Re: [Asterisk-Users] E1 in Brazil
Andrew Kohlsmith wrote: RDSI and ISDN are the same thing. RDSI is ISDN said in portuguese. The E100P does not do ISDN, does it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users PRI ISDN ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and ECHO
What have you specified as rx and txgain? Simon McAuliffe wrote: I've been having the same problem too, except for me it only happens occasionnally. I'm not 100% sure of this, but it seems that for very local calls (eg across the city) I never get echo. For calls that go longer distance (say 500km or more), or through some closer call centres, I'm getting the echo. I don't get the echo on an analogue POTS connection to the same places (it is clearly only happening on our asterisk system). This might indicate some link between echo cancellation and delayed audio, but if so, its sensitive to very small delays. The echo can only be heard at our end, there is no trace of it at the other end. I'm using ATAs doing SIP to Asterisk and through a PRI connection to a Telco. Echo cancellation is turned on and showing as activated on the Zap channels. Echo cancellation is also enabled on the ATAs. - Original Message - From: Brian J. Schrock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 29, 2003 3:16 AM Subject: [Asterisk-Users] SIP and ECHO Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN through the FXO cards I get horrible echo, I have even been able when talking loud enough to get a horrible feedback loop going. I have tried 4 different echo cancellers in the Makefile for the Zap drivers and nonoe of them changed the situation. I have echocancel = (Any where from 1 - 256, I have tried alot of different values), and I have echocanelwhenbridged = yes.I only hear the echo start when the call gets bridged onto the outgoing PSTN lines. Is there anything I can do? Brian J. Schrock ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Fredrik Hedberg Telavox AB Direct: +46 46 6220013 Lilla torg 1 Phone: +46 46 622 S-211 34 Malmo Mobile: +46 70 3323033 Sweden Web: www.telavox.se ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer ATA186
How exactly does you 3Party calling work? ;) Fred ASN wrote: Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Lines
Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: den 8 augusti 2003 08:14 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Lines Instead of using a PCI card is it possible to use an outside SIP service for CO lines?
RE: [Asterisk-Users] Fix for Redhat 9 zombie AGI processes
Having similar problems with Debian Woody. Is this a known issue or merely poorly written AGI scripts from my side? Fredrik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: den 5 augusti 2003 00:39 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fix for Redhat 9 zombie AGI processes Hi all- Thanks to Mark Spencer for finding this patch: If you are experiencing leftover zombie processes from your AGI scripts that have terminated, this is apparently due to a RedHat 9 threading issue introduced in a recent update To get rid of this, try entering the following line before you start asterisk: export LD_ASSUME_KERNEL=2.4.1 It works for me - I'm still checking to see if there are any other side effects - none yet.. -Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo on incoming calls (PRI-SIP) but not on outgoing (SIP-PRI)
Hello all! We have an E400P on a dual Xeon box with four EuroISDN PRI to the PSTN and Cisco ATA-186 as SIP UA. We experience very large amount of echo (on -some- calls) when we're doing PRI-SIP calls but not when doing SIP-PRI calls. We don't think the problem is IP latency related (When calls are made PRI-*-PRI we still experience the echo). What do you think? Regards Fredrik Hedberg Hello all! We have an E400P on a dual Xeon box with four EuroISDN PRI to the PSTN and Cisco ATA-186 as SIP UA. We experience very large amount of echo (on -some- calls) when were doing PRI-SIP calls but not when doing SIP-PRI calls. We dont think the problem is IP latency related (When calls are made PRI-*-PRI we still experience the echo). What do you think? Regards Fredrik Hedberg
[Asterisk-Users] E400P E1 Pin Layout
What is the pin layout of the E1 sockets on the E400Ps? What pins are for the TX and RX pair? Regards Fredrik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
