[asterisk-users] Unable to set TOS to 184?

I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 From what I have read the reason is asterisk can't set TOS if not running in root. Mine is running as asterisk. I found one post that says to run at boot: #!/bin/bash

[asterisk-users] sched_settime: Request to schedule in the past?!?!

Does anybody know what this message means? [2009-10-17 08:11:59] DEBUG[9598]: sched.c:204 sched_settime: Request to schedule in the past?!?! I've tried all I can find on google with no change It seems to happen when call is at a READ() or Background() - What the caller hear is small delays

[asterisk-users] Method to use SOX inside a Dialplan

I'm trying create a feature that allows a callers to add more speech to his recording. I think this can be done inside a dialplan, but I can't find an example of how to do this. Basically,after he records the primary message, a menu would play asking if he wants to append to this message. If

[asterisk-users] DTMF problem during read()

I have a simple dialplan. Using the read cmd, I ask caller for his passcode. If the caller is calling from an plain old analog phone, his dtmf is not heard until the prompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode.

[asterisk-users] Method to downgrade asterisk

I currently have asterisk-1.4.26.2 installed and working. It was sugguested I try asterisk-1.4.25 to see if it fixes my SIP dtmf problems. What is the method to downgrade? Do I just do in the asterisk-1.4.25 folder: make clean ./configure make install Or do I need to 'make clean' in the

[asterisk-users] Appending two voice files

Does anyone know how I can append to different user recorded voice files within a dial plan? For example Asterisk ask caller a question and records the answer, then ask another question record the answer to the end of the first answer - so when it's played back, all the answers are in one

Re: [asterisk-users] Freepbx changes dont reflect in asterisk

With FreePBX you can not modify certain conf files - many are overwritten at reload Bart Pedro Silva wrote: Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed

[asterisk-users] AEL2 Confusion

I just downloaded and installed asterisk-1.2.13 Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected. Where and how do I get current release of AEL2 - Is there some 'How To' somewhere? TIA Bart

[asterisk-users] Match Chat Author?

I stumble on this URL that is Chat Line script written by Steven L. Edwards called 'Match Chat' here: http://bugs.digium.com/file_download.php?file_id=11080type=bug But I can't seem to find any additional info on Author or Applications - I was wondering if you might know more about either?

[asterisk-users] 2 CPU's, Only 1 taking IRQ's

I'm trying to solve a echo problem... The system is Centos 2.6.9-34.0.2.ELsmp (SMP) CentOS release 4.3 (Final). And the Box is Dual Intel Xeon CPU 2.80GHz with 2 GB memory. It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests

Re: [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's

Hmm, this must not be installed: # locate irqbalance # /usr/src/kernels/2.6.9-34.0.2.EL-smp-i686/include/config/irqbalance.h How do I install this? Bart Álvaro Palma wrote: It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for

[asterisk-users] I'm I wrong - No 3-way calling for Single line sets?

It appears the only way to cause a 3-way call (or a screened transfer) is by using conference - nasty This mean SLT would need to transfer to conference than add second party, then add themselves. I've searched and I can't find anything that works in asterisk like the Telco method or am I

Re: [asterisk-users] Caller ID has extra digits to strip

. Bart Bart Fisher wrote: === About 70% of the time, my Local DID provider sends me ANI II digits (see http://www.nanpa.com/number_resource_info/ani_ii_assignments.html) where there will be an extra 2 digits added to the Caller ID - For example 62714222 where '62' = Cell Phone for example

Re: [asterisk-users] Caller ID has extra digits to strip

*DNIS* which is pretty funky. Thanks, Steve Bart Fisher wrote: I was hoping someone might have the answer to this: As an update: even though you can modify the Caller ID in extensions.conf for call handling and use CallerID(number) in you script, asterisk does not honor the modified number

[asterisk-users] Caller ID has extra digits to strip

About 70% of the time, my Local DID provider sends me ANI II digits (see http://www.nanpa.com/number_resource_info/ani_ii_assignments.html) where there will be an extra 2 digits added to the Caller ID - For example 62714222 where '62' = Cell Phone for example.. The problem is, I have not

Re: [asterisk-users] em wink, TE110P, * answers too soon

Well you say too much and not enough about the problem or configuration So, I assume the DID's are on Ports 1 - 24 T1?. If asterisk is missing the first digit, then I'll bet the DID T1 from Telco is set to immediate on their side, not wink - Because dialing should NOT start until after the

[asterisk-users] Strange CLI Output

While the console is in monitor mode (asterisk -r) I see duplicate messages from Asterisk one after the other - But not when I connect via SSH. Example: -- Starting simple switch on Zap/65-1 -- Starting simple switch on Zap/65-1 What could cause this? Thanks Bart

Re: [asterisk-users] Hotels...

LOL - PMS = Property Management System Bart C F wrote: Interesting you said PMS? here is the definition: http://en.wikipedia.org/wiki/PMS On 8/7/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Ideally you'd get billing to work by integrating directly with the property management software.

[asterisk-users] Run a script at certain CLI writes

I'm trying to detect when a T1 goes to Yellow or Red alarm. I noticed these events will be displayed on the CLI. What I'd like to do is cause an email to be sent when from a script on these events, but somehow I would need to capture the CLI outputs to detect messages Message are: wct4xxp:

Re: [asterisk-users] Run a script at certain CLI writes

Oh, good idea - the messages do appear there - I'll check it out Thanks Joey McDonald wrote: Have you looked to see if they're being logged to /var/log/asterisk/full ? That would be much easier to detect. --joey On 8/3/06, *Bart Fisher* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote

[asterisk-users] [ANN] - Coder Needed for Patch

I've posted on GAF (Free Lance Site) a request for bids for modifications to Asterisk PBX source. If you are interest in bidding on this, please view it at http://www.getafreelancer.com/projects/78138.html Thanks you for your time. Bart Fisher [EMAIL PROTECTED

Re: [asterisk-users] IAX over two T1 connections bad quality

Replace IAX with SIP - It solved my problems with several providers including FWD and Teliax Bart Jerry Geis wrote: Help please. I have two systems on the net. one in indiana and one in georgia. connected with IAX. local SIP phones in each office (10 each) are cisco and running sip. TDM04B

[asterisk-users] How do you recompile individual source modules?

How do you recompile individual source modules? I need to make a small change (addition) to chan_zap.c. I read somewhere you can recompile individual module source without the need to recompile the entire asterisk sources each time at change is made. Can someone tell this 'C' noob how to do

Re: [asterisk-users] How do you recompile individual source modules?

If I understand, I cd to asterisk source folder and run make - it take card of rest? Also, when/why should you use astxs? Bart Russell Bryant wrote: - Bart Fisher [EMAIL PROTECTED] wrote: I need to make a small change (addition) to chan_zap.c. I read somewhere you can recompile

[asterisk-users] Urgent source code changes needed

I need someone to patch what I believe to be a simple change to chan_zap.c - I know if I attempt I'll screw it up :) Whom would you approach for doing this? - My requests have received a 'blank stare' from Free Lance sites and I'm running out of time on this install. If you know someone or

[asterisk-users] I need help patching source

I'm trying to provide dial tone on EM Wink type trunks. I found where in source, 'chan_zap.c' where I believe the code needs to be added. Basically I believe I can copy parts used for PRI in to EM and EM Wink signal types. However with my attempts, it fails to compile at chan_zap. And I'm not

[Asterisk-Users] Dial Tone + EM

Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've

[Asterisk-Users] T1 + EM

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these

[Asterisk-Users] EM + Dial tone

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these

Re: [Asterisk-Users] How many TE405 ...

The best I've done is 2 - the thirdcard will not start properly everytime - so I gave up - Forget about trying 4 cards, never happen Bart - Original Message - From: Ard To: asterisk-users@lists.digium.com Sent: Monday, June 05, 2006 2:29 PM Subject:

[Asterisk-Users] ADIT 600 = Asterisk Help

I've been reading the Google searches trying to understand how to tie together Adit 600 to Asterisk to provide 2 way service. I'm about blind from reading. I assume, the answer is using MGCP between the boxes. However, the examples I found don't really explain fully enough to know how to

[Asterisk-Users] What about T400 T1 cards?

Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Bart ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] What about T400 T1 cards?

about T400 T1 cards? On Tuesday 23 May 2006 10:48, Bart Fisher wrote: Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Google for the performance data

[Asterisk-Users] Slash Tone at pstn cut-though?

I'm looking for a method to signal an insideextension (asterisk extension with external dialing appl.) with a DTMF "A" tone to indicate when Asterisk has completed dialing and the voice path has beencut-though on a ZAP T1 Trunk. If this can be done, I'd also like to know if there is a

[Asterisk-Users] EM and Dial tone

I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was

[Asterisk-Users] TE410P = Dialogic D/240SC-T1

I'm trying to connect an Asterisk T1 port to a Dialogic card. The Dialogic side is an external VMS. I setup for ISDN-PRI between systems and have green lights on both card/ports. Zttool shows connection is good also. However, when I tryattempt terminate or originate a call to either

Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover

In these cases, the Transmit and Receive pairs are in different binders. Thus electrically isolating by virtue of how the binders are wrapped with each other and how the pairs are twisted within the binders. Bart - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To:

Re: [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site

Yeah, I've had a project listed at asteriskhelpdesk.com for over a month and it still has 0 bids. I wouldn't waste my time redesigning the pages, they won't come... Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent:

[Asterisk-Users] Voicemail() - Reading exit or return results

Here my script: exten = 230,1,Answer exten = 230,2,NoOpexten = 230,3,Voicemail(u${EXTEN})exten = 230,4,NoOp(Need results from VoiceMail() above - should be non-zero and provided to ARG3) exten = 230,5,NoOpexten = 230,6,GoToIf($[${ARG3} = 0]?s|8) exten =

Re: [Asterisk-Users] On site installtion Tech. wanted

Maybe I could help. Located in Buena Park Bart - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 2:06 PM Subject: [Asterisk-Users] On site installtion Tech. wanted Looking for a Tech. that

[Asterisk-Users] OT: ADIT 600 Manual needed

If you would be willing to make available for download the Adit 600 Install / Configuration manual for this unit I would gladly PayPal you for your time and troubles... TIA Bart [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] Problem with two cards Digium

I'll guess the TE410P is being loaded first - Try swapping entries in zaptel.conf and zapata.conf Bart - Original Message - From: Bartosz Supczinski [EMAIL PROTECTED] To: Asterisk-Users asterisk-users@lists.digium.com; Asterisk-Dev asterisk-dev@lists.digium.com Sent: Tuesday,

[Asterisk-Users] Answering Service Add-on?

Anybody seen some client/server asterisk add-on script for "live" answering services to provide call handling and message taking from an Operator? Bart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To

[Asterisk-Users] Kudzu and Zaptel Cards

Redhat has a 'Hardware Discovery Utility' called Kudzu. When I change cards, kudzu pops up and ask to remove/config the card. Most of the time kudzu has trouble recognizing the Digium Zaptel cards and calls them something wrong, like calling the TDM card a network card. I'm having a

[Asterisk-Users] T1 to T1 dialout problem

I need a few minutes of time to work out a dial out problem. I'm willing to pay for your time. What I have is a system that connect 2 external VMS systems to one of two Telco T1's. Mainly the Telco T1's route inbound calls to one of the two external VM systems depending on the DNIS. This

[Asterisk-Users] Motherboard Selection Assistance

I need some help selecting a motherboard. I'm using 3 TE410P's and 1 TDM card in this system. I followed Digium suggestion and purchased a ASUS NRL-LS533 board. For the life of me, I cannot get all 4 cards to work in this environment. Unless the Digium cards are no good, then I assume it's

[Asterisk-Users] Help with this

I'm trying to get this to work, but it always goes to step 4 - there something I don't understand about LEN with GotoIf: exten = _,1,NoOp,${CALLERIDNUM} ; CID as received exten = _,2,GotoIf([LEN(${CALLERIDNUM}) = 10]?4:3) ; if CID length = 10 then do

[Asterisk-Users] Problem with CallerIDNum

I've been jacking with this for a while but don't understand all thatI'm reading... The problem is sometimes I get ANI II digits from the phone company. These will be two digits that prefix ANI- so some callerid might arrive as only "00" or "007147391234", "00714", "714" or

Re: [Asterisk-Users] Zaptel T1 Timing Source

My understanding there should only be one timing source per TE410. You should use a REAL Telco T1 for a timing source. - Otherwise, do not choose any if for example all PBX T1's installed. The settings is only a priority level for asterisk to obtain the source. Example: 1 = use this source

Re: [Asterisk-Users] Zaptel T1 Timing Source

Telco T1s and not connected to a PBX. Am I to assume that even if they are different providers the timing should be the same? That doesn't make a lot of sense to me. Thanks, Waldo On Nov 9, 2005, at 12:34 PM, Bart Fisher wrote: My understanding there should only be one timing source per

[Asterisk-Users] dial during greeting to access another extension if busy or not available?

Is there some way to allow dialing on top of mailbox greeting during playback to allow caller to move to another extension, and not the operator? Using version 1.2.0 Bart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] Re: Double DTMF with tdm card

the bug and the change will slightly improve TE410P performance Thanks for you help! Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:20 PM Subject

[Asterisk-Users] How does Nightly Downloads work at ftp://ftp.digium.com/pub/nightly

Maybe someone can explain how the Digium Nightly works. At ftp://ftp.digium.com/pub/nightly- As far as I can tell, there is a file posted regardless if any changes were made - True? The strange part is the file dates inside the "tar" - I would expect dates that were more current. For

Re: [Asterisk-Users] Double DTMF with tdm card

. Very frustrating. I've had to discontinue use of my TDM FXS ports until some solution is found. On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said: Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL

Re: [Asterisk-Users] Double DTMF with tdm card

you know Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: Bart Fisher [EMAIL PROTECTED] Cc: Walt Reed [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 9:57 AM Subject: Re

Re: [Asterisk-Users] Re: Double DTMF with tdm card

. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Bart Fisher [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] OK, then... I posted on the Bugs Web Site and markster said: This is a technical support issue. Please pursue through Digium tech support ([EMAIL PROTECTED]) and contact me if you have any issues

Re: [Asterisk-Users] Re: Double DTMF with tdm card

. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Bart Fisher [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] OK, then... I posted on the Bugs Web Site and markster said: This is a technical support issue. Please pursue

Re: [Asterisk-Users] Re: Double DTMF with tdm card

the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Bart Fisher [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] OK, then... I posted

Re: [Asterisk-Users] Double DTMF with tdm card

Did you ever find a solution for this problem? I have it on latest Beta 2 Bart - Original Message - From: Walt Reed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, October 21, 2005 7:26 AM Subject: [Asterisk-Users] Double DTMF with tdm card I have a TDM22B

Re: [Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge

Bump - I'm stuck until I can find a solutions Please help - I'll try anything! Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 01, 2005 5:37 PM Subject

[Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge

I have asterisk sitting in the middle with Telco on one side and Legacy PBX on the other using two T1 ports on a TE410P. I also have the latest Beta 2 installed. My problem is after a call is connected (port to port T1) and the outside user presses a touch tone, asterisk is repeating the

Re: [Asterisk-Users] Double DTMF sent on T1 to T1 Native Bridge

connection. - Make sense? This is a new problem since Asterisk 1.0.9, so I guess it's a bug? Seems there should be some way to make the Telco T1 stop listening and sending DTMF after connection Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk-Users

[Asterisk-Users] Asterisk and Zaptel Versions Command?

Is there a command line for discovery of Asterisk and Zaptel Versions? Bart ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Asterisk and Zaptel Versions Command?

of all the avaiable files that start with chan, you could also do just {TAB} to get a list of all the commands. To get help you could type help command. Hope this helps. On 10/31/05, Bart Fisher [EMAIL PROTECTED] wrote: Is there a command line for discovery of Asterisk and Zaptel Versions? Bart

[Asterisk-Users] I give up - Help with TE410P

I'm trying to install two TE410P's in one box. Would like to get 3 total. I can always get one card to work. If I install only one card, I will get green lights on all ports when loop back plugs installed - everything is perfect... If I install 2 cards, I'll get yellow alarm on span 2 and 6

Re: [Asterisk-Users] I give up - Help with TE410P

On Saturday 29 October 2005 18:06, Bart Fisher wrote: I'm trying to install two TE410P's in one box. Would like to get 3 total. I can always get one card to work. You are adjusting the 'ident' rotary switch on the others, right? -A. ___ --Bandwidth

Re: [Asterisk-Users] I give up - Help with TE410P

] To: asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 3:35 PM Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 18:19, Bart Fisher wrote: Yep - that was easy part :) and these are T1 (D4, AMI, SF, and EM Wink) BTW Ok, well I'll go for the obvious

Re: [Asterisk-Users] I give up - Help with TE410P

Of Andrew Kohlsmith Sent: Saturday, October 29, 2005 4:50 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 19:30, Bart Fisher wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch

[Asterisk-Users] EM to EM Dialing - TE410P

I have a TE410P with two real Telco T1's and the other 2 portsterminate into an in-house voice mail/IVR system. Calls arrive from Telco are routed to the appropriate in-house system based on the DID Digits. This part works perfectly. Now I what to allow the in-house VMS to dial though