Re: [asterisk-users] Skype for Asterisk -- Codec support

for future releases. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

did you install? It is possible you've installed a version which isn't compatible with your CPU and it's trying to execute instructions that your CPU does not support. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

with HPEC, but just use a 'generic' CPU flavor instead of a highly-optimized version. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

and this is not an issue. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] chan_dahdi refuses to build

Boehm, Matthew wrote: Using asterisk 1.6.2.1 and dahdi 2.2.0.2. dadhi-linux installed just fine. Using dahdi_dummy as there is no card in system. Did not install libpri, again, no card. There isn't any 1.6.2.1 release of Asterisk; which version did you try to build? -- Kevin P. Fleming

Re: [asterisk-users] chan_dahdi refuses to build

file which comes from dahdi-tools. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Unable to compile 1.6.0.12

cycle, same problem. Is this just me or are others having this difficulty? It was a mistake in the release process... 1.6.0.13 will be out shortly. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber

Re: [asterisk-users] MixMonitor and Transcoding..

Gordon Henderson wrote: Transcoding is something that's not an option here. Hm. Maybe old fashioned 'monitor' and offline mixing although I'm open to suggestions here.. In general, it is not possible to mix compressed audio; it must be uncompressed first. -- Kevin P. Fleming Digium, Inc

Re: [asterisk-users] Placing a SIP Call on Hold

not offer methods to do those operations right now; I'm sure they could be added, but it hasn't been done. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out

Re: [asterisk-users] Placing a SIP Call on Hold

back off of this 'fake hold', just drop the MOH-playing channel and unmute both of the phone channels so they can talk to each other again. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] T.38 and reinvite

stream to inactive (when placed on hold) or to T.38, it will still be done, regardless of this setting. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out

Re: [asterisk-users] Different codecs for reading and writing

in Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth

Re: [asterisk-users] HPEC VPM ?

Noah Miller wrote: My question for anyone with knowledge on this: would HPEC do a better job than the VPM module (or oslec)? Can HPEC cope with very long echo tails? HPEC and the Digium VPMADT032 use the same algorithms from the same vendor. -- Kevin P. Fleming Digium, Inc. | Director

Re: [asterisk-users] HPEC VPM ?

on very high latency connections, or when the echo is actually acoustically generated by the far end and not by network effects. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com

Re: [asterisk-users] Fax for Asterisk quick question

, only sending and receiving of FAXes from TIFF files on the Asterisk system itself. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com

Re: [asterisk-users] Free Fax for Asterisk -- benchfax utility hangs.

at various different places, then that would point to something very different. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Free Fax for Asterisk -- benchfax utility hangs.

or bzip2 and email it to me directly; that may give us a clue what it was doing. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Sending faxes with T.38 problem. Fax for Asterisk (no SpanDSP) - 1.6.1.1

, with 'core set verbose 10', 'core set debug 10' and 'sip set debug on' (and ensure that the 'debug' logger level is activated for the console log channel in logger.conf). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype

Re: [asterisk-users] T38 negotiation, the last step !

an expected RTP packet, which Asterisk dropped. If your FAX works, then you can ignore this. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com

Re: [asterisk-users] g.722 + loudness

should solve your problem. Upgrading to 1.6.0.10 should give you the fix (and the fix should be noted in the ChangeLog for 1.6.0.10 as well). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] Fax for Asterisk - Fax routing based on TSID

to see if this is possible. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Fax for Asterisk download selector broken?

-specific optimizations enabled in res_fax_digium for maximum performance. For some reason the public downloads site was not updating correctly from its master site; I've re-synced it, and the download selector page is now working properly. -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] g729a compatibility

. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] g729a compatibility

and known format is there any way for Asterisk to negotiate it within an explicit rtpmap? Yes, that is already supported. Asterisk does not require rtpmap entries for well-known (RFC specified) codec mappings. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW

Re: [asterisk-users] g729a compatibility

for you to post at this point would be a 'sip debug on' console trace of this INVITE being handled by Asterisk, so we can both the entire SDP and the messages that Asterisk generates as it parses and processes it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis

Re: [asterisk-users] g729a compatibility

. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] g729a compatibility

know in advance whether transcoding will even be necessary for the call, so there's not much point in dropping g729 from the format list in the case where no transcoder is available. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806

Re: [asterisk-users] g729a compatibility

already. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth

Re: [asterisk-users] g729a compatibility

. Asterisk properly parsed the SDP and understands that the peer supports G.729. None of the concerns about SDP parsing or RFC compliance, as it turns out, were even relevant to this problem :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville

Re: [asterisk-users] g729a compatibility

of the codec name. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth

Re: [asterisk-users] Sounds format: GSM to G.729

Alejandro Cabrera Obed wrote: Because sounds files in /var/lib/asterisk/sounds are a lot as I see. If you are using the Spanish sounds distributed by Digium, they are already available in G.729 format from downloads.asterisk.org. -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] res_cepstral, register existing Cepstral licenses.

Cepstral-supplied licenses will continue to operate, and will be added to any Digium-supplied licenses you purchase and activate. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com

Re: [asterisk-users] Announcement: Howler-optimised G.729A Solution for Asterisk

for commercial products on this *non-commercial* mailing list. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] music on hold file formats

, though. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation

Re: [asterisk-users] Asterisk 1.6.1 and dahdichanname = no

releases require DAHDI, and will only support DAHDI channel names. If you saw any documentation to the contrary please point it out so we can get it fixed. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming

Re: [asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x

list for each module you want to load. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x

:-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation

Re: [asterisk-users] Help building dahdi for debian

to request echo cancellation on the channel, and there is a hardware EC present (and not disabled via module parameters) it will be used. No software EC needs to be configured or even loaded. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] Query about tdm410 cards

of the information and all the photo's are not clear. No, the PWR2400B includes a PCI bracket with cables that connect to the Molex connectors on the cards. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming

Re: [asterisk-users] Digium Fax Driver

on how testing goes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth

Re: [asterisk-users] TDM400P in PCI-X Slot

Michael C. Cambria wrote: Kevin P. Fleming wrote: [delted] PCI-X (not PCI-E) slots are backwards compatible with PCI slots, by definition. Thanks to you both. I knew about the 5v cards, and IIRC the TDM400P isn't available in 5v. The TDM400P (and all other PCI Digium cards except

Re: [asterisk-users] TDM400P in PCI-X Slot

this would be the only option for the TDM400P. PCI-X (not PCI-E) slots are backwards compatible with PCI slots, by definition. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com

Re: [asterisk-users] problem with T.38 media headers

? Asterisk is not a proxy; SIP signaling is never 'passed through'; the two legs of a call are completely separate and Asterisk bridges them together when necessary. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype

Re: [asterisk-users] problem with T.38 media headers

, there are still areas we've identified where our T.38 negotiation needs some additional work, and we'll be trying to address those shortly. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] Can't get G.726 to work.

Chris Maciejewski wrote: Found unknown media description format G726-16 for ID 102 It's right there. And asterisk is replying with 488 Not acceptable here Asterisk does not support G726-16, it only supports G726-32. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445

Re: [asterisk-users] Can't get G.726 to work.

in SDP offer No, that is not relevant. Asterisk's SDP parser does not pay much attention to a:fmtp entries at this time. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check

Re: [asterisk-users] Can't get G.726 to work.

' should show you what formats Asterisk can convert to and from G.726. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Can't get G.726 to work.

of any kind that I see in this log is the call to Playback. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] How to stop a background music

presses DTMF that matches an extension in the context Background was called from. There is no dialplan code executing while Background is running, so where do you want to be able to issue this 'StopBackground' command from? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan

Re: [asterisk-users] interruption in queue

key to exit from waiting for an agent; 'core show application queue' should give you the information you need. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out

Re: [asterisk-users] reg static build

a different C library. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth

Re: [asterisk-users] interruption in queue

didn't look for where that was actually documented. Sorry :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Problem with Free Fax For Asterisk

transfers are involved; that could have some effect on the audio path configuration, especially if transcoding was in use for the voice call before it was transferred. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype

Re: [asterisk-users] Problem with Free Fax For Asterisk

, so its APIs are still subject to change. FFA for Asterisk 1.6.1 *may* work fine with it, or may not. Once Asterisk 1.6.2 reaches the release candidate state, if new FFA modules are required then they will be produced and the FFA download selector page will be updated to reflect that. -- Kevin P

[asterisk-users] Asterisk open source project servers have new names!

addressed as quickly as possible. Thanks for using Asterisk! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Module not loading

cbbs...@hotmail.com wrote: Module 'app_machinedetect.so' did not register itself during load It means that the module you are loading was designed for an older (pre-1.4) version of Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW

Re: [asterisk-users] Digium fax force T38?

to help you. At a minimum, we need the exact versions of Asterisk and FFA you tested with, and a complete console log (including verbosity = 10, debug = 10 and 'sip set debug on') to see what is happening. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW

Re: [asterisk-users] Who has the clever Polycom upgrade system?

customized files are listed *before* the standard sip.cfg. This will make your settings take precedence over the standard settings. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] Digium Fax for Asterisk

. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] BLINDTRANSFER and SIP hardphones

the transferee's number and hears the call ringing, that is not a blind transfer, it is an attended to transfer to a call that hasn't been answered yet. There won't be any variables set for blind transfer, as it isn't one. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445

Re: [asterisk-users] Polycom wideband codecs?

. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation

Re: [asterisk-users] DTMF

is even aware of the DTMF events on the SIP channel. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] 2B Channel Transfer on XO-based T1

and the CDR becomes incomplete. Not everyone wants that behavior. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk G729 codec...

Olle E. Johansson wrote: Is this also available as a manager command? I would really appreciate being able to check license status over manager. It is not today, but I'll make a note to add it to the next builds, which will probably happen next week. -- Kevin P. Fleming Digium, Inc

Re: [asterisk-users] Asterisk G729 codec...

was not able to find any valid license files. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk G729 codec...

/asterisk-1.2' area of the downloads.digium.com site. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk doesn't relay remote MOH during hold

with media' in Asterisk today. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk doesn't relay remote MOH during hold

. Any ideas how to change that? What version of Asterisk are you using? If it's 1.4 or later, this already configurable in sip.conf. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] Help: RED alarm on Wildcard TE122 card

Andreas-Johann Ulvestad wrote: When inserting the cable going into TE122 into an ISDN phone, the phone works perfectly. Ummm... you have a BRI, not a PRI. I've never heard of an ISDN phone with an ISDN PRI port (E1 or T1). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies

Re: [asterisk-users] sip.conf outboundproxy

the proxy. In fact, this is even better when one of your 'internal' phones happens to be registered from a non-'localnet' IP address. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] sip.conf outboundproxy

; there is no 'internal', 'external', 'outbound', 'inbound', at least not in the sense of 'inside my PBX' or 'outside my PBX'. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us

Re: [asterisk-users] SIP trunk with 250 lines

:-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation

Re: [asterisk-users] IAX trunktimestamps and AST_CONTROL_SRCUPDATE

does not transport the native timestamps of the Asterisk frames it is asked to send (it creates its own timestamps), there is no value in sending a SRCUPDATE frame across IAX2, because the receiver of the frame will never see any timestamp discontinuity due to it. -- Kevin P. Fleming Digium, Inc

Re: [asterisk-users] Digium and Sangoma Cards PCI express compatibility

and x16 slots, assuming the machine doesn't restrict the usage of the slots (some 'personal' PCs, for example, have x16 slots but refuse to support anything but a graphics card in them). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] Asterisk and G.726 Codec

Le'an Liu wrote: My questions: 1. G.726 16/24/32/40 supported in asterisk-1.6.0.5? No. Only G726-32 is supported in all Asterisk versions. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber

Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)

channel? exten = 301,n,Dial(SIP/DavidR1,,M(dynamic_features)) [macro-dynamic_features] exten = s,1,Set(DYNAMIC_FEATURES=monkey) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] Digium B410P: misdn v1 or misdn v2 or dahdi + asterisk 1.6 ?

major caveat with chan_dahdi/libpri is that NT-PtMP mode is not supported, but both TE modes are supported and NT-PTP mode is as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] chan_zap.so missing

(at least) the UPGRADE.txt file that explains about the important differences between Asterisk 1.4 and Asterisk 1.6, and also contains a link to the Zaptel-to-DAHDI.txt file which has documentation specifically on the subject you posted about. -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] chan_zap.so missing

-to-DAHDI.txt file in the 1.4.23 source tree *carefully* to understand what you should see when you build this version of Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check

Re: [asterisk-users] Wideband (G722) MeetMe

responses! :o) It was just merged into Asterisk SVN trunk, on its way to becoming part of Asterisk 1.6.2. It's new and still has some limitations, but progress is being made and we welcome additional testing! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW

Re: [asterisk-users] Asterisk 1.6.1-rc1 with OpenAIS and different subnets

; it is not designed to work across routed connections. Russell Bryant has spent some time talking to the OpenAIS developers about this, but so far there doesn't seem to be a good solution. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806

Re: [asterisk-users] Silk for Free

, and acoustic effects of the phone design. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Silk for Free

). It will be interesting to see where this goes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Wideband g711-HD vs. g711.1?

P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Compiling to use IMAP: how?

Ken D'Ambrosio wrote: So: what/how do I need to install to meet this dependency? You need to read the documentation, specifically doc/imapstorage.txt, which is conveniently located in the source tree and named with a name very similar to the feature you are trying to use :-) -- Kevin P

Re: [asterisk-users] How to set PRI line timeout value

phone then I go to extension h and have DIALSTATUS set to CANCEL. This probably the phone initiating a no-answer hangup, many SIP phones do this. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] Help T.38

belong there. Thanks. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth

Re: [asterisk-users] benchg729 - no valid g729 license

from your system, along with the output of the 'asthostid' program (located in the 'register' directory on downloads.digium.com)? I can then determine whether the system should have handled this properly or not. Thanks. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan

Re: [asterisk-users] [asterisk-dev] IAX2 - now known as RFC 5456

media-only transfers, and Asterisk SVN trunk has encryption key rotation support. Those are the only *current* changes that I'm aware of. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem

Re: [asterisk-users] Annoying silence suppression effect on my digium E1 card with the VPMADT032 module

it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation

Re: [asterisk-users] benchg729 - no valid g729 license

g729 license. I have valid licenses on all systems, which show just fine when typing show g729 from CLI. How recently have you re-run the 'register' tool for those licenses? It's possible the license files are in an old format that the new programs don't expect. -- Kevin P. Fleming Digium

Re: [asterisk-users] SLA and Flashing BLF

display a 'hold' state on a line key using the method we use for signaling to them. What brand of phones are you using, and are you using up-to-date firmware for them? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype

Re: [asterisk-users] DTMF not completely muted

not enjoy the additional latency this would create. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk 1.6.x timing API

softmixing bridge functionality that is being worked on. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] PRI Test Lab

, the onboard clock on the card will be the timing source. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] PRI Test Lab

a recovered clock for any spans when they are all on the same card and using the same clock for transmission. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out

Re: [asterisk-users] PRI Test Lab

has been selected to be the clock source for the card, *all* the spans on that card will use that clock source for their transmitted bit streams. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber

Re: [asterisk-users] PRI Test Lab

. I don't know that I know what you mean by 'set timing'; if you transmit a bitstream onto a T1/E1 span, that bitstream includes clocking, by definition. The source of that clocking is what we are talking about, not whether it is present or not. -- Kevin P. Fleming Digium, Inc. | Director

Re: [asterisk-users] PRI Test Lab

to what our cards do, except that it's not controllable on a port-by-port basis. In that situation, which port did you pick as the clock source for the channelbank port, and what was the clock source for it prior to that? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan

Re: [asterisk-users] PRI Test Lab

have been able to do what you needed, and it could very well have been driver bugs or something else like that which kept it from working back then :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber

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