similar to: How to know which party hangup() first when using analogue cards? - Dahdi and Asterisk 1.4.x

Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *"asteriskcdrb"* table and it's pretty much useless in this case as it only logs the duration and

So made my first forray into 1.4 and DAHDI and hit a problem. (Not convinced this is a DAHDI issue though...) Testing an analogue line and asterisk sees the caller ID being passed, but then fails to detect ringing. A plain old analogue phone plugged in rings just fine. Console output: == Starting post polarity CID detection on channel 4 -- Starting simple switch on

Hi Everyone, I want to fine tune the Rx and Tx gain on an analogue Sangoma card by dialing into another server that is running on Sangoma PRI card (both services on Bell network). [mwatt1004khz] exten => s,1,Answer exten => s,n,PlayTones(1004/1000) exten => s,n,Wait(300) If I match the Rx/Tx numbers on both sides by monitoring "ztmonitor X -vv" am I right with my theory of

Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on 1.4.. Nothing special about the hardware - older TDM400 card, 2 red modules fitted... Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels still work OK, but only for one line - the 2nd line causes it to refuse to dial-out no matter which port it's plugged into. The Lines are bog-standard BT

Anyone try this? I saw a post here: http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html But not sure if it's possible. What I am asking is if there are any T1 cards with virtual functions implemented in their drivers to allow pci-passthrough? Kind Regards, Nick.

Hello All; After trying A2Billing and certainly when the trunk is analogue lines (FXO ports), I faced a problem that the channels were not hanged up properly from time to time which cause us to do restart for the dahdi. Without A2Billing, I was able to handle the Dial scenario properly and no hanging for the analogue channels and no need to restart dahdi from time to time.? Really I would if

Guys, I need to use Asterisk to call out PSTN numbers via an analogue line. I understand Digium manufactures these kinds of cards, but can someone tell me which model number it is. I really only need a card with one or 2 analogue ports max. Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. [Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16 [Sep 29

Hello, I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I would like use analogue lines for outboud calls. How is it possibile to detect ANSWER? - answeronpolarityswitch does not seem to work in Italy - call progress does not give safe results, sometimes calls get billed, sometimes not Am I forced to buy an ISDN adapter or could there be a solution (maybe tweaking some

This Wednesday, November 26th, the Toronto Asterisk Users Group invites all in the area to join us for a telephony workshop and talk sponsored by Sangoma Inc.[1] Jose Landivar, co-founder of PaloSanto Solutions[2], creators of Elastix, will be running a "getting started" workshop on Elastix, followed by a talk discussing how it differs from other Asterisk-based distributions, and a

Hi I have 2 digium cards (tdm410) with combination of fxs + multiple fxo ports. I have had a quick look at sangoma B series cards. I was wondering if there is a card out there with hardware echo canceller say max 4 ports (mix of fxs/fxo) g729 encoding onboard Alex -- "More and more of our imports are coming from overseas." - George W. Bush 09/26/2005 On NPR's Morning

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

I have a potential customer who has an existing PBX with analogue FXS ports connected to phones. He wants to allow a single remote worker to be connected to one of the analogue extension ports using VOIP. I know I could do it using Asterisk with an X100P card, but that seems a bit overkill. Does anyone know of an analogue->VOIP adapter that has an FXO port in it instead of just an FXS port?

Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new stack [Jul

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

Is there anyone using * in AU that has successfully extracted the CLID from an incoming analogue PSTN phone call, and would like to spread the word? Note - I am only interested in analogue, not ISDN phones. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you

I try to perform a clustering using an existing dissimilarity matrix that I calculated using distance (analogue) I tried two different things. One of them worked and one not and I don`t understand why. Here the code: not working example library(cluster) library(analogue) iris2<-as.data.frame(iris) str(iris2) 'data.frame': 150 obs. of 5 variables: $ Sepal.Length: num 5.1 4.9 4.7

Hello, on previous versions of asterisk, extension h and H make us know who ended a call (caller or callee). In the last * versions, seems that only h extension is used, as stated here http://www.voip-info.org/wiki/view/Asterisk+standard+extensions In the last versions, how do we know which end terminate a call (SIP, ISDN, Analog, ...) in h extension ? Will the

I'm trying to work out whether I want an ADSI phone or not, and in fact, whether it is ... useful/works with Asterisk. I have decided that for the cost/performance of the various IP based phones, I am not interested, I seem to get significantly better quality from a plain analogue phone using the TDM40B card. Is it beneficial to get an ADSI phone as opposed to a plain old analogue phone from