Re: [Asterisk-Users] Where are chan_capi bug reports and bugfixes sent?

bristuff (same site as above) b) if you have kernel 2.6, use mISDN kernel patches and chan_mISDN, that is, seems, well supported and developed (and works, with a compilation flag, with asterisk stable and asterisk head as well): http://www.beronet.com/?PageID=3017 Best regards Marco Menardi Luis

[Asterisk-Users] New native assisted transfer (atxfer) usage info required

the 3 way calling of my TDM400, but I want to make this feature of asterisk working without any client implementation (that is the goal of atxfer). Thanks a lot Marco Menardi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Asterisk QSIG

system to Asterisk with a QSIG PRI interface ... Thanks a lot. marco -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.302 / Virus Database: 265.7.1 - Release Date: 19/01/2005 ___ Asterisk-Users mailing list

[Asterisk-Users] bri stuff and unknown signalling type

I've downloaded and compiled zaphfc and libpri. To do that i've downloaded bri-stuff and commented out the asterisk related stuff because i've installed it from a debian package. Does this means that i have to rebuild the whole asterisk thing to support zaphfc? thanks ciao

Re: [Asterisk-Users] hfc card and isdn error E001B

In data Thu, 9 Dec 2004 13:43:05 +0100, hai scritto: I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. More info. This is the output from asterisk: -- Registered SIP 'marco' at 192.168.0.5 port 5060 expires 120

[Asterisk-Users] hfc card and isdn error E001B

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten = _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten = _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error

[Asterisk-Users] Snom 220 Problem

the same problem ? Thanks Marco Vescovi --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.793 / Virus Database: 537 - Release Date: 10/11/2004 ___ Asterisk-Users mailing list [EMAIL

RE: [Asterisk-Users] Queue and Agent functionality

. The part that doesn't work is in the [fromcsr] context. My SetGlobalVar(GCSR1=on) seems to have no effect, therefore, making my solution not work. Does anyone have any ideas? Thanks, Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Icide

RE: [Asterisk-Users] Queue and Agent functionality

, what should I do once I determine if there is an agent logged in? Am I able to set an * global variable from within this external perl script? If not, what do you suggest? Thanks, Marco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von

[Asterisk-Users] agents and queues

Any help with this would be appreciated. thanks, Marco -- Marco Nicolayevsky Chief Technology Officer MisterArt.com LP http://www.misterart.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.

[Asterisk-Users] Problems with ISDN (NT-Mode) - Error Messages inside

tell me. :) Thanks, Marco ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] 2x HFC ISDN Cards - SuSE 9.1 - Problems with making calls

happend. TxA, TxB etc. are empty, too. Can someone help me? - I really need some sample configs, too. Which linux distribution runs smoothest with Asterisk? Thanks! Marco Czudej ___ Gesendet von Yahoo

[Asterisk-Users] Asterisk and AVM FritzBox Fon (Germany)

NOTICE[98310]: chan_sip.c:7653 sip_poke_noanswer: Peer '10' is now UNREACHABLE! I would be really pleased for any help :) Regards, Marco canreinvide=no ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB

[Asterisk-Users] Integrating an old PBX with Asterisk

the required extension, configuration changes to old PBX... I know that probably the best way should be to add a digital card to old PBX and havea trunk between two systems, but the PBX is really old and I'm not sure I can still find an expansion card. Any suggestion or tip ??? thanks marco

Re: [Asterisk-Users] Newbie voiceplus + asterisk

: voicepulse May 12 22:31:43 NOTICE[311316]: app_dial.c:527 dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy at this time -- Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Newbie voiceplus + asterisk

ok, I got it voicepulse vs voiceplus more questions coming I am sure... -- Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Variable digit length in national dial plan

someone help us? Best regards, Marco ___ 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] RTP channel

Hi all, i would like to know if it is possible to bridging the rtp traffic over Asterisk... I would like that the RTP flow is not controlled by * but by the endpoint. Is it possible??? Any suggestion to do this??? Thanks Marco Ë^®+$R²f¢–)à–+-Ë^®+$R²X¬¶Çb‚+¦r‰¡¶ÚþX¬¶Çb‚+¦r‰¿™¨¥™©ÿ

[Asterisk-Users] * with external sip proxy

to sip side all work When a try to place a call form sip to h323 nothing happen Does someone try this??? Any suggestion will be appreciate Tnx Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Error chan_oh323.so

liboh323wrap.so Someone has installed and using with success this oh323 package from inaccess networks ??? thanks in advance, Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Error chan_oh323.so

liboh323wrap.so Someone has installed and using with success this oh323 package from inaccess networks ??? thanks in advance, Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Info sip/h.323 interoperability

supports this interoperability ? I have done some test with Vocal to make calls from IP cisco Phone via sip/h323 translator to connect with Netmeeting or other h.323 end points... If this interoparability is supported which module I needs ? Thanks for attention, Marco

[Asterisk-Users] Setting up fax on *

Hello All, I am using an E100P card on a PRI line. I need to setup a FAX extension. Can somebody help me please? Marco

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