[Asterisk-Users] Asterisk rxfax(): One page gets two pages

Hi, so far I had our PRI line connected via Digium TE410P and ZAP channel to asterisk which worked perfectly. I now have a second line coming in through a H.323 gateway and chan_oh323. rxfax() still works and receives faxes with G.711 alaw codec, but I always get one empty first page via H.323

[Asterisk-Users] chan_oh323: busy not correctly signalled

Hi, I have asterisk connected to PSTN via H.323 gateway via chan_oh323. Incoming calls to SIP extensions work, but SIP message 486 busy here from a busy extension isn't correctly forwarded to H.323. As a result, a caller from the H.323 side calling a busy SIP extension gets some rings and then

[Asterisk-Users] H.323 and cause code 'user busy'

Hi all, I just installed chan_h323 to interface to a H.323/ISDN gateway. It works really well after two days learning and testing except one thing somebody of you may have an answer to: If I call in from PSTN to a busy asterisk SIP extension I can see a SIP status 486 BUSY, but don't get it

[Asterisk-Users] Indication Busy to a ZAP ISDN channel

then instead of busy indication. Asterisk seems to know the ext. is busy but doesn't do the correct signalling in the zap ISDN D-channel. What is the correct way to do this, of course without answering the channel and thus producing costs to the caller? Thanks and regards, Jan Baumann

Re: [Asterisk-Users] How to use ZapHFC ?

asterisk to present a dialtone to the isdn phones first. After a Timeout asterisk should hang up. Hi Martin, overlap dial is what you are looking for. Try 'overlapdial=yes' in the channel definition in zapata.conf. Jan Baumann ___ Asterisk-Users

[Asterisk-Users] RxFax/spandsp: file-naming of received faxes

Hi, after successfully having installed RxFax/SpanDSP and some promising tests (great piece of software, Steve!) I wonder if it is possible to avoid overwriting the same tiff file over and over again. Browsing the sourcecode of app_rxfax.c I found a magic '%d' flag being parsed out from the

[Asterisk-Users] Cisco ATA186 SIP transfer

that? Thank you and regards, Jan Baumann [did-from-pstn] exten = 1234531,1,SetVar(ALERT_INFO=1) exten = 1234531,2,LookupCIDName exten = 1234531,3,Dial(SIP/31,20,t) exten = 1234531,4,Voicemail2(u31) exten = 1234531,5,Hangup exten = 1234531,104,Busy exten = 1234532,1,SetVar(ALERT_INFO=1) exten = 1234532,2

[Asterisk-Users] ext.conf for european variable length dialplan ?

with zaptel channels? May I ask what other european * users are using to handle this? Thanks and regards, Jan Baumann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

[Asterisk-Users] zaphfc bri with overlap sending/receiving

are greatly appreciated and very welcome. :) Thank you and regards, Jan Baumann My current config: extensions.conf: ; outbound dialing local calls ; try Enum, then PSTN [local-pstn] exten = _0[1-9]XX.,1,EnumLookup(49821${EXTEN:1}) exten = _0[1-9]XX.,2,SetCallerID(49821xx) exten = _0[1-9]XX

[Asterisk-Users] zaphfc bri: crackling sound

right now. Thank you and king regards, Jan Baumann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

TE410P? Thanks for sharing your experiences. It is greately appreciated. Kind regards, Jan Baumann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Call Queue with no agents - Congestion or voicebox instead of MOH?

Hi all, I have successfully set up a call queue with agents and agentCallbackLogin. Works fine, but if no agent is logged in incoming callers get music-on-hold forever (or until some timeout). Is it possible to play congestion tone without answering the call (and thus causing costs to PSTN

Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea

Ken Alker wrote: Based on several threads I've read on this list, I assume that it would be handy to supply POE (power over ethernet) in an environment without having to purchase POE switches (assumed expensive) and abandon one's existing (familiar/custom/not-yet-expensed/etc.) switches/hubs.

[Asterisk-Users] Asterisk switching ISDN data calls?

it this way. Will my data PBX extensions (PCs with ISDN PCI cards) still be able to dial out through asterisk to place data calls to other PCs? Has anybody tried this (maybe Mr. Junghanns with the 4xBRI card) ? I would very much appreciate reading your experiences. Thank you all, Jan Baumann

[Asterisk-Users] TDM switching between Digium TE410P ports

will be NET5/euroISDN. Many Thanks, Jan Baumann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Multiple phonenumbers on one E1 PRI with Digium TE410P ?

the driver handle this and can I put calls coming in all on the same physical interface put into different contexts based on the dialed prefix? Thanks and Regards, Jan Baumann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

[Asterisk-Users] Securing Cisco SIP gateway

this successfully before and would very much appreciate any hints how to do this best. Many thanks and kind regards, Jan Baumann ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE