[Asterisk-Users] chan_mISDN test release....
Hi Asterisk knights, we are proud to announce that our brand new chan_mISDN channel driver for Asterisk is now official released for testing. You can download it under http://www.beronet.com/index.php?PageID=3017. Please test it ample, and post bugs or feature requests to www.beronet.com/bugs. Have a lot of fun! Best Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow (bei Berlin) FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ 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] Testers for chan_misdn searched
Hello everybody, we've implemented a new Channel Driver for *. It uses the new mISDN isdn4linux architecture and supports bri te and nt mode for now. I assume, there are lots of bugs we didn't found yet, and even mISDN is rarely stable. So we search brave volunteers to test the driver. Get it at: http://www.beronet.com/asterisk/ There you can find also a new asterisk-Application which works like saynumber and saydigit but supports multiple languages. When you find bugs, please send email to [EMAIL PROTECTED] Thanks a lot, CRrichi ___ 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] AGI script will not be terminated
Hi, yes. I answer the call first: exten = _.,1,Answer exten = _.,2,SetVar(FPIN=0) exten = _.,3,AGI(FCall.agi) exten = _.,4,Hangup -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Areski Gesendet: Dienstag, 16. März 2004 16:27 An: Asterisk-Users Mailing-list Betreff: Re: [Asterisk-Users] AGI script will not be terminated Hi Thomas, Do you answer correctly in your dial plan ? I met the same problem but unfortunately I don t remember what I was doing wrong :/ exten = 1112,1,Answer exten = 1112,2,Wait(1) exten = 1112,3,agi,myscript On Tue, 2004-03-16 at 15:58, Thomas Haeger wrote: Hi all, if i answer a call on my astbox and go into an AGI script... then there is somthing happens.(play music or something like that)...and the person who called to the box hangs up the script will never be terminated. The process hangs around self the asterisk process is terminated. Is this a bug, or is there something i'am doing wrong ? Thanks for help, Thomas. ___ 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
AW: [Asterisk-Users] who has German voice files ?
Wait a week and you can have german files from one of our customers, who wants to donate such files. Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jakob Strebel Gesendet: Donnerstag, 11. Marz 2004 14:31 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] who has German voice files ? Hi, I like that my * talks German to the callers. Google does not give me any reference about the availability of german announcement files. Could somebody on this list help me out and make it available to me. Thanks, best regards Jakob ___ 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] callerid will not be set
Hi all, i have a TDM20B in my astbox and i have configured my channels as follows: usecallerid=yes signalling=fxo_ks context=tel1 group=5 callerid=101 channel = 13 callerid=102 channel = 14 But if i make a connection to the manager interface the callerid in the events is not set: Event: Newchannel Channel: Zap/13-1 State: Rsrvd Callerid: unknown Uniqueid: 1077819120.3 Event: Hangup Channel: Zap/13-1 Uniqueid: 1077819120.3 Is this a bug ? Or may i do somthing wrong ? Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow (bei Berlin) FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ 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] nasty segfault with previous strange Zaptel warnings
Hi all,
i try to get help from you guys.
I have wrote a little call generation script that generate calls every few
minutes on a PRI Line which is connected through as E400P...
#! /usr/bin/php4
?
$numbers=file(/home/thaeger/callscript/numbers.list);
while (true) {
foreach ($numbers as $number) {
if (substr($number,0,1) != #) {
// echo ($number);
$fh = fopen(/var/spool/asterisk/outgoing/call.$number,w);
fputs ($fh,Channel: Zap/g2/01033$number\nMaxRetries:
0\nWaitTime: 5\nContext: callgen\nExtension: 1\nPr
fclose($fh);
sleep(5);
}
}
sleep(120);
}
?
On the other side there comes everytime a call rejection, and it should be
so.
The normal log output must look like this:
-- Attempting call on Zap/g2/0103302289319048 for [EMAIL PROTECTED]:1 (Retry 1)
-- Hungup 'Zap/32-1'
Feb 17 13:04:26 NOTICE[376852]: pbx_spool.c:199 attempt_thread: Call failed
to go through, reason 3
But then sometimes there comes strange warnigs:
-- Attempting call on Zap/g2/01033028712747781 for [EMAIL PROTECTED]:1 (Retry 1)
-- Moving call from channel 1 to channel 19
Feb 17 13:04:43 WARNING[442388]: chan_zap.c:595 zt_get_index: Unable to get
index, and nullok is not asserted
Feb 17 13:04:43 WARNING[442388]: chan_zap.c:3425 zt_read: We dont exist?
-- Hungup 'Zap/32-1'
Or like this:
Feb 17 12:52:00 WARNING[589843]: chan_zap.c:3992 zt_new: Channel 32 already
has a Real call
Feb 17 12:52:32 WARNING[688147]: chan_zap.c:595 zt_get_index: Unable to get
index, and nullok is not asserted
Feb 17 12:52:32 WARNING[688147]: chan_zap.c:3425 zt_read: We dont exist?
And after a few minutes, so between 20 and 60, there comes a nasty SEGFAULT
with these messages:
/usr/sbin/safe_asterisk: line 6: 18118 Segmentation fault asterisk
${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
Automatically restarting Asterisk.
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
Automatically restarting Asterisk.
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
Automatically restarting Asterisk.
Disconnected from Asterisk server
Executing last minute cleanups
Have somebody an idea what the problem is ?
The Zap Hardware which is running on my machine is a E400P.
Thanks for help.
Best regards,
Thomas.
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AW: [Asterisk-Users] Need to interface to BRIs
Look at this: http://www.junghanns.net/asterisk/page17.html Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jim Archer Gesendet: Montag, 16. Februar 2004 10:11 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Need to interface to BRIs Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Data call transfer
Hi Tomica,
i had the same problem and here is the solution from Maik Schmitt:
exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
exten = _X.,50,Dial(Zap/g3d/${EXTEN})
exten = _X.,100,Dial(Zap/g3/${EXTEN})
But maybe the dataendpoint would never be reached, and so can try out this:
go to bugs.digium.com and look at bug number 960 at libpri project
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Tomica Crnek
Gesendet: Donnerstag, 5. Februar 2004 10:05
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] Data call transfer
Hi everyone
I have TE410P with one E1 link connected to telecom PSTN, and another E1 to
my internal legacy PBX. On this PBX I have one extension where my RAS server
for both ISDN and analogue calls is located.
Can anyone tell me what has to be done to transfer voice call from one E1 to
another as voice, and if Asterisk detects that the call is a data call to
transfer it further as data?
Tomica
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[Asterisk-Users] WG: Reference projects which using Asterisk !?
Hi all, everyone who deals with Asterisk will know the question from his customers, or folks who want to become a customer ;-) Where is Asterisk in use ? In which larger projects is asterisk in use ? I think it would be nice to collect such hints of such projects, and publish it on the Asterisk website or the asterisk wiki. This would bring trust to the disbelieving folks What do you think ? Or, is there someone who can report something about such projects ? Maybe it is something for the documentation project Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ 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] Data calls (ISDN/64k) through * PRI
i all, is it possible to switch data calls through asterisk with the Dial application? The scenario is as following: PSTN (ISDN 64k) -- Asterisk/PRI(TE410P) --- (same) Asterisk/PRI --- PSTN (ISDN 64k) I tried this with normal Dail, but if you come with ISDN/64k, the outgoing call is an audio call. Any ideas ? Thanks, Thomas. ___ 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] Data calls (ISDN/64k) through * PRI
Thanks Maik,
i try it
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Maik
Schmitt
Gesendet: Donnerstag, 22. Januar 2004 11:21
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Any ideas ?
exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
exten = _X.,50,Dial(Zap/g3d/${EXTEN})
exten = _X.,100,Dial(Zap/g3/${EXTEN})
--
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP
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AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Hi Maik,
is there any special version from libpri or asterisk necessary since it
works ?
I'am runnig version: CVS-11/11/03-11:49:55 and it don't work :-(
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Maik
Schmitt
Gesendet: Donnerstag, 22. Januar 2004 12:45
An: [EMAIL PROTECTED]
Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Any chance one such distinction can be made on incoming calls as well
i.e. branch incoming calls on a single DID depending on whether they are
data or speech?
That's what the first line does.
exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
${CALLTYPE} can be SPEECH, DIGITAL, RESTRICTED_DIGITAL, 31KAUDIO,
7KAUDIO or VIDEO.
--
Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP
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AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data Here my pri debug ouput Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: SETUP (5) Bearer Capability (len= 2) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3328334778' ] Called Number (len=11) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '63494441' ] -- Making new call for cr 5635 -- Processing Q.931 Call Setup -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 108 (Calling Party Number) -- Processing IE 112 (Called Party Number) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 38403/0x9603) (Terminator) Message type: SETUP ACKNOWLEDGE (13) 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: 30 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Accepting call from '3328334778' to '63494441' on channel 30, span 2 -- Executing GotoIf(Zap/61-1, 0?50:100) in new stack -- Goto (pri2,63494441,100) -- Executing Dial(Zap/61-1, Zap/g2/033283077733SPEECH) in new stack -- Making new call for cr 39439 Protocol Discriminator: Q.931 (8) len=50 Call Ref: len= 2 (reference 6671/0x1A0F) (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, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] 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 passed network screening (1) '3328334778' ] Called Number (len=21) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '033283077733SPEECH' ] -- Called g2/033283077733SPEECH Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 39439/0x9A0F) (Terminator) Message type: SETUP ACKNOWLEDGE (13) 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) beroasterisk*CLI Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (Cause) -- Channel 30, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap sending, peerstate Overlap Receiving Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6671/0x1A0F) (Originator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/32-1' == Spawn extension (pri2, 63494441, 100) exited non-zero on 'Zap/61-1' Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 39439/0x9A0F) (Terminator) Message type: RELEASE (77) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release
AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Hi, we tried following scenario: DTAG (S0) at our office Datacall with AVMFritz (PSTN) --- Colo TelesSwitch -- CoLo Asterisk (--- PSTN) I think, no i know that the Teles Switch can route 64k data calls here is the Teles Trace: #08SETUP--| 15:29:40,378 02 01 78 AE | 08 02 03 90 05| Bearer Caps 04 02 88 90 | Channel Id 18 03 A1 83 9B| Calling PN 6C 0C 21 83 33 33 32 38 | 33 33 34 37 37 38 | Called PN70 09 C1 36 33 34 39 34 | 34 34 31 | |--RR #08 | 15:29:40,388 02 01 01 7A |--SETUP ACKNOWLEDGE #08 | 15:29:40,398 00 01 AE 7A |08 02 83 90 0D | Channel Id 18 03 A9 83 9B |--SETUP #12 | 15:29:40,408 00 01 2A D4 |08 02 16 60 05 | Bearer Caps 04 02 88 90 | Channel Id 18 03 A1 83 88 | Calling PN 6C 0C 21 80 33 33 32 38 |33 33 34 37 37 38 | Called PN70 09 81 36 33 34 39 34 |34 34 31 #08 RR--| 15:29:40,408 00 01 01 B0 | #12 RR--| 15:29:40,418 00 01 01 2C | #12SETUP ACKNOWLEDGE--| 15:29:40,418 02 01 D4 2C | 08 02 96 60 0D| Channel Id 18 03 A9 83 88| Progress Ind 1E 02 81 82 | |--RR #12 | 15:29:40,418 02 01 01 D6 #12SETUP--| 15:29:40,428 02 01 D6 2C | 08 02 1A 21 05| Bearer Caps 04 03 88 90 A3| Channel Id 18 03 A1 83 81| Calling PN 6C 0C 41 81 33 33 32 38 | 33 33 34 37 37 38 | Called PN70 0D C1 30 33 33 32 38 | 33 30 37 37 37 33 33 | |--RELEASE COMPLETE #12 | 15:29:40,428 00 01 2C D8 |08 02 9A 21 5A |08 02 80 D8 |[Incompatible destinat |ion] #12 RR--| -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von CW_ASN - Gus Gesendet: Donnerstag, 22. Januar 2004 17:24 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI The incoming call request Unrestricted and 64K, and this looks like ok, but in the SETUP_ACK the called number parameters shows: Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN equipment. In the most of cases, Information transfer rate = to '64 kbit/s', and Info transfer capability = 'real bw required'. Are you sure that the equipment attached to * can be used in 64K? Regards, Gus - Original Message - From: Thomas Haeger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 12:28 PM Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data Here my pri debug ouput Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: SETUP (5) Bearer Capability (len= 2) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3328334778' ] Called Number (len=11) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '63494441' ] -- Making
AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI
Has somebody got it work at all ? I mean data calls (ISDN 64k) through asterisk. Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Donnerstag, 22. Januar 2004 19:07 An: [EMAIL PROTECTED] Betreff: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Hi, we tried following scenario: DTAG (S0) at our office Datacall with AVMFritz (PSTN) --- Colo TelesSwitch -- CoLo Asterisk (--- PSTN) I think, no i know that the Teles Switch can route 64k data calls here is the Teles Trace: #08SETUP--| 15:29:40,378 02 01 78 AE | 08 02 03 90 05| Bearer Caps 04 02 88 90 | Channel Id 18 03 A1 83 9B| Calling PN 6C 0C 21 83 33 33 32 38 | 33 33 34 37 37 38 | Called PN70 09 C1 36 33 34 39 34 | 34 34 31 | |--RR #08 | 15:29:40,388 02 01 01 7A |--SETUP ACKNOWLEDGE #08 | 15:29:40,398 00 01 AE 7A |08 02 83 90 0D | Channel Id 18 03 A9 83 9B |--SETUP #12 | 15:29:40,408 00 01 2A D4 |08 02 16 60 05 | Bearer Caps 04 02 88 90 | Channel Id 18 03 A1 83 88 | Calling PN 6C 0C 21 80 33 33 32 38 |33 33 34 37 37 38 | Called PN70 09 81 36 33 34 39 34 |34 34 31 #08 RR--| 15:29:40,408 00 01 01 B0 | #12 RR--| 15:29:40,418 00 01 01 2C | #12SETUP ACKNOWLEDGE--| 15:29:40,418 02 01 D4 2C | 08 02 96 60 0D| Channel Id 18 03 A9 83 88| Progress Ind 1E 02 81 82 | |--RR #12 | 15:29:40,418 02 01 01 D6 #12SETUP--| 15:29:40,428 02 01 D6 2C | 08 02 1A 21 05| Bearer Caps 04 03 88 90 A3| Channel Id 18 03 A1 83 81| Calling PN 6C 0C 41 81 33 33 32 38 | 33 33 34 37 37 38 | Called PN70 0D C1 30 33 33 32 38 | 33 30 37 37 37 33 33 | |--RELEASE COMPLETE #12 | 15:29:40,428 00 01 2C D8 |08 02 9A 21 5A |08 02 80 D8 |[Incompatible destinat |ion] #12 RR--| -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von CW_ASN - Gus Gesendet: Donnerstag, 22. Januar 2004 17:24 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI The incoming call request Unrestricted and 64K, and this looks like ok, but in the SETUP_ACK the called number parameters shows: Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN equipment. In the most of cases, Information transfer rate = to '64 kbit/s', and Info transfer capability = 'real bw required'. Are you sure that the equipment attached to * can be used in 64K? Regards, Gus - Original Message - From: Thomas Haeger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 22, 2004 12:28 PM Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI Hi , maybe someone knows what's going wrong... The incoming data call will not really identified as ISDN 64k/Data Here my pri debug ouput Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 5635/0x1603) (Originator) Message type: SETUP (5) Bearer Capability (len= 2) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 30 ] Calling Number (len=14
AW: [Asterisk-Users] newbie ISDN question
Hi Thorsten, the E100P is an E1 Card with 30 Channels (PRI), this is not for connecting Phones directly. You can youse the TDM10-40B for Analogphones, or you can use the new BRI Card from kapejod -- http://ns1.jnetdns.de/jn/relaunch/asterisk/page17.html But the driver is alpha stadium ;-), or you can use VoIP Phones like Grandstream BudgetTone 100 -- http://www.grandstream.com/y-product.htm Best regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von FastJack Gesendet: Mittwoch, 14. Januar 2004 10:22 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] newbie ISDN question hi everybody, sorry for posting such a stupid question ;) i've managed to run asterisk* with my AVM fritz2.0 card and a some VOIP-softphones (SIP, H323). the functions of asterisk* really satisfied me ;))) now i want to run asterisk* istead of our old PBX. but it would be great to connect some phones directly to my box. how does a E100P from digium work. can i connect it to my ISDN-line and my internal phones (ISDN)? it would look like this: [PHONE2] / [PC]-[E100P] - [PHONE1] \ [ISDN-LINE] thank you for your help!!! thorsten ___ 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] noises an Zap channel (TDM20B) while hdd activity
Hi all, I have installed one TDM20B on an ITX board in a small cube chassis. When the harddisk is working (when installing something or make a query on a database) i can hear nasty noises (like hdd-head is moving) on the connected phone. Have someone experiences with this manner ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable dial out with the new Oh323 0.5.6
Hi all, i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then i've installed the new chan_oh323 (0.5.6). when i try to make a call with netmeeting through * ( * dial out with Dial,OH323/[EMAIL PROTECTED] ) the call will be blocked. Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7) installed, and it worked. Is here something wrong with this url ? Before i installed the new stuff it worked so. Can somebody help? Thanks, Thomas. Here the trace level 2 log in oh323.log: 0:30.808H323 Listener:80f24d8 H323TCP Started connection: host=172.20.23.206:1752, if=172.20.0.150:1720, handle=11 0:39.475 H225 Answer:80db018 H225Failed to get initial Q.931 PDU, connection not started. 1:12.973H323 Listener:80f24d8 H323TCP Started connection: host=172.20.23.206:1753, if=172.20.0.150:1720, handle=11 1:12.996 H225 Answer:80db018 H225Set remote application name: Microsoft® NetMeeting®3.0 181/21324 1:13.011 H225 Answer:80db018 H323Answering call: AnswerCallPending 1:13.013 ThreadID=0x0004c013 H323Making call to: [EMAIL PROTECTED]:1720 1:13.014 ThreadID=0x0004c013 H323Attempt to use invalid URL [EMAIL PROTECTED]:1720 1:13.014 ThreadID=0x0004c013 H323Could not parse [EMAIL PROTECTED]:1720 1:13.021 H225 Answer:80db018 H225Reading PDUs: callRef=4422 1:13.026 ClearCallT...d:08127268 H225Sending release complete PDU: callRef=4422 1:13.031 H225 Answer:80db018 H225Read error (4): Interrupted system call 1:13.033 H225 Answer:80db018 H225Signal channel closed. 1:13.046 H323 Cleaner H323Connection ip$172.20.23.206:1753/4422 terminated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Unable dial out with the new Oh323 0.5.6
Thanks Michael, for this very special detail :-) Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Freitag, 7. November 2003 14:00 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Unable dial out with the new Oh323 0.5.6 Thomas Haeger wrote: Hi all, i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then i've installed the new chan_oh323 (0.5.6). when i try to make a call with netmeeting through * ( * dial out with Dial,OH323/[EMAIL PROTECTED] ) the call will be blocked. This is a problem of OpenH323 1.12.0. Use this dial string: Dial,OH323/h323:[EMAIL PROTECTED] Or, even better, use the latest (it has been fixed). Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7) installed, and it worked. Is here something wrong with this url ? Before i installed the new stuff it worked so. Can somebody help? Thanks, Thomas. Michael. Here the trace level 2 log in oh323.log: 0:30.808H323 Listener:80f24d8 H323TCP Started connection: host=172.20.23.206:1752, if=172.20.0.150:1720, handle=11 0:39.475 H225 Answer:80db018 H225Failed to get initial Q.931 PDU, connection not started. 1:12.973H323 Listener:80f24d8 H323TCP Started connection: host=172.20.23.206:1753, if=172.20.0.150:1720, handle=11 1:12.996 H225 Answer:80db018 H225Set remote application name: Microsoft® NetMeeting®3.0 181/21324 1:13.011 H225 Answer:80db018 H323Answering call: AnswerCallPending 1:13.013 ThreadID=0x0004c013 H323Making call to: [EMAIL PROTECTED]:1720 1:13.014 ThreadID=0x0004c013 H323Attempt to use invalid URL [EMAIL PROTECTED]:1720 1:13.014 ThreadID=0x0004c013 H323Could not parse [EMAIL PROTECTED]:1720 1:13.021 H225 Answer:80db018 H225Reading PDUs: callRef=4422 1:13.026 ClearCallT...d:08127268 H225Sending release complete PDU: callRef=4422 1:13.031 H225 Answer:80db018 H225Read error (4): Interrupted system call 1:13.033 H225 Answer:80db018 H225Signal channel closed. 1:13.046 H323 Cleaner H323Connection ip$172.20.23.206:1753/4422 terminated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] which channel format number is right?
Hi all, if i enter a show codecs at cli * response with: 1 (1 0) G.723.1 2 (1 1) GSM 4 (1 2) G.711 u-law 8 (1 3) G.711 A-law 16 (1 4) MPEG-2 layer 3 32 (1 5) ADPCM 64 (1 6) 16 bit Signed Linear PCM 128 (1 7) LPC10 256 (1 8) G.729A audio 512 (1 9) SpeeX 1024 (1 10) iLBC 65536 (1 16) JPEG image 131072 (1 17) PNG image 262144 (1 18) H.261 Video 524288 (1 19) H.263 Video If i enter a show channel * response with Name: H323/ip$XXX.XX.XX.XX:3520/25650 Type: H323 UniqueID: 1068111809.487 Caller ID: 0109901 DNID Digits: (N/A) State: Ringing (5) Rings: 0 NativeFormat: 8 WriteFormat: 8 ReadFormat: 8 1st File Descriptor: 366 Frames in: 0 Frames out: 1240 With wich codec is the channel working now, with ALAW or with g.729A And what is the relevant value read/write format or nativeformat ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] which channel format number is right?
Hi, is there anybody who knows this very little detail ??? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Donnerstag, 6. November 2003 10:54 An: Asterisk User Betreff: [Asterisk-Users] which channel format number is right? Hi all, if i enter a show codecs at cli * response with: 1 (1 0) G.723.1 2 (1 1) GSM 4 (1 2) G.711 u-law 8 (1 3) G.711 A-law 16 (1 4) MPEG-2 layer 3 32 (1 5) ADPCM 64 (1 6) 16 bit Signed Linear PCM 128 (1 7) LPC10 256 (1 8) G.729A audio 512 (1 9) SpeeX 1024 (1 10) iLBC 65536 (1 16) JPEG image 131072 (1 17) PNG image 262144 (1 18) H.261 Video 524288 (1 19) H.263 Video If i enter a show channel * response with Name: H323/ip$XXX.XX.XX.XX:3520/25650 Type: H323 UniqueID: 1068111809.487 Caller ID: 0109901 DNID Digits: (N/A) State: Ringing (5) Rings: 0 NativeFormat: 8 WriteFormat: 8 ReadFormat: 8 1st File Descriptor: 366 Frames in: 0 Frames out: 1240 With wich codec is the channel working now, with ALAW or with g.729A And what is the relevant value read/write format or nativeformat ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g.729 codec registration
Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the old binary) But there're a few questions: - should not the codec listed in the codec list when i enter show codecs ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between g729a built in * and the puchased g729b codec? Thanks for help. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g.729 codec registration
Hi i'am again... i have tesed if my * (where the purch. g729 is installed) take calls from a gateway with g.729A codec. The calling mechanism works but there is no voice only bad noises . I'am a little bit confused. On the digium site i bought a g729 codec (without any indication of an a or a b). I thought this codec could take calls with g729.a codec but this seems not to be so. If my fiction is right, how can i take calls with g.729.a codec ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Mittwoch, 5. November 2003 12:15 An: Asterisk User Betreff: [Asterisk-Users] g.729 codec registration Hi all, i have purchased the g.729 codec from digium. The registration was successful. (with the old binary) But there're a few questions: - should not the codec listed in the codec list when i enter show codecs ? - the codec is named with g729b but if i enter show codecs there is a codec g729a listed also the g729b is not installed. what is the difference between g729a built in * and the puchased g729b codec? Thanks for help. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where can i get the g.723 codec?
Hi all, can somebody tell me where i can get the g.723 codec for * ? Thanks. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Where can i get the g.723 codec?
This is the g.729 codec, but i want the g.723 -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Gavin Hamill Gesendet: Montag, 3. November 2003 15:44 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Where can i get the g.723 codec? On Mon, 2003-11-03 at 14:28, Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? http://store.yahoo.com/asteriskpbx/asteriskg729.html $10 per channel. I looked into the licensing costs for another product, and this is damn cheap. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Where can i get the g.723 codec?
Thanks Steve, there is no special reason for me for using g.723. I will take g.729. It seems to be easier :-) Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Steve Underwood Gesendet: Montag, 3. November 2003 17:14 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Where can i get the g.723 codec? Hi Thomas, Unless you have a *very* specific need to use G.723.1 for compatibility with someone else, forget it. It is pretty much an obsolete product. Licencing is also a pain, as there is not patent pool for it. G.729 is expensive to licence, but at least it is relatively strightforward. If you think you will save some bits using G.723.1 instead of G.729, think again. The saving is minute, because of the huge overheads IP imposes. Regards, Steve Thomas Haeger wrote: Hi all, can somebody tell me where i can get the g.723 codec for * ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] probs with loading tor2 and wcusb
Hi all, i tried to load the tor2 driver for the E400P with modprobe tor2 on a P4 2,7Ghz machine. and everything seems to be ok. but after load (after the kernel messages) the machine is freezed. And then i have to reset the whole machine. The same is happend when i try to load the wcusb driver on a VIA EPIA 5000 machine. Any ideas ? Thanks, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] probs with loading tor2 and wcusb
Ok, meanwhile i loaded the tor2 driver with insmod (previous loaded slhc, ppp_generic,zaptel) this works, but when i execute ztcfg the machine is freezed! Now any ideas ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Mittwoch, 29. Oktober 2003 12:32 An: Asterisk User Betreff: [Asterisk-Users] probs with loading tor2 and wcusb Hi all, i tried to load the tor2 driver for the E400P with modprobe tor2 on a P4 2,7Ghz machine. and everything seems to be ok. but after load (after the kernel messages) the machine is freezed. And then i have to reset the whole machine. The same is happend when i try to load the wcusb driver on a VIA EPIA 5000 machine. Any ideas ? Thanks, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] probs with loading tor2 and wcusb
Hi i'am again, here my zapata.conf: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = fr defaultzone=fr If i comment out the span,bchan and dchan values and then ztcfg works. But this is naturally not wished ;-) Any ideas (agian...) ? Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Mittwoch, 29. Oktober 2003 14:05 An: [EMAIL PROTECTED] Betreff: AW: [Asterisk-Users] probs with loading tor2 and wcusb Ok, meanwhile i loaded the tor2 driver with insmod (previous loaded slhc, ppp_generic,zaptel) this works, but when i execute ztcfg the machine is freezed! Now any ideas ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Mittwoch, 29. Oktober 2003 12:32 An: Asterisk User Betreff: [Asterisk-Users] probs with loading tor2 and wcusb Hi all, i tried to load the tor2 driver for the E400P with modprobe tor2 on a P4 2,7Ghz machine. and everything seems to be ok. but after load (after the kernel messages) the machine is freezed. And then i have to reset the whole machine. The same is happend when i try to load the wcusb driver on a VIA EPIA 5000 machine. Any ideas ? Thanks, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] probs with loading tor2 and wcusb
Is an owen irq required ? The card shares one irq with other devices -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Bielicki Gesendet: Mittwoch, 29. Oktober 2003 17:33 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] probs with loading tor2 and wcusb did you check that there are no irq conflicts ? On Wednesday 29 October 2003 3:13 pm, Thomas Haeger wrote: Hi i'am again, here my zapata.conf: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = fr defaultzone=fr If i comment out the span,bchan and dchan values and then ztcfg works. But this is naturally not wished ;-) Any ideas (agian...) ? Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Mittwoch, 29. Oktober 2003 14:05 An: [EMAIL PROTECTED] Betreff: AW: [Asterisk-Users] probs with loading tor2 and wcusb Ok, meanwhile i loaded the tor2 driver with insmod (previous loaded slhc, ppp_generic,zaptel) this works, but when i execute ztcfg the machine is freezed! Now any ideas ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Mittwoch, 29. Oktober 2003 12:32 An: Asterisk User Betreff: [Asterisk-Users] probs with loading tor2 and wcusb Hi all, i tried to load the tor2 driver for the E400P with modprobe tor2 on a P4 2,7Ghz machine. and everything seems to be ok. but after load (after the kernel messages) the machine is freezed. And then i have to reset the whole machine. The same is happend when i try to load the wcusb driver on a VIA EPIA 5000 machine. Any ideas ? Thanks, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] get IP Address from caller using oh323
Hi all (Michael), how it is possible to get the ip address of the calling party ? (i know by using h323... but there're a few unknown segfaults...) and so i want to use oh323, but i have to get the ip from the caller to permit or deny the call with AGI. Is it possible at all ? Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] wcfxs error
Have you another ISDN card in your system ? -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von C M Gesendet: Donnerstag, 23. Oktober 2003 14:06 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] wcfxs error hi guys, i got a TDM400P FXS card an everything is fine except for this when i do modprobe wcfxs , the linux shows 2 TigerJet Network Inc Model 300 128k. i don't know why it is showing 2 of them. or is that what it is? ERROR: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 0 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 0 cshould be 2 Module 1: Not installed ProSLIC on module 2 insane (1) 0 should be 2 Module 2: Not installed ProSLIC on module 3 insane (1) 0 should be 2 Module 3: Not installed /lib/modules/2.4.20-8/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wcfxs.o: insmod /lib/modules/2.4.20-8/misc/wcfxs.o failed /lib/modules/2.4.20-8/misc/wcfxs.o: insmod wcfxs failed i searched the archives and i don't see a slolution for this? somebody help me! = Designs __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and permit specified ip addresses
Hi all, can somebody explain me how exactly the type, host, permit and deny option in sip.conf play together? Where is the difference between user and peer ? I want configure SIP so that it is only from specified net section possible to make a call. I have tried following: [test] type=peer callerid=testaccount context=voipin dtmfmode=inband ; Choices are inband, rfc2833, or info deny=0.0.0.0/0.0.0.0 permit=172.20.0.0/255.255.0.0 Here i thought that it is possible to make a call only for computers from the 172.20.0.0 net section. But this don't work. No computer can make a call. The only thing that worked for me was following: [test] type=peer callerid=testaccount context=voipin host=172.20.23.206 The address 172.20.23.206 is the ip from the client pc. This works, but i want specify a net section so that more computers are allowed to make calls. Can sombody help me with this ? Thank you very much. Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] PRI/E1: machine freeze/dies after a few calls
Hi Scott, thanks for your help. The frame errors which you got i don't got. Yes this is a slow machine, but i greenly thought that * can handle about 30 calls on it without problems Have you ever run 120 simultaneously calls on one machine ? What does you mean with How many instances are you running ? Best regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Scott Stingel Gesendet: Dienstag, 14. Oktober 2003 02:28 An: [EMAIL PROTECTED] Betreff: RE: [Asterisk-Users] PRI/E1: machine freeze/dies after a few calls Hi Thomas- I didn't look closely at your shell script, but I wrote something similar in Perl (and used shell to start each instance of it). I had a few problems too with a similar setup (although no machine lockups) * You are running quite a slow machine to run this script on many lines at once - I found that I needed a P4, 2.4GHz to keep up with 120 channels simultaneously (I had one system to send and one to receive, and very short calls - 3 seconds). How many instances are you running? Are you doing mySQL call logging? * I found that I could only initiate about 18 calls at exactly the same moment without getting failed outbound call errors from asterisk, so I ended up staggering the start times a little. * With lots of new calls, I had tons of framing errors on the receiving end (and occasional D channel restarts) when routing calls through my DMS100 switch - do you have problems like this? I think this problem is specific to the Nortel switch however. Suggest starting with -c and routing all output to a log file...?? regards Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Haeger Sent: Monday, October 13, 2003 8:00 PM To: Asterisk User Subject: [Asterisk-Users] PRI/E1: machine freeze/dies after a few calls Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1# Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kanäle sle=$4 # Timeout bis zum nächsten Versuch if [ -z $4 ]; then sle=0 fi s=1 i=1 while [ $s -le $anz ];do echo $s try started... while [ $i -le $anz2 ];do echo -e Channel: Zap/g1/$n$i\nMaxRetries: 0\nContext: callgen\nExtension: 1\nPriority: 1\nCallerid:334778\n /var/spool/asterisk/outgoing/call.$i.$s sleep 2 i=$((i + 1)) done i=1 echo sleep for $sle sec. sleep $sle s=$((s + 1)) done The calls goes out over the first two ports and through a pri switch (teles) they come back at the other two ports (3 and 4). But after a few calls my machine is completly freezed! So that i have to restart my machine. Here're my extension.conf, zapata.conf and zaptel.conf: extension.conf: [pri1] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(15) exten = _X.,4,Hangup [pri2] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(15) exten = _X.,4,Hangup [pri3] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(60) exten = _X.,4,Hangup [pri4] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(60) exten = _X.,4,Hangup [callgen] exten = 1,1,Wait(90) zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [channels] pridialplan=local switchtype=euroisdn busydetect=yes callprogress=no echocancel=yes echocancelwhenbridged=yes ;callwaitingcallerid=no ;callwaiting=no signalling=pri_net group=1 context=pri1 channel = 1-15,17-31 channel =32-46,48-62 signalling=pri_net group=3 context=pri3 channel = 63-77,79-93 channel = 94-108,110-124 zaptel.conf span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = fr defaultzone=fr Thanks for your help. Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI/E1: machine freeze/dies after a few calls
Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1# Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kanäle sle=$4 # Timeout bis zum nächsten Versuch if [ -z $4 ]; then sle=0 fi s=1 i=1 while [ $s -le $anz ];do echo $s try started... while [ $i -le $anz2 ];do echo -e Channel: Zap/g1/$n$i\nMaxRetries: 0\nContext: callgen\nExtension: 1\nPriority: 1\nCallerid:334778\n /var/spool/asterisk/outgoing/call.$i.$s sleep 2 i=$((i + 1)) done i=1 echo sleep for $sle sec. sleep $sle s=$((s + 1)) done The calls goes out over the first two ports and through a pri switch (teles) they come back at the other two ports (3 and 4). But after a few calls my machine is completly freezed! So that i have to restart my machine. Here're my extension.conf, zapata.conf and zaptel.conf: extension.conf: [pri1] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(15) exten = _X.,4,Hangup [pri2] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(15) exten = _X.,4,Hangup [pri3] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(60) exten = _X.,4,Hangup [pri4] exten = _X.,1,SetAccount(pritest) exten = _X.,2,Answer exten = _X.,3,Wait(60) exten = _X.,4,Hangup [callgen] exten = 1,1,Wait(90) zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [channels] pridialplan=local switchtype=euroisdn busydetect=yes callprogress=no echocancel=yes echocancelwhenbridged=yes ;callwaitingcallerid=no ;callwaiting=no signalling=pri_net group=1 context=pri1 channel = 1-15,17-31 channel =32-46,48-62 signalling=pri_net group=3 context=pri3 channel = 63-77,79-93 channel = 94-108,110-124 zaptel.conf span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = fr defaultzone=fr Thanks for your help. Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modem connection over handy?
Hi all, does anybody know if it is possible to make a modem connection (voice) through * over a handy which is connected to the RS232 port ? I get following messages when * starting: WARNING[16384]: File chan_modem.c, Line 356 (modem_setup): Modem reset failed: (No Response) WARNING[16384]: File chan_modem.c, Line 735 (mkif): Unable to configure modem '/dev/ttyS0' ERROR[16384]: File chan_modem.c, Line 871 (load_module): Unable to register channel '/dev/ttyS0' WARNING[16384]: File loader.c, Line 301 (ast_load_resource): chan_modem.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 347 (load_modules): Loading module chan_modem.so failed! Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to prevent echo ?
Hi all, i have following scenario: * 1 __* 2__ | | | | analog/Zap -- IAX2 DSL --- INTERNET --- Backbone/100Mbit IAX2 --- Zap/pri(E400P) PSTN And, if i make a call from *1 over *2 to PSTN, i can hear an echo in my analog phone, even though echocancel and echocancelwhenbridged is on yes on both sides. Can somebody explain me what i'am doing wrong ? Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sometimes pri channels restart during * is runnig ?
Hi all, i have observed, that sometimes all BChannels on my Zaptel Pri device (E400P) will be restarted. The E400P is connected to another pri switch. In the traces from the other side (pri switch) i can see that libpri request for the channelid is 255. Is this a bug or a feature ...? Or, can it be a bug on the other side (terminator switch) ? Have anyone an idea ? Thanks, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AntiSpam UOL [andersoncbr.sspam@uol.com.br]
Hi, what the hell is this ? Can somebody cancel this user from the list ??? Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX --- PRI --- PSTN call disconnected after a few minutes
Hi all, gave somebody an idea ? I have not set a AbsoluteTimeout or smothing like this. Regards, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0
Hi Jim, i had the same probs, and it seems to be bug/feature of i4l. I can not find anything in the code that would bring these messages to the top of ttyI:-( Or is there somebody who knows it better ??? ;-) Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jim Paraschou Gesendet: Mittwoch, 24. September 2003 21:18 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0 Hi, I use an AVM FRITZ PCI 2.0 to dial out but although it works OK and places the call there is no ring or busy tone. Has someone figured out this problem? Thanks __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to dial a h323 destination ?
Please, can somebody tell me how do a h323 call correctly with the dial app
?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Montag, 22. September 2003 18:26
An: Asterisk User
Betreff: [Asterisk-Users] how to dial a h323 destination ?
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider.
The provider demanded following account data:
H323 ID:XXX-XXX-XX-X
DetinationNumer: XXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?
And how i have to make a call with the dial app ?
I have following config:
exten = _01099X.,1,Dial,OH323/${EXTEN:7}
exten = _01099X.,2,Hangup
I thought it would be enough when i give the destination number if i
registered at the gk, isn't it ?
Or is a ip and something like a userbname necessary ? And if how can i dial
so?
Can somebody help please ?
Thanks,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email: [EMAIL PROTECTED]
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AW: [Asterisk-Users] how to dial a h323 destination ?
OK. This is what i know too...
But this don't work. The gatekeeper tells me everytime caller not
registered.
If i start *, the registration at the gatekeeper is ok.
If i make i call it is not ok. Is there any other info that i have to
send with ?
like : Dial(OH323/[EMAIL PROTECTED]/H323ID or similar like this ?
Thanks for help,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Sergio
Serrano Revuelto
Gesendet: Dienstag, 23. September 2003 11:15
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] how to dial a h323 destination ?
exten=XXX,1,Dial(h323/3|17|tTm)
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 11:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?
Please, can somebody tell me how do a h323 call correctly with the dial
app ?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Montag, 22. September 2003 18:26
An: Asterisk User
Betreff: [Asterisk-Users] how to dial a h323 destination ?
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider. The provider demanded following account data:
H323 ID:XXX-XXX-XX-X
DetinationNumer: XXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?
And how i have to make a call with the dial app ?
I have following config:
exten = _01099X.,1,Dial,OH323/${EXTEN:7}
exten = _01099X.,2,Hangup
I thought it would be enough when i give the destination number if i
registered at the gk, isn't it ? Or is a ip and something like a
userbname necessary ? And if how can i dial so?
Can somebody help please ?
Thanks,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email: [EMAIL PROTECTED]
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RE: [Asterisk-Users] how to dial a h323 destination ?
Here is my oh323.conf ... ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound H.323 connections. ; outboundMax=50 inboundMax=10 ; ; Set tracing options for the H.323 wrapper library. ; libTraceFile can be 'stdout' or a full path name to a logfile ; libTraceLevel=3 ;libTraceFile=stdout libTraceFile=/var/log/asterisk/oh323.log ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; gatekeeper=80.86.166.196 ;gatekeeper=DISABLE ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=Q931 ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; context=voipout ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] alias=BER-BER-GW-1 ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; ;context=all-aliases ;alias=ASTERISK ;alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; ;context=more-aliases ;alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; ;context=all-prefixes ;gwprefix=00 ;gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; ;context=more-stuff ;alias=664 ;gwprefix=02 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; ;codec=G729A ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=20 ;codec=G72316K3 ;codec=G72315K3 ;codec=G7231A6K3 ;codec=G7231A5K3 codec=G711A ;codec=G711U ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] how to dial a h323 destination ?
What is the gwprefix ? I try to connect the gk directly from our * gw -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Sergio Serrano Revuelto Gesendet: Dienstag, 23. September 2003 12:27 An: [EMAIL PROTECTED] Betreff: RE: [Asterisk-Users] how to dial a h323 destination ? Try to add gwprefix in oh323.conf after your alias. You must know that you can configure * gw in gnugk.ini or in oh323.conf. I recommend you put in your oh323.conf. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Thomas Haeger Enviado el: martes, 23 de septiembre de 2003 12:07 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] how to dial a h323 destination ? Here is my oh323.conf ... ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound H.323 connections. ; outboundMax=50 inboundMax=10 ; ; Set tracing options for the H.323 wrapper library. ; libTraceFile can be 'stdout' or a full path name to a logfile ; libTraceLevel=3 ;libTraceFile=stdout libTraceFile=/var/log/asterisk/oh323.log ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; gatekeeper=80.86.166.196 ;gatekeeper=DISABLE ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=Q931 ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; context=voipout ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] alias=BER-BER-GW-1 ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; ;context=all-aliases ;alias=ASTERISK ;alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; ;context=more-aliases ;alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; ;context=all-prefixes ;gwprefix=00 ;gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; ;context=more-stuff ;alias=664 ;gwprefix=02 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; ;codec=G729A ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=20 ;codec=G72316K3 ;codec=G72315K3 ;codec=G7231A6K3 ;codec=G7231A5K3 codec=G711A ;codec=G711U ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list
[Asterisk-Users] Dial over IAX ahngs up after 3 rings
Hi all, can somebody explain this ? Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Dial over IAX ahngs up after 3 rings
I have tried it with a timeout and without... here the * output for the first side: -- Starting simple switch on 'Zap/3-1' -- Executing Dial(Zap/3-1, IAX2/useranme:[EMAIL PROTECTED]/99033283077731) in new stack -- Called thaeger:[EMAIL PROTECTED]/99033283077731 -- Call accepted by 62.180.50.212 (format ALAW) -- Format for call is ALAW -- Hungup 'IAX2[62.180.50.212:4569]/2' == No one is available to answer at this time -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' and here from the other side: -- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8, actual format = 8 -- Executing SetCallerID([EMAIL PROTECTED]:4569]/1, 033283077731) in new stack -- Executing Dial([EMAIL PROTECTED]:4569]/1, Zap/g3/033283077731) in new stack -- Called g3/033283077731 -- Channel 1, span 3 got hangup -- Hungup 'Zap/63-1' == No one is available to answer at this time -- Executing Hangup([EMAIL PROTECTED]:4569]/1, ) in new stack == Spawn extension (voipout, 99033283077731, 3) exited non-zero on '[EMAIL PROTECTED]:4569]/1' -- Hungup '[EMAIL PROTECTED]:4569]/1' -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Steven Critchfield Gesendet: Dienstag, 23. September 2003 17:13 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote: Hi all, can somebody explain this ? Do you have something like a |15 in the dial string? Do you have logs to show what asterisk did? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk stops responding
I have tried it with a timeout and without... here the * output for the first side: -- Starting simple switch on 'Zap/3-1' -- Executing Dial(Zap/3-1, IAX2/useranme:[EMAIL PROTECTED]/99033283077731) in new stack -- Called thaeger:[EMAIL PROTECTED]/99033283077731 -- Call accepted by 62.180.50.212 (format ALAW) -- Format for call is ALAW -- Hungup 'IAX2[62.180.50.212:4569]/2' == No one is available to answer at this time -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' and here from the other side: -- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8, actual format = 8 -- Executing SetCallerID([EMAIL PROTECTED]:4569]/1, 033283077731) in new stack -- Executing Dial([EMAIL PROTECTED]:4569]/1, Zap/g3/033283077731) in new stack -- Called g3/033283077731 -- Channel 1, span 3 got hangup -- Hungup 'Zap/63-1' == No one is available to answer at this time -- Executing Hangup([EMAIL PROTECTED]:4569]/1, ) in new stack == Spawn extension (voipout, 99033283077731, 3) exited non-zero on '[EMAIL PROTECTED]:4569]/1' -- Hungup '[EMAIL PROTECTED]:4569]/1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings
I have tried it with a timeout and without... here the * output for the first side: -- Starting simple switch on 'Zap/3-1' -- Executing Dial(Zap/3-1, IAX2/useranme:[EMAIL PROTECTED]/99033283077731) in new stack -- Called thaeger:[EMAIL PROTECTED]/99033283077731 -- Call accepted by 62.180.50.212 (format ALAW) -- Format for call is ALAW -- Hungup 'IAX2[62.180.50.212:4569]/2' == No one is available to answer at this time -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' and here from the other side: -- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8, actual format = 8 -- Executing SetCallerID([EMAIL PROTECTED]:4569]/1, 033283077731) in new stack -- Executing Dial([EMAIL PROTECTED]:4569]/1, Zap/g3/033283077731) in new stack -- Called g3/033283077731 -- Channel 1, span 3 got hangup -- Hungup 'Zap/63-1' == No one is available to answer at this time -- Executing Hangup([EMAIL PROTECTED]:4569]/1, ) in new stack == Spawn extension (voipout, 99033283077731, 3) exited non-zero on '[EMAIL PROTECTED]:4569]/1' -- Hungup '[EMAIL PROTECTED]:4569]/1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial over IAX ahngs up after 3 rings
I have tried it with a timeout and without... here the * output for the first side: -- Starting simple switch on 'Zap/3-1' -- Executing Dial(Zap/3-1, IAX2/useranme:[EMAIL PROTECTED]/99033283077731) in new stack -- Called thaeger:[EMAIL PROTECTED]/99033283077731 -- Call accepted by 62.180.50.212 (format ALAW) -- Format for call is ALAW -- Hungup 'IAX2[62.180.50.212:4569]/2' == No one is available to answer at this time -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (guersel, 033283077731, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' and here from the other side: -- Accepting AUTHENTICATED call from 217.81.111.2, requested format = 8, actual format = 8 -- Executing SetCallerID([EMAIL PROTECTED]:4569]/1, 033283077731) in new stack -- Executing Dial([EMAIL PROTECTED]:4569]/1, Zap/g3/033283077731) in new stack -- Called g3/033283077731 -- Channel 1, span 3 got hangup -- Hungup 'Zap/63-1' == No one is available to answer at this time -- Executing Hangup([EMAIL PROTECTED]:4569]/1, ) in new stack == Spawn extension (voipout, 99033283077731, 3) exited non-zero on '[EMAIL PROTECTED]:4569]/1' -- Hungup '[EMAIL PROTECTED]:4569]/1' -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Steven Critchfield Gesendet: Dienstag, 23. September 2003 17:13 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Dial over IAX ahngs up after 3 rings On Tue, 2003-09-23 at 09:55, Thomas Haeger wrote: Hi all, can somebody explain this ? Do you have something like a |15 in the dial string? Do you have logs to show what asterisk did? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to dial a h323 destination ?
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider.
The provider demanded following account data:
H323 ID:XXX-XXX-XX-X
DetinationNumer: XXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?
And how i have to make a call with the dial app ?
I have following config:
exten = _01099X.,1,Dial,OH323/${EXTEN:7}
exten = _01099X.,2,Hangup
I thought it would be enough when i give the destination number if i
registered at the gk, isn't it ?
Or is a ip and something like a userbname necessary ? And if how can i dial
so?
Can somebody help please ?
Thanks,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email: [EMAIL PROTECTED]
***
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AW: [Asterisk-Users] No sound on PSTN -- */PRI
Hi Grzegorz, this dont seem to work, too :-( -Ursprngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Grzegorz Nosek Gesendet: Samstag, 20. September 2003 16:10 An: [EMAIL PROTECTED] Betreff: RE: [Asterisk-Users] No sound on PSTN -- */PRI On Fri, 19 Sep 2003 18:10:37 +0100, Scott Stingel wrote Have you tried starting asterisk with -c? It should give you some detail as to what is happening with the call. Scott M. Stingel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Haeger Sent: Friday, September 19, 2003 3:40 PM To: Asterisk User Subject: [Asterisk-Users] No sound on PSTN -- */PRI Hi all, i tried to make a call from public pstn in our */E100P. Config is following: exten = _X.,1,Playback(testgsm) But what i hear is one dtmf tone and then nothing... Any ideas ? Regards, Thomas. [snip] have you tried answering the channel first? as in: exten=_X.,1,Answer ; a 1-2 sec wait here maybe? exten=_X.,2,Playback(testgsm) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No sound on PSTN -- */PRI
Now it's going confusing if i call to this extension the call will be answered but i hear no sound. If i make a call at the same time another channel will be created, which play back the mp3 file at the position where the other channel have to be. And now, if i hang up the first channel, the sound on the scond one vanishes. My conf: exten = _X.,1,Answer exten = _X.,2,Playback(outofdark) ;(mp3 file) Can somebody help? Thanks, Thomas. -Ursprngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Samstag, 20. September 2003 17:11 An: [EMAIL PROTECTED] Betreff: AW: [Asterisk-Users] No sound on PSTN -- */PRI Hi Grzegorz, this dont seem to work, too :-( -Ursprngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Grzegorz Nosek Gesendet: Samstag, 20. September 2003 16:10 An: [EMAIL PROTECTED] Betreff: RE: [Asterisk-Users] No sound on PSTN -- */PRI On Fri, 19 Sep 2003 18:10:37 +0100, Scott Stingel wrote Have you tried starting asterisk with -c? It should give you some detail as to what is happening with the call. Scott M. Stingel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Haeger Sent: Friday, September 19, 2003 3:40 PM To: Asterisk User Subject: [Asterisk-Users] No sound on PSTN -- */PRI Hi all, i tried to make a call from public pstn in our */E100P. Config is following: exten = _X.,1,Playback(testgsm) But what i hear is one dtmf tone and then nothing... Any ideas ? Regards, Thomas. [snip] have you tried answering the channel first? as in: exten=_X.,1,Answer ; a 1-2 sec wait here maybe? exten=_X.,2,Playback(testgsm) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codec probs wit g723.1
Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog --- IAX -INTERNET-IAX---OH323GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ringing tone on analog Zap channel question
Hi all, can somebody explain me why i can't hear a ringing tone (alerting) if i'am going to connect to my destination end point? Is it basically so that i have to configure like: exten = xxx,1,Dial,ChanTec/number|timout|r Is it really nessesary to use the r option everytime if i want to indicate a ringing tone? This suggest a wrong call flow for the user ... Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No sound on PSTN -- */PRI
Hi all, i tried to make a call from public pstn in our */E100P. Config is following: exten = _X.,1,Playback(testgsm) But what i hear is one dtmf tone and then nothing... Any ideas ? Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no ring tone analog Zap -- I4L
Hi all, i have noticed that i can't hear a ring tone if i make a call from my TDM40B to a chan_modem_i4l endpoint. I had a look in the code from chan_modem_i4l and there is a function calling i4l_handle_escape that gives a AST_CONTROL_RINGING frame back. But this seems not work ...(or i4l is not signaling it ?) Til now i have used the Dail app like Dial, Zap/g1:XX|60|r so it is no wonder that i never noticed that the ring tone not working Have anybody an idea ? Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed
Hi Michael, registration is working now, it dials out the phone is ringing but then comes a hang up I'am i lttle newbe on h323 :-) Can you take a look on the log file ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Dienstag, 16. September 2003 15:54 An: [EMAIL PROTECTED] Betreff: Re: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed Thomas Haeger wrote: No. I have installed the versions wich your special friend has recommended. Shall i try to update to the newest versions ? (But then wouldn't work the chan_h323.so further...) I don't know what are the problems with that driver, but, yes, you should install the latest versions. Before this, check the configuration of the remote gatekeeper (if this is possible) and see if there are special requirements for the registration. Michael. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Dienstag, 16. September 2003 13:53 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] h323 gatekeeper registration failed Thomas Haeger wrote: Hi Michael, this gatekeeper works without a password but with a H323-ID, but this will be send with the dial command, i think. No, this id is provided during registration. Here is the trace with trace level 10 (?) Unfortunately, the GK rejects the registration attempt with an undefined reason (!). Did you try it with the latest OpenH323/pwlib ? Regards, Thomas. Michael. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Dienstag, 16. September 2003 12:22 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] h323 gatekeeper registration failed If the gatekeeper requires a password and you don't provide one during the registration, then it will fail. In oh323.conf use the gatekeeperPassword to provide the passwd. If this is not the case enable tracing info in oh323.conf, rerun and send me the trace file to take a look. Michael. Thomas Haeger wrote: Hi all, i have tried to connect to a clarent gatekeeper. I have used both of h323 drivers chan_h323.so and chan_oh323.so. But no one can register to this gatekeeper. Our ip is activated on this gatekeeper. Maybe, i do wrong anything I have only set the gatekeeper option in the h323.conf or oh323.conf to the ip address from the gatekeeper. gatekeeper=x.x.x.x But no one of the both driver can register to this gateway. Is there another thing that i have to keep ? I need yours help urgently. We want to go online with our *-gateway as soon as possible. Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users oh323.log Description: Binary data
AW: AW: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed
Ahh... you mean it's a codec problem? This can be... I ask my provider :-). If this was not the prob, i would get in touch with you. Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Donnerstag, 18. September 2003 17:48 An: [EMAIL PROTECTED] Betreff: Re: AW: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed Thomas Haeger wrote: Hi Michael, registration is working now, it dials out the phone is ringing but then comes a hang up I'am i lttle newbe on h323 :-) Can you take a look on the log file ? Your connection attempt terminates with a EndedByRefusal reason. My guess is that you are not allowed to use the codec you are trying to use (e.g. if you are using a g.711 try to switch to a lower bit-rate one). Or some other reason? Thanks, Thomas. Michael. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Dienstag, 16. September 2003 15:54 An: [EMAIL PROTECTED] Betreff: Re: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed Thomas Haeger wrote: No. I have installed the versions wich your special friend has recommended. Shall i try to update to the newest versions ? (But then wouldn't work the chan_h323.so further...) I don't know what are the problems with that driver, but, yes, you should install the latest versions. Before this, check the configuration of the remote gatekeeper (if this is possible) and see if there are special requirements for the registration. Michael. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Dienstag, 16. September 2003 13:53 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] h323 gatekeeper registration failed Thomas Haeger wrote: Hi Michael, this gatekeeper works without a password but with a H323-ID, but this will be send with the dial command, i think. No, this id is provided during registration. Here is the trace with trace level 10 (?) Unfortunately, the GK rejects the registration attempt with an undefined reason (!). Did you try it with the latest OpenH323/pwlib ? Regards, Thomas. Michael. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Dienstag, 16. September 2003 12:22 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] h323 gatekeeper registration failed If the gatekeeper requires a password and you don't provide one during the registration, then it will fail. In oh323.conf use the gatekeeperPassword to provide the passwd. If this is not the case enable tracing info in oh323.conf, rerun and send me the trace file to take a look. Michael. Thomas Haeger wrote: Hi all, i have tried to connect to a clarent gatekeeper. I have used both of h323 drivers chan_h323.so and chan_oh323.so. But no one can register to this gatekeeper. Our ip is activated on this gatekeeper. Maybe, i do wrong anything I have only set the gatekeeper option in the h323.conf or oh323.conf to the ip address from the gatekeeper. gatekeeper=x.x.x.x But no one of the both driver can register to this gateway. Is there another thing that i have to keep ? I need yours help urgently. We want to go online with our *-gateway as soon as possible. Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 gatekeeper registration failed
Hi all, i have tried to connect to a clarent gatekeeper. I have used both of h323 drivers chan_h323.so and chan_oh323.so. But no one can register to this gatekeeper. Our ip is activated on this gatekeeper. Maybe, i do wrong anything I have only set the gatekeeper option in the h323.conf or oh323.conf to the ip address from the gatekeeper. gatekeeper=x.x.x.x But no one of the both driver can register to this gateway. Is there another thing that i have to keep ? I need yours help urgently. We want to go online with our *-gateway as soon as possible. Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] h323 gatekeeper registration failed
Hi Michael, this gatekeeper works without a password but with a H323-ID, but this will be send with the dial command, i think. Here is the trace with trace level 10 (?) Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Dienstag, 16. September 2003 12:22 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] h323 gatekeeper registration failed If the gatekeeper requires a password and you don't provide one during the registration, then it will fail. In oh323.conf use the gatekeeperPassword to provide the passwd. If this is not the case enable tracing info in oh323.conf, rerun and send me the trace file to take a look. Michael. Thomas Haeger wrote: Hi all, i have tried to connect to a clarent gatekeeper. I have used both of h323 drivers chan_h323.so and chan_oh323.so. But no one can register to this gatekeeper. Our ip is activated on this gatekeeper. Maybe, i do wrong anything I have only set the gatekeeper option in the h323.conf or oh323.conf to the ip address from the gatekeeper. gatekeeper=x.x.x.x But no one of the both driver can register to this gateway. Is there another thing that i have to keep ? I need yours help urgently. We want to go online with our *-gateway as soon as possible. Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users oh323.log Description: Binary data
[Asterisk-Users] PPP over ISDN BRI (modem_i4l) ?
Hi all, is this possible ? Make an incoming data call with ppp ? (like ZapRas...) Thanks for help, Thomas *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX sound probs
Hi all together, i have following configuration: ISDN Phone --- ASTERISK1/PRI --- ASTERISK1/IAX --- INTERNET ---INTERNET ROUTER (Port 5036 nat) --- ASTERISK2/FXO/ANALOG DEV The call flows fine, but no sound will be transfered. On ASTERISK1 a message like stopped sounds occurs. What' s wrong? Is there another port wich i have to nat ? Regards, thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Call script after hangup
Hi
Frank,
why
you so complicated ?
Try
following:
[incoming]
exten =
s,1,Playback,welcome
exten =
s,2,Record,msgfile:gsm
exten = h,1,System(/home/frank/callscript.pl)
as
sample ... :-)
Regards,
Thomas.
-Ursprüngliche Nachricht-Von:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]Im Auftrag von Frank
N.Gesendet: Donnerstag, 4. September 2003 14:54An:
[EMAIL PROTECTED]Betreff: [Asterisk-Users] Call
script after hangup
Beginner: How can a script be called after a
calling user hangup?
What's wrong with this:
[incoming]
exten = s,1,Playback,welcome
exten = s,2,Record,msgfile:gsm
exten = h,1,Goto(callscript,1,1)
[callscript]
exten = 1,1,Wait,5
exten = 1,2,System("SomeScript")
Thank you
[Asterisk-Users] some pri questions...
Hi all, i have a few questions about PRI/ISDN: 1. Are supplementary services like conferencing, call brokering or call forwarding supported by * ? 2. Is there a way to switch calls transparent through * from one port to another port ? 3. Is it possible to configure the * so that * detecting dtmf during a call ? Thanks for answering questions, regards, Thomas. :-) *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP: Maximum retries exceeded
Hi all, this message occurs if i was connected or not: WARNING[213006]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Response) If i was connected, the call will be disconnected after a few seconds. What does it means ? I don't see anything to configure like Max retries Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] some pri questions...
Hi Martin,
in EuroISDN it is possible to hold the call and take another. This is part
of ISDN suplementary services.
Or make three-way-calling ...
And the question was if the * if you dial in
like your config:
exten = _X.,1,Dial,Zap/g2/${EXTEN}
these services will be tranfered to the Network Endpoint (NT part) of the
telcom provider.
And what is with the ISDN- Services (Audio3.1 Speech Data and so on...)
would these services go through those configuration between to ISDN
endpoints through * ?
The main question, i think, is can * be a transparent ISDN switch ?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Martin
Pycko
Gesendet: Montag, 1. September 2003 16:15
An: Asterisk User
Betreff: Re: [Asterisk-Users] some pri questions...
1. Are supplementary services like conferencing, call brokering or call
forwarding supported by * ?
Conferencing (check MeetMe application), cal brokering ??? call forwarding
you can do that by having a little script in extensions.conf (unless
you're using FXS ports, where you can use *code for that)
2. Is there a way to switch calls transparent through * from one port
to
another port ?
Sure,
simply you have to have something similar to this line in extensions.conf
and group the channels to a certain context and group (g2) in zapata.conf
[incoming]
exten = _X.,1,Dial,Zap/g2/${EXTEN}
3. Is it possible to configure the * so that * detecting dtmf during a
call
?
Yes, but you'd have to make some changes to chan_zap.c PSTN channel
driver.
regards
Martin
Thanks for answering questions, regards,
Thomas. :-)
***
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[Asterisk-Users] additional digit in front of the dialed extenesion by outgoing pri/E1 call
Hi all,
i have configured incoming voip traffic as follows:
[voipin]
exten = _X.,1,SetCallerID(033283077734)
exten = _X.,2,Dial,Zap/g4/${EXTEN}
exten = _X.,3,Hangup
If the call going out the pri dials with an additional '0' before the dialed
number.
This is for caller number AND called number. But i can't see anything that
says set a '0' more in front of the dialed extension.
I've solved the problem temorary with:
[voipin]
exten = _X.,1,SetCallerID(33283077734)
exten = _X.,2,Dial,Zap/g4/${EXTEN:1}
exten = _X.,3,Hangup
But this is not realley nice, bacause i don't know where the additional
digit comes from.
My zaptel.conf say:
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
loadzone = fr
defaultzone=fr
Can anybody help me ?
Thank you very much,
Thomas.
***
beroNet technologies GmbH
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Potsdamer Str. 18 A
14513 Teltow
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AW: [Asterisk-Users] ADSI Programs
Hi, one question: What you mean with unlocked ? -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von jerk face Gesendet: Mittwoch, 27. August 2003 18:31 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] ADSI Programs I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] go on in current context after destination channels hung up ?
Hi all, is it possible to go on in the current context after the dest channel hung up? For example: exten = 111,1,Dial,Zap/4 If the originating channel is connected to Zap/4 and the destination channel (Zap/4) hangs up, both channels will be destroyed. Is there any option or whatever for preventing the hangup for the originating channel and go on in the current context ? Or, is there a way to implement this feature ? I think it make sense to implement such a feature, but i don't know where i can prove. Any ideas ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] go on in current context after destination channels hung up ?
Where i can say n for next ? I can not see an option n in your description in app_dial.c Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Mark Spencer Gesendet: Montag, 28. Juli 2003 15:52 An: Asterisk User Betreff: Re: [Asterisk-Users] go on in current context after destination channels hung up ? Is there any option or whatever for preventing the hangup for the originating channel and go on in the current context ? Not currently. Or, is there a way to implement this feature ? Absolutely. It is only necessary not to return -1 if the hangup was on the other side. This could be enabled by an option to dial, say, n for next. I think it make sense to implement such a feature, but i don't know where i can prove. Sure. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] go on in current context after destination channels hung up ?
Aehmm... (Mark,) :-) i think i understand now ... i have to make the n option by myself. I'am not the best in the english language and i don't know the niceties in their. Forgive me, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Mark Spencer Gesendet: Montag, 28. Juli 2003 15:52 An: Asterisk User Betreff: Re: [Asterisk-Users] go on in current context after destination channels hung up ? Is there any option or whatever for preventing the hangup for the originating channel and go on in the current context ? Not currently. Or, is there a way to implement this feature ? Absolutely. It is only necessary not to return -1 if the hangup was on the other side. This could be enabled by an option to dial, say, n for next. I think it make sense to implement such a feature, but i don't know where i can prove. Sure. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] go on in context after the destination channel hung up?
Hi all, is it possible to go on in the context after the dest channel hung up? For example: exten = 111,1,Dial,Zap/4 If the originating channel is connected to Zap/4 and the destination channel (Zap/4) hangs up, both channels will be destroyed. Is there any option or whatever for preventing the hangup for the originating channel and go on in the current context ? like : exten = 111,1,Dial,Zap/4 (after Zap/4 has hung up) exten = 111,2,whatever ... Any ideas ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Please can anybody help me with this, have anybody experiences with the tor2 driver? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 11. Juli 2003 13:23 An: Asterisk User Betreff: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. Is this normal ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Steve, thanks for your explanation. This is the cause for the fact that if i change the pci slot, the problem is blown away, i think. Maybe the IRQ sharing is the cause ... Thanks a lot and best regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Steve Underwood Gesendet: Montag, 14. Juli 2003 16:27 An: [EMAIL PROTECTED] Betreff: Re: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) It is normal. What you see depends on which version of various things are on your system. The tor2 driver spends a lot of time in the interrupt service routine (about 60% of the time on the 700MHz Athlon I use). Whether the interrupt service times shows up as system usage, or falls down a hole without being reported at all, as I said, depends on which versions of things you have on your machine. Regards, Steve Steven Critchfield wrote: No it isn't normal. I have a machine with a T400P in it and I don't even see that load continuously on my machine even with calls being routed. On Mon, 2003-07-14 at 03:08, Thomas Haeger wrote: Please can anybody help me with this, have anybody experiences with the tor2 driver? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 11. Juli 2003 13:23 An: Asterisk User Betreff: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System) Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Wildcard E100P resellers in Europe ?
Here you can get the reseller data -- http://www.digium.com/index.php?menu=resellers Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Nicolas Cartron Gesendet: Freitag, 11. Juli 2003 09:53 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Wildcard E100P resellers in Europe ? All, I'd like to buy an E100P Wildcard frm Digium, but i prefer to buy it in Europe (costs, ...). Could somebody point me to an european reseller ? Thanks in advance. -- Nicolas Cartron [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. Is this normal ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] PGSQL app and pbx parsing :-(
Hi Adam,
i think the real problem is the ,.
This will be allways replaced through |.
Regrads,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Adam
Goryachev
Gesendet: Dienstag, 1. Juli 2003 10:12
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] PGSQL app and pbx parsing :-(
See README.variables or similar in the root of the asterisk source. This
should answer your questions on how to quote the various parameters needed
for sql and other functions.
Regards,
Adam
Please help.
this is generally a problem with arguments that contain ,
or (, i
think.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Sonntag, 29. Juni 2003 15:18
An: Asterisk User
Betreff: [Asterisk-Users] PGSQL app and pbx parsing :-(
Hi all,
i try to use PGSQL app effectively, but there are two problems:
1. If i use PGSQL like
exten = s,3,PGSQL,Query resultid ${connid} SELECT
count(*) FROM credit
WHERE callerid=${callerid};
an error occurs with breaking query string at ( (first bracket).
then somebody told me that i should use PGSQL like follow:
exten = s,3,PGSQL,(Query resultid ${connid} SELECT coun(*)
FROM credit
WHERE callerid=${callerid})
this works BUT,
2. If i use PGSQL like
exten = s,3,PGSQL,(Query resultid ${connid} SELECT
username,credit FROM
credit WHERE callerid=${callerid})
the query string will be broken at the first ,.
I think this is a parsing problem in pbx.c
this prog exchange , into | and parse string for ( to
affect the
beginning from arguments for the app.
How can i solve this problem, is there a trick i can use ?
Or ist this just a BUG ?
Thanks for Help,
Thomas.
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AW: [Asterisk-Users] PGSQL app and pbx parsing :-(
Please help.
this is generally a problem with arguments that contain , or (, i
think.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Sonntag, 29. Juni 2003 15:18
An: Asterisk User
Betreff: [Asterisk-Users] PGSQL app and pbx parsing :-(
Hi all,
i try to use PGSQL app effectively, but there are two problems:
1. If i use PGSQL like
exten = s,3,PGSQL,Query resultid ${connid} SELECT count(*) FROM credit
WHERE callerid=${callerid};
an error occurs with breaking query string at ( (first bracket).
then somebody told me that i should use PGSQL like follow:
exten = s,3,PGSQL,(Query resultid ${connid} SELECT coun(*) FROM credit
WHERE callerid=${callerid})
this works BUT,
2. If i use PGSQL like
exten = s,3,PGSQL,(Query resultid ${connid} SELECT username,credit FROM
credit WHERE callerid=${callerid})
the query string will be broken at the first ,.
I think this is a parsing problem in pbx.c
this prog exchange , into | and parse string for ( to affect the
beginning from arguments for the app.
How can i solve this problem, is there a trick i can use ?
Or ist this just a BUG ?
Thanks for Help,
Thomas.
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[Asterisk-Users] PGSQL app and pbx parsing :-(
Hi all,
i try to use PGSQL app effectively, but there are two problems:
1. If i use PGSQL like
exten = s,3,PGSQL,Query resultid ${connid} SELECT count(*) FROM credit
WHERE callerid=${callerid};
an error occurs with breaking query string at ( (first bracket).
then somebody told me that i should use PGSQL like follow:
exten = s,3,PGSQL,(Query resultid ${connid} SELECT coun(*) FROM credit
WHERE callerid=${callerid})
this works BUT,
2. If i use PGSQL like
exten = s,3,PGSQL,(Query resultid ${connid} SELECT username,credit FROM
credit WHERE callerid=${callerid})
the query string will be broken at the first ,.
I think this is a parsing problem in pbx.c
this prog exchange , into | and parse string for ( to affect the
beginning from arguments for the app.
How can i solve this problem, is there a trick i can use ?
Or ist this just a BUG ?
Thanks for Help,
Thomas.
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AW: [Asterisk-Users] bug in cdr ?
Please, can anybody help me with this ?
Thanks,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Donnerstag, 26. Juni 2003 19:00
An: Asterisk User
Betreff: [Asterisk-Users] bug in cdr ?
Hi all,
i have a TDM40B cinfigured with immediate=yes.
My extension.conf says:
[tel1]
exten = s,1,GotoIf($[${FREI1} = 1]?s|4:s|2)
exten = s,2,Playback,gesperrt
exten = s,3,Hangup
exten = s,4,Background(frei)
exten = _X.,1,Dial..
exten = _X.,2,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
after entering s,4,Background the user can dial digits, i think.
If the user was connected the called id is s and not the dialed number.
Thanks for help,
Thomas.
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[Asterisk-Users] Congestion or Busy app using I4L indicate ringing
Hi all, i want to indicate a congestion or a busy for incoming calls on a i4l modem line. My config entry looks like this: [isdn] exten = _X.,1,Busy (or Congestion) But if i make a call in, i hear a ringing tone. Whats wrong? Works this only for Zap channels ? Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to identify user using chan_H323
Hi all, i have two *. One makes call to another over chan_H323. At the called side i have defined a user (friend) like -- [thaeger] type=friend host=172.20.23.100 context=incomingh323 At the calling side i have following entry in extensions.conf exten = _X.,1,Dial,H323/[EMAIL PROTECTED] Now i have two questions: 1. how will be transfered the callerid (normaly exten = _X.,1,Dial,H323/[EMAIL PROTECTED] i think? But where is the username then?) 2. why the called party says ERROR[16401]: File chan_h323.c, Line 963 (setup_incoming_call): Call from user 'root' rejected due to no default context even though i gave thaeger as username? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bug in cdr ?
Hi all,
i have a TDM40B cinfigured with immediate=yes.
My extension.conf says:
[tel1]
exten = s,1,GotoIf($[${FREI1} = 1]?s|4:s|2)
exten = s,2,Playback,gesperrt
exten = s,3,Hangup
exten = s,4,Background(frei)
exten = _X.,1,Dial..
exten = _X.,2,Hangup
exten = t,1,Hangup
exten = i,1,Hangup
after entering s,4,Background the user can dial digits, i think.
If the user was connected the called id is s and not the dialed number.
Thanks for help,
Thomas.
***
beroNet technologies GmbH
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Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
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AW: [Asterisk-Users] how to identify user using chan_H323
Why not ? We want use * as Viop gateway to other carriers... And there we have to identify us i think. Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jeremy McNamara Gesendet: Donnerstag, 26. Juni 2003 20:26 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] how to identify user using chan_H323 Thomas Haeger wrote: i have two *. One makes call to another over chan_H323. IN GOD'S NAME WHY?Use IAX2. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] no sound pri -- h323
Hi Michael, i have tried both. H323 and OH323. But meanwhile i've found out that the problem is at the teles-pbx. Thanks, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Mittwoch, 25. Juni 2003 15:49 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] no sound pri -- h323 Thomas Haeger wrote: hi all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Which H.323 channel driver do you use? Also, some Asterisk log messages would be more useful. Michael. Can somebody help me? Thanks, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NoOp gives an ringing indication ?
Hi all,
i want lock Zap channels via global var FREE1
if FREE1 = 1 then call should go on with nothing and waiting for digits to
go in _X.
Otherwise hangup the channel
But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication !?
The immediate property in the zapat.conf is yes
[tel1]
exten = s,1,GotoIf($[${FREE1} = 1]?s|4:s|2)
exten = s,2,Playback,gesperrt
exten = s,3,Hangup
exten = s,4,NoOp
exten = _X.,1,Dial,Modem/g1:${EXTEN}
exten = _X.,102,Hangup
What can i do ?
Thanks for help,
Thomas.
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AW: [Asterisk-Users] parsing bug? (using PGSQL)
OK. I see.
This works.
Thank you,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Martin
Pycko
Gesendet: Dienstag, 24. Juni 2003 18:03
An: Asterisk User
Betreff: Re: [Asterisk-Users] parsing bug? (using PGSQL)
If you use brackets () then you need to call it like this
PGSQL(blabla(bla)bla)
That should work
regards
Martin
On Tue, 24 Jun 2003, Thomas Haeger wrote:
Hi all again,
if i make a query with
...
exten = _X.,2,PGSQL,Query resultid ${connid} SELECT
getdest('${EXTEN}');
...
an error like
WARNING[32785]: File pbx.c, Line 1126 (pbx_extension_helper): No
application
'PGSQL,Query resultid ${connid} 'SELECT getdest'
for extension (tel1, 00905888, 2)
occurs.
This looks like an parsing bug. As if the brackets be cut.
Can somebody help?
Thanks,
Thomas.
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[Asterisk-Users] help with pri configuration..
Hi all, can somebody help me with pri configuration? Here my zapata.conf: ; Zapata telephony interface ; ; Configuration file [channels] switchtype=euroisdn signalling=pri_cpe ;group=1 channel = 1-15,17-31 ;group=2 channel =32-46,48-62 ;group=3 channel = 63-77,79-93 ;group=4 channel = 94-108,110-124 And here my zaptel.conf: zaptel.conf [] 0 L:[ 1+ 0 1/ 18] *(0 / 227b)= . 10 0x0A span=1,0,0,ccs,hdb3 #,crc4 span=2,0,0,ccs,hdb3 #,crc4 span=3,0,0,ccs,hdb3 #,crc4 span=4,0,0,ccs,hdb3 #,crc4 bchan=1-15,,32-46,63-77,94-108 dchan=16,47,78,109 bchan=17-31,48-62,79-93,110-124 And here the messages after starting astersik: loadzone = fr defaultzone=us [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Registered channel 3, PRI Signalling signalling -- Registered channel 4, PRI Signalling signalling -- Registered channel 5, PRI Signalling signalling -- Registered channel 6, PRI Signalling signalling -- Registered channel 7, PRI Signalling signalling -- Registered channel 8, PRI Signalling signalling -- Registered channel 9, PRI Signalling signalling -- Registered channel 10, PRI Signalling signalling -- Registered channel 11, PRI Signalling signalling -- Registered channel 12, PRI Signalling signalling -- Registered channel 13, PRI Signalling signalling -- Registered channel 14, PRI Signalling signalling -- Registered channel 15, PRI Signalling signalling -- Registered channel 17, PRI Signalling signalling -- Registered channel 18, PRI Signalling signalling -- Registered channel 19, PRI Signalling signalling -- Registered channel 20, PRI Signalling signalling -- Registered channel 21, PRI Signalling signalling -- Registered channel 22, PRI Signalling signalling -- Registered channel 23, PRI Signalling signalling -- Registered channel 24, PRI Signalling signalling -- Registered channel 25, PRI Signalling signalling -- Registered channel 26, PRI Signalling signalling -- Registered channel 27, PRI Signalling signalling -- Registered channel 28, PRI Signalling signalling -- Registered channel 29, PRI Signalling signalling -- Registered channel 30, PRI Signalling signalling ERROR[1024]: File chan_zap.c, Line 4757 (mkintf): Signalling requested is PRI Signalling but line is in Unkn own signalling 896 signalling ERROR[1024]: File chan_zap.c, Line 6403 (load_module): Unable to register channel '1-15' WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! Whats wrong ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] help with pri configuration..
The problem before is solved. But now gives another problem ... == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Starting D-Channel on span 1 ERROR[1024]: File chan_zap.c, Line 5947 (start_pri): Unable to open D-channel 47 (Device or resource busy) ERROR[1024]: File chan_zap.c, Line 6682 (load_module): Unable to start D-channel on span 2 WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! All channels are registered successfully before. But then this error occur. I tried to deactivate following ports (spans) 1. second one but then the same message occure with the next dchannel and so on. Only the first one works. What's wrong now? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Montag, 23. Juni 2003 11:34 An: Asterisk User Betreff: [Asterisk-Users] help with pri configuration.. Hi all, can somebody help me with pri configuration? Here my zapata.conf: ; Zapata telephony interface ; ; Configuration file [channels] switchtype=euroisdn signalling=pri_cpe ;group=1 channel = 1-15,17-31 ;group=2 channel =32-46,48-62 ;group=3 channel = 63-77,79-93 ;group=4 channel = 94-108,110-124 And here my zaptel.conf: zaptel.conf [] 0 L:[ 1+ 0 1/ 18] *(0 / 227b)= . 10 0x0A span=1,0,0,ccs,hdb3 #,crc4 span=2,0,0,ccs,hdb3 #,crc4 span=3,0,0,ccs,hdb3 #,crc4 span=4,0,0,ccs,hdb3 #,crc4 bchan=1-15,,32-46,63-77,94-108 dchan=16,47,78,109 bchan=17-31,48-62,79-93,110-124 And here the messages after starting astersik: loadzone = fr defaultzone=us [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Registered channel 3, PRI Signalling signalling -- Registered channel 4, PRI Signalling signalling -- Registered channel 5, PRI Signalling signalling -- Registered channel 6, PRI Signalling signalling -- Registered channel 7, PRI Signalling signalling -- Registered channel 8, PRI Signalling signalling -- Registered channel 9, PRI Signalling signalling -- Registered channel 10, PRI Signalling signalling -- Registered channel 11, PRI Signalling signalling -- Registered channel 12, PRI Signalling signalling -- Registered channel 13, PRI Signalling signalling -- Registered channel 14, PRI Signalling signalling -- Registered channel 15, PRI Signalling signalling -- Registered channel 17, PRI Signalling signalling -- Registered channel 18, PRI Signalling signalling -- Registered channel 19, PRI Signalling signalling -- Registered channel 20, PRI Signalling signalling -- Registered channel 21, PRI Signalling signalling -- Registered channel 22, PRI Signalling signalling -- Registered channel 23, PRI Signalling signalling -- Registered channel 24, PRI Signalling signalling -- Registered channel 25, PRI Signalling signalling -- Registered channel 26, PRI Signalling signalling -- Registered channel 27, PRI Signalling signalling -- Registered channel 28, PRI Signalling signalling -- Registered channel 29, PRI Signalling signalling -- Registered channel 30, PRI Signalling signalling ERROR[1024]: File chan_zap.c, Line 4757 (mkintf): Signalling requested is PRI Signalling but line is in Unkn own signalling 896 signalling ERROR[1024]: File chan_zap.c, Line 6403 (load_module): Unable to register channel '1-15' WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! Whats wrong ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] where to get adsi phones in europe ?
Hi all, have anybody an idea where to get adsi phones in europe ? Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] number of digits from incoming msn on i4l modem
Hi all, is it possible to define the number of digits in the modem.conf ? I think now it is so that i can set a * for any number and concrete numbers for defined numbers. But, i want to define the modem so that it goes into the extensions if the incoming number have a defind number of digits. Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with outgoing spool...
Hi all, i 've written a little Callgen script for generating calls through the outgoing spool directory. The calls goes over 8 ttyI devices to another pbx and come in through other 8 ttyI devices. But when i generate the calls, sometimes * register the calls but never initiate them. Especially when the files come to fast into the outgoing dir. What can be wrong ? Is it possible that the i4l driver have a bug ? Or maybe the spooler is defective ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Problem with outgoing spool...
OK, sorry for my deficient description... Scenario is as follwes: One 4 BRI card - ttyI0 - ttyI7 for outgoing second 4 BRI card - ttyI8 - ttyI15 for Incoming My script generates 8 calls with a distance from 2 seconds. The files in the outgoing dir will be generated with an echo command. If the distance is smaller than 2 seconds sometimes calls never be initiated but i can see the verbose messages for the calls. Then i can see channels if i enter show channels in the CLI with state DOWN. But nothing else matters. This is the point in time i have to restart * and my script. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 13. Juni 2003 15:23 An: Asterisk User Betreff: [Asterisk-Users] Problem with outgoing spool... Hi all, i 've written a little Callgen script for generating calls through the outgoing spool directory. The calls goes over 8 ttyI devices to another pbx and come in through other 8 ttyI devices. But when i generate the calls, sometimes * register the calls but never initiate them. Especially when the files come to fast into the outgoing dir. What can be wrong ? Is it possible that the i4l driver have a bug ? Or maybe the spooler is defective ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] s extension don't work on TDM40B
Hi all, i have read in the * whitepaper the following: s: The start extension. A call which does not have digits associated with it (for example, a loopstart analog line) begins at the s extension. I think this means the s extension will be execute when the phone is picked up. But when i pick up the phone the s extension will be never executed. Whats wrong ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
