[Asterisk-Users] checkout ztdummy
Hi, how can I checkout ztdummy? Thank for you help. Felix Deierlein ___ 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
AW: [Asterisk-Users] checkout ztdummy
Hi, thanks. >> how can I checkout ztdummy? >> Thank for you help. >Checkout of cvs the zaptel source then follow these instructions: >http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy I have tried to follow, but I did not know, wich modul I had to check out.. Bye Felix ___ 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] PRI & immediate=no
Hi Thomas, > I have got the following problem (E100P, pri_cpe): > My number range is <6digits>xyz. (e.g. 123456-999) > >From ISDN phones, everything's fine, but calling in from analogue > >phones causes the > following problem: Asterisk only receives the first 6 digits. Do you have overlapdial=yes in your zapata.conf? Cheers Felix ___ 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] Compiling zaptel under 9.1 Suse
Hi, at SuSE 9.0 helped: > > I am not able to compile zaptel... > > Could you give me a hint? > Have you tried the following, which is suggested in the output? > 'make cloneconfig && make dep' in /usr/src/linux/ Felix > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Michael George > Sent: Thursday, June 24, 2004 8:53 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Compiling zaptel under 9.1 Suse > > Try building the kernel and the build the zaptel drivers. > That worked for me. > > On Jun 24, 2004, at 1:20 PM, Tony Nichols wrote: > > Still no go I have asked Digium tech support to look into it. I > > need the later cvs to get around a bug with the latest tdm400 card > > (load driver - unload driver - load driver again to make it work. > > t o n y > > On Thu, 2004-06-24 at 08:15, Tony Nichols wrote: > >> On Wed, 2004-06-23 at 14:32, asterisk wrote: > >>> Have some errors with the above. > >>> > >>> I have tried make and make linux26 > >>> > >>> Anyone got any clues ? I've googled but only got the > "make linux26" > >>> help > >>> > >>> Asterisk compiles and runs great, libpri compiles with no > problems. > >>> > >>> TIA > >>> > >>> Julian. > >>> > >>> pbx:~ # cd /usr/src/zaptel > >>> pbx:/usr/src/zaptel # make linux26 > >>> make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules > >>> make[1]: Entering directory `/usr/src/linux-2.6.4-52' > >>> CHK include/linux/version.h > >>> *** Warning: Overriding SUBDIRS on the command line can cause > >>> *** inconsistencies > >>> make[2]: `arch/i386/kernel/asm-offsets.s' is up to date. > >>> CC [M] /usr/src/zaptel/zaptel.o > >>> /usr/src/zaptel/zaptel.c: In function `zt_net_open': > >>> /usr/src/zaptel/zaptel.c:1166: warning: passing arg 1 of > `hdlc_open' > >>> from > >>> incompatible pointer type > >>> /usr/src/zaptel/zaptel.c: In function `zt_net_stop': > >>> /usr/src/zaptel/zaptel.c:1238: warning: passing arg 1 of > >>> `hdlc_close' from incompatible pointer type > >>> /usr/src/zaptel/zaptel.c: In function `zt_xmit': > >>> /usr/src/zaptel/zaptel.c:1294: error: structure has no > member named > >>> `netdev' > >>> /usr/src/zaptel/zaptel.c:1294: warning: type defaults to `int' in > >> > >> > >> This happened to me too (same dist/kernel) with cvs head > 6/21/2004 - > >> older version 4/24/2004 worked ok. I'm going to try latest > cvs today > >> and see if it works. > >> t o n y > >> > >> ___ > >> 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 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 > > > > > > -Michael > > ___ > 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 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] Which Linux ?
Hi, > >From recent experience: > If you want to use digium hardware dont use suse 9.0. It > seems to think the E1 card is a tigerjet bri card and the > kernel hangs on ztcfg. I have a WT405P running under SuSE 9.0 and it works great. But I had only choosen SuSE because I also need capi... Bye FElix ___ 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] Which Linux ?
> Mike, > > I've been trying to install under SuSE 9.1, but cannot compile zaptel > > What's the secret incantation ?? > > TIA I was helped with: > > I am not able to compile zaptel... > > Could you give me a hint? > Have you tried the following, which is suggested in the output? > 'make cloneconfig && make dep' in /usr/src/linux/ Felix ___ 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] chan_capi problem - hangup???
Hi Tobi, > I installed Asterisk with CAPI support. Everything works fine > while starting Asterisk, but when a call comes in Asterisk > hangsup the call after two times of ringing. > > The output is like: > > Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: > CONNECT_IND ID=002 #0x011d LEN=0048 >Controller/PLCI/NCCI= 0x101 >CIPValue= 0x10 >CalledPartyNumber = **some_number** >CallingPartyNumber = <21 83>**some_number** >CalledPartySubaddress = default >CallingPartySubaddress = default >BC = <80 90 a3> >LLC = default >HLC = <91 81> >AdditionalInfo = default > >== CONNECT_IND > (PLCI=0x101,DID=**some_number**,CID=**some_number**,CIP=0x10,C > ONTROLLER=0x1) > Jun 24 22:19:49 WARNING[1086696368]: pbx.c:1819 ast_pbx_run: > Channel 'CAPI[contr1/**some_number**]/0' sent into invalid > extension 's' in context 'default', but no invalid handler > -- CAPI Hangingup > > activehangingup > -- started pbx on channel (callgroup=0)! > -- INFO_IND ID=002 #0x011e LEN=0023 >Controller/PLCI/NCCI= 0x101 >InfoNumber = 0x70 >InfoElement = **some_number** > > > I read in the mailing list archives of commenting out line > 2615 in chan_capi.c, but that did not change anything. > > Has anybody got an idea what the error: > > "Channel 'CAPI[contr1/**some_number**]/0' sent into invalid > extension 's' in context 'default', but no invalid handler" Do you have DIDs (PTP-ISDN)? Bye Felix ___ 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] chan_capi problem - hangup???
Tobi, > > Do you have DIDs (PTP-ISDN)? > yes then I guess that I have the same problem. If I get a overlaped dial from PSTN, i get only the first did-digit as extension , p.e: my number 8993-12 then it goes to 89931 and that extension does not exist If I get a call from ISDN (or maybe mobile) with block transfer, I get 899312 and it works. For me it seems that chan_capi does not supply inbound overlap-dial. Could anybody clearify that, please? Bye Felix ___ 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] Asterisk Eating Digits
Hello,
do you have overlapdial=yes in your zapata.conf?
Felix
> When I call a PBX system and enter digits, Asterisk is eating
> away some digits. For example when I call AT&T and when the
> system prompts me to enter my phone number, Asterisk eats
> away some digits, so AT&T does not get the number that I
> entered. I am using the extensions.conf as it came from the
> install with some additions. I added longdistance to the
> default context. Please help!
>
>
> [default]
> include => mainmenu
> include => longdistance
>
> exten => _9X.,1,Dial(ZAP/1/${EXTEN:1})
>
>
> Thank you,
> Naren
>
>
>
>
>
> __
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[Asterisk-Users] Chan_Capi Down
Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein ___ 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] RE: Chan_Capi Down
Hi all, are you able to see incoming calls at the isdnlog? I have guessed I have a problem with the capi/isdn/card itsself and not really with asterisk. Felix > Thanks I will give that a try. > > Looks like this may need a bug report? We are all getting the > same errors. > > Outgoing is fine for me. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Andreas Anderson > Sent: 28 June 2004 23:26 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] RE: Chan_Capi Down > > Same here :-( > > asterisk show's this error in the same moment i'm trying to > pick up an incoming call: > > Jun 23 13:14:03 ERROR[-1284076624]: chan_capi.c:881 > capi_write: dont know how to write subclass 64 > > This problem starts with cvs update -D "6/21/04 21:00:00 CET" > > If i revert back to cvs update -D "6/21/04 18:00:00 CET" the > problem is gone. > > -- original message -- > > I am also having the same problem. Latest CVS & Latest Capi > > When it does work and you pick up the phone, CAPI disconnects > the call. > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > ePyron Felix Deierlein > Sent: 28 June 2004 18:34 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Chan_Capi Down > > Hi all, > > * was running ... I have a WT405P and an AVM C4 with > chan_capi 0.3.4a Today chan_capi stopped working, without any > changings at the system. > It seems, that not * is the reason, because isdn-log also > shows no calls. > > If I try to call * from outside via capi, I only get a busy. > > That is the try from inside to outside: > stern01*CLI> > -- data = @89930:0107901723168212 > -- capi request omsn = @89930 > == found capi with omsn = 89930 > == CAPI Call CAPI[contr1/89930]/2 == CAPI Call > CAPI[contr1/89930]/2 > -- CONNECT_CONF ID=003 #0x000d LEN=0014 > Controller/PLCI/NCCI= 0x101 > Info= 0x0 > > == received CONNECT_CONF PLCI = 0x101 INFO = 0 > -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 > Controller/PLCI/NCCI= 0x101 > Reason = 0x3302 > > == DISCONNECT_IND PLCI=0x101 REASON=0x3302 > == Spawn extension (OutDial-Dial, 01723168212, 2) exited > non-zero on 'SIP/ePfd-7515' > -- data = @89930:01079h > -- capi request omsn = @89930 > == found capi with omsn = 89930 > == CAPI Call CAPI[contr1/89930]/3 == CAPI Call > CAPI[contr1/89930]/3 > -- CONNECT_CONF ID=003 #0x000e LEN=0014 > Controller/PLCI/NCCI= 0x101 > Info= 0x0 > > == received CONNECT_CONF PLCI = 0x101 INFO = 0 > -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 > Controller/PLCI/NCCI= 0x > Info= 0x2002 > > -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 > Controller/PLCI/NCCI= 0x101 > Reason = 0x3302 > > == DISCONNECT_IND PLCI=0x101 REASON=0x3302 > == Spawn extension (OutDial-Dial, h, 1) exited non-zero on > 'SIP/ePfd-7515' > > > dmesg shows: > > isdn_dc2minor: di(0) ch(-1072539760) invalid > capidrv-1: now up (2 B channels) > capidrv-1: D2 trace enabled > capi: controller 1 up > kcapi: notify up contr 2 > capidrv: controller 2 up > isdn_dc2minor: di(1) ch(-1072539760) invalid > capidrv-2: now up (2 B channels) > capidrv-2: D2 trace enabled > capi: controller 2 up > kcapi: notify up contr 3 > capidrv: controller 3 up > isdn_dc2minor: di(2) ch(-1072539760) invalid > capidrv-3: now up (2 B channels) > capidrv-3: D2 trace enabled > capi: controller 3 up > kcapi: notify up contr 4 > capidrv: controller 4 up > isdn_dc2minor: di(3) ch(-1072539760) invalid > capidrv-4: now up (2 B channels) > capidrv-4: D2 trace enabled > capi: controller 4 up > > > I hope, that you could help me... > > Thanks > > > Felix Deierlein > > _ > Listen to music online with the Xtra Broadband Channel > http://xtra.co.nz/broadband > > ___ > 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 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 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] CallPres screening DDI
Hello, we had a running configruation where asterisk passed the phone number and the ddi to the pstn (i.e. 595-431) Now only the rootnumber arrives: 5950 I do not know, what to do. I tried to use callingpres (now i am just hiding every number, because 595-0 is no valid extension..) but that did not worked. > Protocol Discriminator: Q.931 (8) len=44> Call Ref: len= 2 (reference 28/0x1C) (Originator)> Message type: SETUP (5)> [04 03 80 90 a3]> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)> Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)> Ext: 1 User information layer 1: A-Law (35)> [18 03 a9 83 81]> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0> ChanSel: Reserved> Ext: 1 Coding: 0 Number Specified Channel Type: 3> Ext: 1 Channel: 1 ]> [6c 08 21 80 35 39 35 34 33 31]> Calling Number (len=10) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)> Presentation: Presentation permitted, user number not screened (0) '595431' ]> [70 11 c1 30 31 30 37 39 30 31 37 32 33 31 36 38 32 31 32]> Called Number (len=19) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0107901723168212' ] -- Called g1/0107901723168212< Protocol Discriminator: Q.931 (8) len=10< Call Ref: len= 2 (reference 32796/0x801C) (Terminator)< Message type: SETUP ACKNOWLEDGE (13)< [18 03 a9 83 81]< Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0< ChanSel: Reserved< Ext: 1 Coding: 0 Number Specified Channel Type: 3< Ext: 1 Channel: 1 ]-- Processing IE 24 (cs0, Channel Identification) With kind regards Felix Deierlein
RE: [Asterisk-Users] avm c4, ptmp
Hi,
could you post your capi.conf..
Regards
Felix
> >I would set the MSN's to 855285 and 859609 > >They do not
> usually include the area code.
> >
>
> [local]
> exten => _9XX.,1,Dial,CAPI/855285:bBYEXTENSION:1
> exten => _9XX.,2,Congestion
> exten => _9XX.,3,Hangup
>
> ;
> ; CAPI config
> ;
> ;
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
>
> [interfaces]
>
> [controller1]
> msn=855285,859609
> incomingmsn=*
> controller=1,2,3,4
> softdtmf=1
> accountcode=
> context=local
> ;echosquelch=1
> ;echocancel=yes
> ;echotail=64
> ;callgroup=1
> ;deflect=12345678
> devices=2
> mode=immediate
> isdnmode=p2mp
> ;
> ;--
>
>
> Aug 3 12:02:28 DEBUG[1145346992]: chan_sip.c:4423
> build_route: build_route: Contact hop:
> -- Executing Dial("SIP/sip1-0167",
> "CAPI/855285:bBYEXTENSION:1") in new stack
> -- data = 855285:b90721950396:1
> -- capi request omsn = 855285
> Aug 3 12:02:28 NOTICE[1224625072]: chan_capi.c:1172
> capi_request: didn't find capi device with outgoing msn =
> 855285. you should check your config!
> Aug 3 12:02:28 NOTICE[1224625072]: app_dial.c:714 dial_exec:
> Unable to create channel of type 'CAPI'
> == Everyone is busy/congested at this time
>
> - --
> Maurizio Marini GSM +39-335-8259739
> Work: +39-0721-855285 Fax +39-0721-859609
> Home: +39-0721-950396 IAXTel: (700) 350-1234
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.0.7 (GNU/Linux)
>
> iD8DBQFBD2W24Q/49nIJTlwRAi0cAJ4/ckdwqJMDbWVYYsMU8wj9zksbugCeJfl5
> lh2CHTrKNg7WOhqfFf/B1Zo=
> =LVNs
> -END PGP SIGNATURE-
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RE: [Asterisk-Users] CallPres screening DDI
Sorry for the HTML-Messages, I have simply forgotten to change it before sending. Hello, we had a running configruation where asterisk passed the phone number and the ddi to the pstn (i.e. 595-431) Now only the rootnumber arrives: 5950 I do not know, what to do. I tried to use callingpres (now i am just hiding every number, because 595-0 is no valid extension..) but that did not worked. > Protocol Discriminator: Q.931 (8) len=44 > Call Ref: len= 2 (reference 28/0x1C) (Originator) > Message type: SETUP (5) > [04 03 80 90 a3] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) > Ext: 1 User information layer 1: A-Law (35) > [18 03 a9 83 81] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 >ChanSel: Reserved > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 1 ] > [6c 08 21 80 35 39 35 34 33 31] > Calling Number (len=10) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) > Presentation: Presentation permitted, user number not screened (0) '595431' ] > [70 11 c1 30 31 30 37 39 30 31 37 32 33 31 36 38 32 31 32] > Called Number (len=19) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0107901723168212' ] -- Called g1/0107901723168212 < Protocol Discriminator: Q.931 (8) len=10 < Call Ref: len= 2 (reference 32796/0x801C) (Terminator) < Message type: SETUP ACKNOWLEDGE (13) < [18 03 a9 83 81] < Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
[Asterisk-Users] opencall.org down?
Hello, it seems that opencall.org is down. Could anybody send me the instructions and sources for fax? (pm: [EMAIL PROTECTED]) Thanks Felix Deierlein ___ 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
AW: [Asterisk-Users] PC based Switchboard application
Hello Pertti, we would be interessted to, if you could send further informations... Thanks Regards Felix Deierlein [EMAIL PROTECTED] -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Pertti Pikkarainen Gesendet: Samstag, 10. April 2004 11:26 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] PC based Switchboard application We have switchboard application ( PC+browser+Java ) with quite a rich feature set. It talks to * via manager port. Works as a call center too. However, it is not open source. If you are interested in, please contact me directly. Best regards Pertti Keith D'Atrio wrote: > Hello All > I am looking for a PC based switchboard application. Cisco > CallManager has a web attendant console that allows you to use the PC > to transfer calls and the like and I was wondering if there was a > similar program compatible with *. > Thank you in advance > Keith D'Atrio ___ 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 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] German sound files available
Hello, first: thanks. The normal prompts are working. But I am still not sure, where I sould place the german digits, letters and phonems. First I placed everything under sounds/de/.. but then digits did not work, then I linked it to /sounds/digits/de/ now I have german digits but saynumber is still english. The question where to place the subdirectories. In the wiki is not a real answer.. Bye Felix > Hi there, > > today I made the German language prompts available for download: > http://www.karl.aegee.org/asterisk.nsf/HT/sound-de > > Be aware: Asterisk doesn't yet fully support languages other > than English, there are still (smaller) issues with voicemail > and date/time announcements that require a patch. > > Cheers, Philipp ___ 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] No outbound calls at a PRI possible
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without alarms and inbound
calls are working fine (from both: Carrier and Siemens).
But I am not able to dial wether outbound nor to the Siemens PBX.
I allways get the message:
== Everyone is busy at this time
After hours of googling and reading and trying I seek help...
Thank you very much.
Felix Deierlein
My extension.conf (only important parts):
[AtInternal]
;exten => 402,1,Macro(stdexten,402,Zap/g2/402)
exten => 402,1,Dial(Zap/g2/595402)
[ePInternal]
include=>system
include=>test
include=>AtInternal
exten => 812,1,Macro(stdexten,812,${ePFfd})
exten => 814,1,Macro(stdexten,814,${ePFjw})
exten => 854,1,Macro(stdexten,854,${ePFch})
exten => 5950,1,Macro(stdexten,812,${ePFfd})
exten => _0.,1,Dial(Zap/g1/${EXTEN:1},60)
[zapata.conf]
[channels]
language=en
context=default
switchtype=euroisdn
;pridialplan=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
;pridialplan=national
switchtype = euroisdn
signalling = pri_cpe
group = 1
channel => 1-15
channel => 17-31
immediate=no
switchtype = euroisdn
signalling = pri_net
group = 2
callgroup=2
pickupgroup=2
channel => 32-46
my zaptel.conf
#amt (carrier)
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
#hicom (siemens)
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
loadzone=uk
defaultzone=uk
channel => 48-62
PRI Debugging Infos:
Call to Carrier: (Destination was 899312)
-- Executing Dial("SIP/ePfd-b455", "Zap/1/899312|60") in new stack
-- Making new call for cr 32774
> Protocol Discriminator: Q.931 (8) len=40
> Call Ref: len= 2 (reference 6/0x6) (Originator)
> Message type: SETUP (5)
> Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
> Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
> Ext: 1 User information layer 1: A-Law (35)
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
>ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel Type:
3
> Ext: 1 Channel: 1 ]
> Display (len= 6) [ 1Felix ]
> Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> Presentation: Presentation permitted, user
number not screened (0) '812' ]
> Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
> Sending Complete (len= 0)
-- Called 1/899312
< Protocol Discriminator: Q.931 (8) len=14
< Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
< Message type: STATUS (125)
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Public network serving the local user (2)
< Ext: 1 Cause: Info. element nonexist or not implemented
(99), class = Protocol Error (6) ]
< Cause data 0: 14 (20)
< Cause data 1: 01 (1)
< Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state:
Call Initiated (1)
-- Processing IE 8 (Cause)
-- Processing IE 20 (Call State)
< Protocol Discriminator: Q.931 (8) len=10
< Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
< Message type: CALL PROCEEDING (2)
< Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
Protocol Discriminator: Q.931 (8) len=9
> Call Ref: len= 2 (reference 6/0x6) (Originator)
> Message type: RELEASE (77)
> Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0)
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
> Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
> Presentation: Unknown (67) '' ]
> Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
What pridialplan should I use with an
E1 with Euroisdn from the German Telekom (DTAG or T-Com).
Thanks
Felix
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> ePyron Felix Deierlein
> Sent: Sunday, May 09, 2004 6:48 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] No outbound calls at a PRI possible
>
> Hello all,
>
> the scenario:
>
> Carrier S2M-- * -S2M--Siemens
> |
> |
> SIP Clients
> and many other features
>
> With much help from the list, the PRI links are without
> alarms and inbound calls are working fine (from both: Carrier
> and Siemens).
>
> But I am not able to dial wether outbound nor to the Siemens PBX.
> I allways get the message:
> == Everyone is busy at this time
>
>
> After hours of googling and reading and trying I seek help...
>
> Thank you very much.
>
> Felix Deierlein
>
>
> My extension.conf (only important parts):
> [AtInternal]
> ;exten => 402,1,Macro(stdexten,402,Zap/g2/402)
> exten => 402,1,Dial(Zap/g2/595402)
>
> [ePInternal]
> include=>system
> include=>test
> include=>AtInternal
>
> exten => 812,1,Macro(stdexten,812,${ePFfd})
> exten => 814,1,Macro(stdexten,814,${ePFjw})
> exten => 854,1,Macro(stdexten,854,${ePFch})
> exten => 5950,1,Macro(stdexten,812,${ePFfd})
> exten => _0.,1,Dial(Zap/g1/${EXTEN:1},60)
>
>
> [zapata.conf]
> [channels]
> language=en
> context=default
> switchtype=euroisdn
> ;pridialplan=national
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> immediate=no
>
> ;pridialplan=national
> switchtype = euroisdn
> signalling = pri_cpe
> group = 1
> channel => 1-15
> channel => 17-31
>
>
> immediate=no
>
> switchtype = euroisdn
> signalling = pri_net
> group = 2
> callgroup=2
> pickupgroup=2
> channel => 32-46
>
> my zaptel.conf
> #amt (carrier)
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-15
> dchan=16
> bchan=17-31
> #hicom (siemens)
> span=2,1,0,ccs,hdb3,crc4
> bchan=32-46
> dchan=47
> bchan=48-62
> loadzone=uk
> defaultzone=uk
> channel => 48-62
>
>
> PRI Debugging Infos:
> Call to Carrier: (Destination was 899312)
> -- Executing Dial("SIP/ePfd-b455", "Zap/1/899312|60") in new stack
> -- Making new call for cr 32774
> > Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 2
> (reference
> > 6/0x6) (Originator) Message type: SETUP (5) Bearer
> Capability (len= 3)
> > [ Ext: 1 Q.931 Std: 0 Info transfer
> capability: Speech (0)
> > Ext: 1 Trans mode/rate: 64kbps,
> > circuit-mode
> (16)
> > Ext: 1 User information layer
> 1: A-Law
> > (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
> > Exclusive
> Dchan: 0
> >ChanSel: Reserved
> > Ext: 1 Coding: 0 Number Specified
> Channel Type:
> 3
> > Ext: 1 Channel: 1 ] Display (len= 6)
> [ 1Felix ]
> > Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> > Presentation: Presentation permitted, user
> number not screened (0) '812' ]
> > Called Number (len= 9) [ Ext: 1 TON: National Number (2) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
> > Sending Complete (len= 0)
> -- Called 1/899312
> < Protocol Discriminator: Q.931 (8) len=14 < Call Ref: len=
> 2 (reference 32774/0x8006) (Terminator) < Message type: STATUS (125)
> < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0)
> 0: 0 Location:
> Public network serving the local user (2)
> < Ext: 1 Cause: Info. element nonexist or
> not impl
RE: [Asterisk-Users] No outbound calls at a PRI possible
Hello Darren,
> The "error" messages that you reported in your last e-mail
> are actually outbound Q.931 call setup messages that are
> being sent to DTAG from your Asterisk machine. The direction
> of the message is indicated in the first column of the trace
> output in the form of > or <. Although these are not error
> messages I am surprised to see those particular messages
> being generated with your current zapata.conf settings; with
> pridialplan=local I would have expected something similar to
> the following messages during call
> setup:
>
> > Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> > Presentation: Presentation permitted, user
> number not screened (0) 'X58777' ]
> > Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'X986600' ]
>
> (I have inserted X in the PSTN numbers above to protect
> the innocent Calling and Called parties.)
>
> Please retry pridialplan=local and pridialplan=unknown in
> zapata.conf and post the trace results so we compare results.
> With pridialplan=local in zapata.conf the outbound call setup
> from Asterisk to DTAG should look ideal.
I will try again in the late evening (the pri is in production use in
another Detewe...)
> On a different subject, how are your results with telephony
> calls from the Asterisk machine to your Hicom PBX? I would
> have expected the zaptel.conf entry to have been:
>
> > #hicom (siemens)
> > span=2,0,0,ccs,hdb3,crc4
>
> ...so that your Asterisk provides clocking/timing information
> for the Hicom.
> If this configuration is not set correctly you could find
> that the systems seem to communicate well at first but after
> a while you might see strange PRI errors (every hour or so)
> that relate to clock synchronisation problems.
The Hicom has been switched to secondary clocking... We had some problems
with the cables, so we tried everything possible..
I guess we will change it back later on, so that we could use the Hicom
without * if asterisk stops (could that be?:)
But there is also another problem, if I try to dial out via Hicom to DTAG,
the Hicom sends digit after digit.
My dial line is:
exten => _0.,1,Dial(Zap/g1/${EXTEN:1},60)
and that works fine with SIP and IAX. But with the Hicom I get only the
first two digits and then it trys to dial out: error.
Does I have to use schemes like exten => _0XXX
But I guess that the german numbers have differnt lengths.
Thanke you.
Felix
> Hi Felix,
>
> on some UK public switches I have seen similar bad call setup
> problems with a release cause of 28 (Invalid number format)
> when using:
>
> pridialplan=national
>
> Have you tried:
>
> pridialplan=unknown
>
> in zapata.conf?
>
> It seems as though the omission of the pridialplan= statement
> in zapata.conf is treated by Asterisk as pridialplan=national.
>
> We could probably give you more relevant suggestions if you
> would enable a more verbose level of output and post the call
> setup trace results here. Try the following command from the
> Asterisk CLI before making your next call:
>
> pri debug span x
>
> Where x = single integer digit for the PRI span that will be
> used to make the outgoing call. (Eg. 1)
>
> Please drop a note to the list (either way) with your results.
>
> HTH
>
> Darren
> --
> Comgate
> Telco>Internet
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of
> ePyron Felix Deierlein
> Sent: 09 May 2004 20:32
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] No outbound calls at a PRI possible
>
>
> Hello,
>
> i guess the problem ist pridialplan from zapata.conf
>
> with
>
> pridialplan = local
>
> it works :-). But I still get the error messages:
>
> > Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
> Unknown Number Plan (0)
> > Presentation: Unknown (67) '' ] Called
> > Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '899312' ]
>
> What pridialplan should I use with an
> E1 with Euroisdn from the German Telekom (DTAG or T-Com).
>
>
> Thanks
>
>
> Felix
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of ePyron
> > Felix Deierlein
> > Sent: Sunday, May 09, 2004 6:48 PM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] No out
[Asterisk-Users] Quality differences of codecs from PRI to SIP
Hello all, I have googled a bit, but was not able to a definite answer (maybe there is not one..) The question is, how different would be the voice qualitiy, if you let translate * from alaw (PRI) to gsm instead of using alaw as codec for sip. And also how would echo and the processor load be affected? The point is, I really would like to use IAX Phone, but is has no alaw codec... (it seems that there is not any win iax client with alaw/mylaw)... I hope you have some ideas and hits Thanks Bye Felix Deierlein ___ 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] * & ISDN-BRI-PTP & DID & ISDN4Linux does not show incoming number
Hi Andreas, I guess it is better to buy a B1 or C2 :-). They are not very expensive at ebay. Or you buy digium hardware, it surely runs with *... Or have a look at www.junghanns.net (author of chan_capi) He sells a 4 Port BRI ... Bye Felix > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Andreas Frackowiak > Sent: Friday, May 07, 2004 11:05 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] * & ISDN-BRI-PTP & DID & > ISDN4Linux does not show incoming number > > Hallo Felix, > > > it seems that the FAQ only describes windows & co. > > Just try to use the capi driver, I guess you would get much more > > support for capi here... > > Well now I am sure: The AVM-Fritz-CAPI does not work with PTP. > > o I have tried it and it doesn't work > o I asked AVM and they answered that the Fritz > CAPI-Software (Windows + Linux) does not support > DDI/PTP-Mode. > o I found a lot of messages in old archives of this list > and the i4l-list which also say that PTP with > Fritz CAPI does not work. > > Also "mISDN" (ISDN4Linux successor with CAPI20) maybe will > support P2P with Fritz Card sometime, but not today. > > And so it seems that my problem between ISDN4Linux and the > chan_modem_i4l driver remains an unsolved mystery. > > So maybe I have to buy an AVM B1 or C2 card to circumvent > this problem or use something else than asterisk. > > thanks and regards > Andreas > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf > Of Andreas > > > Frackowiak > > > Sent: Wednesday, May 05, 2004 8:08 PM > > > To: [EMAIL PROTECTED] > > > Subject: Re: [Asterisk-Users] * & ISDN-BRI-PTP & DID & ISDN4Linux > > > does not show incoming number > > > > > > Hi Felix, > > > > > > > > I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux > > > (and a Fritz > > > > > Card PnP). > > > > > The ISDN-BRI is in PTP-Mode (Point to Point "german: > > > > > Anlagenanschluss") which is enabled within I4L with > "hisaxctrl > > > > > fcpcipnp0 7 1". > > > > are you shure, that the capi does not support PTP? > > > > I have an AVM C4 card, but it should be the same with > the fritz.. > > > > > > Well, I am not sure, but AVM says in: > > > http://www.avm.de/de/Service/FAQs/FAQ_Sammlung/2671.php3 > > > that only the B1-family of cards and the C2 and C4 Controllers > > > support PTP. > > > > > > I would be very happy if someone has a Fritz with CAPI > working with > > > a PTP und could proove that I am wrong. > > > > > > I also would be very happy if someone could help me with the > > > original question, why I4L does not give the called > number / MSN to > > > Asterisk (and help me fix it, of course :) > > > > > > Thanks > > > Andreas > > > > > > ___ > > > 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 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 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 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] indications.conf
Hello Vit, just try the indications from the UK. That worked fine in Germany. Bye Felix > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Dudlik > Sent: Monday, May 17, 2004 9:20 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] indications.conf > > Hello > > I am looking for Czech (Czech Republic) country support to > indications.conf Have you ever seen it anywhere ? > We are a small country in middle Europe :) > > > thank you > > -- > Vit Bohacek > ___ > 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 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] CallCenter setup
Hi, > > > enough get redirected to human consultant. There should be > > > possibility for supervisors to connect to ongoing conversation. > > > Expected traffic will not exceed 30 concurrent calls. > > Look at "ZapBarge" for the listening-in. As usual the Wiki is > your friend. Also I assume you'll want to look at this: > > http://www.voip-info.org/wiki-Astguiclient > > By the way: If you can do give Asterisk a life of its own > with an E1 ISDN card and do not put it behind the Siemens > HiPath, that'll make things easier. That would permit you to > avoid the rather evil H.323 protocol ... > > > > Now my problem is which interface to choose? Will voip be > good enough? > > > Wont it introduce to much latency? Or should I insist on > buying ISDN > > > interface for asterisk box? What hardware would You recommend for > > > this setup? > > Before ordering any equipment you should first of all test > the H.323 setup/connectivity between HiPath and Asterisk. You also could place * between PSTN and the HIPATH. Have a look at the Wildcard TE4xxP from Digium. Regards Felix ___ 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] E1 Connection breaks
Hello, we have a connection to a leagacy pbx (Siemens Hicom 150E) via PRI (E1). Everything works really fine, but the connection breaks sometimes (there is not really a time scheme), so that you could not dial from the hicom to * or from * to hicom. The only one to get the connection back is to reboot *. Today is our first production day, so it is rellay bad. The connection to the PSTN (E1 to DTAG) works without any problems. * runs at a SuSE 9.0 with 2x PIII 1000MHz and 512 MB RAM. Hardware: 1 WCT405p 1 AVM C4 (with chan_capi) * Version: CVS-05/29/04-17:36:55 stable Zaptel.conf #amt span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 #hicom span=2,2,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 loadzone=uk defaultzone=uk zapata.conf [channels] language=de ;context=default switchtype=euroisdn ;pridialplan=national usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no overlapdial=yes pridialplan = local context = Amt595xxx-In switchtype = euroisdn signalling = pri_cpe group = 1 channel => 1-15 channel => 17-31 context = Hicom-In switchtype = euroisdn signalling = pri_net group = 2 callgroup=2 pickupgroup=2 channel => 32-46 channel => 48-62 I have created a logfile (via putty) with pri intense debug. I have attached it zipped. I hope that is okay? -> It seems not: so I have uploadet it to: http://ePyron.de/log.zip I would be very happy with any help you could provide. Thanks Felix Deierlein ___ 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] New Firefly version
Hello Adam, Hi Adam, two features I would really like to have: - the textbox from "Dial a URL" in the normal client (maybe optionally) so that you could easily copy and paste numbers in - a function that replaces +49 or wathever to 00. maybe it would be also possible, to recognize that +49 (333) is not a local number, so that another 0 should be added (or a 9). Regards Felix > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart > Sent: Monday, May 31, 2004 3:01 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] New Firefly version > > As Promised, I've released a new version of Firefly (ver 1.8) > with IAX & SIP support back in. > > Get it from Virbiage site or here's the direct link > http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > > If it crashes on startup, export your Firefly tree from the > registry (current user -> software -> firefly), then delete > tree from your registry. If that fixes it, send me your > exported reg file, there's a bug left to do with some wierd > reg entry but everyone just deletes it instead of sending it to me :| > > Transfers will be in the next version - email me any > comments, requested features, bugs and I'll see what I can do > > -Adam > ___ > 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 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] E1 Connection breaks
Hello Jason, > >Everything works really fine, but the connection breaks sometimes > >(there is not really a time scheme), so that you could not dial from > >the hicom to * or from * to hicom. > I see from your config file you are using the hicom as the > second timing source make sure the hicom is not clocking off > of this line > Jason I have allready tried it with 0 and with 1. Normally the Hicom should give the timing, but it does not matter. It works for hours or only for minutes and then it crashes. I cannot close * and have to reboot the machine. Felix ___ 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] Fax Recognizion without Answer? How to Supress this?
Hello,
we have a PRI (E1) to a carrier and a second one to a legacy PBX:
DTAG ---pri * -- Hicmo
(PSTN) |
|
Sip
and
more
Many normal inbound calls are direcly routed to the hicom.
Outbound calls from the Hicom go through LCR and then to PSTN.
Inbound faxes are working, but outbound faxes from hicom to pstn are
recognized as faxes and * tries to forward the call to "fax". I do not
answer this calls...
== Spawn extension (Amt595xxx-In, 595164, 1) exited non-zero on 'Zap/14-1'
-- Hungup 'Zap/14-1'
-- Starting simple switch on 'Zap/62-1'
-- Accepting overlap call from '595457' to '034491' on channel 31,
span 2
-- Executing SetVar("Zap/62-1", "Out=Zap/g1/") in new stack
-- Executing Goto("Zap/62-1", "OutDial-LCR|BYEXTENSION|1") in new stack
-- Goto (OutDial-LCR,034491***,1)
-- Executing SetVar("Zap/62-1", "LCR=01081") in new stack
-- Executing Goto("Zap/62-1", "OutDial-Dial|BYEXTENSION|1") in new stack
-- Goto (OutDial-Dial,034491,1)
-- Executing Dial("Zap/62-1", "Zap/g1/0108103|30|TrH") in new stack
-- Called g1/010810344918***
-- Redirecting Zap/62-1 to fax extension
-- Hungup 'Zap/1-1'
== Spawn extension (OutDial-Dial, fax, 0) exited non-zero on 'Zap/62-1'
-- Executing Dial("Zap/62-1", "Zap/g1/01081fax|30|TrH") in new stack
-- Called g1/01081fax
-- Channel 2, span 1 got hangup
-- Hungup 'Zap/2-1'
What have I to change? Could I supress that?
Thanks
Felix Deierlein
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RE: [Asterisk-Users] Fax Recognizion without Answer? How to Supress this?
Hi,
I have really googled and read the wiki but I still no idea, how to supress
the fax recognizion.
Our users are not able to fax and that is bad... Could you give me an hint,
please?
Thanks
Felix
>
> Hello,
>
> we have a PRI (E1) to a carrier and a second one to a legacy PBX:
>
> DTAG ---pri * -- Hicmo
> (PSTN) |
> |
>Sip
>and
>more
>
> Many normal inbound calls are direcly routed to the hicom.
> Outbound calls from the Hicom go through LCR and then to PSTN.
>
> Inbound faxes are working, but outbound faxes from hicom to
> pstn are recognized as faxes and * tries to forward the call
> to "fax". I do not answer this calls...
>
>
> == Spawn extension (Amt595xxx-In, 595164, 1) exited
> non-zero on 'Zap/14-1'
> -- Hungup 'Zap/14-1'
> -- Starting simple switch on 'Zap/62-1'
> -- Accepting overlap call from '595457' to '034491'
> on channel 31, span 2
> -- Executing SetVar("Zap/62-1", "Out=Zap/g1/") in new stack
> -- Executing Goto("Zap/62-1",
> "OutDial-LCR|BYEXTENSION|1") in new stack
> -- Goto (OutDial-LCR,034491***,1)
> -- Executing SetVar("Zap/62-1", "LCR=01081") in new stack
> -- Executing Goto("Zap/62-1",
> "OutDial-Dial|BYEXTENSION|1") in new stack
> -- Goto (OutDial-Dial,034491,1)
> -- Executing Dial("Zap/62-1",
> "Zap/g1/0108103|30|TrH") in new stack
> -- Called g1/010810344918***
> -- Redirecting Zap/62-1 to fax extension
> -- Hungup 'Zap/1-1'
> == Spawn extension (OutDial-Dial, fax, 0) exited non-zero
> on 'Zap/62-1'
> -- Executing Dial("Zap/62-1", "Zap/g1/01081fax|30|TrH")
> in new stack
> -- Called g1/01081fax
> -- Channel 2, span 1 got hangup
> -- Hungup 'Zap/2-1'
>
> What have I to change? Could I supress that?
>
> Thanks
>
> Felix Deierlein
>
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RE: [Asterisk-Users] chan_capi and DDI (Anlagenanschluss)
Hello Holger, I guess that you must configure your /etc/capi.conf options = p2p.. Bye Felix > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Holger Schurig > Sent: Monday, June 07, 2004 5:04 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] chan_capi and DDI (Anlagenanschluss) > > > I remember that it's not possible to have an AVM Fritz card > on an PTP > > mode ISDN line. I think cards with HFC chipset are able to > do so. Of > > cause you could also use an active card with CAPI driver ;-) > > I read something like this in the mailing list archive, but > they were referring to isdn4linux, so I thought they where > using ISDN via chan_modem_i4l. > > > I already ordered an HFC-S based ISDN card > two-and-a-half-week ago, but this card has not yet been arrived. > > ___ > 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 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] Integration with a Siemens HiCom 150E / HiPath 3750
Hello Martin, how would you like to integrate? PRI (E1) or BRI (ISDN)? We have a running integration with PRI and a Hicom 150.. If you have any questions... Bye Felix > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Martin Mielke > Sent: Tuesday, June 08, 2004 4:05 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Integration with a Siemens HiCom > 150E / HiPath 3750 > > Hi * :-) > > I found in the online WiKi docs some information on how to > integrate Asterisk with "old PBX"... > > http://www.voip-info.org/wiki-Asterisk+legacy+integration > > ...but I couldn't find anything on integration with a Siemens > HiCom 150E. Later on we'll migrate to a HiPath 3750 so > information covering this model would be nice too... > > Do you know if any of the PBX listed on the link above are > similar somehow to the Siemens I mention in terms of > integration with Asterisk? > > Answers much appreciated. > > > Martin > > ___ > 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 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] Integration with a Siemens HiCom 150E / HiPath 3750
Hello Martin, > >how would you like to integrate? PRI (E1) or BRI (ISDN)? > Besides of making calls with VoIP from PC to PC, we'd like > that our people abroad could dial company internal extensions > through Asterisk using a SIP client. On a second approach, > the same people abroad could dial the PSTN using the same method... That should not affect your integration with the legacy pbx. Our scenario is: DTAG -- * HICOM PRI | PRI | SIP > Please tell me the magical receipt on a step-by-step basis, > as I'm not much into this telco world ;) Sorry, that is not that easy because the receipt depends much on the circumstances. What connection do you have between pstn and hicom? And you should read everything about the leagacy integration, so you will get an idea, what you want to have. Bye Felix ___ 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] Fax detected, but no fax extension
Hi Patrick, could you please give us a feedback if that have worked? Because I have hacked the source to disable fax.. Thanks Felix > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Nicolas Gudino > Sent: Wednesday, June 09, 2004 8:48 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Fax detected, but no fax extension > > Hi Patrick > > Patrick J. Conroy wrote: > > > Hello all, > > > > I have a fax machine attached to one of the FXS ports on my channel > > bank running into one of the spans of my TE405P. Every > time I try to > > send a fax, I get the error "Fax detected, but no fax > extension" in asterisk. > > Does anyone know why this would happen? The only other reference I > > have found that relates to this in the list said to enable > > OLD_DSP_ROUTINES and rebuild and reinstall asterisk. I have done > > that, but there is no change. > > If you used CVS-HEAD there is a new "faxdetect" parameter for > zapata.conf . I have not tried, but it might solve your problem. > > ;faxdetect=both > ;faxdetect=incoming > ;faxdetect=outgoing > ;faxdetect=no > > > -- > Nicolas Gudino > House Internet S.R.L. > Buenos Aires - Argentina > ___ > 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 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] New version of DIAX (0.9.8a) available nowfor free download
Hi Dan, could you support alaw/mlaw? Is that a big problem? Regards Felix ___ 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] Integration with SIEMENS HIPATH PBX
Hi, you can integrate it via PRI or BRI. Regards Felix From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Friday, June 11, 2004 7:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Integration with SIEMENS HIPATH PBX Hi, I would like to know if Asterisk is able to be integrated with a Siemens HIPATH PBX by VoIP or other ways. Best regards, Ronaldo S. Pereira PRI Telemática. ___ 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] TE410P in Germany
Hello, we have a TE405P running at DTAG. Zapata.conf: stern01:/etc/asterisk # cat zapata.conf [channels] faxdetect=no language=de usecallerid=yes hidecallerid=no restrictid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no usecallingpres=yes overlapdial=yes pridialplan = local context = Amt595xxx-In switchtype = euroisdn signalling = pri_cpe group = 1 channel => 1-15 channel => 17-31 Regards Felix Deierlein From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henrik Pfluger Sent: Tuesday, September 07, 2004 6:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] TE410P in Germany Is there anyone successfully using the TE410P with a German "PMX-Anschluss"? Please just drop me a note mentioning the carrier you use. We are having problems making the card work, although configuration is correct (Posted this before). Our carrier blames the card for this. We would just need some evidence that it really works. Thanks, Henrik ___ 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
