Re: [asterisk-users] Functions not autoloading

to 1.8.3.0 and I do not see any issues in /var/log/asterisk/messages ? No, this is not expected behavior. -- 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] asterisk's SDP

, but the other endpoint is not obligated to send them if it doesn't want 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

Re: [asterisk-users] asterisk's SDP

. If the Sonus device sent fmtp:101 0-15 in its SDP, then Asterisk should not send 'event 16' events to it. If it does, that's a bug, although standard programming practices would mean that it wouldn't be harmful, it would just be ignored by the Sonus device. -- Kevin P. Fleming Digium, Inc. | Director

Re: [asterisk-users] FAX with SIP

one vendor is (unfortunately) likely to have problems, whether any version of Asterisk is involved 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

Re: [asterisk-users] FAX with SIP

On 07/21/2011 04:43 PM, Israel Gottlieb wrote: On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 07/21/2011 04:34 PM, Joaquin Sosa wrote: On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com

Re: [asterisk-users] libss7 variables

On 07/18/2011 05:05 PM, Elliot Murdock wrote: I am wondering if the Libss7 add-on for Asterisk also translates ss7 variables into the dialplans for routing, accounting, etc? What are 'ss7 variables'? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem

Re: [asterisk-users] AsteriskNow install addons despite license conflict with FFA and G.729

for PostgreSQL and FreeTDS (Microsoft SQL Server), and also generic ODBC support which can be used to connect to MySQL if you wish. -- 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] max one sip peer to register

call that sip user, both sip clients will ring. No, it's not. Asterisk does not support multiple registrations to the same SIP AoR. -- 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] Multiple Asterisk Sessions on same machine

instance to listen on another port? It would be much easier to install a SIP proxy to listen on the second port and forward requests over to Asterisk on the standard port. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com

Re: [asterisk-users] Multiple Asterisk Sessions on same machine

On 07/19/2011 01:16 PM, Alex Balashov wrote: On 07/19/2011 02:15 PM, Kevin P. Fleming wrote: Actually, you can do this with one installation of Asterisk, and a separate set of config files and data directories. When the Asterisk executable is started, the '-C' option can be used to point

Re: [asterisk-users] SS7 and PRI compatibility

incompatible signalling? They are completely incompatible above the physical and HDLC layers. -- 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] libss7 variables

for pbx_builtin_setvar_helper() function calls where the variable name starts with SS7_. If you have more specific questions about Asterisk's support for SS7, join the asterisk-ss7 mailing list and ask there. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem

Re: [asterisk-users] Multiple Asterisk Sessions on same machine

Asterisk dialplan, just via different ports). The lightest weight solution for this problem is a stateless SIP proxy. -- 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

Re: [asterisk-users] Seg Faults with 1.6.2.19

If it is a regression introduced in 1.6.2.19, then it should still be fixed. At least I believe that's the rules. That should be the case, yes. -- 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] Asterisk binaries on CentOS version 6

to get RPMs properly built and tested. -- 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] skype for asterisk usage in the future

more. It would be best to plan for it being non-functional after the two year support period is over. -- 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] Benchmarking AGI performance in C, PHP, and Perl

time is all that's important, right? OT: Take a look at 'systemd'; this is exactly what's happening there, and Fedora is likely to incorporate it into Fedora 16, and it will make its way into other distros after that. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] Blind Transfer Connected

is 'not answered'. However, writing such a dialplan would indeed be non-trivial :-) -- 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] Blind Transfer Connected

. -- 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] ReceiveFax to G.711

claims to support T.38, then ensure you are running an updated version of Asterisk, and if you still can't make FAX work over T.38 with them, post debug logs here and we can try to help you figure why it's not working. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] ReceiveFax to G.711

to be used? -- 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] Problem with detecting fax on PRI/DAHDI channels

can do it, and most of the add-on answering machine detection applications can do it as well. -- 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] Problem with detecting fax on PRI/DAHDI channels

to a different destination in the dialplan). Now that chan_sip has 'faxdetect' as well, many usages of 'outgoing' in chan_dahdi are no longer necessary. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming

Re: [asterisk-users] Asterisk call limitation

message had three lines of content and 30+ lines of non-content. -- 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] Google Voice receiving call problem

everything and fix it when Google changes the protocol. -- 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 set a HA8 board + B400M in NT mode ?

in the manual for the Hx8 cards: change the 'te' in the 'span' line to 'nt'. -- 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] issues.asterisk.org/jira not working

and the appropriate permissions granted to the user. What is probably happening here is that Safari does not handle the 'optional' client certificate request from the server properly. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem

Re: [asterisk-users] benefits of asterisk 1.8

Switchvox isn't really designed for. -- 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] ConfBridge for 1.8 ?

in Asterisk 1.8 is very different from the one in trunk (what will become Asterisk 1.10). -- 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 1.4.40.2 Now Available

. 1.4.40.2 was released so that 1.4.40/1.4.40.1 users could get a security fix regression resolved without having to move to 1.4.41 if they are not ready to do so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype

Re: [asterisk-users] T38 fax detection using g729

On 04/20/2011 04:55 AM, Niccolò Belli wrote: Il 19/04/2011 23:41, Kevin P. Fleming ha scritto: If you are the receiver of the call (and thus they are the sender of the call), it is *your* system's responsibility to initiate the switch to T.38, not theirs. Are you sure? So what's faxdetect=t38

Re: [asterisk-users] dtmf payload type problem during faxing..

to be able to determine what might be happening. The quick answer, though, is that Asterisk will use whatever payload number for RFC2833 DTMF that the other end requests. The message you are seeing has nothing to do with DTMF. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] T38 fax detection using g729

on 'faxdetect' at all; this would allow you to use G.729 for your voice calls. -- 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] dahdi and linux-2.6.38

only in userspace, and has nothing to do with anything in the kernel. -- 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] Load Asterisk Module with parameters?

for parameters to be passed to a module. -- 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] dtmf_2833_1.pcap: what PCM codec? ulaw or alaw?

read RFC 2833 or RFC 4733; they explain how digits are sent over RTP. -- 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_dahdi unknown dependency 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 www.asterisk.org

Re: [asterisk-users] chan_dahdi unknown dependency problem

means you *must* have them installed. Have you made any changes to the Asterisk source code? -- 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] dtmf_2833_1.pcap: what PCM codec? ulaw or alaw?

interpret PCAP files? -- 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] Issues with Digum Repos / AsteriskNOW Bad Packages

load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? Yes... but this indicates that the module that was built appears to be broken. I'll let the package maintainer know. -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] IAX Call token revisited

is not incrementing it's OSeqNo in the REGREQ packets you showed in the capture, I would agree that it appears that the replies from Asterisk are not being received by the phone. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem

Re: [asterisk-users] IAX Call token revisited

of the traffic to/from the phone and then use Wireshark to look at what is going on. -- 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] One PRI card with 2 (or more) Telcos

, in which case it could cause the call to fail). For your sanity, I would strongly suggest that you don't connect spans from multiple telcos/networks/etc. on a single card, but keep each span provider on their own card. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] Some errors

no peer named 'h' or is that an IP address or DNS name. It should have failed a little more cleanly than it did, but I'm sure that at least part of the problem is attempting to dial a SIP endpoint that doesn't exist (and dialing out from the 'h' extension as well). -- Kevin P. Fleming Digium

Re: [asterisk-users] Ast 1.8_CentOS5.5 with timerfd as timing source

to occur, and the cause has not yet been found. -- 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 send Hold invite from asterisk to other

On 03/15/2011 04:18 AM, Nikhil wrote: how to send SIP HOLD Invite from asterisk to other sip client/server.? Asterisk's chan_sip does not yet have the ability to *send* 'hold' re-INVITEs. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP

Re: [asterisk-users] Multiple SIP endpoint registrations

an account and make a Local/234@somecontext which dials SIP/234-fooSIP/234-bar. Why do you need a Local channel to do this? If extension 234 exists in some context, the Dial() statement in that extension can dial SIP/234-foo and SIP/234-bar itself. -- Kevin P. Fleming Digium, Inc. | Director

Re: [asterisk-users] Asterisk 1.8 paging with ploycom

1.8 has a built-in Page() application you can use from the dialplan to achieve what it appears you were trying to achieve with your AGI script. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan

Re: [asterisk-users] Asterisk 1.8 paging with ploycom

phones answering a call at the same instant is a *lot* for Asterisk to handle. This is why multicast paging is preferred, but as others have pointed out, it doesn't appear that Polycom phones support that type of paging. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer

. I'm not sure if this is a chan_sip.c problem or if this is a dial plan problem. If your version string is 'SVN-trunk-r309404', you are not using 1.8, you are using 'trunk'. If you want to follow the 1.8 Subversion branch, you need to checkout that branch, not trunk. -- Kevin P. Fleming Digium

Re: [asterisk-users] Error loading module 'res_fax_digium.so'

[13429] res_fax_digium.c: Copyright (C) 1998-2008 The OpenSSL Project/ How can I fix this WARNING error? You can follow the instructions with the product and ensure that res_fax.so is loaded before res_fax_digium.so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber

Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add

unselected? no clue where to start looking Have you specified any '-march' or '-mcpu' options to the compiler? This sort of thing can occur if you are building for a plain-jane i386 processor or something similar. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem

Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer

echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using jitterbuffer cause this? This is probably acoustic echo from your phone. The jitterbuffer has nothing to do with this. -- Kevin P. Fleming Digium, Inc

Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add

On 03/07/2011 04:31 PM, RR wrote: On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: Please do not reply directly to posters on the mailing list unless they request it. On 03/07/2011 03:35 PM, RR wrote: Hello all

Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add

to all of those questions is probably 'yes', but that's why I said someone with SPARC experience would have to chime 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

Re: [asterisk-users] asterisk security....again

server and customers) can spoof the IP addresses of your server(s) in order to get the remote endpoints to at least accept an INVITE (they can't place a successful call through them using spoofing though). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW

Re: [asterisk-users] AGI script dies after receivefax

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

Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints

(which it can optionally generate for DND being enabled and disabled). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] SIP Hardphone that works well with asterisk

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

Re: [asterisk-users] Hide the plain text password

isn't supposed to have access to it, not the system's normal user. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Fax Woes

in a single sentence :-) Clarity and completeness make it much easier for people to understand what you are trying to express. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us

Re: [asterisk-users] Hide the plain text password

the modified source code, and thus the same problem arises. Security through obscurity does not work with open source software. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us

Re: [asterisk-users] Hide the plain text password

to figure out a way around the obscuring mechanism(s), but if enough people are interested in doing so, they will. With open source software, pretty much anyone can get around such mechanisms in a short period of time. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan

Re: [asterisk-users] further action after caller in a queue hangs up

, although there are some pretty creative people out there, so who knows :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Hide the plain text password

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

Re: [asterisk-users] Hide the plain text password

passwords using 'md5secret', but all other protocols that Asterisk supports need the password in plaintext to be able to perform the authentication process required by that protocol. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] Fax for Asterisk SIP-TDM

not currently support T.38-TDM gateway mode for FAX, although there is a patch on the issue tracker to add support for it, and it's in the works for Asterisk 1.10. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming

Re: [asterisk-users] [modules.conf] Modules still loaded after noload

is what you're expecting). This is correct. 'reload' is not 'restart', it only tells all the currently-loaded modules to 'reload' themselves (which generally means they will reparse their configuration files to look for changes). -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] digium te220

; if you didn't receive a printed copy when you purchased it, you can read it online on www.digium.com. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com

Re: [asterisk-users] Early audio SIP sequence order question

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

Re: [asterisk-users] Early audio SIP sequence order question

of the problem is quite different. This can of course cause complications if Dial() is used to dial multiple endpoints... because then there could be multiple audio streams received from them as the call proceeds towards one of them answering. -- Kevin P. Fleming Digium, Inc. | Director

Re: [asterisk-users] sip trunk balancing

in asterisk the current calls number on sip trunk alfa? 1) set call-limit in sip.conf. then in the dialplan sip show peer inuse|grep alfa - parse - if numcalls 25 then dial(sip/delta) 2) groupcount ? 3) what else? GROUP() would be the way to go, for sure. -- Kevin P. Fleming Digium, Inc. | Director

Re: [asterisk-users] RTP keepalive doesn't work

after 1.6.2 was branched (so only 1.8.0 and trunk are missing the code). Leif Madsen entered an issue on Mantis as a blocker for any more 1.8.x releases until this is resolved, as it is clearly a regression in the 1.8.x series. -- Kevin P. Fleming Digium, Inc. | Director of Software

Re: [asterisk-users] Disabling Music On Hold

On 01/31/2011 02:06 AM, Urs Buob wrote: On 01/28/2011 Kevin P. Fleming wrote: Loading or not loading a MOH provider is not going to change Asterisk's behavior regarding hold/unhold of endpoints; if you want Asterisk to pass through hold/unhold indications over SIP, unfortunately

Re: [asterisk-users] res_fax

On 01/31/2011 02:08 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Thursday, January 27, 2011 3:08 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01

Re: [asterisk-users] res_fax

be fairly easy to manually make the changes when you install a new release. Now that you have a working system, it would be really nice to get some debugging information like I asked for before, so we can try to figure out why T.38 negotiation is failing with your provider. -- Kevin P. Fleming

Re: [asterisk-users] Lots of warnings: SUBSCRIBE failure: no Accept header: pvt

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

Re: [asterisk-users] RTP keepalive doesn't work

investigation to find out when it was removed and why, because the configuration option should have been removed if the keepalive support was removed on purpose. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming

Re: [asterisk-users] RTP keepalive doesn't work

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

Re: [asterisk-users] Disabling Music On Hold

SIP, unfortunately it can't do that yet... although most of the code has been written, it has not quite been finished. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us

Re: [asterisk-users] res_fax

is pretty full today, but I will take another look over the code and see what might be going on. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com

Re: [asterisk-users] res_fax

304599 should fix this (and I also changed the option letter from 'n' to 'F' since it really means 'force audio'). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out

Re: [asterisk-users] res_fax

, with 'core set debug 10' and 'core set verbose 10' and all logger levels (including 'fax') enabled. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com

Re: [asterisk-users] ReceiveFAX issue.

for the initial debugging steps. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] res_fax

On 01/26/2011 01:12 PM, Tom Rymes wrote: On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Steve did not write res_fax (which where SendFAX and ReceiveFAX come from) snip I am personally a little confused here, because I have a ReceiveFAX application when I unload the res_fax module

Re: [asterisk-users] res_fax

On 01/26/2011 01:21 PM, Tom Rymes wrote: On 01/26/2011 2:16 PM, Kevin P. Fleming wrote: On 01/26/2011 01:12 PM, Tom Rymes wrote: On 01/26/2011 1:49 PM, Kevin P. Fleming wrote: snip Am I correct to infer that using app_fax.so is no longer recommended and that res_fax.so

Re: [asterisk-users] res_fax

On 01/26/2011 01:19 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 1:50 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01

Re: [asterisk-users] Asterisk 1.8.2.3 Now Available

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

Re: [asterisk-users] res_fax

using t.30 No, unfortunately there isn't a way to do that that I can see. It wouldn't be terribly hard to add to res_fax.c, but I don't think we ever thought of doing that before. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806

Re: [asterisk-users] res_fax

On 01/26/2011 04:36 PM, Bryant Zimmerman wrote: *From*: Kevin P. Fleming kpflem...@digium.com *Sent*: Wednesday, January 26, 2011 5:21 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] res_fax On 01

Re: [asterisk-users] SIP, IAX2 and ISDN ISUP data

servers. There is an SMS application, but it is an SMS endpoint, not a router. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk on Debian Lenny with timerfd

scary) or glibc (very scary) instead. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] Asterisk on Debian Lenny with timerfd

On 01/24/2011 12:46 PM, RR wrote: On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 01/24/2011 07:29 AM, RR wrote: On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com mailto:ranjt...@gmail.com

Re: [asterisk-users] res_fax

out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem

Re: [asterisk-users] ReceiveFax

to the second caller? Of course ReceiveFAX can be run on multiple channels at once. What makes you think it cannot? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out

Re: [asterisk-users] Mailing list question

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

Re: [asterisk-users] Introducing easySysAdmin - automated security and telecom fraud protection

-COMMERCIAL* usage. Please do not post advertisements for commercial products to this list. Thank you. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out

Re: [asterisk-users] Can I know if a call is transffered to asterisk

to get the 'redirecting number' (the number that redirected the call to you). Note that this is *not* a transfer (which is a manually initiated operation), but a call forward/redirection. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] res_fax_digium.so crashing

that the 1.2.x modules should not be used with Asterisk 1.8.2. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org

Re: [asterisk-users] AST-2011-001: Stack buffer overflow in SIP channel driver

an endpoint is registered or not has nothing to do with whether it can place calls; registration is for delivery of calls to the endpoint. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem

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