similar to: DAHDI FXO port only recognizes the "S" extension‏

I'm moving asterisk to a laptop, so can't use the dahdi board. Is there any supported USB dahdi device ? I see the Sangoma USBfxo device, but the dahdi driver no longer supports it. Anything else ? sean

Calls that come in on DAHDI FXO ports are routed to [context], extension 's' INSTEAD, I would like to route specific ports to specific extensions, For example: I want DAHDI/1-1 to go to 1234 I want DAHDI/1-2 to go to 2345 I want DAHDI/1-3 to go to 3456 ...etc What is the CLEANEST way to do this? Yes, I can create a private context for each DAHDI channel but that seems messy and

Dear listers, I'm facing a puzzling situation with Digium TDM2400 card (12 FXO / 12 FXS). Once a day or so we detect 1 or 2 zombie FXO channels. These can be either outbound or inbound calls. I thought this could be related to obsolete DAHDI or Asterisk versions, so I upgraded to 2.4.0 and 1.6.2.15 respectively (OS: Ubuntu 10.04 64 bits). To no avail; the zombie channels keep showing up.

Hi, I've just installed DAHDI at two PBXs as follows: *PBX-1 PBX-2* FXO ------------- FXS When I try to send calls from PBX-1 to PBX-2 I just receive the message: "Starting simple switch on 'DAHDI/1-1" It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at chan_dahdi.conf at PBX-2 dialplan is executed at PBX-2 but nothing is heard at

Hi, can anybody please show me a valid dial plan for a dahdi card with a bri port? I can not get asterisk 1.6 to dial a number. I want to receive one call on the first b channel and dial another number on the second b channel of the same isdn port. I tried something like Dial(DAHDI/g1/123456789) but that does not work. It always says that the channel is busy. Kind regards,

Hi, I'm using Asterisk 1.6.1.18 with latest Dahdi Linux and Tools (rev 8854) and libpri 1.4.10.2. During installation, I'm facing this : # dahdi_genconf -v system Default parameters from /etc/dahdi/genconf_parameters Empty configuration -- no spans Of course, I've got span descriptions in /etc/dahdi/genconf_parameters : .... pri_termtype SPAN/1 TE SPAN/2

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Dears; I am facing the following problem: Already we requested from the service provider to enable the auto jumping service for our analoge telephone lines, so because we have 4 telephone lines from the service provider, then if you called line # 1 and it was busy, then the call will be sent to any available line #2 or #3 or #4, and if you call line # 3 and it was busy then the call will be sent

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 -

On 12/14/19 11:29 AM, Greg Troxel wrote: > sean darcy <seandarcy2 at gmail.com> writes: > >>> There is also the ObiHai OBi202 with an OBiLine, which provides an FXO >>> port remoted over SIP. (I am not sure if this is discontinued.) >> >> "FXO port remoted over SIP"? >> >> I have an analog phone system. I can use the obi202 to

Hi All, I am new to Asterisk. Now I got one question on the identifier parameter of the Dial() command. I saw as below: exten => 20,1,Dia(Zap/3/5551234). Would you please let me know the meaning of "5551234"? Thanks, Songtao -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, My costumers are logged in on my Asterisk PBX through XLite Softphone (SIP). My server is connected to PSTN. Problem is when SIP phone calls ordinary phone via dahdi I get DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is start counting. Is it normal behavior ? Can I change that ? So channel gets in ANSWERED state and billsec starts as soon as line starts to ring even if no

So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3

Hi, Could someone explain me the following questions: Where are differences between Para-Virtualization and Full-Virtualization in the aspects of memory management, CPU and Device I/O, Network? How is the performance difference between them? Thanks Songtao _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.11.1 DAHDI-Tools-v2.11.1 dahdi-linux-complete-2.11.1+2.11.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Notable changes: Raised E1

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.11.1 DAHDI-Tools-v2.11.1 dahdi-linux-complete-2.11.1+2.11.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Notable changes: Raised E1

hello , i try to install dahdi-linux/dahdi-linux-current.tar.gz ,dahdi-tools/dahdi-tools-current.tar.gz and libpri/libpri-1.4-current.tar.gz but got these error msg make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.1/drivers/dahdi/firmware' make -C

Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want

I've got a single TDM 400P board with two internal ports and 1 external. chan_dahdi.conf: context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks not auto Use FXO signalling for FXS group=0 channel => 1 ; Telephone attached to port 1 channel => 2 ; Telephone attached to port 2 context=incoming-pstn-line ; Incoming calls