search for: hwit

Just curious, I noticed that with SetLanguage() you can change it into a lot of other languages. Yes one can record them easily enough with "record", but don't like to re-invent the wheel.. Browsed through a lot of google-pages but failed to find any other languages (except for FR and ES on the digium site) Any pointers for german, dutch, greek, italian, .... prompts? Hans --

Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOME (not all) people phone me (isdn-incoming) DTMF is not recognized. How come? Either it works

Morning! This may be the "day of the dead" in some regions, but we expect the usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This Sunday, I believe the USA falls back to Standard time. Future VUC are still at 12 Noon EST. Info site: http://voipusersconference.org PSTN (724) 444-7444 enter 22622# 1#

Hi all, Perhaps someone has dealt with it before. I want to activate a bunch of my own scripts after someone has registred om my asterisk, or when his cient has de-registerded. have been skimming through AGI and AMI, and seen a lot of nice features, but not the (de-)registering events. Kind regards, Hans

Hello For customers who need a small IP PBX to handle up to four ISDN lines (in France, so I guess that means EuroISDN) instead of a PC + Asterisk and an ISDN gateway box, has someone already played with the Atcom IP-4B? www.atcom.cn/IP-BRIM.html Any feedback appreciated.

My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part

Hi List, i have to put an * between two other SIP gateways and due to some circumstances, i have to use sip over tcp. With 1.6.2.6 this is working fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B (ocs) and that's about it. In the other direction however (ocs -> me -> deverto4) the call setup is complete but there is no audio. I can see the audio in the form of

I am sure many of you have seen my post asking question that I cannot seem to resolve. While the responses i have been getting have been helpful i still cannot seem to get this working 100%. So I have waving the white flag here. I give up. I need someone to come to my office and help me get this working. If anyone is interested the office is in Lexington KY. If someone is interested we can

I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if

On the Digium's site it says: The Wildcard TDM400P is a half-length PCI 2.2-compliant card while for other cards it says: The TE411P is for use only with a 3.3 volt PCI slot. Does the TDM400 not only fits, but also functions in a 3.3V only slot? >From what i detected so far, is that some MOBO manufactures have pci-slots that provide 3.3 Volt AND 5.0 Volt, thus can handle all kind of

Can anyone inform me if voip can be used on a IPv6 network? Does any hard phones/soft phones/Asterisk support it? Google told me that there was/is a bounty on it, but that expired august last year. Furthermore, there used to be a patch (Bernhard Schmidt), but that one is about a year old. I presume it can't be used on recent versions of "*" Hans -- pgp-id: 926EBB12

Hi all, Just curious, Quite a while a go, i was checking for supported SW-platform. AFAIR, it was RHES and SLES Now it's only RHES-4 and FC-3 or FC-4. Not a single syllable about CentOS or SLES-9 or SLES-10 It probably just runs fine, but any chance of getting support for their *-enterprise version? (just in case of, if one needs it) Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF

Hi Cosmin I saw in the archives that you achieved getting msn & did working. Perhaps you can help me a bit on it. Currently i have a running 1.2.19 with latest misdn. I can generate sip => isdn call, which uses the default number connected to the line. No incoming calls are detected (yet) On my isdn-line i have multiple MSN's What i want is: 1) when user "X" places an

Just noticed that packman has precompiled versions of amr codec. Both wideband and narrowband. Can these be used for asterisk? Heard some nice about AMR (in general) If so, any one around with experience with either?? hw

Hi all, As stated in the subject, slightly off-topic, as it is not directly a Asterisk issue, but more SIP in general Because security in general, and specifically identification becomes more and more a subject for more concern, and Asterisk is capable of doing sip/TLS, i was wondering what more could be done to improve security. Specially softphones, might it be possible to employ etokens or

Hi, I'm going to have a go with realtime mysql. Just wondering, most examples i came across while googling, was with 1.6 systems. So any drastic changes with 1.8.3, table-layout? other pitfalls? hw

Hi all, I've got something strange, that got me searching for quite awhile. Configuration as followed: Linphone on a laptop, that is connected via openvpn to a proxy. That proxy is connected with iax to another asterisk. On the second one i have several hard and softphones. Behaviour at first glance: >From the softphone i can allways set up a connection, But the otherway round fails 9

Hi all, Trying to find where i got wrong in my config.... Is the "realm" parameter in sip.conf only used for possible autentication? The thing is, i got my box more-or-less working as i wanted, but i can only reach internal functions (like echo-test and so on) and other sip-clients if i dial "1234 at fqdn", while i was expected to be able to just dial "1234" I

Hi all, Perhaps a no-brainer, but i think i am making my dialplan on my proxy too complicated. Normally, what you find in the examples is that you have to dial a specific number, other "9" or "0" for an external line. What i want to do is this: If you pre-pend a number with something like "*" then you can dial local defined numbers, otherwise everything goes

Hi all, In order to get icecast working i found that i had to install libao-0.8.4 libogg-1.1 libvorbis-1.0.1 flac-1.1.0 speex-1.0.3 vorbistools-1.0.1 icecast-2.0.0 So i grapped the source files. I copied all the files to an 32-bit Intel machine, and *there* all files configured, compiled and installed OK But on the intended target machine (SUN blade server) i have a problem... libao and libogg