Re: [asterisk-users] MOH Oddity

On Friday 05 March 2010 17:19:06 Matt wrote: > For some reason I have to set the type to 'files' if I set it to 'quietmp3' > I get nothing, even though the files are valid MP3 files that play on > another asterisk system... does that mean I've got something installed > wrong? Mostly likely you did

Re: [asterisk-users] MOH Oddity

Un-top-posting... >> On Fri, Mar 5, 2010 at 11:29 PM, Matt wrote: >>> >>> I'm trying to setup my asterisk system for the least overhead as >> possible. On Fri, 5 Mar 2010, Matt wrote: > For some reason I have to set the type to 'files' if I set it to 'quietmp3' > I get nothing, even though the

Re: [asterisk-users] MOH Oddity

For some reason I have to set the type to 'files' if I set it to 'quietmp3' I get nothing, even though the files are valid MP3 files that play on another asterisk system... does that mean I've got something installed wrong? 2010/3/5 Håkon Nessjøen > On Fri, Mar 5, 2010 at 11:29 PM, Matt wrote:

Re: [asterisk-users] MOH Oddity

On Fri, Mar 5, 2010 at 11:29 PM, Matt wrote: > > I'm trying to setup my asterisk system for the least overhead as possible. > > My understanding (and experience with other systems) leads me to believe I > can run any MOH using a certain class through a single 'player' as opposed to > starting an

Re: [asterisk-users] Having problems with BLF

Clarification: > Not sure, haven't seen that before. Anyone? This comment refers to: >> Looks like the PBX isn't sending the SIP messages- I notice the >> previous NOTIFY messages said (queued)- does >> this mean anything? Philipp -- _

Re: [asterisk-users] MOH Oddity

Forgot to include: I'm running 1.6.2.5 On Fri, Mar 5, 2010 at 5:29 PM, Matt wrote: > I'm trying to setup my asterisk system for the least overhead as possible. > > My understanding (and experience with other systems) leads me to believe I > can run any MOH using a certain class through a single

[asterisk-users] MOH Oddity

I'm trying to setup my asterisk system for the least overhead as possible. My understanding (and experience with other systems) leads me to believe I can run any MOH using a certain class through a single 'player' as opposed to starting an independent stream for each MOH instance. However, try as

Re: [asterisk-users] Having problems with BLF

Hi! > > PBX*CLI> sip show subscriptions > > Peer             User        Call ID      Extension        Last state   > > Type            Mailbox 192.168.13.114   222         3c26707958d > >  ...@default      Idle   dialog-info+xml 1 active SIP > > subscription > > The phone is behind natted route

Re: [asterisk-users] Asterisk 1.4 Followme Question

Isnt that the point of the FMFM – to allow the call to come back into the asterisk server and have your voicemail managed in one location? If not wanted, I guess remove the voicemail step from the FMFM config and just have it end on the forwarded cellphone. -- *From

Re: [asterisk-users] Playback in h extension

2010/3/5 Danny Nicholas : > Not possible.  H exten is called by a hangup. Well - sometimes not both parties hang up at the same time. ;-) If you want to play something to the originating party after die Dial()ed party hangs up use the option "g" in the Dial command to get more commands executed af

Re: [asterisk-users] Hardware requirements question.

Yes, this machine will be enough for that task. Performance wise. The other good thing is that it is not very likely that someone will steal your PBX. As far as I remember it is a 7 rack unit box which weights approx. one metric ton. ;-) But remember - if anything dies in the box and you have to g

[asterisk-users] app_confbridge production ready?

I have an existing conference bridge running on Asterisk 1.4.2 using MeetMe and it's been pretty much rock solid since it was installed. We do around 460,000 minutes on it monthly and peak at about 150 simultaneous sip channels. I'm adding a second bridge for redundancy purposes into another fac

Re: [asterisk-users] NeoSpeech & Asterisk?

Byron, How about Cepstral TTS software for text to speech playout. I use it for short messages ( less than 5 mins). -Irfan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Byron J. Lee Sent: Friday, March 05, 20

Re: [asterisk-users] Having problems with BLF

> PBX*CLI> sip show subscriptions > Peer             User        Call ID      Extension        Last state >   Type            Mailbox > 192.168.13.114   222         3c26707958d  ...@default      Idle >   dialog-info+xml > 1 active SIP subscription > The phone is behind natted router on a private

Re: [asterisk-users] Uverse, Asterisk and SIP

On Mar 5, 2010, at 1:01 PM, sean darcy wrote: > The issues are that sip doesn't work, What does "doesn't work" mean? In / Out? Both? Do you have a sip trace? > even though this same set up > worked with POTS dsl. IAX does (but gives lousy audio quality) so I > don't believe all udp ports are b

Re: [asterisk-users] Observation about DAHDI, FAX and Echo cancellation

- "Håkon Nessjøen" escreveu: > Hi, > > I have read that DAHDI automagically turns of echo cansellation when > it sees that it is a FAX. > > So I checked this out. I have a fax call into asterisk which is > immediately called out to an external fax machine via DAHDI again.. > > For example,

Re: [asterisk-users] Uverse, Asterisk and SIP

On Wed, Mar 3, 2010 at 1:23 PM, Fred Posner wrote: > > On Mar 3, 2010, at 1:03 PM, sean darcy wrote: > >> Well at least my RG doesn't let you use DMZplus _unless_ you've chosen >> dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh >> into my router from the internet. >> >> Anybo

[asterisk-users] Observation about DAHDI, FAX and Echo cancellation

Hi, I have read that DAHDI automagically turns of echo cansellation when it sees that it is a FAX. So I checked this out. I have a fax call into asterisk which is immediately called out to an external fax machine via DAHDI again.. For example, the result is: DAHDI/1-1 = incoming call, DAHDI/2-1

Re: [asterisk-users] State of 64 bits applications in Asterisk

I've used FFA briefly but successfully on Asterisk 1.6.2 x64. Jordan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: 05 March 2010 17:00 To: Asterisk-Users Subject: [asterisk-users]

Re: [asterisk-users] AMI logs

AMI does not create any log files itself. If actions cause actions that would otherwise get logged then those of course are logged. You need to receive the response from your action to see if there were problems. The AMI protocol calls for sending action packets and then receiving a response or

Re: [asterisk-users] Hardware requirements question.

On Fri, 5 Mar 2010, Steve Edwards wrote: > On Fri, 5 Mar 2010, Gordon Henderson wrote: > >> On Fri, 5 Mar 2010, David Little wrote: >> >>> I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, >>> SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop >>> an aste

[asterisk-users] Regarding - P-Asserted identity

Hi All, We have two servers, one server (SIP asterisk server) sending calls to the second server(has PRI) which goes our through the PRI's (using TE 412p). When the pprivacy is enabled: P-Asserted-Identity Header, privacy "id" are sent in the header of SIP invite packet to the second server, how c

Re: [asterisk-users] FollowMe / Asterisk 1.4 Question

On Fri, Mar 5, 2010 at 9:33 AM, Cory Andrews wrote: > Is there a way to strip the normal features out of FollowMe (call > acceptance, etc), and just set followme up to to blind transfer any call to > an extension's associated cell number if it is not answered on the extension > after 4 rings? U

[asterisk-users] State of 64 bits applications in Asterisk

Hi, what is the state at this time for 64bits applications and compatibility with 1.6.2 Mainly speaking about FFA, SFA, G729. Thanks for any information -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-

Re: [asterisk-users] 30 mins GSM file

On Fri, 5 Mar 2010, David @ULC wrote: > Sorry if you guys find this silly, > > for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm >>> * /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ; done* > > I need to enter above lines in my root prompt ? Yes. Your system will run muc

Re: [asterisk-users] Hardware requirements question.

On Fri, 5 Mar 2010, Gordon Henderson wrote: > On Fri, 5 Mar 2010, David Little wrote: > >> I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, >> SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop >> an asterisk pbx with 4 POTS lines in and 16 analog extens

Re: [asterisk-users] Hardware requirements question.

On Fri, 5 Mar 2010, David Little wrote: > I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, > SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop > an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). > I also will install a sound card

Re: [asterisk-users] Denial of Service Attack

On Friday 05 March 2010 10:17:27 Dan Journo wrote: > I currently have a dedicated server with a hosting provider for my voip and > the provider is currently experiencing a DOS attack. I have been looking at > purchasing a number of servers and creating my own VOIP setup with > redundancy built in.

Re: [asterisk-users] Hardware requirements question.

- "David Little" wrote: > I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz > processors, > SCSI controller with four 9MB drives and 1 GB of RAM. I want to > develop > an asterisk pbx with 4 POTS lines in and 16 analog extensions (no > VOIP). > I also will install a sound card for

Re: [asterisk-users] 30 mins GSM file

Sorry if you guys find this silly, for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm >>* /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ; done* I need to enter above lines in my root prompt ? for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm * /va

Re: [asterisk-users] InterPBX communication using SIP

OK Guys i got fixed the phones i was using were registered in both servers which is not good, once i removed them it started working! 2010/3/4 khalid touati > Hi Guys, > i am using the following config in pbx1: > register => pbx1:endop...@172.16.200.175 > [pbx2] > type=friend > host=dynamic > t

[asterisk-users] NeoSpeech & Asterisk?

I am working on a project where a caller would call my PBX and get a menu of categories, sub-categories, and even more sub-categories. The reason for this is to determine what text-file a user wants to read out a large pile of them. I need some help deciding how to impliment this system. Firstl

[asterisk-users] Denial of Service Attack

Hi, I currently have a dedicated server with a hosting provider for my voip and the provider is currently experiencing a DOS attack. I have been looking at purchasing a number of servers and creating my own VOIP setup with redundancy built in. However, how I can design the system to ensure serv

Re: [asterisk-users] iLBC installation problem

Hi! > I would like to install iLBC codec. I have found a "HOW TO ..." here > http://www.voip-info.org/wiki/view/iLBC. Unlucky when I compile with > make I get the following errors. After you ran the script to obtain the iLBC code, you need to go into asterisk/contrib/scripts/codecs/ilbc and cop

[asterisk-users] AMI logs

Hi, I'm executing some commands using AMI... I suppose the log is saved in some place, but I don't know where... where is it saved?More details: I'm executing a UpdateConfig in the voicemail.conf file, but the file is not updated, so I would like to know why...Thanks, Anahi Anahi Ludueña

[asterisk-users] Hardware requirements question.

I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient

Re: [asterisk-users] 30 mins GSM file

- "Jeff LaCoursiere" escreveu: > On Thu, 4 Mar 2010, Steve Howes wrote: > > > > > On 4 Mar 2010, at 23:11, Steve Edwards wrote: > >> On Thu, 4 Mar 2010, Steve Edwards wrote: > >>> On Fri, 5 Mar 2010, David @ULC wrote: > >>> > I need to create 30 mins of GSM file for Asterisk . > >

Re: [asterisk-users] Deadlock in Asterisk 1.4.29.1

Hi Moises, Thanks for the URL. I hope to have a feedback before open an issue. If there isn’t I will do that. Regards, Adrien .L De : Moises Silva [mailto:moises.si...@gmail.com] Envoyé : vendredi 5 mars 2010 16:02 À : alemo...@legos.fr; Asterisk Users Mailing List - Non-Co

[asterisk-users] FollowMe / Asterisk 1.4 Question

Is there a way to strip the normal features out of FollowMe (call acceptance, etc), and just set followme up to to blind transfer any call to an extension's associated cell number if it is not answered on the extension after 4 rings? Users want followme calls to wind up in their cellphone voic

Re: [asterisk-users] 30 mins GSM file

On Thu, 4 Mar 2010, Steve Howes wrote: > > On 4 Mar 2010, at 23:11, Steve Edwards wrote: >> On Thu, 4 Mar 2010, Steve Edwards wrote: >>> On Fri, 5 Mar 2010, David @ULC wrote: >>> I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way

Re: [asterisk-users] SIP / Echo Cancellation

- "Steve Underwood" escreveu: > On 03/05/2010 02:45 PM, Vineet Bhojnagarwala wrote: > > Very informative post Vinícius ! > > > > 2010/3/5 Vinícius Fontes > > > > > > - "Chandrakant Solanki" > > escreveu: > > >

[asterisk-users] Asterisk 1.4 Followme Question

I have a question related to FollowMe on Asterisk 1.4. Is there a way to force Asterisk to always leave VM on the forwarded extension's cell phone, as opposed to pulling the call back from the forward to cell and depositing in Asterisk voicemail? Thanks in Advance! -- *Cory J Andrews* 725 Pow

Re: [asterisk-users] Having problems with BLF

Yes- followed all 3 wiki instructions. Thanks for naming tips! Does this log help at all? Looks like the PBX isn't sending the SIP messages- I notice the previous NOTIFY messages said (queued)- does this mean anything? John PBX*CLI> sip show subscriptions Peer UserCall ID

Re: [asterisk-users] Deadlock in Asterisk 1.4.29.1

If you want to open a bug report the proper place to do it is at http://issues.asterisk.org/ Compile with DEBUG_THREADS and DETECT_DEADLOCKS (see make menuselect compiler flags). -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3

[asterisk-users] Follow-up to CALLERID(num) not working

I sent a question yesterday about having problems setting the caller ID. I turned on pri debug for both a good and bad call and I see this in the good call [2010-03-05 05:58:20.743] > [6c 0c 21 80 30 30 30 30 30 30 30 30 30 30] [2010-03-05 05:58:20.744] > Calling Number (len=14) [ Ext: 0 TON: N

Re: [asterisk-users] Having problems with BLF

Hi! > I'm having a problem getting a snom 300 to work with BLF (extension > 222). I've set it to watch extension 220 in the function key config > pages as per the wiki (BLF, ) but I can't get the > light to come on when 220 is ringing. The SIP trace page doesn't show > anything coming from my PBX

Re: [asterisk-users] Playback in h extension

Not possible. H exten is called by a hangup. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, March 05, 2010 8:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Playback in h ext

Re: [asterisk-users] SIP / Echo Cancellation

On 03/05/2010 02:45 PM, Vineet Bhojnagarwala wrote: > Very informative post Vinícius ! > > 2010/3/5 Vinícius Fontes > > > - "Chandrakant Solanki" > escreveu: > > > Hello > > > > I have successfully compil

[asterisk-users] Playback in h extension

Hi people, I'm trying to execute the PlayBack command in the h extension... but it is not played... is it possible to do that?Thanks, Anahi Anahi Ludueña _ Ahora Messenger en tu Blac

Re: [asterisk-users] 30 mins GSM file

Score another top-notch tip for Tilghman!!! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Thursday, March 04, 2010 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subj

Re: [asterisk-users] Asterisk Management API

At an Asterisk CLI use the command "manager show commands". -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 5, 2010, at 1:50 AM, Peter Childs wrote: > Is there a list of input's / out puts from the management API together > with there parameters, there meanings and

Re: [asterisk-users] FAX configuration for DAHDI lines

Then it's simple, set up an exten in the extensions.conf like exten => 123456789,1,Dial(DAHDI/2,,rtT) exten => 123456789,n,Hangup() replace the 123456789 with the public phone number and the DAHDI/2 with the channel you are using. Regards, Peter 2010.03.05. 14:07 keltezéssel, Gopalakrishna

Re: [asterisk-users] Caller ID in Asterisk

Hi,   Well, if you replicate the line that set the callerid for every extension than you can set each one manually.   Jimmy -Original Message-From: venui...@motorola.comSent: Fri, 5 Mar 2010 14:54:56 +0800To: venui...@motorola.com, asterisk-users@lists.digium.comSubject: Re: [asterisk-

Re: [asterisk-users] Caller ID in Asterisk

As far as I know, you should set up the callerid in the chan_dahdi.conf with the usecallerid=yes and the callerid=8001234001 options where you are setting the each channels. Regards, Peter Gelencser 2010.03.05. 7:54 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: > Hi All, > > Finally

Re: [asterisk-users] FAX configuration for DAHDI lines

Hi Petr, I would like this fax to be reached from a public number. I will replace the existing analog phone and replace the same with a fax. Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 **

Re: [asterisk-users] FAX configuration for DAHDI lines

Hi, How would you like it to work? It would be an inner extension or this fax should be reached from a public phone number? Best regards, Peter Gelencser 2010.03.05. 13:06 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta: > Hi Experts, > I have an asterisk machine with DAHDI and i want to

[asterisk-users] MGCP FXO endpoint

I have a fxo endpoint installed in a Cisco router. I would like in my dialplan to get an extension call a telephone number through that fxo endpoint. Since with zaptel channels it is done like: exten => 0999,1,Dial(DAHDI/2-1/111) --> being 111 the phone number I want to call. I thought that for

[asterisk-users] FAX configuration for DAHDI lines

Hi Experts, I have an asterisk machine with DAHDI and i want to connect analog fax machines to asterisk.I already have TDM800 card where i am using analog telephone lines to make calls.Kindly let me know how to configure fax for dahdi lines.Where all do i need to modify my configurations.

[asterisk-users] Deadlock in Asterisk 1.4.29.1

Hello, I have previously open a topic on the mailing list about deadlocking on Asterisk 1.2.35. After upgrading to 1.4.29.1 we still experienced the same problem : Mar 5 12:05:56] DEBUG[8647] channel.c: Avoiding initial deadlock for channel '0xb7689840' [Mar 5 12:06:41] DEBUG[7130] cha

[asterisk-users] Having problems with BLF

Hi, I'm having a problem getting a snom 300 to work with BLF (extension 222). I've set it to watch extension 220 in the function key config pages as per the wiki (BLF, ) but I can't get the light to come on when 220 is ringing. The SIP trace page doesn't show anything coming from my PBX when 220 i

[asterisk-users] Asterisk Management API

Is there a list of input's / out puts from the management API together with there parameters, there meanings and which are required and what they do/mean. Its just all the docs I've found seam to be rather sketchy and gathered by trial and error, not really up to what I would call a protocol stand

[asterisk-users] iLBC installation problem

Hi, I would like to install iLBC codec. I have found a "HOW TO ..." here http://www.voip-info.org/wiki/view/iLBC. Unlucky when I compile with make I get the following errors. [code] Generating embedded module rules ... [CC] codec_ilbc.c -> codec_ilbc.o codec_ilbc.c:40:30: error: ilbc/iLBC_enco