search for: a400e

Hello, I'm preparing a setup before installing it within the next few days. In this setup, I'm using a SPA112 as an ATA for an analog phone. The target phone is a Gigaset A400 DECT handset. In my lab, I've got another A400 handset and an old Matracom 46 handset. When I connect my Matracom 46 handset to my SPA112, I can send and receive calls. When I connect my A400 handset to the

I'm a long time user of Digium carts and stupid me i wanted to give Sangoma a try. We got Sangoma A400 with 6 FXO ports. Asterisk version: 1.4.35 Zaptel version: 1.4.11 Wanpipe version: 3.5.11 we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and

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 Network controller: Tiger J...

I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM

Hello Chris, On Thu, 01 Feb 2018 17:00:03 +0000 Chris Murphy <lists at colorremedies.com> wrote: > You can to use efibootmgr for this. NVRAM boot entry is what changed, not > the contents of the EFI System partition. > > efibootmgr -v > > Will list all entries and Boot Order. You need to use --bootorder to make > sure the CentOS entry is first. Interesting.. thanks

Hi everybody, I have a 1.6.0.5 asterisk system and I?m having many problems with my pstn cards. This system has a Digium B410P with 4 BRIs configured, and an OpenVox A400E. The issue is that it?s impossible to make the system recognize the A400E, despite the ?lspci? output is: 0a:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 11:00.0 ISDN controller: Digium, Inc. Wildcard B410 quad-BRI card (rev 01) And ?dahdi_hardware? output...

Hi list, I've just compiled and installed Asterisk 1.4.16 and when I try to run "zap show" I get the message "*No such command 'zap show'*". I have a Sangoma Remora A400D with 2 FXS / 10 FXO ports, I've installed the latest wanpipe too. zaptel-1.4.7.1 was compiled from the wanpipe installation, so I don't know what's happening here!!! Any ideas???

Is there any option in Asterisk distribution of this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120531/8fac6a22/attachment.htm>

Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1

Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance.

Hello, I am having problems, getting my asterisk-domU to work properly. It consists of the following components: - Debian Etch under Xen-3.1 with a 2.6.18-kernel - Asterisk 1.2.24 - mISDN-1.1.5 I have 2 HFC-ISDN-cards, which I pass through to the Asterisk-DomU in permissive mode. This is working fine. The strange problem is, that the two HFC-ISDN-cards are not beeing initialized by the

Hi ! I've fetched a spare AVM B1 card from the cellar, and installed it. After "modprobe b1pci" I did "capiinit" and capiinit moaned about a missing t1.b4. So I search the web and found one at http://www.avm.de/ftp/cardware/b1/x_misc/ddi/. When I now look at the controller, I finally see p2p-mode: # cat /proc/capi/controllers/1 name b1pciv4-a400 io

Hi list, I'm about to install Asterisk on an Old HP NetServer LC2000 Server (year 2001), it has 2 Pentium III 1GHz CPUs (Coppermine FSB 133MHz 256K L2 Cache), 768MB PC-133 ECC RAM, 3 UltraSCSI LVD2 18.2GB 10K RPM HDD in RAID5, 100Mb NIC for server. This Server will support 35 SIP phones (users) and 10 FXO ports (for telco lines) and 2 FXS ports (internal analog phones) with a Sangoma Remora

Hi list, I'm having problems transferring certain calls made by the attendant between the PSTN and to an internal extension. Although, transfers between the majority of the calls ends successfully. Debugin this, I've found that calls made to certain "numbers" (Telephony Providers), aren't detected as ANSWERED in the CDR, so they are not properly accounted (for billing),

I have a Sangoma A400 card with two FXS ports. They work fine, however as I have analog phones connected, I have no way of telling the phone I am done dialing. Pressing # works fine, but then Asterisk passes that # over to the POTS line, and about every 5th call, for some reason that is causing the call on the POTS line to fail. The suspect the trailing # is also going to get in the way

supervision. Verify if for those "numbers" the CO revert the line polarity when callee answer. callprogress=no is a good test too. Jorge Ra??l G??mez C. wrote: > Hi list, > > I'm having problems transferring certain calls made by the attendant > between the PSTN and to an internal extension. Although, transfers > between the majority of the calls ends successfully.

Hi list, I'm wondering if it's possible to transfer a call that is still ringing??? Actually, the problem is that my telco provider doesn't offer an uniform method for answer/disconnection supervision, and by that I mean, some of it's line (I think) offer a polarity reversal, but other lines (of the same service provider) do not offer anything at all, so the answer of a call

On Thu, Feb 1, 2018 at 10:13 AM, wwp <subscript at free.fr> wrote: > Hello Chris, > > > On Thu, 01 Feb 2018 17:00:03 +0000 Chris Murphy <lists at colorremedies.com> wrote: > >> You can to use efibootmgr for this. NVRAM boot entry is what changed, not >> the contents of the EFI System partition. >> >> efibootmgr -v >> >> Will list all

hi everybody a windows kvm guest would not start, process gets killed with: Out of memory: Kill process 21984 (qemu-kvm) score 44 or sacrifice child I really don't know where/what I might be missing, config seems fine, everything looks ok - I only am not sure, do I need to first stub a SR-IOV device like regular passthrough? I'm trying sr-iov, having one NIC left to the host and the

I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the