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Hello,

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

On 4/12/2017 12:38 PM, Jerry Geis wrote: > Sorry for the extra email. It send to quickly. > > procmail: Assigning "SUBJECT= Tornado Monday, 03/27/2017 at 20:27:02. The > Point BB.OBSURGRH is" sounds like your issue is procmail then... I just used your exact command to send myself a message form a bone stock C6 system (sendmail as the email server) and recieved... (domain

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them

file. I don''t know why it will work for other hosts but these steps do not work for this.. there are no extra outputs from using --debug On 4/9/07, Atom Powers <atom.powers at gmail.com> wrote: > > On 4/9/07, Mike Zupan <hijinks at gmail.com> wrote: > > I recently had a working puppet server serving around 4-5 clients. One > of > > the clients needed to

-------------------------------------------------------------------------------------------------------- dev*CLI> zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 dev*CLI> zap show channels Chan Extension Context Language MOH Interpret pseudo

an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM

2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell at digium.com> * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes

Leg/Transaction Does Not Exist" and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db To: <sip:sip:7531 at

1. You pass in half the samples as the 'bits' arg. Speex looks at 1 frame worth of those bits and decodes them, decoded result in 'pcm'. 2. You pass in exactly 1 frame of data as the 'bits' arg. Speex looks at 1 frame worth of those bits (which is all there, exactly), decodes them, stores decoded result in 'pcm'. 3. You pass in 2 frames of data as

Perhaps everything isn't as spiffy as I thought When running zttool the card still reports as internally clocked Zaptel.conf: # Global data span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=se defaultzone=se And as pointed out by Peter I do get a lot of D-channel warnings ... Aug 9 16:21:25 NOTICE[1350]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 And

supported by Asterisk for Video. I also find that video_caps branch has a fix for this problem, please can someone share more information about this and where i can find it ? I do not want those fmtp lines to be stripped. Suggestions to change the Asterisk config files, to achieve this are also welcome. Thank you. Best regards, Simith ------=_Part_17870_6007467.1218041254938 Content-Type:

1. Is it normal to see : # lsmod Module Size Used by dahdi_dummy 3236 0 Shouldn't it be used by asterisk or is this 0 value meaning something specific ? 2. How can you check dahdi is running ? Here, "ps aux | grep dahdi " replies "grep dahdi". Cheers ------=_Part_2692_19661943.1228286635399 Content-Type: text/html; charset=ISO-8859-1

*Update Jul 2007:* For a T.38 gateway you can use Asterisk 1.4's T.38pass-through support in combination with the new OPAL (Open Phone Abstraction Library) - using t38modem (currently CVS) which now supports SIP (and not just H.323) to terminate T.38 calls. You can also use OPAL and chan_woomera to do essentially the same. Where can you find this t38modem stuff ? Google replies things that

pages to mean "dial a new call to the mentioned phone number". Is there any pointer explaining available options ? I'm after something meaning "transfer ongoing call to the mentioned phone extension" instead of "dial a new call". Google replied me this : http://msdn2.microsoft.com/en-us/library/ms709071.aspx Anything else relevant ? Regards

Then, providing Linux compliant drivers should be Supermicro's problem. Anyway, how can you check in advance these compliance issues ? Maybe, a note in www.voip-info.org would avoid such issues. Regards ------=_Part_16952_4150825.1191306919994 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline Hello Matt,<br><br>Do you mean

user, password from user_sensitive_data_table into dovecot-sql.conf, but I'll live with that. You most probably had your reasons, and ultimately I agree - security first ;-) -- Chaos greets U ------=_Part_57551_1009602.1160777305352 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline <br><br><div><span