similar to: how to selectively disable callerid block?

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

Upgrading an ancient customer installation... was running 1.4.23.1 (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 1.8.24.0, Dahdi 2.7.0.1, and a new Sangoma A104D. The single active span is an RBS T1 B8ZS/ESF/E&M Wink. I tried to move one span over one

Hi everyone! I have a Digium Card TDM410 But, it appear for me this massege chan_dahdi.c:8061 handle_alarms: Detected alarm on channel 3: Red Alarm But my line is ok! But sometimes it back sig_analog.c:3807 analog_handle_init_event: Alarm cleared on channel 2 But it again back to red alarm. What can be happen? My lines is all ok! But when I put on Digium Card TDM410 is very inconsistent

Humm, if I put a filter in this lines, maybe back? 2016-01-05 12:36 GMT-02:00, Ryan, Travis <RyanT at oscarwinski.com>: > >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- >> bounces at lists.digium.com] On Behalf Of Vitor Mazuco >> Sent: Tuesday, January 05, 2016 9:21 AM >> To: Asterisk Users Mailing

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

I'd like to have my Asterisk system pick up a certain extension on an existing Panasonic PBX when it rings. (It's connected to some proprietary Panasonic doorphones that I haven't replaced yet.) I connected that extension to an FXO port on a Digium AEX410 card, and set that channel to have the context "doorphone". The problem is that the extension is never executed. With

Hi all! I'm getting an error when I try to start asterisk with chan_misdn. I patched my kernel (2.6.12.4), and compiled the whole stuff (kernel, mISDNuser, asterisk, chan_misdn). I got mISDN from http://isdn.jolly.de/download/v3.0/ I'm using a CVS Snapshot of asterisk, which was checked out about 5 hours ago. This is the error: [chan_misdn.so]Aug 15 14:13:29 WARNING[4929]:

The Asterisk Development Team would like to announce the release of Asterisk 18.19.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.19.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Asterisk 18.19.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.19.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Asterisk 20.4.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.4.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Asterisk 20.4.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.4.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

Hi, I am trying to upgrade my system (running 2.4 kernel) from 1.0.9 to 1.2.6, everything upgraded fine, however asterisk is not seeing any zap/sip/iax2 channels. I compiled in this order: libpri/zaptel/asterisk. Zaptel comes up fine... ztcfg -vv shows all of my channels, however asterisk lacks the 'zap show' 'sip show' or 'iax2 show' commands, further, if I try to force

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate

Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L

I am trying to use chan_skinny but when loading the module I get: [ Booting....../usr/lib/asterisk/modules/chan_skinny.so: undefined symbol: ast_pickup_call I am using CVS 07/23 I can't get chan_sccp2 to compile, it gives me parse errors, or I'd be using that. :-/