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
[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
RE: [Asterisk-Users] avm c4, ptmp
Hi, could you post your capi.conf.. Regards Felix I would set the MSN's to 855285 and 859609They 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: sip:[EMAIL PROTECTED]:5060 -- 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- ___ 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] 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 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 ___ 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] 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] 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] 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 snip 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 = c1**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 = c1**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] PRI immediate=no
Hi Thomas, I have got the following problem (E100P, pri_cpe): My number range is 6digitsxyz. (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] 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] 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 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] 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
[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 ___ 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 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 ___ 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 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
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
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] 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
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
TelcoInternetBroadcast
-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 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
[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
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type:
3
Ext: 1 Channel: 1 ]
-- Processing IE 24 (Channel Identification)
Protocol Discriminator: Q.931 (8) len=13
Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
Message type: DISCONNECT (69)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Public network serving the local user (2)
Ext: 1 Cause: Invalid number format (28), class = Normal
Event (1) ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0:
0 Location: Public network serving the local user (2)
Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
-- Processing IE 8 (Cause)
-- Processing IE 30 (Progress Indicator)
-- Channel 1, span 1 got hangup
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request
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) 0: 0 Location:
Private network serving the local user
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 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
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified
Channel Type:
3
Ext: 1 Channel: 1 ]
-- Processing IE 24 (Channel Identification) Protocol
Discriminator: Q.931 (8) len=13 Call Ref: len= 2
(reference
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
[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
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
[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
