similar to: CID debugging help needed

I have an Asterisk 1.4 setup behind a TDM800P with 5 incoming lines, which happen to be inside my university's PBX. I have a problem wherein the university's PBX is apparently sending its CID spill after the second ring for a certain class of incoming calls (those from outside the PBX). I've temporarily hacked around this with a goto in chan_zap.c to restart CID processing if a good

We have gotten stuck trying to get a highly available Asterisk cluster fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's behind the virtual public IP. I got as far as getting phones registered and being able to place calls that rang and you could answer, but there was no audio. So, I enabled RTP debugging and discovered Asterisk was still attempting to send the audio

There was a flurry of "Vonage is going to unlock SIP" activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you

On Aug 12, 2019, at 12:18 AM, Sami Ketola via dovecot <dovecot at dovecot.org> wrote: > > "1. What's the first mail's saved-timestamp? > doveadm fetch -u user date.saved mailbox Junk 1 Well... this is giving me strange results. For myself, when I run this command, I get: [~]# doveadm fetch -u cepheid date.saved mailbox Junk 1 date.saved: 2019-07-15 01:42:56

Hello group, I am installing Asterisk on a new pc running Ubuntu 12.04.3 Server. ?I have dahdi-complete-2.7.0.1+2.7.0.1, and have tried 2.6.1 and 2.8.0-rc1 and 2.8.0-rc2. ?I am following along in the ASTERISK The Definitive Guide 4th Edition and on page 116 when I issue the command $ sudo /etc/init.d/dahdi start I get the following error: Loading DAHDI hardware modules: ? ?wctdm24xxp: done

I just installed a TE820 octal span T1 card, and it's not showing up in dahdi_hardware output. This was installed into a test machine that already has a TDM800P card in it, and that one is showing up and working fine. Is there some kernel module that I'm missing? Lspci: 05:04.0 Ethernet controller: Digium, Inc. Wildcard TDM800P 8-port analog card (rev 11) 21:08.0 Communication

Hi All, I have a server running Fedora 14, kernel 2.6.31.14, Asterisk 1.6.2.17.2 and Dahdi 2.4.1. I have the wctdm24xxp+ loaded with a Wildcard TDM800P with 8 FXO ports When a call is placed extension to extension, there is no problem... When an extension is used to dial out, the called number answers, but the server only detects the answer about 3 seconds later .. The outgoing

Hello! I have the autoexpunge settings defined inside protocol imap thus (and it works): protocol imap { ? mail_plugins = quota notify replication imap_quota imap_sieve ? namespace inbox { ??? location = ??? mailbox Ham { ????? autoexpunge = 365 days ??? } ??? mailbox Spam { ????? autoexpunge = 365 days ??? } ??? mailbox Trash { ????? autoexpunge = 180 days ??? } ??? prefix = ? }

On 8 Aug 2019, at 13:03, Amir Caspi <cepheid at 3phase.com> wrote: > IMHO the setting should apply regardless of protocol, but is that actually the case in practice? It seems to be broken. I have namespace inbox { inbox = yes location = mailbox Drafts { special_use = \Drafts } mailbox Junk { autoexpunge = 14 days auto = subscribe special_use = \Junk } ? } and I have messages in my

I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? TDM800P and HPEC The TDM800P is an 8-port analog telephony interface card, so it fills the gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines are going to be with us for some time, and demand for support for them remains high. The

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

Dear all I have asterisk 1.4.11 on Cent 4.3 i have faceing some problem i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make outgoing call it is working fine but incomming call not handling ...when i call from outside to this line it is rinning but no one call land on my asterisk no debug in asterisk some time it land but most of time not .....

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part

Hello everyone, How can I let Digium Wildcard TDM800P work successfully with DAHDI? Because the Centos recognizes the card but I can't get the analog card working with DAHDI. Thanks in advance, Gilberto -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130309/d82ddd47/attachment.htm>

Hi All, This is a request for testing for users of Asterisk 1.4 or 1.6 with any of the following Digium VoiceBus based cards: TDM2400P, AEX2400, AEX800, TDM800P, TDM410, TE120P, TE121, and/or TE122. From a practical standpoint, this branch should allow these boards to work in more systems / configurations where IRQ misses caused problems in the past. The updated wctdm24xxp and wcte12xp

The Asterisk.org development team has announced the release of Zaptel versions 1.2.20 and 1.4.5. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. This release also contains support for Digium's new 32 channel hardware echo canceler

The Asterisk.org development team has announced the release of Zaptel versions 1.2.20 and 1.4.5. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. This release also contains support for Digium's new 32 channel hardware echo canceler

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number

On Aug 12, 2019, at 8:54 PM, Thomas Zajic via dovecot <dovecot at dovecot.org> wrote: > > * Amir Caspi via dovecot, 12.08.19 22:01 > >> [~]# doveadm mailbox status -u cepheid firstsaved Junk >> Junk firstsaved=1563154976 >> >> I can't tell how that timestamp corresponds to a human-readable date, however. > > [zlatko at disclosure:~]$ date -d

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd