similar to: IVR: playing multiple streams simultaneously?

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem

Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josu?

Hi all, I have thoroughly read the available documentation and I can't seem to find a workaround for my setup... I'm trying to create a phone conference line that users would call using a unique phone number (no matter if they are moderators or just plain users). I use Asterisk 1.2.6 The available conferences are defined as follows: conf => 1000,user pin1, moderator pin1 conf =>

Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec ) before the call gets dialled i assume that the problem would be due to the dial plan in PAP2 if so please help me changing it

HI I have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ? i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody, I?ve been pulling my hair for a week now over a problem, and I really don?t know where to look anymore. Here?s my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323 and openH323, and on the same server I also setup GnuGK (10.253.30.1). I

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a

Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. But coming in via the IAX2 route leaves me with a silent phone. The prompts all work still letting me navigate the menu. But just can't hear anything. This is with A@H 2.8 (Asterisk 1.2.7.1, with FreePBX 2.1.0 also installed) Any thoughts on where to

Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf

Hi, I'm speech recognition researcher and would like to do some research on recognition robustness in echo distortion of speech signal. Since VOIP is becoming wide spread, I'd like to simulate (one or more) common echo distortions that mostly appear in voip communications ? Any example, FIR or IIR filter or acoustical system response ? Any other distortion worth researching ? Thanks

Hi i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any large scale deployment of asterisk to date ? thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 18 April 2016 at 16:33, Silviu Baranga <Silviu.Baranga at arm.com> wrote: > Doing a grep "eabi" * -R | grep darwin in llvm I found the test divmod-eabi.ll > which uses the triple armv7-apple-darwin-eabi. What format does that have? Certainly not ELF. :) But I didn't mean "has eabi on triple", but "is in none-eabi mode", which may have to check a

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

I''m playing around with foreman for the moment. I can''t seem to figure out how to make the dashboard look like in this screenshot http://theforeman.org/wiki/foreman/Screenshots#Dashboard In order for those statistics to work what should be done? I have activated the rrdgraph reports in puppetd, uncommented the line ":rrd_report_url: report/" in config/settings.yaml

On 18 April 2016 at 16:18, Silviu Baranga <Silviu.Baranga at arm.com> wrote: > This doesn't look like something ACLE specific (I can't find it in the ACLE doc). Sorry, I didn't mean it was ACLE, only that you guys were fiddling with macros. :) > This seems to be a generic macro. I think it would make sense to define it > if we know we're emitting ELF. Since the

Hi, Created a zpool with 64k recordsize and enabled dedupe on it. zpool create -O recordsize=64k TestPool device1 zfs set dedup=on TestPool I copied files onto this pool over nfs from a windows client. Here is the output of zpool list Prompt:~# zpool list NAME SIZE ALLOC FREE CAP DEDUP HEALTH ALTROOT TestPool 696G 19.1G 677G 2% 1.13x ONLINE - When I ran a