[asterisk-users] Asterisk - SIP client latency

Heya all, what is the acceptable latency for VoIP calling? 200ms? 300ms? Best Regards, Freddy Setiawan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Asterisk - SIP client latency

to that. We try to engineer our on net to sub 100 of course our echo cans tell us the PRI to the PSTN regularly hit over 150ms which is ridiculous, and keep getting worse On Aug 19, 2006, at 12:04 AM, Freddy Setiawan wrote: Heya all, what is the acceptable latency for VoIP calling? 200ms? 300ms

Re: [asterisk-users] buy X100p card in singapore

Thanks for the informations. Regards, Freddy Go to http://www.voxzone.com/osCommerce This is from Singapore. Hi, Does anyone know where I can get the X100P card and VoIP phones in Singapore? I really want to buy it. ___

[asterisk-users] buy X100p card in singapore

Hi, Does anyone know where I can get the X100P card and VoIP phones in Singapore? I really want to buy it. Regards, Freddy Setiawan Developer @ Simpleware Solution ___ --Bandwidth and Colocation provided by Easynews.com

[Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

Hello, I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk as followed: [SIP_BD1] type=peer qualify=yes host=192.168.0.254 disallow=all context=from-pstn allow=h723 and inside the quantum I change the config sip useragent to 5060. Up to this part if I run sip show peers,

RE: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

Thanks. Gonna try today. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Neill Wilkinson Sent: Monday, June 26, 2006 3:54 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode In the quintum also check you

RE: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

I tried to change the codec to ulaw but still cannot do anything. I got this on my Asterisk box: - Found RTP audio format 0 Peer audio RTP is at port 192.168.0.254:10240 Found description format pcmu

RE: [Asterisk-Users] SPA-1001 behind NAT - Internet Asterisk box --BOUNTY!

Well, just make sure the sip.conf for your extension has the nat=yes, and dont forget to open the firewall hole for the service to run. Best Regards, Freddy Setiawan Senior Programmer Simpleware Solution [EMAIL PROTECTED] Yahoo Messenger: [EMAIL PROTECTED] Msn Messenger: [EMAIL

[Asterisk-Users] conf from pgsql database

Hai there, Is it possible to save all the configuration of the asterisk on the pgsql database? or just the cdr record? best regards, Freddy Setiawan ~SimpleWare Solusion~

[Asterisk-Users] cannot make app_prepaid

/ld: cannot find -lpq collect2: ld returned 1 exit status make[1]: *** [app_prepaid.so] Error 1 make[1]: Leaving directory '/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Any sugestion what should i do? Best Regards, Freddy Setiawan ___ Asterisk

RE: [Asterisk-Users] cannot make app_prepaid

Eureka... i must edit the Makefile hehe... yes correct, i must mantion about the psql lib -L /usr/local/pgsql/lib Thnx. Regards, Freddy Setiawan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joshua McClintock Sent: Tuesday, June 29, 2004 12:56 PM

RE: [Asterisk-Users] please mail me wave.cc and tts.scm

content extensions.conf of mine : [local] exten = 4001,1,Dial(SIP/4000) exten = 4002,1,Dial(SIP/4001) exten = 555,1,Answer exten = 555,2,Festival(good morning) exten = 555,3,Wait(2) exten = 555,4,Hangup should be alright. Regards, Freddy Setiawan ::Simple is Everything,Nothing is Complex

RE: [Asterisk-Users] app_dial broken

Yeah its broken... i just re-checkout the * then try to compile and it show error regards, Freddy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Stenton Sent: Monday, June 21, 2004 8:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] app_dial

RE: [Asterisk-Users] Festival and asterisk

the festival folder to be festival-1.4.3 then apply the patch... Done... (*)sorry for my bad english Best Regards, Freddy Setiawan ::Simple is Everything, Nothing is Complex:: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Totaro Sent: Sunday, June 20

[Asterisk-Users] please mail me wave.cc and tts.scm

Can someone email me pached file festival/lib/tts.scm and festival/src/arch/festival/wave.cc (for festival version 1.4.3)? my mail is [EMAIL PROTECTED] Thanks in advance. Best Regards, Freddy Setiawan ___ Asterisk-Users mailing list [EMAIL PROTECTED

RE: [Asterisk-Users] please mail me wave.cc and tts.scm

that.. patch and compile it manually. gentoo users: export USE=+asterisk emerge festival bkw - Original Message - From: Freddy Setiawan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 10:52 PM Subject: [Asterisk-Users] please mail me wave.cc and tts.scm Can someone email me

RE: [Asterisk-Users] Festival and asterisk

i've try to remove the quote marks but still not working the * console still return : Executing Answer(SIP/4001-bf1a,) in new stack Executing Festival(SIP/4001-bf1a,good morning) in new stack Parsing '/etc/asterisk/festival.conf' : Found Spawn extension (local,555,2)

[Asterisk-Users] Festival and asterisk

') exten = 555,3,Wait(2) exten = 555,4,Hangup What's the problem I'm facing? Thanks in advance. Best regards, Freddy Setiawan ::Simple is Everything, Nothing is Complex:: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http