[asterisk-users] snmp commands not working with the Asterisk MIB ( Asterisk 1.4)

Hi, I have installed the Asterisk 1.4 package which support snmp operation for Asterisk MIB. The Asterisk server is running smoothly .Fine . But I can't perform any snmp operation in the Asterisk MIB. Please let me know if any other files need to be configured so that I can perform the s

Re: [asterisk-users] Asterisk integration with IBM Sametime

Hi Ram, Thanks for the suggestion. What would be the form of communication between Sametime server and the conference bridge on asterisk? My key concern is that Sametime server doesnt have a SIP interface (because its a trial version) so how would it talk to Asterisk? Regards, Reshmi _

Re: [asterisk-users] Dedicated Codec Conversion Server

The way I would accomplish this is to have 2 Asterisk boxes. Your conversion server would just have a dial plan to forward all calls to the Asterisk box that has the PSTN interface. Once the PSTN Asterisk Server receives the calls it just routes the call based on dial plan rules. {Internet -> VP

Re: [asterisk-users] Asterisk Shutting Down

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Totaro wrote: > Matt Riddell wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Steve Totaro wrote: >> >>> Matt Riddell wrote: >>> -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Schall wrote: >

[asterisk-users] Dedicated Codec Conversion Server

I have a need for a large number of remote phones. I want to use GSM between the phones and the conversion server which will transcode to ulaw eventually send or recieve calls via the PSTN (ulaw). I am curious is anyone has any ideas on the easiest way to create a dedicated codec conversion bo

[asterisk-users] Polycom Provisioning Tool

Not sure if one exists, but someone had asked me for this a while ago. Here it is! My Polycom Provisioning Tool. Notice the version is 0.0.1. Just a concept program (but it works well). I am open for suggestions, feature additions, and bug fixes. Email me with any requests. I want to improve th

Re: [asterisk-users] Asterisk Shutting Down

Matt Riddell wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Steve Totaro wrote: > >> Matt Riddell wrote: >> >>> -BEGIN PGP SIGNED MESSAGE- >>> Hash: SHA1 >>> >>> Rob Schall wrote: >>> >>> We've experienced the same problem twice now in the past month. Th

Re: [asterisk-users] TE210P issues

Jerry Geis wrote: > I have a box with a TE210P. Things work for a while then stop when > making call files. > I get NOANSWER as the return code (right away). > > I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1 > > When I try to update to newer zaptel the machine locks when loading the

Re: [asterisk-users] Asterisk Shutting Down

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Totaro wrote: > Matt Riddell wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Rob Schall wrote: >> >>> We've experienced the same problem twice now in the past month. The >>> asterisk pid stops responding. We aren't able to con

Re: [asterisk-users] Compatibility Issues with dell poweredge 195 and TE110P card

We had issues with TE110p cards in Dell 860's, but TE120p's fixed the problem. PaulH On Wed, 2007-10-24 at 21:12 -0400, Joseph Begumisa wrote: > > > > Has anyone had any compatibility issues with a TE110P card installed > > on a Dell Poweredge 1950? I noted the following error on the LCD > > d

[asterisk-users] Gentilini, Paul is out of the office.

I will be out of the office starting 10/24/2007 and will not return until 10/29/2007. I will be on vacation Thursday and Friday and traveling on Business 10/29 - 11/2. I will make every effort to respond to emails in a timely fashion and emergency requests should be directed to the IntelliCare

Re: [asterisk-users] Asterisk Shutting Down

Matt Riddell wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Rob Schall wrote: > >> We've experienced the same problem twice now in the past month. The >> asterisk pid stops responding. We aren't able to connect to the CLI and >> all calls are dropped. The lots are pretty bare as we

Re: [asterisk-users] TE210P issues

I use TE212P, it shoudl work without errors. I use it with Asterisk 1.2.18 + zaptel-1.2.17.1 On RHEL 4.4 On Dell PowerEdge 850 It may be that the card is bad, try contacting Asterisk support. I had one bad card when I first got it, the 2nd one worked . -- Deepak Jerry Geis <[EMAIL PROTECTE

Re: [asterisk-users] Compatibility Issues with dell poweredge 195 and TE110P card

> > Has anyone had any compatibility issues with a TE110P card installed > on a Dell Poweredge 1950?  I noted the following error on the LCD > display of the Dell Poweredge 1950: > > > > E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. >Yes, I have had this problem with a dell PE1650, 1850,

Re: [asterisk-users] Asterisk Shutting Down

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Schall wrote: > We've experienced the same problem twice now in the past month. The > asterisk pid stops responding. We aren't able to connect to the CLI and > all calls are dropped. The lots are pretty bare as well. > Asterisk 1.2.13-r1 Unfortuna

Re: [asterisk-users] How to tune Asterisk AMD - Answering Machine Detection "hacks"

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Costa Dinoteli wrote: > Hello Everyone, > > Can someone point me to reliable links on how to tweak Asterisk AMD > I am calling a number and have to two files to play depending if it is a > real person or an > answering machine. > Most everytime Asteri

Re: [asterisk-users] Asterisk under VMWare

> Our testing has yielded pretty good results. We had 10 simultaneous > calls with ulaw and quality was very good. We are pure VOIP also. How many VMs were you running at the time, and what load were they under? We've setups running between 3 and 5 VMs on a single box (multi-core, lots of RAM, e

[asterisk-users] RTP multicast support

Anyone know what SIP phones support RTP multicast intercom or MOH. I am working on a project that a client needs to page 150 phones at the same time. I have clients that have 40 phones working with a custom script that I wrote that checks to see if there on the phone and if not puts them in a meetm

Re: [asterisk-users] Voicemail playback on iPhone

Am Mittwoch, den 24.10.2007, 11:19 -0500 schrieb Peder @ NetworkOblivion: > This is semi-related, but I have a Tmobile MDA and I couldn't play the > files either. The issue was not a codec issue, it was an email encoding > issue. If I sent the message to an email account and it was then > down

Re: [asterisk-users] AstManProxy Host Prefix?

Did you count the number of $'s? ;-) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Bright Sent: Wednesday, October 24, 2007 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AstManProxy Ho

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

On Wed, Oct 24, 2007 at 06:32:45PM -0500, Jonn R Taylor wrote: > Jason Parker wrote: > > Will the madness never end? Aparantly, not. The message I have quoted told me three times how to unsubscribe from the mailing list (not counting the fourth one added to the post by the mailman after posting).

Re: [asterisk-users] How to get TCP access to CDR Master.csv

>#!/bin/bash >while true; do > tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l >done Thank you John, this bash script is exactly what I was looking for. Very simple, yet works. As for doing this with insert into database and then polling for it... well I don‘t like po

Re: [asterisk-users] How to get TCP access to CDR Master.csv

On Wed, Oct 24, 2007 at 10:29:41PM -, [EMAIL PROTECTED] wrote: > Hi. > > I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time > from a remote connection coming in on TCP. Basically what I have is a > Windows application that is used to process incoming, outgoing and mis

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

Jason Parker wrote: > See response in-random-lined. > > David Gomillion wrote: >> >> On 10/24/07, *David Gomillion* <[EMAIL PROTECTED] >> > wrote: >> >> On 10/24/07, *Steve Totaro* <[EMAIL PROTECTED] >> > wrote: >> >> Let me screw

Re: [asterisk-users] Voicemail playback on iPhone

I tested this again, and wav files do play as attachments with firmware 1.1.1. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Eric Chamberlain > Sent:

Re: [asterisk-users] How to get TCP access to CDR Master.csv

I’m no expert in this field bit I would have though logging the calls to MySQL and then queering the MySQL database would be the best not to mention the easiest way to get the details you are looking for. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Hass Sent: Thursday,

Re: [asterisk-users] How to get TCP access to CDR Master.csv

Hello, I am not sure if I totally understand the question but if your looking to stream the connection you could create a simple bash script like this #!/bin/bash while true; do tail -f /var/log/asterisk/cdr-custom/Master.csv | nc -p 1024 -l done There probably is a better solution the

[asterisk-users] How to get TCP access to CDR Master.csv

Hi. I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time from a remote connection coming in on TCP. Basically what I have is a Windows application that is used to process incoming, outgoing and missed call records putting them into a database for some analysing etc. This

Re: [asterisk-users] Voicemail playback on iPhone

Convert the voicemail to a mp3 file. As of firmware version 1.1.1, the iPhone mail application will recognize, but not play wav attachments. But the mail application does, recognize and play mp3 file attachments. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

See response in-random-lined. David Gomillion wrote: > > > On 10/24/07, *David Gomillion* <[EMAIL PROTECTED] > > wrote: > > On 10/24/07, *Steve Totaro* <[EMAIL PROTECTED] > > wrote: > > Let me screw this thread up by top posti

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

On 10/24/07, David Gomillion <[EMAIL PROTECTED]> wrote: > > On 10/24/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > > > > Let me screw this thread up by top posting now. > > > > Could echo be caused by late packets if jitterbuffer is on or something > > or would that just cause lag? > > > > Thanks,

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

On 10/24/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > > Let me screw this thread up by top posting now. > > Could echo be caused by late packets if jitterbuffer is on or something > or would that just cause lag? > > Thanks, > Steve So, does this qualify as an in-line reply, or a top post? Maybe

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

Let me screw this thread up by top posting now. Could echo be caused by late packets if jitterbuffer is on or something or would that just cause lag? Thanks, Steve kevin bergner wrote: > On 10/24/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > >> Jonn Taylor wrote: >> >>> Eric

Re: [asterisk-users] reload manager.conf

On Wed, 2007-10-24 at 13:31 -0700, Richard Lyman wrote: > every time there is a new connection to the asterisk manager > interface, the manager.conf file is reread. > (meaning, it reloads itself) Great. Thanks for your help! ___ --Bandwidth and Colocati

Re: [asterisk-users] TE210P issues

Calling Digium. Post your /var/log/messages and /var/log/asterisk/full (just anything that looks relevant). Try a Sangoma card. Thanks, Steve Jerry Geis wrote: > I have a box with a TE210P. Things work for a while then stop when > making call files. > I get NOANSWER as the return code (right

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

On 10/24/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > Jonn Taylor wrote: > > Eric "ManxPower" Wieling wrote: > >> Any echo you hear on pure IP calls is caused by the endpoint phone. You > >> cannot do ANYTHING about it on Asterisk. > >> > >> > >> Jonn Taylor wrote: > >> > >>> Any ideas

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

Alan, I'm glad to see that you are able to run zaptel and OSLEC following my tweak! Some days ago I've sent to David Rowe a little patch that preserves the echo cancel status between calls. I'm using it since several weeks with my TDM400P home based PBX and I think that's a really effective solu

[asterisk-users] TE210P issues

I have a box with a TE210P. Things work for a while then stop when making call files. I get NOANSWER as the return code (right away). I am running asterisk 1.2.12.1, libpri 1.2.3 and zap 1.2.9.1 When I try to update to newer zaptel the machine locks when loading the zaptel drivers. I tried to

Re: [asterisk-users] reload manager.conf

Bob Pierce wrote: > I've made a change to my manager.conf file in asterisk 1.2.18 > > Is there a way to reload that config file from the CLI without > restarting asterisk? > > Bob > > every time there is a new connection to the asterisk manager interface, the manager.conf file is reread. (meani

Re: [asterisk-users] bugs.digium.com

On 22:06, Wed 24 Oct 07, Michiel van Baak wrote: > On 15:38, Wed 24 Oct 07, Doug Lytle wrote: > > > > The bug tracker seems to be down. > > And so is the public svn and downloads.digium.com and > ftp.digium.com and the websvn. > They are working on it. And it's working again for me -- Michiel

Re: [asterisk-users] AstManProxy Host Prefix?

I can do it for $10,000 On 10/24/07, asterisk <[EMAIL PROTECTED]> wrote: > > What would be nice if it you could specify the host per user in > astmanproy.users > Anyone interested in making the change? $$$ > > Doug > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

Jonn Taylor wrote: > Eric "ManxPower" Wieling wrote: >> Any echo you hear on pure IP calls is caused by the endpoint phone. You >> cannot do ANYTHING about it on Asterisk. >> >> >> Jonn Taylor wrote: >> >>> Any ideas ? >>> >>> Jonn >>> >>> Original Message >>> Subject:

Re: [asterisk-users] bugs.digium.com

On 15:38, Wed 24 Oct 07, Doug Lytle wrote: > > The bug tracker seems to be down. And so is the public svn and downloads.digium.com and ftp.digium.com and the websvn. They are working on it. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/l

[asterisk-users] reload manager.conf

I've made a change to my manager.conf file in asterisk 1.2.18 Is there a way to reload that config file from the CLI without restarting asterisk? Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list T

Re: [asterisk-users] AstManProxy Host Prefix?

What would be nice if it you could specify the host per user in astmanproy.users Anyone interested in making the change? $$$ Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Lyman Sent: Wednesday, October 24, 2007 1:45 PM To: Asterisk Users

[asterisk-users] bugs.digium.com

The bug tracker seems to be down. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation Provided by http://www.api-digit

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

Tzafrir Cohen wrote: > On Wed, Oct 24, 2007 at 01:20:31PM -0400, Dave Fullerton wrote: > >> This looks like it is isolated to 1.4.5.x. > > Right. > >> It looks like digium added > > Just to set the record straight, it was me who added it, and thus caused > hte changed behaviour you noticed h

Re: [asterisk-users] AMI ActionID.... Doesn't work

Douglas Garstang wrote: > Is it well known that setting the ActionID when connecting to AMI has > absolutely no effect? > Is this fixed in Asterisk 1.4? > > If you add an ActionID to your Originate command for example, it looks like > the only events that come back with an ActionID associated a

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

Buy a Polycom 301 off ebay and see if it echos on your LAN. Thanks, Steve Totaro Jonn Taylor wrote: > Any ideas ? > > Jonn > > Original Message > Subject: [asterisk-users] Internal LAN echo problem > Date: Wed, 24 Oct 2007 08:34:32 -0500 > From: Jonn R T

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

On Wed, Oct 24, 2007 at 01:20:31PM -0400, Dave Fullerton wrote: > This looks like it is isolated to 1.4.5.x. Right. > It looks like digium added Just to set the record straight, it was me who added it, and thus caused hte changed behaviour you noticed here. The behaviour was restored in late

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

On Wed, Oct 24, 2007 at 03:03:01PM +0100, Alan Lord wrote: > Hi all, > > After reading great things about the OSLEC Echo Canceller > (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of > people who have tried it on a recent Trixbox thread > (http://www.trixbox.org/forums/trixb

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

Eric "ManxPower" Wieling wrote: > Any echo you hear on pure IP calls is caused by the endpoint phone. You > cannot do ANYTHING about it on Asterisk. > > > Jonn Taylor wrote: > >> Any ideas ? >> >> Jonn >> >> Original Message >> Subject: [asterisk-users] Internal LAN e

Re: [asterisk-users] libdundi?

> I would have thought an LGPL version wouldn't be out of the question. > I hope not! LGPL is perfect for library-ish FOSS. Releasing libraries under standard GPL, while making Richard Stallman's heart go pitter-patter, limits what they can do since they can only go into other GPL projects.

Re: [asterisk-users] [Fwd: Internal LAN echo problem]

Any echo you hear on pure IP calls is caused by the endpoint phone. You cannot do ANYTHING about it on Asterisk. Jonn Taylor wrote: > Any ideas ? > > Jonn > > Original Message > Subject: [asterisk-users] Internal LAN echo problem > Date: Wed, 24 Oct 2007 08:

[asterisk-users] [Fwd: Internal LAN echo problem]

Any ideas ? Jonn Original Message Subject:[asterisk-users] Internal LAN echo problem Date: Wed, 24 Oct 2007 08:34:32 -0500 From: Jonn R Taylor <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mai

[asterisk-users] AMI ActionID.... Doesn't work

Is it well known that setting the ActionID when connecting to AMI has absolutely no effect? Is this fixed in Asterisk 1.4? If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSu

Re: [asterisk-users] AstManProxy Host Prefix?

Thanks, just realised that... - Original Message From: Richard Lyman <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, October 24, 2007 10:45:25 AM Subject: Re: [asterisk-users] AstManProxy Host Prefix? Douglas Garstang wrote: > Can the A

Re: [asterisk-users] Unusual DTMF behavior

We have Digium PRI cards, TE110 and TE420 (with hardware echo cancellation). On 10/24/07, John Meksavan <[EMAIL PROTECTED]> wrote: > > What is your setup, hardware wise? > > If you have the digium cards- FXO or FXS, you must make sure you tune > them. I had issues with DTMF's, when I went live

Re: [asterisk-users] AstManProxy Host Prefix?

Douglas Garstang wrote: > Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output > applies to, to the start of each line? If you are proxying multiple systems, > how can it uniquely identify the output from each system? > > Thanks, > Doug. > > each Event block should hav

Re: [asterisk-users] Remote provisioning for ATA's

Your best bet may be to write your own. That's what we ended up doing and it isn't that hard. On 10/24/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote: > > Hi all, > I need a fully developed web based remote provisioning system. I cant find > anything reliable on the internet. Have already checked at

Re: [asterisk-users] Voicemail playback on iPhone

On Oct 24, 2007, at 12:25 PM, [EMAIL PROTECTED] wrote: This is semi-related, but I have a Tmobile MDA and I couldn't play the files either. The issue was not a codec issue, it was an email encoding issue. If I sent the message to an email account and it was then downloaded to my desktop v

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

marcotasto wrote: > Hi Alan. > I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the > 1.4 version and I have had the same problem. > Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV > is defined by default and this prevents the echo selection from

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

marcotasto wrote: > Hi Alan. > I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the > 1.4 version and I have had the same problem. > Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV > is defined by default and this prevents the echo selection from

[asterisk-users] SOLVED Re: Voicemail playback on iPhone

Seems the answer was simple enough - set format=wav and it works fine. Mine was set at wav49. On 24-Oct-07, at 1:02 PM, Jason Lixfeld wrote: > It plays wav, but as far as I understand, * encodes the wav using > something like ulaw which iPhone doesn't support. If I can switch the > codec to p

Re: [asterisk-users] Backport Func_ODBC question

On Wednesday 24 October 2007 10:50:47 JR Richardson wrote: > Ingnorant question, how do you apply the backport func_odbc to 1.2 branch? ASTSRC=/path/to/downloaded/asterisk/source make install -- Tilghman ___ --Bandwidth and Colocation Provided by http

Re: [asterisk-users] Voicemail playback on iPhone

It plays wav, but as far as I understand, * encodes the wav using something like ulaw which iPhone doesn't support. If I can switch the codec to pcm, that may work - is that possible? On 24-Oct-07, at 12:25 PM, Darrick Hartman (lists) wrote: > Jason Lixfeld wrote: >> I guess what I'm asking

[asterisk-users] AstManProxy Host Prefix?

Can the Asterisk Manager Proxy, AstManProxy, prefix the host name that output applies to, to the start of each line? If you are proxying multiple systems, how can it uniquely identify the output from each system? Thanks, Doug. __ Do You Yahoo!?

Re: [asterisk-users] Voicemail playback on iPhone

Jason Lixfeld wrote: > I guess what I'm asking is if there is a recipe anyone has used to > allow a voicemail message (once recorded by asterisk) to playback on > iPhone when said recorded voicemail is received as an email > attachment. I understand you can convert using sox, so I guess that

Re: [asterisk-users] Voicemail playback on iPhone

This is semi-related, but I have a Tmobile MDA and I couldn't play the files either. The issue was not a codec issue, it was an email encoding issue. If I sent the message to an email account and it was then downloaded to my desktop via outlook and then forwarded on to my phone, I can listen

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

Hi Alan. I've installed OSLEC with zaptel-1.4.5.1 applying the patches made for the 1.4 version and I have had the same problem. Looking at the compiler options I've found that the symbol ECHO_CAN_FROMENV is defined by default and this prevents the echo selection from zconfig.h. I've solved chang

Re: [asterisk-users] Asterisk integration with IBM Sametime

On 10/24/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hi, > > I am trying to setup a conference between Sametime users using > conferencing infrastructure of asterisk. > > Sametime server has a component called TCSPI, which we can implement to > talk to any PBX, including asterisk (as per

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

Dave Fullerton wrote: > Replies/Comments inline... Ditto :-) > Alan Lord wrote: >> Hi all, >> >> After reading great things about the OSLEC Echo Canceller >> (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of >> people who have tried it on a recent Trixbox thread >> (http://

Re: [asterisk-users] Voicemail playback on iPhone

I guess what I'm asking is if there is a recipe anyone has used to allow a voicemail message (once recorded by asterisk) to playback on iPhone when said recorded voicemail is received as an email attachment. I understand you can convert using sox, so I guess that's the ingredient and some

[asterisk-users] Backport Func_ODBC question

Hi All, Ingnorant question, how do you apply the backport func_odbc to 1.2 branch? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCR

[asterisk-users] Two DTMF tones on keypress with Handsfree cell

Hello, I am using Asterisk SVN, a cellular phone, and chan_mobile to run a small home PBX with two analog telephones connected to a Linksys ATA using SIP. It works great (except for some Bluetooth adapter bugs that I am still trying to beat...seems the misaligned audio detection still needs work),

[asterisk-users] Asterisk Shutting Down

We've experienced the same problem twice now in the past month. The asterisk pid stops responding. We aren't able to connect to the CLI and all calls are dropped. The lots are pretty bare as well. This is the message log: Oct 24 09:12:35 WARNING[20711] channel.c: Avoided initial deadlock for '0x84

[asterisk-users] Question about outgoing callerid

Hi I have an ISDN connection with 100 DIDs assigned to it... What I'm trying to achieve is set the proper outgoing callerID while showing the local caller's extension in the CDR. There is a behaviour that I just can't explain. the callerid field in sip.conf is set as : callerid="Jean-Yves/E" <3

Re: [asterisk-users] Unusual DTMF behavior

What is your setup, hardware wise? If you have the digium cards- FXO or FXS, you must make sure you tune them. I had issues with DTMF's, when I went live with my Asterisk system. Once I tune them, everything worked great. Date: Wed, 24 Oct 2007 09:05:35 -0500 From: [EMAIL PROTECTED] To:

[asterisk-users] Remote provisioning for ATA's

Hi all, I need a fully developed web based remote provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -

Re: [asterisk-users] Grandstream GXP-2000's and Asterisk.

Thomas Kenyon wrote: Thomas Kenyon wrote: I have 5 GXP 2000's with firmware 1.1.4.25 running with Asterisk 1.4.13. Is anyone else getting the following error in the asterisk console: [Oct 22 11:39:01] WARNING[7100]: rtp.c:1142 ast_rtp_read: RTP Read too short every couple of seconds when a

Re: [asterisk-users] How to tune Asterisk AMD - Answering Machine Detection "hacks"

On 10/24/07, Costa Dinoteli <[EMAIL PROTECTED]> wrote: > > Most everytime Asterisk calls it thinks it is an Answering Machine and it > starts playing > the AMD message, instead of the delivering the "1st real message" > Why is it thinking that it's a machine? If you're on the console at verbose

Re: [asterisk-users] Voicemail playback on iPhone

It is doable. The iPhone uses a subset of the Apple OS. Sometime ago I reviewed the file structure of the iPhone. It is just a matter of placing the voicemail files from * into the voicemail folder of the iPhone. Somebody with more time than me though :) CS -Original Message- From:

Re: [asterisk-users] Voicemail playback on iPhone

Jason, I think there is a bit of terminology confusion here, you can easily convert from format to another using sox, so if your * server is going to record and email you a voicemail file, it can surely "sox" the file to whatever format the iphone needs it in and then send the email. It

[asterisk-users] HD Voice

Video on HD Voice. Worth a watch but nothing you wouldn't already know about. http://www.eweek.com/article2/0,1895,2193922,00.asp My question however is this - when are ITSP's going to start offering digital voice services with HD codecs? It's crazy that calls to my clients via skype are bett

Re: [asterisk-users] Voicemail playback on iPhone

Sorry, it's clear my question was far too vague. To clarify, is there a recipe to make * record voicemail in a format that allows playback on iPhone's media/music player playback for voicemails that are received say, in an email message. It seems the * voicemail defaults don't work. This lin

Re: [asterisk-users] whisper chanspy in asterisk 1.2

Carles Pina i Estany wrote: > Hello, > > I would like to have "whisper" channel spy (not private whisper) in > Asterisk 1.2. I see here: > http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html > That is only available for Asterisk 1.4. > > I wonder if there is any way to emulate it i

Re: [asterisk-users] OSLEC and zaptel-1.4.5.1

Replies/Comments inline... Alan Lord wrote: > Hi all, > > After reading great things about the OSLEC Echo Canceller > (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of > people who have tried it on a recent Trixbox thread > (http://www.trixbox.org/forums/trixbox-forums/open

[asterisk-users] How to tune Asterisk AMD - Answering Machine Detection "hacks"

Hello Everyone, Can someone point me to reliable links on how to tweak Asterisk AMD I am calling a number and have to two files to play depending if it is a real person or an answering machine. Most everytime Asterisk calls it thinks it is an Answering Machine and it starts playing the AMD messag

[asterisk-users] whisper chanspy in asterisk 1.2

Hello, I would like to have "whisper" channel spy (not private whisper) in Asterisk 1.2. I see here: http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html That is only available for Asterisk 1.4. I wonder if there is any way to emulate it in Asterisk 1.2. For example, to "join" tw

[asterisk-users] Unusual DTMF behavior

We are having an issue where DTMF is not being sent out right away and the tone duration is inconsistent. For a test we send a '5', then a second later we send a '9', and then five seconds later we send a '5'. If you look at the logs below you can see the first '5' is played right away, then the

[asterisk-users] OSLEC and zaptel-1.4.5.1

Hi all, After reading great things about the OSLEC Echo Canceller (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of people who have tried it on a recent Trixbox thread (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems), it sounds like

Re: [asterisk-users] Polycom Phone and bitmaps

Shaun wrote: >I've been trying to get the polycom 550 phones to show a idle display bitmap >but have not been successful. Anybody have any experience with this? The >manual gives instructions >(http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soun

Re: [asterisk-users] asterisk and Skype - your experiences please

randulo said on 24.10.2007 10:17: > From time to time, various ways of connecting asterisk SIP channels to > skype has been discussed here. This Friday, one of the subjects of our > weekly Voip Users Conference will be about trying to connect our > asterisk pbx with Skype. Has anyone tried http://

[asterisk-users] Internal LAN echo problem

Hi all, I have an internal echo problem on my LAN only. I replaced the LAN switch with a new linksys 2024 with QOS and seemed to help but not fix the problem. Any ideas? Here in my setup - Dell PE6400 Dual 700, Asterisk 1.2.24/FreePBX, 2-NIC cards, one with a public ip and one with an internal

Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card

On 10/24/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > > Joseph Begumisa wrote: > > > > Has anyone had any compatibility issues with a TE110P card installed > > on a Dell Poweredge 1950? I noted the following error on the LCD > > display of the Dell Poweredge 1950: > > > > > > > > E1711 PCI PErr S

Re: [asterisk-users] Asterisk integration with IBM Sametime

Hi, I am trying to setup a conference between Sametime users using conferencing infrastructure of asterisk. Sametime server has a component called TCSPI, which we can implement to talk to any PBX, including asterisk (as per documentation). I was trying to implement the TCSPI for Asterisk. Re

Re: [asterisk-users] Asterisk integration with IBM Sametime

On 10/24/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hi, > > I wanted to know if anyone has experience in integration asterisk with IBM > Sametime server (by implementing TCSPI). > Any pointers for this would be very helpful. > > Have been reading/googling around a bit and I get to unders

Re: [asterisk-users] Snom 360 lights not working on subscription / fail to extend xx to xx error

Dear guys, many people have been using Snom with Subscription/notify lights I tried almost every tip in the google. But there's one thing related to the snom phones and asterisk I didn't see in any forum The Asterisk console show very often a message like: fail to extend from xx to xxx This

Re: [asterisk-users] Help with loop counting?

Phil Knighton wrote: > exten => i,1,Set(trips=$[${trips} + 1]) > exten => i,2,Goto(s,7) i=invalid, t=timeout exten => t,1,Set(trips=$[${trips} + 1]) You'll also want to initialize ${trips} with a Set(trips=0) at the beginning of your routine. Doug -- Ben Franklin quote: "Those who would

Re: [asterisk-users] register => to let Asterisk register to another softswitch via SIP

Dear Alex; Thanks for your great help and nice replies. I would like to confirm that I understood your request very well, so please advise me for the following: 1) If no need for registering asterisk with the softswitch, then no need to use register => but we will configure the section with type

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