[Asterisk-Users] french sounds in asterisk

Hi all i want to know where i can find french sounds for asterisk. I don't have any studio to register good sounds. Bests regards Serge ___ Nouveau :

Re: [Asterisk-Users] Dumb question (hang up detection/Zapata.conf)

On Mar 14, 2006, at 3:53 AM, artifex maximus wrote: My asterisk system seems to have problems detecting hangups. I am getting a LOT of voicemails with dialtone or silence. I am using an external gateway (wellgate 3701a) and don't have zaptel at all. I think your 3701a don't understand

[Asterisk-Users] OT: any one in Stockholm for quick piece of advice

Apologies for off-topic, so I'll make this quick: any one in Stockholm Sweden willing to take an email off-list for a quick question. Most appreciated. Tackar s mycket. Jason SJOBECK ___ --Bandwidth and Colocation provided by Easynews.com --

[Asterisk-Users] Echo/Milliwatt Test Numbers in Oz ?

Is anyone aware of an milliwat/echo test number for telstra or similar? I want to fiddle a bit with gains but can't seem to get a hold of a miliwatt test number Im aware of a number they distribute for line quality testing for modems for BigPond, but I can't track it down -- Adrian Carter

Re: [Asterisk-Users] Creating a voip network... use asterisk?

Mark Hayward wrote: I wish to create a voip phone system used by different people accross the internet. I want certain people dotted around the country to be able to connect via voip to our main office. At first this will be using software phones but could extend to hardware based phones if it

[Asterisk-Users] asterisk configurations

I'm lerning to make my custom configurations. In extensions.conf, there is #include extensions_custom.conf [from-trunk]; just an alias since VoIP shouldn't be called PSTN include = from-pstn [from-pstn] include = from-pstn-custom

[Asterisk-Users] Numbered Voicemails when you still delete them.

Hello, I am using Asterisk 1.2.5 and the voice mail application. I don't actually want to keep the voicemails on the server so I am emailing them. However if I have the line in my voicemail.conf 1234 = 4242,Temp_User,voicemail,,delete=1 Then my voicemails are all sent with the same subject

[Asterisk-Users] OT: reset LinkSys 941 to factory defaults howto config' via TFTP

Dear All, 1. any one know how to reset the 941 to factory defaults? 2. any one know how to config' the 941 via flat configuration file via TFTP? sample file? URL? (LinkSys tech support has been just a hair above worthless to me thus far) Thanks very much. www.sjobeck.com

Re: [Asterisk-Users] module load order for Junghanns qozap and TDMcard

Hello, I would be delighted to know if you successfully bridged fax calls between both cards, as I suppose : - the Junghanns QuadBRI is connected to PSTN, - at least, one analog fax is connected to the Digium TDM card. Personnally, I had ongoing problems with fax calls crossing Junghanns BRI

[Asterisk-Users] Re: regexten

In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: A few days ago I didn't realise that the phones only registered to one Asterisk box. If I did, I wouldn't have spent hours today trying to get something working. I would have thought that it was a function of the phone how

Re: [Asterisk-Users] Numbered Voicemails when you still delete them.

Is there anyway, to delete a message, but still have some sort of incremental counter for the message id? Not that I am aware of. At least unless you script something yourself. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily

[Asterisk-Users] automatic fax detection in asteriskathome

How is working the automatic fax detection? I'm making tests in asteriskathome and the ivr plays, the fax sends little bips but asterisk don't detects it as a fax. (for testing I routed one caller id to the ivr). -- Alejandro Vargas ___ --Bandwidth and

Re: [Asterisk-Users] french sounds in asterisk

serge messa wrote: Hi all i want to know where i can find french sounds for asterisk. I don't have any studio to register good sounds. Free sounds available now (recorded in a studio by my wife): http://www.sineapps.com/FrenchPrompts.tar.gz - ready to go in GSM format Also available:

Re: [Asterisk-Users] voip-info.... again

Greg Oliver schrieb: On Thu, 2006-03-16 at 18:39 -0500, Alexander Lopez wrote: I have offered but I don't think he (owner) id open to that. This matter was discussed earlier, the problems seems to be in the difficulties of mirrowing an database-driven wiki. Tobias

RE: [Asterisk-Users] Queues Not Reporting Estimated Hold Time

Nope, never removed them, they are still there. It doesn't report an error either, it just never says playback . If this works for someone please let me know, otherwise I will report it to the bug tracker. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

[Asterisk-Users] embedded hardware for Asterisk?

Hi, Is there any specific made embedded hardware designed VoIP software or Asterisk? I want to build a router that have VoIP enabled, so that I can use it connect to a VoIP ISP. Thanks Sam ___ --Bandwidth and Colocation provided by Easynews.com --

[Asterisk-Users] TFTP problems on FC4

Greetings to all. I am hoping someone can help me out with a problem I am having getting my Cisco phones, 7960s and 7940s, to download the appropriate files from our TFTP server. The TFTP server is running on Fedora Core 4. The TFTP server appears to be setup properly: service tftp {

Re: [Asterisk-Users] Avaya IP Office 412

Hi Mark, Do you have the right cable? You need a cross-over T1 cable and NOT a cross-over ethernet cable that people commonly try. This should satify the electrical requirements and turn the lights green. Yes, i build up my own cable with the schemas available on Net, but it was not

Re: [Asterisk-Users] UK Caller ID - Asterisk 1.2.5 - TDM4 Card

On Thu, Mar 16, 2006 at 11:28:34PM -, Magnus Kelly wrote: Has any one found any issues (bug?) with ver 1.2.5 as CallerID generated on an outgoing FXS port (to the handset) fails when UK tones are used, with a message 'Didn't finish Caller-ID spill. Cancelling.' Any tips on getting this

Re: [Asterisk-Users] saydigits

snip I tried say digits 123 and saydigits 123 both gave no application error /snip 1)its saydigits as in one word and not two 2)As with a lot of functions in asterisk thre data that you are working with has to be in parentheses i.e. Exten = 123,1,Answer Exten = 123,2,Saydigits(1234567890)

[Asterisk-Users] Is LDAP working on 1.2.X

Hello, Does anyone know if LDAP is working or broken on 1.2.X ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

RE: [Asterisk-Users] OT: Unblocking bloced CID

Thanks a lot for all the info guys. --- Shanon Swafford [EMAIL PROTECTED] wrote: I can't remember the details but several years ago in my former life as a CLEC, we had an issue where a gov't official whose caller ID was blocked called his daughter who was a customer of ours. She

RE: [Asterisk-Users] TFTP problems on FC4

Hi Joe, Maybe there it is a problem related to the rights on the specific files. Please check if you have read rights for everybody for the files under /tftpboot. Also check that tftpboot have r+x rights for everybody. Regards, Ioan. www.modulo.ro -Original Message- From: [EMAIL

Re: [Asterisk-Users] Outbound paging dialplan example?

For setting the oncall solution you can do as follows Play the following message to reach the on call tech/doctor etc. press 1 Then have the on call person set the number that they want to be reached at. [SetOnCallTech] Exten = _1NX,1,Answer Exten =

[Asterisk-Users] asterisk and skype - asterisk newbie

Hi all, I just set up a small asterisk box at home and it works as expected, I have some relatives in USA, I usually use skype to speak to them, is there anyway to connect skype and asterisk. I'd like to use skpe as an extension or a channel with asterisk. Thanks in advance for any suggestion.

Re: [Asterisk-Users] Echo canceller data-points

On 3/16/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Wednesday 15 March 2006 16:46, Steve Davies wrote: I thought I would try the 1.2 trunk/HEAD version of MG2 with the extra knobs and whistles, but found 2 problems. This version trains even a normally clean line in about 10 seconds,

[Asterisk-Users] One-Way SIP Audio with SVN Codebase

Please tell me the obvious mistake I'm making here. (And yes, I well know about NAT and one-way audio problems in general.) I want to try the new T.38 passthrough stuff, downloaded it, built it, tested it with an SPA-2100 and can hear announcements fine but echo test shows no audio outbound

Re: [Asterisk-Users] Question about advanced IVR

I Would reccomend learning asterisk bit by bit. It's the only way to do it. Here are several tools that you can use. 1)Get [EMAIL PROTECTED] Its a great resource for starters. You can learn a lot from it. You can get it at http://asteriskathome.lists.digium.com 2)Read the book Asterisk, The future

[Asterisk-Users] Call pickup between different protocols

Hi, I'm unable to pickup a call (*8) directed to a SIP phone from a IAX2 phone. Is it normal? I don't see ant pickupgroup/callgroup setting in iax.conf... -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] All calls in queue go to agent that is down??

snip and the user had forgotten to log out of the queue all calls go to this agent that is no longer connected to the system. I have tried training and retraining but I need some way to fix this. /snip Well you can do what some did here a while back. If you forget to log off, each time

[Asterisk-Users] Best budget IP phone at the moment?

Hi, I am looking for a budget IP phone that can use preferably iLBC or GSM codecs.. Suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] asterisk and skype - asterisk newbie

2006/3/17, adriano ghezzi [EMAIL PROTECTED]: Hi all, I just set up a small asterisk box at home and it works as expected, I have some relatives in USA, I usually use skype to speak to them, is there anyway to connect skype and asterisk. I'd like to use skpe as an extension or a channel with

Re: [Asterisk-Users] Best budget IP phone at the moment?

We are using the Grandstream GXP-2000 which works very well. On Fri, 2006-03-17 at 12:11, WipeOut wrote: Hi, I am looking for a budget IP phone that can use preferably iLBC or GSM codecs.. Suggestions? ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] asterisk and skype - asterisk newbie

2006/3/17, adriano ghezzi [EMAIL PROTECTED]: I'd like to use skpe as an extension or a channel with asterisk. By the way, check this: http://www.voip-info.org/wiki-bounty+skype -- Alejandro Vargas ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] Queues Not Reporting Estimated Hold Time

I'm getting the same thing since upgrading from 1.0.x to 1.2.x - no queue hold time announcements. There are other oddities in queues in 1.2.x compared to 1.0.x too. But I'm always afraid to raise them as bugs in case they are not, and 1.0.x was going things the wrong way and 1.2.x is going

Re: [Asterisk-Users] ISDN BRI and UK Premium Rate Numbers

This is pretty standard Asterisk behaviour exten = whatever,1,NoOp exten = whatever,2,Dial(SIP/nSIP/n+1SIP/n+2) exten = whatever,3,Hangup The incoming ISDN call will ring the specified SIP phones, and will not be answered until one of them picks up. As simple as that? Thanks!!

Re: [Asterisk-Users] OT: Unblocking bloced CID

On Thursday 16 March 2006 22:56, Jonathan k. Creasy wrote: Well, whether I SHOULD get it or not may be totally irrelevant to whether I CAN or DO get it. The caller ID info is most definitely there and it shows up in my CDR records. However, it is not displayed on the device because only the

[Asterisk-Users] How to install the cdr_odbc module.

Hi. I want to the CDR detail to be logged to a database backend. For that I need the cdr_odbc module. Executing the CLI command show modules I realized I don't have the cdr_odbc module loaded. So, I tried to load it using load cdr_odbc.so, but it didn't work because I don't have it. Where

[Asterisk-Users] Asterisk on hosted server

I'd planning on install Asterisk on a hosted Linux box we're setting up. The hosting provider that seems to offer the best deal can install either Debian 3.1 or SUSE 9.x or 10.0 in either 32 or 64 bit editions (running on AMD 64 bit). My experience has been gained on RedHat to date, but I do

[Asterisk-Users] Keeping the user name in sip INVITE with fixed IP host routing.

Hi, I would like to set a sip phone for a user with fixed IP but I would like also to keep the user name in the invite, how is that possible? Is there any sip.conf setting that can be used for that? My sip.conf ; [rui] type=friend secret=oit host=127.98.12.88 canreinvite=no

[Asterisk-Users] [FOLLOWUP]: Calls not tearing down properly

Title: [FOLLOWUP]: Calls not tearing down properly As a follow-up, I have traced the PRI connection and discovered that calls hung up by a local BCM voice station (analog, digital, whatever) send a disconnect with a clear cause of 1 (Number not assigned) instead of a normal clear cause of 16.

Re: [Asterisk-Users] Question about advanced IVR

2)Read the book Asterisk, The future of telephony. You can buy it or download it for free. I dont have the link to it but if some one else does please post it. http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 ___ --Bandwidth and

[Asterisk-Users] Set CallerID to a specific Queue Member

Hi all I have my asterisk connected to different SIP Providers. I had to idea to create something like a Helpdesk queue ringing all my friends connected to those SIP providers. Now my problem is, that I have to set the correct callerID if a call goes out to one of those sip providers. I can

[Asterisk-Users] Analog POTS line - Rhino FXO Channel Bank - No Hangup

Hello list, I have recently deployed Asterisk as the phone system for my office. So far, everything has been going really well, except for one little thorn in my side. I have a set of 6 analog lines that are connected to a TE411P via a Rhino FXO Channel bank. If I call the analog

[Asterisk-Users] D4 AMI - No Caller ID

I currently have Asterisk deployed in my office with a TE411P. On the first port of this card is a T1 from the telco setup for D4 AMI. Unfortunately, I'm not receiving caller ID on inbound calls from this line. The caller ID information is arriving in the form *ANI*DNIS*. In zapata.conf, I

Re: [Asterisk-Users] Analog POTS line - Rhino FXO Channel Bank - No Hangup

[EMAIL PROTECTED] wrote: If so, is there a way to detect the hangup? Check out http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html for some possible clues. - Mike ___ --Bandwidth and Colocation

[Asterisk-Users] TDM 2400 With 24 FXO

Hi, Have someone there tried the TDM 2400 with 24 FXO? Have had echo problems? or any other problem ? Recommendations? Optional echo cancellation modules are necessary? TIA, Fernando ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] embedded hardware for Asterisk?

Hi, Is there any specific made embedded hardware designed VoIP software or Asterisk? I want to build a router that have VoIP enabled, so that I can use it connect to a VoIP ISP. Thanks Sam There are plenty of small form factor boards you could start with (http://www.linuxdevices.com/),

[Asterisk-Users] (no subject)

Does anyone have a DISA alternative? I currently use the line: exten = s,16,DISA(no-password|from-internal) however that just drops a user at a dial tone, what I would like to do is prompt user for number to dial, followed by the # key, and then have asterisk dial out. Can this be

[Asterisk-Users] Voice volume using Monitor application

When you say that you tweaked the volumes, is that modifying the Asterisk code to call sox and soxmix or are you mixing outside of Asterisk. Also, I used Sox to increase the volume and then Soxmix to mix the two audio files. Is there a way to just use soxmix to increase the volume on one

[Asterisk-Users] Asterilink?!?!

Hi, Anyone know anything about asterlink? I've heard good things about them. So I thought I'd give them a try. I went to sign-up however, I haven't heard from them, I've called them several times between 9am and 5pm EST and no responce. Called today again after 9am EST, no responce. Tried

[Asterisk-Users] choppy recorded sounds in asterisk

I have installed asterisk on numerous servers. Every install was done on Fedora and (White box Linux). I now have zap channels in one of the boxes (T-1). No matter what type of channel I call on (sip or zap) I get some really strange artifacts in the sound, almost like a skip in the

RE: [Asterisk-Users] automatic fax detection in asteriskathome

I had a similar problem related to rx gain. You can use ztmonitor to make sure you're getting a reasonable number of '#'s during the call to rule that out. Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Friday,

RE: [Asterisk-Users] RE: DUNDi .... Halfway and CLUSTERING

At the moment I'm out of the office, but when I return I'll be certain to do that. Note that my solution is different from what you are working on with regexten, though I suspect some of the challenges that I've faced and overcome are not. I'm actually using UltraMonkey for load-balancing and

[Asterisk-Users] RE: choppy recorded sounds in asterisk

I'm sure it's unrelated, but this reminds me how softphones on my HP notebook do this. Everything else is fine, but any kind of softphone flakes out a bit on SIP calls. Other devices do not display this behavior. Have you tested different phones against your set-up? Bob McDowell

RE: [Asterisk-Users] DISA alternative

You can do something like this: answer background(enter-number-wish-dial) Waitforexten(10) Dial({TECH}/${EXTEN} or Goto (outsidetruck,${EXTEN},1) where outsidetrunk is a context that checks te number and routes the call. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On

[Asterisk-Users] Re: DUNDi .... Halfway and CLUSTERING

Brad, How are you able to overcome the Call-ID stickiness problem when loadbalancing with Ultramonky? As I understand it LVS does not properly support SIP in that it doesn't always use the same source path. regards, David On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote: At the moment I'm

RE: [Asterisk-Users] Question about advanced IVR

There are some good examples of IVR on the wiki (insert prepared response to comment about the availability of said wiki)... http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu The wiki also says you can use 'ControlPlayback' to satisfy parts of 3c. Note I've never tried it...

Re: [Asterisk-Users] New ncurses Asterisk Manager Interface

The build problem is there isn't berkeley db headers installed on your system. I'll try to find the correct method of detecting this. In the mean time you can edit assman/Makefile and add -I/usr/include/db3 (or whereever your db.h is located) to the CFLAGS variable.On 3/16/06, El Flynn [EMAIL

[Asterisk-Users] problems with emailing voicemail

Hi, I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to our configuration I bet I'm expereiencing a Linux problem rather

Re: [Asterisk-Users] New ncurses Asterisk Manager Interface

Excuse me, change the lines in the makefile like so. (db3 is not new enough) db4.2 is requried.depending on how your distro packages these headers you'll find more than one db.h (Gentoo does atleast)CFLAGS=-I/usr/include/db4.2 -I../inc -Wall LDFLAGS=-ldb-4.2On 3/17/06, Sig Lange [EMAIL PROTECTED]

[Asterisk-Users] Re: problems with emailing voicemail

In article [EMAIL PROTECTED], hugolivude [EMAIL PROTECTED] wrote: I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to

Re: [Asterisk-Users] Advice on configuration

Hi Peter, Sorry it took me so long to get back to you. Most of your config worked great but I obviousley did something wrong on the Sipdiscount side :-( I am getting my contexts confused in extensions.conf Mar 17 14:25:26 WARNING[7607]: pbx_config.c:1753 pbx_load_module: Unable to include

[Asterisk-Users] Asterisk and PacketCable

Hi group!I'm very interested on VoIP and trying to make some tests with 2 eMTA motorola (SBV5120) but with the PackectCable architecture I need a softswitch. My question is: is Asterisk and PacketCable compatible? If they are, how can I configure it to act as a Call Management Server into my

[Asterisk-Users] Re: Asterilink?!?!

Just wanted to post a follow-up. Just got a call from Brian West, who was very helpful. Apparently their sales/support guy is at Von this week, and they forgot to forward the phone. But he called me within 30 minutes of me sending this e-mail so kudos to Asterlink.. and I hope the rest of my

Re: [Asterisk-Users] embedded hardware for Asterisk?

Adam Moffett wrote: Hi, Is there any specific made embedded hardware designed VoIP software or Asterisk? I want to build a router that have VoIP enabled, so that I can use it connect to a VoIP ISP. Thanks Sam There are plenty of small form factor boards you could start with

RE: [Asterisk-Users] problems with emailing voicemail

Title: Message I'm not real knowledgably in Linux, but have you loaded Webmin so you canlook at the status andmessages in sendmail? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hugolivudeSent: Friday, March 17, 2006 9:06 AMTo:

Re: [Asterisk-Users] problems with emailing voicemail

On Fri, 2006-03-17 at 10:05 -0500, hugolivude wrote: Hi, I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to our

[Asterisk-Users] RE: Clustering

Date: Thu, 16 Mar 2006 21:35:44 -0800 (PST) From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Clustering To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Quick question, you said a lot of your clients are using

RE: [Asterisk-Users] New one on me: How to UN-transfer

app_pickup2.c in this won't compile until I get app_pickup.c. The docs say that it is a technology independent way to pick up calls, but it is bundled in BRIStuff. Damned if I could find it last night, it's not in the bristuff from Junghann's. anyone know how to get app_pickup.c ? -Original

[Asterisk-Users] CRM + Phones

Friends, I have Asterisk up and working with SugarCRM and want to be able to use a directory-like phonebook in a voip phone. What phones have people used that integrate easily with SugarCRM? I know I could use the Cisco ones but am trying to avoid such a high price. Wired or wireless

[Asterisk-Users] Countries supporting SMS on PSTN (ISDN)

Hi does anyone have a definitive list of countries other than those listed on the wiki which are supporting app_SMS on landlines using ETSI ES 201 912 ?? Thanks Tim Robinson Basingstoke, UK ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5

jitterbufferfor svn trunk + jitterbuffer jitterbuffer-1.2 for 1.2 + jitterbuffer test-this-branchfor the test branch with a lot of cool stuff including the jitterbuffer I installed the jitterbuffer-1.2 branch and I have a few questions. First

Re: [Asterisk-Users] D4 AMI - No Caller ID

Hello, Been there, not fun, just got rid of my last D4/AMI T1 last month. Are you receiving **DNIS* from your telco, or *1234567890*DNIS*? Is that what you mean by not receiving any CallerID? With RBS(Ribbed Bit Signalling) sending digits is the only way to send callerID, and no name is sent

[Asterisk-Users] Sticky Problem SER/Asterisk

Trying to find a solution to a sticky problem here. We have 3 OpenSER systems. Phones register with the OpenSER systems, and after they authenticate the user, pass the registration info using OpenSER's send() command to all Asterisk boxes sitting behind them. Each asterisk system then knows

RE: [Asterisk-Users] D4 AMI - No Caller ID

RBS would be 'Robbed Bit Signalling'. 'Ribbed Bit Signalling' is a much more ticklish subject. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Florell Sent: 17 March 2006 16:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

RE: [Asterisk-Users] problems with emailing voicemail

Try sending email from the command prompt first... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hugolivudeSent: Friday, March 17, 2006 10:06 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] problems with emailing voicemail Hi, I'm

RE: [Asterisk-Users] choppy recorded sounds in asterisk

This may not solve your problem but I think it is a good place to start. Have you tried using Asterisk native sounds so that you are not doing any transcoding? -Original Message- From: Jordan Novak [mailto:[EMAIL PROTECTED] Sent: Friday, March 17, 2006 6:13 AM To:

[Asterisk-Users] caller unable to transfer

Hey all, posted this the other day, but re-read it realized I didn't give enough info to be useful.. so I thought I'd try again.. I'm using AAH 2.0 (* 1.2.0) and am unable to transfer a call when I initiate the outgoing call. In AMPs general settings, I've tried changing the Dial command using

Re: [Asterisk-Users] Asterisk RealTime Question, Please help

I need many contexts because I have around 1000 DID's each with 5-10 Extensions. These DID numbers are changed or added very frequently and whenever there is a change I have to change Extensions.conf manually. So please tell me how can I do this dynamically without changing

[Asterisk-Users] Disappearing voicemail

Asterisk 1.2, Fedora Core 4: When I leave a voicemail message, it writes the necessary files to the INBOX: msg.gsm msg.txt msg.wav msg.WAV When I hang up, the files are erased. There is no indication of anything untoward in the logs: -- x=0, open writing:

RE: [Asterisk-Users] Re: regexten

Any thing better than five-nines is a much better sale pitch. LoL. BTW. My Cisco 7960s alows for a entry of a backup proxy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Friday, March 17, 2006 4:04 AM To:

RE: [Asterisk-Users] Re: problems with emailing voicemail

Do a tail -f /var/log/maillog which will give you a real-time view of your mail server activity, then while that's running, leave yourself a voicemail. From: Tony Mountifield [mailto:[EMAIL PROTECTED] Sent: Fri 3/17/2006 10:13 AM To:

RE: [Asterisk-Users] DISA alternative

Mine looks like this now: exten = s,1,Answer exten = s,2,Authenticate(1234) exten = s,3,Background(pls-entr-num-uwish2-call) exten = s,4,Read(EXTENArea=$EXTENArea|10|2|15) exten = s,5,Noop(EXTENArea=${EXTENArea=${EXTENArea}} exten = s,6,Goto(outsidetruck,${EXTENArea},1)

Re: Re: [Asterisk-Users] D4 AMI - No Caller ID

It appears that I'm getting **DNIS*, as my setup in extensions.conf uses the DNIS, and that all works fine. Is there some debug logging I can turn on to see if that is in fact what I'm getting? Thanks, James From: Matt Florell [EMAIL PROTECTED] Date: 2006/03/17 Fri AM 11:08:14 EST To:

[Asterisk-Users] RE: Disappearing voicemail

Never mind. The system was configured to delete the messages once they have been emailed. Sorry. Please ignore the previous message. Asterisk 1.2, Fedora Core 4: When I leave a voicemail message, it writes the necessary files to the INBOX: msg.gsm msg.txt msg.wav

Re: [Asterisk-Users] Development news :: T38 passthrough support

On Wed, Mar 15, 2006 at 11:26:45AM +, Paul Hayes wrote: The SPA-2100 is the only one to support T.38 at the moment though. SPA-2002 has the ability to support t.38 (i.e. it has the processing power required) but the firmware support isn't there yet. Per our sales contact at Cisco for

[Asterisk-Users] Aastra Questions

Hi, Does anyone have experience with Asterisk and the aastra 9112i or 9133i phones? I am looking at purchasing some, and was curious how quality, and stability was with them. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Transfer problems revisited

Hey all, an odd update to my previous note about not being able to transfer if I'm the 'caller'... I set the dial command (in amp) to T only... now, If I call into the pbx from outside and reach an extension, I CAN hit # on the calling phone (outside, from-pstn) and get the 'transfer' message

Re: [Asterisk-Users] [EMAIL PROTECTED] V's Asterisk

You can do that as well. Its really not that difficult to add to the dialplan. On 3/17/06, Ira [EMAIL PROTECTED] wrote: At 09:04 PM 03/16/2006, you wrote: 6 different companies with 6 different IVR's and different ring groups and options. All of the companies no nothing about the other

[Asterisk-Users] problem with tdm22b

Hi everyone, I have a problem installing interface card tdm22b in a debian etch machine. First I added manually the zaptel module: apt-get install zaptel-source kernel-headers-`uname -r` m-a a-i zaptel Then I do tumiwall:/usr/src# ztcfg -vv Zaptel Configuration ==

RE: [Asterisk-Users] Aastra Questions

I have one 9133i in a test environment and prefer it to the Polycom 301. I find the 9133i very similar to the Norstar sets we will be replacing. This is especially true in the way it presents lines. I have a hard time explaining it, but in the Aastra/Norstar world when you need another line, it

RE: [Asterisk-Users] New ncurses Asterisk Manager Interface

Your manager interface is not so bad: simple but working as a charm. I love ncurses interfaces! A goodexercise of programming with Asterisk Manager API. Keep up the good work. Mimmus ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] Re: problems with emailing voicemail

Wow, Thanks so much for all your help. I tried Steve's suggestion using tail and found: from=[EMAIL PROTECTED], ... stat=Deferred: 450 [EMAIL PROTECTED]: Sender address rejected: Domain not found So it looks like the sender email is no longer acceptable. This worked fine b4, so perhaps the ISP

Re: [Asterisk-Users] Voice volume using Monitor application

Jeff Hoppe wrote: When you say that you tweaked the volumes, is that modifying the Asterisk code to call sox and soxmix or are you mixing outside of Asterisk. Also, I used Sox to increase the volume and then Soxmix to mix the two audio files. Is there a way to just use soxmix to increase the

Re: [Asterisk-Users] Aastra Questions

On Fri, 2006-03-17 at 12:11 -0500, Matt wrote: Hi, Does anyone have experience with Asterisk and the aastra 9112i or 9133i phones? I am looking at purchasing some, and was curious how quality, and stability was with them. I'm more than happy with my 9112i, handset feels good, the screen is

Re: [Asterisk-Users] problem with tdm22b

Diego: I have a tdm22b and the first two ports are FXS and the next two FXO. So the zaptel.conf must be as: fxoks=1,2 fxsks=3,4 Saludos Daniel Pizarro Infobox Peru ex alumno de la PUCP On 3/17/06, Diego Quintana Cruz [EMAIL PROTECTED] wrote: Hi everyone,I have a problem installing interface

Re: Re: [Asterisk-Users] D4 AMI - No Caller ID

If you set debug 41 in Asterisk CLI and watch a call come in your should be able to se the extension it is sending it in as. MATT--- On 3/17/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: It appears that I'm getting **DNIS*, as my setup in extensions.conf uses the DNIS, and that all works

Re: [Asterisk-Users] Re: problems with emailing voicemail

in /etc/hosts edit the file and add your external IP or dyndns name to the 127.0.0.1 address: 127.0.0.1externalip asterisk localhost so just add it as the first entry after 127.0.0.1 Then restart sendmail and see if it picks up the new address from which to send. If not just reboot the

Re: [Asterisk-Users] Aastra Questions

Very Good, We are actually replacing a Norhell phone system with these, so that's why I liked the look. I don't know that I understand the statement it just sort of gives you a line when you need it ? On 3/17/06, Bob McDowell [EMAIL PROTECTED] wrote: I have one 9133i in a test environment and

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