Re: [asterisk-users] voicemail - digits/1F does not exist in any format

I've seen this before, in an ISDN card (can't recall which one) that defaults the incoming language to german. Since you don't have german, it defaults to english files but voicemail still runs through the german logic (e.g. 1F for femail). I reported a bug against this, it was silently killing

Re: [asterisk-users] error in FreePBX

something is broken in your configuration. dialparties is returning with no extension to dial (which could be DND, could be the phone is occupied and no CW active, or several other factors). And your call to voicemail is failing which implies something is broken in that setup since it wouldn't

Re: [asterisk-users] Answer Confirmation with SIP/AIX channels

I have implemented the requested call confirmaiton feature in the freepbx followme and ringgroup applications (asterisk 1.2 for now). You can select to have confirmation and by default any external call (e.g. cellphone) will require such confirmation, any internal phone will not (unless you

RE: [asterisk-users] Follow me on multiple numbers..

in Trixbox but when the option ‘confirm’ is selected in the follow me properties screen, no code is generated and the call goes dead. Is there a trick to get the code generated? - From: Philippe Lindheimer [mailto:[EMAIL PROTECTED] Sent

Re: [asterisk-users] Follow me on multiple numbers..

On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell

Re: [asterisk-users] Retain call control: Avoid letting call get

Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears

[asterisk-users] talking caller ID

Christian wrote:<[EMAIL PROTECTED]>From: "Christian" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 08 Nov 2006 20:10:02 +0100Subject: [asterisk-users] talking caller ID Hi all,Lets say I have my incoming calls transfered to my mobile phone. When a call comes in, Asterisk will

[asterisk-users] OT - Polycom https provisioning

Hi,I've setup polycom https provisioning with an apache/linux server. However the log files aren't saved because there is nothing to process the http PUTS polycom uses. Does anyone have a secure solution they are using in this scenario so the phone log files can be saved?philippe Cheap Talk?

Re: [asterisk-users] Follow Me problems

<[EMAIL PROTECTED]>From: "Time Bandit" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Tue, 7 Nov 2006 08:53:51 -0500Subject: Re: [asterisk-users] Follow Me problems Today we appear to have discovered our first bug. We have an

RE: [asterisk-users] Asterisk in Seattle

The mail system somewhere seems to have eaten some of the digetst versions of this list that are sent to me (jumped from 24 to 29). So - in case this didn't make it out, just expressing my interest. Were there many others around here who responded that I must have missed?philippePhilippe

RE: [asterisk-users] Asterisk in Seattle

If anyone wants to try to start a users group in the Seattle Area, I'm interested. (Although not in July).philippe<[EMAIL PROTECTED]>From: "calvis" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Wed, 5 Jul 2006 15:16:06

Re: [Asterisk-Users] Digium Hardware Reliability

I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with that many systems, which makes this really concerning. I've started a thread on the

Re: [Asterisk-Users] Digium Hardware Reliability

are finding.pFrom: Andrew Kohlsmith [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Fri, 30 Jun 2006 08:49:07 -0400Subject: Re: [Asterisk-Users] Digium Hardware Reliability On Friday 30 June 2006 02:24, Philippe Lindheimer wrote: I would love to see some feedback on this as well. I've lost exact

RE: [Asterisk-Users] need help troubleshooting clipping and garbledVOIP calls

Interesting web tool, but the results are completely misleading and wrong on my system that I just tested it on 5-6 times. And this is a connection that I use regularly for VoIP traffic, multiple channels (although just a few) and I am very familiar with the characteristics of the internet paths

RE: [Asterisk-Users] Trixbox maunual configuration

As pointed out, just build your own system. If you understand the Freepbx dialplan, you can usually do almost anything you want in _custom files including redefining contexts in such a way that upgrades do not wipe them out. It's simply a matter of spending some time to see what is being done and

Re: [Asterisk-Users] Remote employees using Polycom 501 lose

The Polycom's need to have their registration time lowered. Set it to 60 seconds which will re-register every 30 seconds. The polycom doesn't have any sort of 'keep alive' feature to keep the NAT holes open. There is information on the wiki fruther describing this and how to set it up if you don't

Re: [Asterisk-Users] Asterisk home on VMWare time sync issues

Take a look at /etc/grub.conf and on the line(s) that look something like:kernel /vmlinuz-2.6.9-34.0.1.EL ro root=LABEL=/ add clock=pit so that it looks something like:kernel /vmlinuz-2.6.9-34.0.1.EL ro root=LABEL=/ clock=pit You will also want to install VMWare Tools and then in your

[Asterisk-Users] Passing DID to external number?

You already posted this. I answered it yesterday also?p<[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "Brian McCarey" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Fri, 23 Jun 2006 10:27:19 +0100Subject:

Subject: [Asterisk-Users] Passing DID to external number?

For the DID's the easiest way for you to trasmit the incoming DID is to create custom extensions for the external numbers that access the external trunk directly. (e.g. they should NOT go to Loca/. or it will not retain the orignal CID in freepbx- which is effectively how it is being sent when

Re: [Asterisk-Users] Canreinvite

How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd on the specific channel to see where the rtp streams are going. Or ... if this is the only active

Re: [Asterisk-Users] MOS Scores and LCR

I don't have any links, but there has been work done to 'measure' MOS scores and I believe they are a little more sophisticated than simply tracking latency, jitter and packet loss. An example of one box that does measure/predict MOS is Edgewater Network's Edgemarc. (I have no experience with it

Re: [Asterisk-Users] FreePBX 2.1.0: Manually rewriting

do you have selinux enabled? It should not be.pp.s. - if it comes to re-installing, you can backup all your settings with the freepbx backup utility and then restore so that you don't have to re-enter everything.From: "Lachek Butalek" [EMAIL PROTECTED]To: "Asterisk Users Mailing List -

Re: [Asterisk-Users] Recommended Web Interface

Dakota,freepbx is a web application and associated core dialplan that allows you to do many things on top of asterisk by generating the dialplan customizations ontop of the base that it provides. Once you spend some time understanding it, you can usually do most things that you want within the

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 4

Kevin,since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't that why it is failing when it hits the voicemail system? (sure

Re: [Asterisk-Users] Re: G729, voicemail, no codec_g729

Sorry for the repost - forgot to put the proper subject last time.Kevin,since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't

Re: [Asterisk-Users] Converting Voicemail wav to mp3

Aaron,any chance you've gotten that mp3 email file such that a blackberry unit can listen to it? (I've experimented but the blackberry just doesn't like mp3 attachments, just links?)thanks,philippe From: Aaron Daniel [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial

Re: [Asterisk-Users] Handset recommendations

I am also a VERY happy with the Polycom 501. I will disagree with the comment "They Just Work." when it comes to NAT. Once they are setup properly (e.g. set registration timeout to 60 sec so it registers every 30 seconds and keeps NAT holes open) then they work fine. There is good info on the wiki

Re: [Asterisk-Users] Handset recommendations

, Philippe Lindheimer wrote: I am also a VERY happy with the Polycom 501. I will disagree with the comment "They Just Work." when it comes to NAT. Once they are setup properly (e.g. set registration timeout to 60 sec so it registers every 30 seconds and keeps NAT holes open) then they

[Asterisk-Users] Ring-Answer with Polycom 501 and Asterisk

Peter,the configurations that I have seen do not auto-answer on CID. You need to set the Alertinfo field in the sip header in order to make this work. The polycoms do have an ability to customize the ring based on the caller which is set in the telephone's inernal directory. You may be able to

RE: [Asterisk-Users] macro-dial

Domenico,as I mentioned: "...and extensions are not necessarily what you think they areeither." AMP/Freepbx 'virtualizes' extensions. The basic concept is that there are users and then there are devices. A user can have multiple devices. The default shipping mode provides the 'extensions' tab

RE: [Asterisk-Users] macro-dial

I understand, seems like it might be easier to write a new dialplan from scratch though, vs. running into all sorts of strange issues? On the other hand, doing it your way will make you understand what freepbx is doing, which migh provide for your own ideas on how to do or not to do things in your

RE: [Asterisk-Users] FreePBX virtualization

A revision of what? of Freepbx? Can you elaborate?p To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comDate: Thu, 25 May 2006 14:34:00 -0400Subject: RE: [Asterisk-Users] FreePBX virtualizationWe have a revision of this that we use in house. We are

Re: [Asterisk-Users] Placing call files in

actually it sounds like a permission issue. You said you are doing it as root, but what is asterisk running as. I've found it is very sensitive, even to ownership. Make sure the owner:group is set to what Asterisk is running as before copying. Then, I've never had problems copying vs. moving -

Re: [Asterisk-Users] macro-dial

It's not that simple. dialparties is fundamental to the whole dialplan in AMP/freepbx and accomplishes a lot of the features such as hunt groups, DND, etc. And extensions are not necessarily what you think they are either. If you don't like it, you'd probably be better off writing your own

Re: [Asterisk-Users] macro-dial

It also provides the 'hunt' functionality and implements the different ring strategies.From: Avi Miller [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comDate: Thu, 25 May 2006 07:29:46 +1000Subject: Re: [Asterisk-Users] macro-dialMimmus

Re: [Asterisk-Users] Default dialplan??

Aaron,There are probably plenty of ways to do this, off the top of my head, if you add a 'include = go-to-pbx' context within the context where your Asterisk patterns are, and there is no match, Asterisk will then begin to check the 'include' contexts in order. (It does not even look at them

[Asterisk-Users] Re: DISA SPA3000 issues

Just tried it on mine, worked fine:Cellphone Call - POTS - SPA3000 - Asterisk - DISA - TelasipAs an FYI, I have my SPA3000 setup with INFO for the DTMF. When I originally installed it, I couldn't get the DTMF digits to work coming in using AUTO, which is why I have it using INFO (needs to

[Asterisk-Users] How to tell if RTP stream is has been reinvited?

I do a sip debug on the appropriate channel or IP address and look at the SIP messages. Would be great if there were an easier way though?pFrom: "Brent Torrenga" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Mon, 15 May 2006 12:52:19 -0500Subject: [Asterisk-Users] How to tell

Re: [Asterisk-Users] number that starts with star on PAP2

n't have my PAP2 under hand, but this is all I did, changedevery *xx to **xx and it worked.Something that may help you ishttp://www.netphonedirectory.com/pap2_dialplan.htm Philippe Lindheimer wrote: Yes - that's your problem. You need to porgram the dialpan in the PAP2 appropriately, disable functions

RE: [Asterisk-Users] Call Center Phone with Auto Answer

I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of

Re: [Asterisk-Users] number that starts with star on PAP2

In the PAP2's setup there are all of these "Vertical Service Activation Codes" that start with star and "Outbound Call Codec Selection Codes", also the setup menu is accessed by pressing star four times, could they be intefering with dialing numbers that start with a star? And is there any

Re: [Asterisk-Users] FreePBX in production?

Rich Adamson wrote: Let's see if I can summarize various recent postings relative to the broader topic of whether FreePBX/AAH is production-ready.It's not proper to put FreePBX/AAH in the same breath. AAH puts FreePBX ontop of their build, along with a bunch of other software. Although

Re: [Asterisk-Users] Polycom NTP issue

Kerry Garrison wrote: I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they are pointed to our the offset, i am getting the correct time, but 24 hours ahead. So for today it is showing Friday April 28 but with the correct