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unusable.<br> Is anyone using it successfully ?<br> <br> This kind of poll would be very useful to estimate is a code rewrite has a chance to disturb a running system. <br> I we get no "successful report", it would help developers to consider a code rewrite as patch instead of a new feature.<br> <br> So please, do not hesitate to report

Olle ?) aiming to unify logging, eventing, monitoring (AMI, SNMP, ...) APIs. I think that thread occurred when it was decided to include a version number in Manager interface. I agree this is an interesting idea ... The use case that made me ask this is here : I've got a running system which is working ok up to a moment it stops to dial out on ISDN-BRI spans (incoming calls are ok). When

connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. > > > Right now, I would not preclude the possibility that NT-PTMP support > might be added, but I could not give you a concrete time at which

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

described (stop accepting calls and shut down when all calls have completed). If you don't want to stop accepting calls, but still want to stop Asterisk when there are no active calls, you can use "stop when convenient". The same qualifiers ("gracefully" and "when convenient") can be applied to the "restart" command. Cheers, AR On Dec 10, 2007 7:29 AM,

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

the Telco, I can make calls in. What I am trying to get though is how to pass through the DID range. The E1 that I am connecting to the Telco with, used to connect direct to the NEC system and already has hunt group calling enabled for all 30 channels. Also, I was given a 100 number indial range (from 00 -> 99). If the E1 is connected to the NEC directly, I can call 5555 7320 and the NEC

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

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

Hi,

- ActivaTSP can't work with Astmanproxy as Asmanproxy needs to be patched, - Asttapi wouldn't terminate a completed call. Which option would you pick ? Is there any other option (free or commercial) for Outlook click2call ? Best regards ------=_Part_283_12644120.1196417210166 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline

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

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. > >

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

tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at

Content-Type: multipart/mixed; boundary="===============0382030028==" MIME-Version: 1.0 From: Albert Dengg <a_d@gmx.at> To: Debian Bug Tracking System <395305@bugs.debian.org> Subject: xen-utils-common: here is a small patch to correctly handle long domU names in parseln Message-ID: <20070317182707.16837.48971.reportbug@audhumbla.dengg.priv.at> X-Mailer: reportbug 3.31

selected for the great NETVC project.It complicates a little the future of the nhw codec, and it is furthermore nearly to end of life, with more than 3 years on the Theora channel without catching attention. I still think that the NHW codec could be useful for portable devices, as it is very fast with a low power consumption, and with more neatness/sharpness.If you are developing such a project

selected for the great NETVC project.It complicates a little the future of the nhw codec, and it is furthermore nearly to end of life, with more than 3 years on the Theora channel without catching attention. I still think that the NHW codec could be useful for portable devices, as it is very fast with a low power consumption, and with more neatness/sharpness.If you are developing such a project