kleis-asterisk-dev@tiscali.it
2004-Jul-26 07:35 UTC
[Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the latest bri-stuff.0.1.0-RC2g (released just today!) and started the installation. My extensions.conf is: [default] ...cut... ignorepat => 9 exten => 9,1,Dial(Zap/g1/) ; direct outbound dialing exten => 9,2,Congestion Here's my zaptel.conf: loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 Here's my zapata.conf: [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode signalling = bri_cpe_ptmp pridialplan=local prilocaldialplan=local echocancel=yes immediate=yes group = 1 context=default channel => 1-2 Here's my channels map: Zaptel Configuration ===================== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. Finally, here's my sip.conf: [general] context=default ; Default context for incoming calls ...cut... disallow=all ; First disallow all codecs allow=gsm ; Allow codecs in order of preference allow=ulaw allow=alaw ...cut... [alessandro] type=friend username=alessandro secret=bissoli host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm allow=alaw allow=ulaw When I run asterisk and dial from the SIP phone I get this error: -- Executing Dial("SIP/alessandro-0f69", "Zap/g1/") in new stack Jul 26 14:27:31 NOTICE[311313]: app_dial.c:711 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Congestion("SIP/alessandro-0f69", "") in new stack == Spawn extension (default, 9, 2) exited non-zero on 'SIP/alessandro-0f69' I really don't know what is wrong! Do you have any hints, please? Alex __________________________________________________________________ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27 luglio. Per te gratis il modem in comodato e l'attivazione. In piu' navighi a soli 1,5 euro l'ora per i primi tre mesi. Cosa aspetti? Attivala subito! http://abbonati.tiscali.it/adsl/prodotti/640Kbps/
Sjaakie Helderhorst
2004-Jul-26 08:13 UTC
[Asterisk-Users] Re: Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
I got things running with ISDN4Linux See configuration example below, I found it exploring the WIKI-site. (to make an outgoing call users need to press 0*[number to call]) Hope this is useful. modem.conf: ------------- [interfaces] context=remote noload => chan_modem.so driver=i4l language=nl type=autodetect dialtype=tone mode=immediate msn=[your MSN] incomingmsn=[MSN1],[MSN2],... device => /dev/ttyI0 device => /dev/ttyI1 extensions.conf --------------- [remote] include => local-sip exten => s,1,Wait(20) exten => s,2,Answer exten => s,3,DigitTimeout(10) ; Set Digit Timeout to 10 seconds exten => s,4,ResponseTimeout(20) ; Set Response Timeout to 20 seconds exten => s,5,Background(vm-extension) ; Ask them for the extension they want [dial-via-isdn] exten => _0*XXX.,1,Dial(Modem/ttyI0:${EXTEN:2},30,r) exten => _0*XXX.,2,Playback(asterisk-sounds/sounds/the-party-you-are-calling) exten => _0*XXX.,3,Playback(asterisk-sounds/sounds/is-curntly-unavail) exten => _0*XXX.,4,Wait(2.5) exten => _0*XXX.,5,Playtones(congestion) exten => _0*XXX.,102,Playback(asterisk-sounds/sounds/the-party-you-are-calling) exten => _0*XXX.,103,Playback(asterisk-sounds/sounds/is-curntly-busy) exten => _0*XXX.,104,Wait(2.5) exten => _0*XXX.,105,Playtones(congestion) exten => _0*XXX.,106,Hangup <kleis-asterisk-dev@tiscali.it> schreef in bericht news:4104C030000024DD@mail-3.tiscali.it... Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the latest bri-stuff.0.1.0-RC2g (released just today!) and started the installation.
Robinson Tim-W10277
2004-Jul-26 08:20 UTC
[Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
Try ignorepat => 9 exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) ; direct outbound dialing exten => _9.,2,Congestion Best Regards Tim Robinson Tools Development Manager Motorola Ltd Midpoint Alencon Link BASINGSTOKE RG21 7PL United Kingdom Tel. +44 1256 790472 Fax +44 1256 790190 Mobile +44 7785 300316 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of kleis-asterisk-dev@tiscali.it Sent: 26 July 2004 15:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap' Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the latest bri-stuff.0.1.0-RC2g (released just today!) and started the installation. My extensions.conf is: [default] ...cut... ignorepat => 9 exten => 9,1,Dial(Zap/g1/) ; direct outbound dialing exten => 9,2,Congestion Here's my zaptel.conf: loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3
Matteo Brancaleoni
2004-Jul-26 08:21 UTC
[Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
Hi Il lun, 2004-07-26 alle 16:35, kleis-asterisk-dev@tiscali.it ha scritto:> -- Executing Dial("SIP/alessandro-0f69", "Zap/g1/") in new stack Jul > 26 14:27:31 NOTICE[311313]: app_dial.c:711 dial_exec: Unable to create channel > of type 'Zap' > == Everyone is busy/congested at this time > -- Executing Congestion("SIP/alessandro-0f69", "") in new stack > == Spawn extension (default, 9, 2) exited non-zero on 'SIP/alessandro-0f69' > > I really don't know what is wrong! Do you have any hints, please?just read what you see: unable to create channel of type Zap. that seems that asterisk isn't loading chan_zap... what says zap show channels on the cli? or you see chan_zap.so in show modules ? also, your extension.conf is wrong. instead of ignorepat => 9 exten => 9,1,Dial(Zap/g1/) ; direct outbound dialing exten => 9,2,Congestion use exten => _9X.,1,Dial(Zap/g1/${EXTEN:1}) ; direct outbound dialing exten => _9X.,2,Congestion then read the docs! Matteo -- **************************************** Matteo Brancaleoni System Administrator mbrancaleoni@espia.it **************************************** EspiA Srl - e*solution provider Via Pascoli, 37 20129 Milano - Italy SIP:matteo@sip.voismart.it Tel. +39 0270633354 Fax. +39 0245487890 IAXTEL: 17005662458 http://www.espia.it ****************************************
Alessandro Bissoli
2004-Jul-28 01:11 UTC
[Asterisk-Users] Re: Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Sjaakie Helderhorst > Sent: Monday, July 26, 2004 5:14 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Re: Can't dial SIP<->EuroISDN (HFC-S basedPCI> ISDN card): Unable to create channel of type 'Zap' > > I got things running with ISDN4Linux > See configuration example below, I found it exploring the WIKI-site. > (to make an outgoing call users need to press 0*[number to call]) > Hope this is useful.It seems an interesting solution, but I need echo cancellation and so I have to use zaphfc. Thanks, Alessandro
Alessandro Bissoli
2004-Jul-28 02:01 UTC
[Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of kleis-asterisk-dev@tiscali.it > Sent: Monday, July 26, 2004 4:36 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCIISDN> card): Unable to create channel of type 'Zap' > > Hi, > > I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linuxbox> (customized kernel version 2.4.24). I want calls from my SIPsoft-phones> to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a > cheap > HFC-S based PCI ISDN card connected to the NT1+ interface, so I need > zaphfc. > I've read everything I've found at www.voip-info.org, then I'vedownloaded> the latest bri-stuff.0.1.0-RC2g (released just today!) and started the > installation.I still have the problem! I really have no idea about what to do! Any suggestion would be greatly appreciated. Thanks, Alex
Alessandro Bissoli
2004-Jul-28 04:23 UTC
[Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of kleis-asterisk-dev@tiscali.it > Sent: Monday, July 26, 2004 4:36 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCIISDN> card): Unable to create channel of type 'Zap'Hi, The following is from Asterisk's log (asterisk -vvvvgc | tee asterisk.log): [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found Jul 28 12:34:29 WARNING[16384]: chan_skinny.c:2584 reload_conf ig: Unable to get our IP address, Skinny disabled == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Jul 28 12:34:29 WARNING[163850]: chan_oss.c:238 sound_thread : Read error on sound device: Resource temporarily unavailable Do you think that such warnigs may be somehow related to "Unable to create channel of type 'Zap'"? (Soundcard is an onboard VIA chipset based card) Thanks, Alex