similar to: outgoing caller id from SIP to isdn (p2p)

hello, i have a problem starting asterisk with one x100p digium and two hfc chipset isdn cards with bri-stuff.0.0.2. "ztcfg -vv" shows me a this info: Zaptel Configuration ====================== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03:

hi all! Is it possible to trigger some external device trough parallel port or. serial on some caller id ? or if it's possible tu run some external (linux) program on specific caller id ? OR if anyone have idea how to open the doors with asterisk :) by, Thomas

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making

hi, when i call asterisk (on x100p) i got this : CLI> -- Starting simple switch on 'Zap/7-1' Jul 13 15:03:34 ERROR[311316]: callerid.c:192 callerid_feed: fsk_serie made mylen < 0 (-9) Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4735 ss_thread: CallerID feed failed: Success Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4777 ss_thread: CallerID returned with error on channel

Hi, I can't make outgoing calls with CAPI (passive ISDN Fritz card). See Asterisk error below. Incoming calls and SIP to SIP calls do work. It looks like a msn mismatch in extensions.conf and capi.conf, but I can't find it. Can anyone help me find the problem? Thanks, Rob *CLI> -- Executing Dial("SIP/8112-1be9", "CAPI/356666666:BYEXTENSION") in new stack

hi, I have a problem with passing caller id information from telco (isdn) to sip client (grandstream). i see callerid in asterisk verbose console but on grandstream (sip) phone is just internal (own-gs) 101 number. Isdn line is connected with hfc card and p2p , asterisk is latest CVS in extensions.conf i have: exten => 2442242,1,Dial,SIP/101,r|T and in console is this: -- Executing

hello, anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode? what kernel ? and second question mISDN driver .. anyone has working solution with mISDN and maybe fritz card? what you suggest for DDI -> point2point mode (card,kernel,chan_..., ...) ? thank you, Tomaz

Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in

I'm trying to set caller ids on outgoing calls. I have a quad BRI B410P card connected to my telephony provider. I know the list of DID numbers the provider assigned to my company. If I don't set the caller id then the callee always sees the same "top-level" number. If I set the caller id to a specific DID number we own, the callee keeps seeing the "top-level" number,

Is any way to limit music on hold (or ringtones) to specified time ? I need it to play it ~ for 7 seconds . How to do this ? in dial plan i have: exten => _021XXXXXX,4,Dial,Zap/1/BYEXTENSION||r when go to this extension it rings once! and then asterisk say : -- Zap/1-1 answered Modem[i4l]/ttyI0 and it stop ringing ;) becouse mean that other end is ringing :) .. BUT when the other

The situation is that I want to have certain extensions that can only call nationwide and others that can make international calls (with a different provider). I suppose, you could also see this as a scenario that could involve multiple internal extensions that get matched to specific outbound trunks. I have tried to find appropriate commands in the Asterisk documentation, but could find nothing

hello! I'd like to use my asterisk-box on a p2p-line from my alcatel pbx. isdn4linux and capi works fine on a s0-line from the pbx, but i need more then 8 msns, cause i'd like to implement a voice-mail system and an extension to our "old" pbx with voip-phones. had anyone success using isdn4linux on a t0 (p2p) line (of course with an inexpensive passive card like my avm

Hi there! I've set up an asterisk@home with AVM C2 P2P ... Everything works fine ... BUT ... If someone is dialing the PBX head number without any extension, asterisk can't handle this ... the DID in this case is empty .... Any ideas how to handle this? Regards, Marcus Hofbauer -- |** realit?t ist da wo der pizzamann herkommt **|

Hi! I'm looking for someone who has successfuly setup an asterisk in austria with isdn in p2p mode and chan_capi. There is is a special problem in austria with DID. If someone is dialing the phone number of the asterisk pbx like 12345-0, zero is passed as an DID, but in Austria u can dial 12345, and the DID which is passed is empty. It seems that asterisk cant handle this. Any ideas?

For outgoing calls made on our PRI circuit we are setting the Caller ID using the format Exten => _9XXXXXXX,1,SetCallerID(1601XXXXXXX) The monitor shows that the CallerID is being set to the specified number, but yet when the call is received on the user end the ID is always the base number of our DID. For example we have 8600-8650 as DID's but the callerid is always 8600 regardless of

Hello there I'm quite a newbie in the IP Telephony area. I'm playing a little around with a setup with one linux box with a e100 p card installed, which functions as an Asterisk gateway and a oh323 GK(Gnu Gatekeeper). I have two h323 phones, Welltech WellGate 1501 and 3502. So far I've managed to get the two IP phones and Asterisk connected to the GK. I can place calls from one

hi ---------modem.conf :---------- msn=240862922 incomingmsn=240866365,6365 device => /dev/ttyI2 group=1 device => /dev/ttyI1 ; ttyI3, ttyI4 ---------extensions.conf ;------- [sip] exten => _XXXXXXXXXX,1,Dial,Modem/g1:BYEXTENSION (Sjphpone) Call to : 024076xxxx result : --Executing Dial(Sip/roseau-6163","Modem/g1:BYEXTENION") in new stack -- Called g1:024076xxxx --

It might have a noticeable effect on clients. I encountered (probably triggered by this in some way?) that I was unable to het the 'read' bit set in macOS Mail.app. Maybe (as I am doing HA with round robin) the Mail.app client got to one dovecot repository on one tcp connection and then on the other. Is there a reason why syncing tis move from new to cur is a bad idea? Gerben Wierda

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

I can confirm this in a slightly different setting, but still using two-way sync between two dovecots. On e is 2.3.19.1 running on macOS Monterey, the other is 2.3.20 running in an alpine container on Ubuntu. Gerben Wierda (LinkedIn <https://www.linkedin.com/in/gerbenwierda>) R&A IT Strategy <https://ea.rna.nl/> (main site) Book: Chess and the Art of Enterprise?Architecture