R: [Asterisk-Users] Outgoing quality
OK .. what about ilbc ... could it be a decent choice?? -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] conto di Mojo with Horan Company, LLC Inviato: lunedì 10 ottobre 2005 22.00 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Outgoing quality Are you calling from a soft- or hardphone on a network with a high amount of latency? If your (for example) SIP phone can't deliver voice packets to asterisk in time for asterisk to put them where they belong in the Zap channel, things like this might happen. Usually the interruptions could be described as clicks or crackles. In this case, you could reduce the network traffic by utilizing a codec with a smaller bandwidth usage, like g729 or gsm if your phone supports it. Fabrizio Mazzoni wrote: I'm having slight problems with outgoing audio quality on Zap channels. People hear an interrupted voice. Can anyone help..? Regards, Fabrizio Mazzoni Macron SPA ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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 !DSPAM:434ac9d542161952810880! ___ --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
[Asterisk-Users] Outgoing quality
I'm having slight problems with outgoing audio quality on Zap channels. People hear an interrupted voice. Can anyone help..? Regards, Fabrizio Mazzoni Macron SPA ___ --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
[Asterisk-Users] No audio non channels and choopy sound to PSTN network
Hello all, I just set up an asterisk box (1.0.9). Its running on slackware linux 10.2 on a dual xeon 2.8 with 2 gb of ram and scsi disks. I have 2 tdm 04b (8 fxo modules) on this box and attached there are 40 SIP peers (12 GXP 2000 and 28 budgetone 486) The problem is that when people call from the PSTN net they sometimes hear a choppy intro sound and sometimes they hear it with a low volume. Another problem is that when people dial an extension (SIP) when they call the asterisk box, sometimes the SIP peer and the caller cannot hear each other or they have a flaky audio. This basically never happens when i dial internally from SIP to SIP. The load on the machine never exceeds 0.6 The cables that connect the PSTN to the TDM cards are about 26-27 mt long but they are brand new. Can anyone help me solve my problem? Could it be also an internal QOS problem? Because when there are less peers connected everything seems better (but noe good enough). Best Regards, Fabrizio Mazzoni Macron SPA -- Fabrizio Mazzoni http://macron.com ___ --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
R: [Asterisk-Users] Problem setting up TDM22B card
-Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] conto di somesh s Inviato: venerdì 23 settembre 2005 11.49 A: Asterisk Users Cc: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] Problem setting up TDM22B card Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wcfxo.o: insmod /lib/modules/2.4.20-8/misc/wcfxo.o failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed I tried to put the card in another PCI slot also...but same result! What should I do? Please help me in this regard. Regards, Somesh S. Shanbhag __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 !DSPAM:4333d11e129761748210392! ___ --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
R: [Asterisk-Users] Problem setting up TDM22B card
Try using modprobe wctdm Regards, Fabrizio Mazzoni -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] conto di somesh s Inviato: venerdì 23 settembre 2005 11.49 A: Asterisk Users Cc: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] Problem setting up TDM22B card Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wcfxo.o: insmod /lib/modules/2.4.20-8/misc/wcfxo.o failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed I tried to put the card in another PCI slot also...but same result! What should I do? Please help me in this regard. Regards, Somesh S. Shanbhag __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 !DSPAM:4333d11e129761748210392! ___ --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
[Asterisk-Users] Storing a number to Dial
Hello all, I have a problem that i cannot resove so I'm seeking for some help. Basically i want a user to dial from the PSTN line into our * box, dial a certain extension, give him a voice propmt asking him to insert a number and store this number. I need to do this because i want to give the possibility to our employees to dial to the free voipbuster countries from home so that they do not spend money. I cannot understand how to store the number they whish to dial. Must this be done with a context or a macro?? If needed i can show my dialplan. Regards, Fabrizio Mazzoni ___ 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
[Asterisk-Users] TDM400P slow getting line tone
Hello all, I just installed a TDM400P with 2 FXO modules on my asterisk server. The card works perfectly. To get users to ring out from my SIP phones i setup an extension with 0 that basically does something like this: extension = 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels extension = 0,2,Hangup This works exactly as i want so users basically can dial 0, wait for the dialtone and then dial the requested number. The only problem that i have is that from when a user dial 0 to when i get the dialtone from the telephone line, something like 5 seconds pass... is it possible to pull this wait time down to about 1 second? or even less?? I already set in zapata.conf the immediate=yes property.. Can someone help me out? Best Regards, Fabrizio Mazzoni Macron Srl ___ 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
