Hi Guys! I have a problems with some sipuras 841 and asterisk 1.0.9. Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with steve's unicall. Everything compiled fine and in fact I can make and receive calls but I have a problem with bad sound when the sipuras call the outside E1's lines. I can listen to the caller without problems but they heard me with a choppy sound as if you were losing frames or cutting off. Calls between internal sipuras sound good (eventhough the speaker and headset sound comes and goes, for example, when you start talking, seems as if the sipuras takes a few seconds to catch up with you on volume so the remote user listen to you as if the first words and the last were at low volume and the conversation in the middle sound good, any had that problem?) So, internal calls sound good between 841's but sound volume is weird at the start and end of a sentence. Calling the outside lines via E1's, I can listen to people without problems but they heard me as choppy or cut off. Anybody had issues like this? Is it asterisk or the phones or what? Hope you can help Guys, Im really banging my head against the wall here.
Have you tried upgrading the firmware? I had several problems with the outbound volume of these phones until I upgraded them. On Tuesday 20 September 2005 20:46, Anton Krall wrote:> Hi Guys! > > I have a problems with some sipuras 841 and asterisk 1.0.9. > > Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with > steve's unicall. > > Everything compiled fine and in fact I can make and receive calls but I > have a problem with bad sound when the sipuras call the outside E1's lines. > I can listen to the caller without problems but they heard me with a choppy > sound as if you were losing frames or cutting off. Calls between internal > sipuras sound good (eventhough the speaker and headset sound comes and > goes, for example, when you start talking, seems as if the sipuras takes a > few seconds to catch up with you on volume so the remote user listen to you > as if the first words and the last were at low volume and the conversation > in the middle sound good, any had that problem?) > > So, internal calls sound good between 841's but sound volume is weird at > the start and end of a sentence. > Calling the outside lines via E1's, I can listen to people without problems > but they heard me as choppy or cut off. > > Anybody had issues like this? Is it asterisk or the phones or what? > > Hope you can help Guys, Im really banging my head against the wall here. > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Juan Jose Comellas (juanjo@comellas.com.ar)
Seems upgrading the firmware fixed the problems with internal calls. Im still having problems sith those phones and unicall r2mfc for making outside calls. |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Anton Krall |Sent: Martes, 20 de Septiembre de 2005 06:47 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] sipuras 841 bad sound | |Hi Guys! | |I have a problems with some sipuras 841 and asterisk 1.0.9. | |Im using 841 with asterisk 1.0.9 with a digium card (single e1 |span) with steve's unicall. | |Everything compiled fine and in fact I can make and receive |calls but I have a problem with bad sound when the sipuras |call the outside E1's lines. I can listen to the caller |without problems but they heard me with a choppy sound as if |you were losing frames or cutting off. Calls between internal |sipuras sound good (eventhough the speaker and headset sound |comes and goes, for example, when you start talking, seems as |if the sipuras takes a few seconds to catch up with you on |volume so the remote user listen to you as if the first words |and the last were at low volume and the conversation in the |middle sound good, any had that problem?) | |So, internal calls sound good between 841's but sound volume |is weird at the start and end of a sentence. |Calling the outside lines via E1's, I can listen to people |without problems but they heard me as choppy or cut off. | |Anybody had issues like this? Is it asterisk or the phones or what? | |Hope you can help Guys, Im really banging my head against the |wall here. | |_______________________________________________ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |
> Re: sipuras 841 bad sound (Juan Jose Comellas)> On Tuesday 20 September 2005 20:46, Anton Krall wrote: > > I have a problems with some sipuras 841 and asterisk 1.0.9. > > (upgrade the firmware was suggested and completed, and didn't fix the > problem.)There are a few little configuration details which are hard to catch on the SPA-841, which can affect sound quality. * RTP packet size: 0.20 On the "SIP" tab of the Advanced Admin page, the "RTP packet size" is shown, measured in seconds. It defaults to 0.03, however Asterisk is hardcoded to use 0.02. This mismatch can cause sound issues. * Silence Supp Enable: Off On the "Ext1" and "Ext2" tabs of Advanced Admin, the "Silence Supp Enable" option must be turned off. This is Silence Suppression, which causes the phone to stop sending RTP packets when the phone detects silence in the handset. Asterisk 1.0.9 does not support silence suppression, so this option must be turned off, or audio stream timing will fail a lot. We have a bunch of SPA-841's in service, and we're just finishing working out the bugs in the system. Our latest audio issue, as far as we can tell, was caused by a Duplex Mismatch between the ethernet port on the Asterisk server, and the ethernet port on the switch it was connected to. When one is set to full duplex and the other half duplex, you get random, intermittant periods of massive packet loss/jitter, which messes up audio something fierce. I've found http://www.voiptroubleshooter.com/ to be a good source of info on diagnosing random "audio is bad" issues. It has sound clips of the different kinds of "audio is bad" problems, along with info on what might cause that kind of problem. Alan