestions/recommendations greatly appreciated. Much thanks, -- Ron [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-941 (and others ) Transmit Sound Quality
I too seem to have the same problem, dont know about poor quality but its certainly not loud enough, I have to put my mouth to the microphone, otherwise the other end reports they cannot hear me. This does however seem to do a good job to cancel out the background noise On 11/10/06, Ron Winograd [EMAIL PROTECTED] wrote: Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the box to provide hardware timing. The use case is calling from the SIP phones (which are extensions registered with the * 1.2 box) to a VoIP termination service which routes the call to a PSTN number. Everything sounds great on the SIP phone, but the sound on the other end of the line is distant and missing bass, most especially so on the SPA-941 (which is the phone we really want to use). If I use the default handset mic gain value of 0db, the sound is so loud for the other person they have to hold the phone away from their ear. If I set it to -6db, it is still too quiet. The Aastra 9113i sounds a little better, and the Uniden 5.4 GHz Cordless sounds actually very good, so I'm pretty sure my network setup is capable of transmitting good sound. Using the speaker-phone on the SPA-941 sounds significantly better than using the handset. But we need the handset to also sound good. I've tried different providers etc. and always come back to the phone. I'm using G711u codec in all cases and silence suppresion is off. I saw a previous thread that mentioned changing the RTP from .03 to .02, however the post was regarding a MeetMe issue. I tried anyway, and it introduced an echo on the line. I've seen many rave reviews regarding the sound quality on the SPA-941, so I'm wondering if maybe I got a bum handset? Would anyone be willing to receive/place a call to tell me if it sounds the way its supposed to or if there is indeed a problem? All suggestions/recommendations greatly appreciated. Much thanks, -- Ron [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-941 (and others ) Transmit Sound Quality
Andrew Joakimsen wrote: I too seem to have the same problem, dont know about poor quality but its certainly not loud enough, I have to put my mouth to the microphone, otherwise the other end reports they cannot hear me. This does however seem to do a good job to cancel out the background noise In the SIPura setup change the packet size from .3 to .2. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA-941 (and others ) Transmit Sound Quality
Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the box to provide hardware timing. The use case is calling from the SIP phones (which are extensions registered with the * 1.2 box) to a VoIP termination service which routes the call to a PSTN number. Everything sounds great on the SIP phone, but the sound on the other end of the line is distant and missing bass, most especially so on the SPA-941 (which is the phone we really want to use). If I use the default handset mic gain value of 0db, the sound is so loud for the other person they have to hold the phone away from their ear. If I set it to -6db, it is still too quiet. The Aastra 9113i sounds a little better, and the Uniden 5.4 GHz Cordless sounds actually very good, so I'm pretty sure my network setup is capable of transmitting good sound. Using the speaker-phone on the SPA-941 sounds significantly better than using the handset. But we need the handset to also sound good. I've tried different providers etc. and always come back to the phone. I'm using G711u codec in all cases and silence suppresion is off. I saw a previous thread that mentioned changing the RTP from .03 to .02, however the post was regarding a MeetMe issue. I tried anyway, and it introduced an echo on the line. I've seen many rave reviews regarding the sound quality on the SPA-941, so I'm wondering if maybe I got a bum handset? Would anyone be willing to receive/place a call to tell me if it sounds the way its supposed to or if there is indeed a problem? All suggestions/recommendations greatly appreciated. Much thanks, -- Ron [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
