Re: [asterisk-users] RFC List

2012-08-08 Thread Kevin P. Fleming
usually it's better to ask about the specific features you are interested in. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

[asterisk-users] So long, and thanks for all the fish!

2012-07-31 Thread Kevin P. Fleming
for the opportunity that Digium and the Asterisk community provided me to learn, grow and find the place where my skills and experience are the most valuable (to both myself and my employer). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem

Re: [asterisk-users] best PRI gateway?

2012-07-29 Thread Kevin P. Fleming
G100/G200 gateways. We've had really good feedback from users since they were released, and they certainly meet all of your needs listed above (and there are a few more, like T.38 FAX support on all channels). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem

Re: [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?

2012-07-27 Thread Kevin P. Fleming
echo cancellers? That's a terrible idea. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?

2012-07-27 Thread Kevin P. Fleming
the world! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

2012-07-26 Thread Kevin P. Fleming
phones are the statuses that Asterisk itself generates based on the phones' activity. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] Less good call quality using Asterisk

2012-07-25 Thread Kevin P. Fleming
without overloading its CPU. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] T.38 Gateway

2012-07-23 Thread Kevin P. Fleming
it has been enabled) will realize that the DAHDI channel is not T.38 capable, and it will step in to provide T.38 gateway services. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive

Re: [asterisk-users] Less good call quality using Asterisk

2012-07-23 Thread Kevin P. Fleming
? If both legs of the call are using the same codec, then normally Asterisk would not modify the audio in any way at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville

Re: [asterisk-users] T.38 (PRI) Fax Debugging

2012-07-20 Thread Kevin P. Fleming
properly, because that's what we used when we developed this 'audio capture' feature. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] Voicemail Emails

2012-07-20 Thread Kevin P. Fleming
to do), the email client that receives the HTML is going to treat it as plain text, which is what you saw. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806

Re: [asterisk-users] Voicemail Emails

2012-07-20 Thread Kevin P. Fleming
using standard email libraries in many scripting languages. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com

Re: [asterisk-users] Channel is rsrvd and does not turn off

2012-07-19 Thread Kevin P. Fleming
the person(s) who made the channel driver you are using, since it's not part of Asterisk itself. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us

Re: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway

2012-07-18 Thread Kevin P. Fleming
. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium

2012-07-18 Thread Kevin P. Fleming
.34 not supported, will be ignored. Is v34 only supported with SpanDSP? Those docs are in error. V.34 is not supported. I'll notify our documentation people. Thanks for the report. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem

Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium

2012-07-18 Thread Kevin P. Fleming
of the main Asterisk log directory. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used

2012-07-12 Thread Kevin P. Fleming
On 07/12/2012 09:19 AM, Benny Amorsen wrote: Kevin P. Fleming kpflem...@digium.com writes: That's quite interesting; can you describe a scenario where this occurs? Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24 and eth1 with 10.0.2.1/24. Further imagine that you wish

Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used

2012-07-12 Thread Kevin P. Fleming
this simple in order to cause the IP stack on the Asterisk server to choose the wrong source IP address for outbound packets. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW

Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used

2012-07-12 Thread Kevin P. Fleming
, this is the situation I referred to earlier. In your case, it's all on one interface, but the server has multiple addresses on the *same* network, and thus it cannot know (without help) with address should be used for outbound packets. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used

2012-07-12 Thread Kevin P. Fleming
. Benny, are you aware of some other method to accomplish this? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com

Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used

2012-07-11 Thread Kevin P. Fleming
On 07/11/2012 07:51 AM, Olle E. Johansson wrote: 10 jul 2012 kl. 20:50 skrev Kevin P. Fleming: On 07/10/2012 03:24 AM, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug

Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash

2012-07-10 Thread Kevin P. Fleming
it which extension it should 'be'... when the user returns home, they can 'steal' the extension back from the office. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville

Re: [asterisk-users] sip.conf and binaddr issue

2012-07-10 Thread Kevin P. Fleming
it does not specify the source IP address at all. If this is occurring, it must involve the operating system's IP stack in some fashion. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive

Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash

2012-07-10 Thread Kevin P. Fleming
On 07/10/2012 01:50 PM, Carlos Alvarez wrote: On Tue, Jul 10, 2012 at 11:46 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: This can be done using Digium phones; they have built-in support for selecting which 'user' they should be when they are reconfigured

Re: [asterisk-users] Asterisk trying to call a queue with no members

2012-07-06 Thread Kevin P. Fleming
as members are currently connected to Asterisk or not. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com

Re: [asterisk-users] Asterisk trying to call a queue with no members

2012-07-06 Thread Kevin P. Fleming
pretty uncommon for users to have 'turn on' their phones. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Outbound Asterisk calls default directmedia specifications

2012-07-05 Thread Kevin P. Fleming
or not Asterisk will respond appropriately to a re-INVITE received *from* a SIP endpoint (to which Asterisk should always respond properly, unless the re-INVITE is malformed in some way or is unacceptable). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP

Re: [asterisk-users] SendFAX timestamp

2012-06-28 Thread Kevin P. Fleming
On 06/27/2012 09:30 PM, David Cunningham wrote: Would anyone else know if Asterisk allows use of SpanDSP's time zone conversion? No, SendFAX (in res_fax) doesn't currently offer the ability to do what you are asking about. -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] Digium IP Phones D40

2012-06-25 Thread Kevin P. Fleming
damage to users' ears), and the Digium phones are no exception. If you are finding that the volume produced by common SIP phones is too low and you can't make it loud enough, I'd bet that the problem is not in the phones, but in your environment or your ears :-) -- Kevin P. Fleming Digium, Inc

Re: [asterisk-users] Getting channel 'SIP/192.168.1.69-00000000' refused to negotiate T.38

2012-06-22 Thread Kevin P. Fleming
-INVITE from Asterisk that wanted to switch the SIP channel to T.38 mode. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] SIP over SSL TCP or SRTP?

2012-06-22 Thread Kevin P. Fleming
to VPN. SIP over TLS (what used to be called SSL) is what secures the SIP signaling. SRTP is for securing media streams. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW

Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

2012-06-20 Thread Kevin P. Fleming
I have the analog lines working). Have you read any of the O'Reilly Asterisk books? They will help you learn quite a lot about Asterisk, and they are available online. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com

Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone

2012-06-20 Thread Kevin P. Fleming
-by-step instructions on setting up an FXS port for use with an analog telephone. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] Clipping issue with SIP over satellite

2012-06-19 Thread Kevin P. Fleming
that have a proper jitter buffer, there is no issue. If you send the call over SIP to this 'SV8300' device and still have audio issues, that would imply that this device does not have a jitter buffer capable of handling this level of jitter. -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] Clipping issue with SIP over satellite

2012-06-18 Thread Kevin P. Fleming
like the lack of a proper jitter buffer (of adequate size) is the issue here, since when the audio is directed at endpoints outside of Asterisk that have them, the audio is as you'd expect it to be. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com

Re: [asterisk-users] Digium IP Phones - Teleworker Capability?

2012-06-15 Thread Kevin P. Fleming
On 06/14/2012 05:23 PM, asterisk users wrote: Is there a detailed application note in the Digium wiki (or anywhere else for that matter) about these implementing features under Asterisk/Switchvox? Not yet, I don't believe. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies

Re: [asterisk-users] Local Channel Resource Limit

2012-06-14 Thread Kevin P. Fleming
On 06/14/2012 04:20 AM, [Digital^Dude] ® wrote: How can I set a hard limit to the number of Local channels asterisk can spawn? chan_local does not have a mechanism to do this. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem

Re: [asterisk-users] Digium IP Phones - Teleworker Capability?

2012-06-14 Thread Kevin P. Fleming
/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Most of it, I think. Give them a try! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com

Re: [asterisk-users] Deleting an inadvertent message

2012-06-13 Thread Kevin P. Fleming
Thanks in Advance!! No, there is not any way to do that. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com

Re: [asterisk-users] Get voicemail box password from dialplan?

2012-06-11 Thread Kevin P. Fleming
for doing this. I have used VMAuthenticate but I would like more flexibility than what this offers What do you need that VMAuthenticate does not offer? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445

Re: [asterisk-users] Which Digium cards to select: AEX with EF or E or P or B?

2012-06-11 Thread Kevin P. Fleming
, they will all produce identical signal levels ('voice volume') when plugged into your telephony circuit(s). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Re: [asterisk-users] Sangoma Card Issue SOLVED

2012-06-06 Thread Kevin P. Fleming
On 06/06/2012 09:46 AM, Eric Wieling wrote: For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it to zapata.conf it worked. That means Asterisk (1.4.4.x?) was built against Zaptel, not DAHDI. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] BRI Installation

2012-06-05 Thread Kevin P. Fleming
that would be greatly appreciated. The Hx8 User's Manual (here: http://docs.digium.com/H8/hx8_series_manual.pdf) has an entire chapter on software installation and configuration, including DAHDI, libpri and Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] Invite + decreasing sequence number = 500 Error?

2012-05-31 Thread Kevin P. Fleming
. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Loss of RTP stream during DTMF collection

2012-05-30 Thread Kevin P. Fleming
On 05/25/2012 06:30 PM, Dave George wrote: How can I enable the option to allow asterisk to maintain the RTP stream during DTMF collection? If it's the problem I hypothesized it was, you can set 'transmit_silence=yes' in your asterisk.conf file. -- Kevin P. Fleming Digium, Inc. | Director

Re: [asterisk-users] Dual- or Quad ISDN cards for PCI-X Slots

2012-05-29 Thread Kevin P. Fleming
available in your price range. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] axfer with simple CDR

2012-05-29 Thread Kevin P. Fleming
in CDRs. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] How to tell VPM presence without restarting?

2012-05-28 Thread Kevin P. Fleming
(since 2.4, I think) has made this information available in /proc/dahdi. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

[asterisk-users] Digium's new Community Support Manager - Rusty Newton

2012-05-25 Thread Kevin P. Fleming
across the community! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Dual- or Quad ISDN cards for PCI-X Slots

2012-05-25 Thread Kevin P. Fleming
should fit just fine into PCI-X slots. Do you mean PCI-Express instead? That's very different. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us

Re: [asterisk-users] Loss of RTP stream during DTMF collection

2012-05-25 Thread Kevin P. Fleming
and waiting for input from the caller; if your version is older than this, then that could explain what you are seeing. That's just a mildly-educated guess though. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype

Re: [asterisk-users] Detecting Fax Tones over IAX2

2012-05-24 Thread Kevin P. Fleming
know what version of Asterisk you are using. However, in Asterisk 10, there is a channel-agnostic FAX detection function that can be applied to any channel type, so at a minimum that is one way to solve your problem. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] Detecting Fax Tones over IAX2

2012-05-24 Thread Kevin P. Fleming
is of utmost relevance: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Absolutely correct. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Re: [asterisk-users] T.38 debug logs

2012-05-24 Thread Kevin P. Fleming
sending and see somehow its debugs? Or I should just be better off with capturing sip data through tcpdump? This will depend on what you are asking the Asterisk 10 system to *do* with T.38. Are you sending FAXes from it, or receiving FAXes into it, or something else entirely? -- Kevin P. Fleming

Re: [asterisk-users] twenty thousands (20, 000) users, which asterisk and how many servers?

2012-05-23 Thread Kevin P. Fleming
a failure. You'd be much better off to at least split the load across two machines, both of which should be large enough to handle the entire load when necessary. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com

Re: [asterisk-users] gr-303

2012-05-22 Thread Kevin P. Fleming
documentation. There is no support in chan_dahdi to make Asterisk behave *as* a GR-303 channel bank. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us

Re: [asterisk-users] SET SIP_CODEC and Video issues

2012-05-22 Thread Kevin P. Fleming
is going to consume vastly more bandwidth than the audio stream. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com

Re: [asterisk-users] DPMA for Digium Phones

2012-05-21 Thread Kevin P. Fleming
. There is no 'buying' licenses, they are free. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] DPMA for Digium Phones

2012-05-21 Thread Kevin P. Fleming
we have to pay for some license? Your questions are answered on the Certified Asterisk page on the Digium website: http://www1.digium.com/en/products/asterisk/certified-asterisk -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem

Re: [asterisk-users] Asterisk and the media path

2012-05-21 Thread Kevin P. Fleming
.). It seems like you have the configuration set up (mostly) properly, so in order to know what is going on you're going to have to post a more complete log snippet, including 'sip debug' output. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP

Re: [asterisk-users] Asterisk and the media path

2012-05-21 Thread Kevin P. Fleming
or later, you can use 'directmedia' on all of them. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk and the media path

2012-05-21 Thread Kevin P. Fleming
are hoping to transfer media to? No. 3) How long into the call before the media is transferred over? It should happen quite quickly after the call is answered. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype

Re: [asterisk-users] Asterisk 1.8 canreinvite

2012-05-18 Thread Kevin P. Fleming
* feature. If the document(s) you read didn't make that clear, the authors did you a disservice. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check

Re: [asterisk-users] Fwd: RTP stats explaination

2012-05-18 Thread Kevin P. Fleming
on the table with another cell and being stunned with the magnitude of the delay and that most people manage to carry on conversations without noticing. Yes, cellular networks have largish latencies, but no jitter. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem

Re: [asterisk-users] realtime configuration for /etc/dahdi/system.conf

2012-05-17 Thread Kevin P. Fleming
be useful. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] R-Series with NON-DIGIUM card on servers

2012-05-15 Thread Kevin P. Fleming
. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] R-Series with NON-DIGIUM card on servers

2012-05-12 Thread Kevin P. Fleming
-series failover appliances will work with any device that uses the appropriate type of PSTN circuits (digital or analog, depending on the R-series model), even a legacy PBX. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com

Re: [asterisk-users] Digium IP Phones

2012-05-12 Thread Kevin P. Fleming
released. The D40 and D50 have the same size screen, but it is smaller than the one on the D70. The D40 and D50 do not have a hard 'Apps' button, but they do have an on-screen softkey for access to applications. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem

Re: [asterisk-users] Least Machine Specs to run a production asterisk server

2012-05-11 Thread Kevin P. Fleming
of tiny, low power embedded devices. As Carlos already said, *any* decent x86 box produced in the last five years would be able to handle this without any noticeable CPU load at all. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem

Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

2012-05-11 Thread Kevin P. Fleming
) to the equation. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Replacing PBX with Asterisk, need feedback on my new architecture.

2012-05-11 Thread Kevin P. Fleming
of registrations. However, it would not surprise me in the least if Asterisk 1.8.x and later handled that volume of registrations without much of a problem. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype

Re: [asterisk-users] Digium IP Phones

2012-05-10 Thread Kevin P. Fleming
for building apps is usable? Can you build powerfull apps? Examples? The phone app SDK has not been released yet, it's still under development. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis

Re: [asterisk-users] where can i find code documentation

2012-05-10 Thread Kevin P. Fleming
tree. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] enabling dialing by sip uri

2012-05-10 Thread Kevin P. Fleming
at all (it should go to wherever the URI resolves to). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] looking for solid state like PC suitable for Asterisk

2012-05-10 Thread Kevin P. Fleming
:-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] British Telecom ISDN BRI line issues

2012-05-10 Thread Kevin P. Fleming
releases of DAHDI and Asterisk should have support for it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com

Re: [asterisk-users] enabling dialing by sip uri

2012-05-10 Thread Kevin P. Fleming
, but extensions don't get rejected. Extensions can be 'not found', but that's very different from rejected. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806

Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P

2012-05-09 Thread Kevin P. Fleming
, libpri and Asterisk working together. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk 1.8 Transfer CallerID

2012-05-08 Thread Kevin P. Fleming
by using the CONNECTEDLINE() dialplan function to send anything desired to a phone that is in a call with Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville

Re: [asterisk-users] TDM400P: Lifetime Replacement

2012-05-06 Thread Kevin P. Fleming
except the TDM2400P. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] End-To-End Secured Communications

2012-05-03 Thread Kevin P. Fleming
, etc.) Given that, what you really want is a pure SIP proxy like Kamailio or OpenSIPs. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] Playing Multiple File ( simultaneously ) on Channel

2012-04-27 Thread Kevin P. Fleming
( and there so louder ) on the channel! without stoping MOH? This is an interesting idea, but at this time Asterisk's app_queue has no ability to do what you are asking for. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem

Re: [asterisk-users] Playing Multiple File ( simultaneously ) on Channel

2012-04-27 Thread Kevin P. Fleming
exist inside the Queue. or maybe an application which written from scratch... can be help full. It's hard to parse what you are saying, but yes... it would be possible for someone to write code to do what you want to do. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] Hangup Cause and SIP Response Code

2012-04-27 Thread Kevin P. Fleming
On 04/25/2012 05:29 PM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 25, 2012 6:25 PM To: asterisk-users@lists.digium.com Subject: Re

Re: [asterisk-users] Master Registrations?

2012-04-27 Thread Kevin P. Fleming
, instead the target device is located when a call requests it. There are many examples of this on the Internet... use your favorite search engine. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan

Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

2012-04-25 Thread Kevin P. Fleming
to merge patches that added support for either of those mechanisms. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com

Re: [asterisk-users] Hangup Cause and SIP Response Code

2012-04-25 Thread Kevin P. Fleming
of Asterisk you are using. In some versions there is a SIP_CAUSE feature that can be used to extract that information (although this has been reimplemented for Asterisk 11 in a way that doesn't affect performance as much as the old method did). -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] Hangup Cause and SIP Response Code

2012-04-25 Thread Kevin P. Fleming
). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6

2012-04-23 Thread Kevin P. Fleming
from that branch. One final note: please don't reply to list posters' personal email addresses unless they ask you to do so. The list is configured to force replies to go back to the list, and that's done for a reason. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6 on Ubuntu

2012-04-20 Thread Kevin P. Fleming
file. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6

2012-04-20 Thread Kevin P. Fleming
and what it was not able to find, but the menuselect information is a reasonable next step. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us

Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6

2012-04-19 Thread Kevin P. Fleming
. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] g729 freezes 1.8

2012-04-19 Thread Kevin P. Fleming
' on the G.729 download selector page here: http://www.digium.com/en/docs/G729/g729-download.php Did you use that download selector, or go directly to the downloads.digium.com site to grab the files? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com

Re: [asterisk-users] g729 freezes 1.8

2012-04-19 Thread Kevin P. Fleming
. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

2012-04-18 Thread Kevin P. Fleming
On 04/18/2012 06:08 AM, Niccol├▓ Belli wrote: Hi, Il 18/04/2012 00:39, Kevin P. Fleming ha scritto: You guys know that it works in Asterisk 10, but you say you can't use Asterisk 10 for some reason that I don't understand. 1) No Debian packages for v10. If you have to maintain lots of servers

Re: [asterisk-users] g729 freezes 1.8

2012-04-18 Thread Kevin P. Fleming
commercial product and you are entitled to technical support. The simple answer to your question is no, there are no known incompatibilities between Asterisk 1.8 and Digium's G.729 codec modules (if there were, we'd fix them). -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

2012-04-18 Thread Kevin P. Fleming
On 04/18/2012 11:23 AM, Dan Austin wrote: Kevin P. Fleming wrote: This is a valid point, and we'll get this corrected. Our package repository should have packages for Asterisk 10, but it doesn't. How likely is it that a Centos 6 repo might be setup at the same time? It's on our list

Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

2012-04-17 Thread Kevin P. Fleming
to use an old (and clearly broken) patch for adding T.38 gateway support to Asterisk 1.8. You guys know that it works in Asterisk 10, but you say you can't use Asterisk 10 for some reason that I don't understand. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

2012-04-16 Thread Kevin P. Fleming
occur, or that it would be easy to backport. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] T.30 Fax and Echo Cancelation

2012-04-13 Thread Kevin P. Fleming
at the same time, then yes, you have some other type of problem. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com

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