similar to: dahdi wcb4xxp and fax

Hello all, I have an asterisk box running in a customer with Hylafax, iaxmodem, asterisk 1.2.18. The service can receive faxes, from a lot of fax machines, but there are a couple of them that asterisk Hylafax cannot complete. This calls arrive the asterisk box, asterisk detect that this calls are fax, asterisk answer the call, and then Hangup the call. But hylafax do not receive nothing. When

hello: thanks for Tzafrir Cohen for dahdi testing. I installed dahdi-2.1-r3c svn code and asterisk1-6 for testing OpenVox B400P and junghans card. i fund that there is bug (i think) to dectect NT or TE mode. actually on the board, i set it as TE mode, but after start wcb4xxp, but it show the port is NT mode. to detect the TE mode, I modefy the code in base.c

Hi, I'd like to know what's the most "popular" method for automatic fax/voice detection for incoming calls on mISDN cards such as the B410P (hfcmulti). I'm running: kernel 2.6.17 misdn 1.1.3 asterisk 1.4.21.2 B410P card I'm using iaxmodem and hylafax with asterisk (the setup works for zap channels). I've used the following options in /etc/asterisk/misdn.conf:

Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1

Hi, 1. Has someone met any success at all, connecting a Digium B410P to a Patton Smartnode 4638 (with latest 5.3 firmware) ? 2. If positive, then, which signalling was used on both sides ? My project's goal is use a Patton Smartnode 4638 to act as telco BRI lines, from a B410P-enabled asterisk box. In my testings, I can see channel is respectively up (with patton web server status page)

OK... so I'm not sure what I'm looking at. I've had the good old echo problems on my Rev C FXO again this morning, so I thought I'd attempt some debugging, though I'm not sure what I'm looking at. This call has echo. Channel: 2 File Descriptor: 20 Span: 1I> Extension: Dialing: no Context: incoming Caller ID string: "External Call" <99999999> Destroy:

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap

Dear all, I know this is more a Digium hardware than an Asterisk issue. Already posted a question at Digium, however also like to see whether anyone in the Asterisk community has encountered the following situation: I installed a Digium B410P BRI PCI card on my new asterisk server, following the steps specified in the manual. I can see the PCI card is available using the lspci command: ...

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

Hi, I'm sending twice the same file using Hylafax's sendfax app. The first time I'm "dialing" a DID attached to the ReceiveFAX application. The second time I'm dialing an internal extension attached to the same ReceiveFAX application : 1. sendfax/hylafax/iaxmodem ----> asterisk ----> dadhi ----> dahdi ----> asterisk ----> spandsp 2.

What are the meaning of these Zap show channel output? Caller ID string: Owner: <None> Real: <None> Callwait: <None> Threeway: <None> Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0,

Hello, In my system, dmesg outputs : [ 13.039788] wcb4xxp 0000:01:0a.0: probe called for b4xx... [ 13.039788] ACPI: PCI Interrupt 0000:01:0a.0[A] -> GSI 21 (level, low) -> IRQ 21 [ 13.039788] wcb4xxp 0000:01:0a.0: Identified HFC-8S Junghanns.NET octoBRI PCI (controller rev 1) at 0001dfe0, IRQ 21 [ 13.041202] wcb4xxp 0000:01:0a.0: NOTE: hardware echo cancellation has been disabled [

Hi, I'm trying to make a dialup internet connection through my asterisk PBX. When I bipass the Asterisk box, I can get 51600bps. When I run through the asterisk box, I'm limited to about 21600bps. I have a TDM31B card. Any help on speeding these connections up would be good - I was on the understanding that if you bridged the channels, then the call should essentially flow straight

My office is investigating using an Asterisk PBX and also going to a VOIP provider for our main phone connections, but one of the tricky things is that we need to have outbound and inbound modem calls (fax too). I see a lot of talk about faxes but no mention of modems on this list. I get the impression that modems will be a problem for us. Is that true?

Hello everyone, I have a Atcom AX-1600P(1) card with a FXO module and I can't configure it. I have four extension with this PJSIP settings: --- /etc/asterisk/pjsip.conf --- [transport-udp] type=transport protocol=udp bind=0.0.0.0 [6001] type=endpoint transport=transport-udp context=from-internal disallow=all allow=ulaw auth=6001 aors=6001 direct_media=no rtp_symmetric=yes force_rport=yes

Hello, Is it possible, for testing, to connect an cat5 straight patch cord between 2 ports of a Digium B410P card and use these 2 ports as a normal dahdi trunk ? I've tried this: One port is set as NT, the other as TE. I would expect timing to come for system hardware so I choose in /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Results: - both ports lights are green -

Hi everybody, In an Asterisk 1.6.0.5, dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2, with a Digium B410P and an OpenVox A400E, I can't make "dahdi_scan" to recognize the OpenVox. This card was working correctly but suddenly stopped working and I cannot make it work again. Both ?lspci? and ?dahdi_hardware? detect it but ?dahdi_scan? not and I cannot use it. >lsppci: *0a:00.0

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn