Re: [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine
I do not think so. Nuance Vocalizer follows RealSpeak (4.5) which based on some older products. I never have used RealSpeak on the command line, but I think the standard tool is what you should be looking for. Some years ago Nuance bought Rhetorical with their rVoice TTS. This had a command line utility named tts-it on board. Hope this helps. Regards, Christophorus 2009/10/21 Christophorus Laube christophorus.la...@semanticedge.de mailto:christophorus.la...@semanticedge.de I think you should use the nvcmdline utility Is this nvcmdline bundled with every Nuance TTS ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 4531 (20091022) Information __ Diese E-Mail wurde vom NOD32 antivirus system geprüft http://www.nod32.com -- Christophorus Laube Systemadministrator christophorus.la...@semanticedge.de SemanticEdge GmbH Kaiserin-Augusta-Allee 10-11 10553 Berlin Deutschland Tel +49-30-345077-58 Fax +49-30-345077-77 http://www.semanticedge.de Geschäftsführer : Dr.Ralf Köhrbrück, Dr. Lupo Pape HRB 84682 Amtsgericht Charlottenburg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine
Hi, I think you should use the nvcmdline utility to synthesize your prompt to a certain file to be specified. Afterwards, you could play that on your asterisk, for example a wav file. But this could be some kind of long lasting as the TTS synthesizes in realtime, i.e. the longer the prompt is the longer you have to wait for the file to play. So, using AGI should be worthwile to take a look at. Using the nvcmdline utility you should use bash AGI or something more scripty. If there is a Java API for Nuance Vocalizer (I do not know that) you also could use that. Regards, Christophorus Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cascaded pickup
Hi list,
I am struggeling with getting an out-of-office redirected call picked
up. That is if one extension gets called and is not picked up in a
certain time the call is redirected to the central phone by local
channel. Both times when the Dial command is executed a pickup mark is
set before:
exten = XXX,1,Set(PICKUPMARK=${EXTEN:6})
exten = XXX,2,Dial(SIP/${EXTEN}Local/centralph...@e1,,tT)
exten = centralphone,1,Wait(15)
exten = centralphone,n,Set(PICKUPMARK=0)
exten = centralphone,n,Dial(SIP/XXY,,tT)
If someone from another extension tries to pickup the original extension
it works. But if the centralphone starts ringing it is not possible to
pickup the centralphone extension.
Can anyone of you imagine what my mistake is in that case?
Regards, Christophorus
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[asterisk-users] asterisk and mISDN on Solaris
Hi, I read that installing asterisk on Solaris is supported. Does anyone of you actually have experiences with that? And especially, does anyone of you have experiences in runnning asterisk with misdn unter Solaris? Thanks and regards, Christophorus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.6 and mISDN
Hi on the list, does anyone of you have experience with asterisk 1.6 and mISDN, pri primarily? Thanks in advance Regards, Christophorus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ANI with Pickup application
Hi, does anyone of you have made it to get the ANI also picked up? I mean: if I fetch a foreign call to me by using the pickup application I want to see the callerID/ANI of the caller to the foreign extension. Is that possible and if yes - how do I achieve that? Regards, Christophorus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ANI with Pickup application
Hallo Ralf, das ist die Antwort von der Liste. Klingt etwas vage und nicht absolut erfolgversprechend... Derzeit ist kein Upgrade auf Asterisk 1.6 geplant und nach allem, was ich darüber bisher gelesen habe, kann die Umstellung auch etwas größer werden (Wählplansyntax etc.). Beronet empfiehlt derzeit auch nach wie vor 1.4, von 1.6 auf Produktivsystemen wird abgeraten. Gruß, Christophorus 2009/3/16 Christophorus Laube christophorus.la...@semanticedge.de Hi, does anyone of you have made it to get the ANI also picked up? I mean: if I fetch a foreign call to me by using the pickup application I want to see the callerID/ANI of the caller to the foreign extension. Is that possible and if yes - how do I achieve that? using SIP P-asserted ids and asterisk 1.6.1, this shoulld be possible to get CallerID (I've never tried it yet). Regards, Christophorus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dipl.-Ling. Christophorus Laube Systemadministrator SemanticEdge GmbH Kaiserin-Augusta-Allee 10-11 10553 Berlin Deutschland Tel +49-30-345077-58 Fax +49-30-345077-77 christophorus.la...@semanticedge.de Geschäftsführer : Dr.Ralf Köhrbrück, Dr. Lupo Pape HRB 84682 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no dial to busy sip line
Hi,
thanks a lot. That helped me going most of my intended way. The only
thing is it still calls even busy lines (shown in use by show
queues) with either roundrobin (which is marked deprecated) or rrmemory
method. Did I miss something while reading howtos?
Thanks in advance. Regards, Christophorus
How about a call queue using the roundrobin strategy?
http://www.voip-info.org/wiki/view/Asterisk+call+queues
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus
Laube
Sent: Friday, November 14, 2008 11:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] no dial to busy sip line
Hi list,
is it possible to get in the running dialplan the status of (SIP) lines
without using AGI or anything like that? What I want is a stepwise
calling: I have several SIP lines (let's say they are three) which I
want to dial to alternatingly. But I do not want to dial to a already
busy line and catch the busy. Instead I do not want to dial to that peer
but to the next one. I want to have a kind of a adaptive dialplan.
Using AGI and such things just makes it slower in my opinion (if I call
an AGI script that does an asterisk -rx 'sip show channels' |gawk -F
{' print $1 '}, for example). Does anyone of you have an idea of how
to do that?
Thanks in advance. Best regards,
Christophorus Laube
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[asterisk-users] no dial to busy sip line
Hi list,
is it possible to get in the running dialplan the status of (SIP) lines
without using AGI or anything like that? What I want is a stepwise
calling: I have several SIP lines (let's say they are three) which I
want to dial to alternatingly. But I do not want to dial to a already
busy line and catch the busy. Instead I do not want to dial to that peer
but to the next one. I want to have a kind of a adaptive dialplan.
Using AGI and such things just makes it slower in my opinion (if I call
an AGI script that does an asterisk -rx 'sip show channels' |gawk -F
{' print $1 '}, for example). Does anyone of you have an idea of how
to do that?
Thanks in advance. Best regards,
Christophorus Laube
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[asterisk-users] Cisco 79X1 speaker issue
Hi list, I have a couple of Cisco 79X1 running well behind an asterisk. My question is more dedicated to the ones of you knowing some tricks with these phones. Does anyone of you know if there is a possibility to use the speaker and the microphone of the handset? For now, when I activate the speaker the microphone of the handset is forced off and cannot be used anymore. Thanks in advance. Regards, Christophorus Laube ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
I do have to answer to your suggestion of renaming the CTLSEPmac.tlv to SEPmac. The phone is still requesting CTLSEPmac.tlv and as it cannot find that it goes into a loop. I also let the phone do that the whole weekend so there should be no iterative process in requesting the files as I read in some howtos. Any further ideas? I also read that it is possible to connect and configure the phone by ssh. So after flashing the phone with a SIP image there should be some default username/password combination which I did not manage to find out yet. Does anyone know? I now am going to revert to an older release to try that. I will report any success as well as misses. Thanks again, Christophorus This should result in the same problem. The CTLSEPmac file is the first that is requested on the TFTP server. But I am going to try that. Regards and thanks, Christophorus Try naming the empty file: SEP0019E7D16CD6.tlv Not CTLSEP0019E7D16CD6.tlv -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus Laube Sent: Friday, January 04, 2008 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Thanks for the hint. I just tried that although I only see my worries coming true: the CTLSEPmac.tlv file is the first one the phone requests when booting, no possibility to set something different as the SEPmac.cnf.xml should be loaded after the successful load of the CTL file. And thus the phone is looping with Configuring IP and CTLFile failure. Can I set this option by ssh? Thanks a lot and in advance, Christophorus In your SEPmac.cnf.xml file look for the setting below and set it to 0: deviceSecurityMode0/deviceSecurityMode -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn Cobb Sent: Friday, January 04, 2008 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Here is a little more info... I hooked up the 7971G-GE to my pc and grabbed this with tera-term. Its the console output during the CTL update process. I am using SIP70.8-3-3. NOT 09:28:45.969295 DHCP: Restart - delay = 1 NOT 09:28:45.981198 DHCP: Sending Release... NOT 09:28:49.000449 DHCP: dhcpSendReq: status 0x12301000 NOT 09:28:49.001281 DHCP: Sending Request... NOT 09:28:49.015673 DHCP: ACK received NOT 09:28:49.016517 DHCP: Succeeded NOT 09:28:49.058273 DHCP: IP Address -- 10.10.10.247 NOT 09:28:49.059129 DHCP: Subnet Mask - 255.255.255.0 NOT 09:28:49.059960 DHCP: Default Gwy - NOT 09:28:49.073169 PAE: SIGIPCFG received... NOT 09:28:49.075897 ESP: send ADMIN, logging = 1, shell = 0, ipconfig = 1 WRN 09:28:49.120127 SECD: WARN:getCTLInfo: ** phone has no CTL WRN 09:28:49.127292 SECD: WARN:getCTLInfo: ** phone has no CTL NOT 09:28:49.140946 CDP-D: catchipcfg getdhcpinfo IP: a0a0af7 Chng:1 NOT 09:28:49.152532 tftpClient: request server 0 --- 10.10.10.10 NOT 09:28:49.178685 tftpClient: request server 1 --- NOT 09:28:49.201261 tftpClient: request server 0 --- 10.10.10.10 NOT 09:28:49.204518 ESP: server 0 = 10.10.10.10 NOT 09:28:49.228784 tftpClient: request server 1 --- NOT 09:28:49.233253 ESP: server 1 = NOT 09:28:49.319960 SECD: updateCTL: starting CTL update NOT 09:28:49.323284 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:28:49.324525 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:28:49.327942 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:28:49.331598 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:28:49.332439 tftpClient: look up server - 0 WRN 09:28:49.335498 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:28:49.339140 tftpClient: secVal = 0xa NOT 09:28:49.340260 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:28:49.341141 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:28:49.341897 tftpClient: retval = 10 NOT 09:28:49.342678 tftpClient: Non secure file requested NOT 09:28:49.356155 TFTP: [26]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:28:49.359594 TFTP: [26]:Finished -- rcvd 1 bytes NOT 09:28:49.363943 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:28:49.365631 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:28:49.367522 SECD: updateCTL: finished CTL update ERR 09:28:49.368469 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:28:53.768028 SECD: updateCTL: starting CTL update NOT 09:28:53.772517 SECD: ctlRequestFile: Socket 7
Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
Update and revision: I now downloaded the oldest gettable SIP firmware for 7941/61, i.e. 8.0.2. I always get the same behaviour. But I realized it never got to the SIP image completely loaded status. I bought this phone and it had - no wonder - an SCCP image installed. When plugging that into an ethernet port the first thing it does is requesting an IP address and afterwards the CTLSEPmac.tlv file. In the status section I see an SCCP firmware entry. When I do a factory reset (that should be the right way to get the SIP firmware on such a phone, right?) it now loads the term41.default.loads and some other files and then reboots and requests the CTLSEPmac.tlv file. The firmware entry in the status section now says term41.default.loads. Getting over this CTLSEP step should bring the phone to load the SIP41XXX.loads file, I assume. But as I am not getting over this step it stays in the term41.default.loads step, unfortunately. Does that ring a bell to anyone? Does anyone of you have had the same situation? In which state did you get the 7961G? SCCP? And how did you manage to load SIP firmware onto it? Christophorus I do have to answer to your suggestion of renaming the CTLSEPmac.tlv to SEPmac. The phone is still requesting CTLSEPmac.tlv and as it cannot find that it goes into a loop. I also let the phone do that the whole weekend so there should be no iterative process in requesting the files as I read in some howtos. Any further ideas? I also read that it is possible to connect and configure the phone by ssh. So after flashing the phone with a SIP image there should be some default username/password combination which I did not manage to find out yet. Does anyone know? I now am going to revert to an older release to try that. I will report any success as well as misses. Thanks again, Christophorus This should result in the same problem. The CTLSEPmac file is the first that is requested on the TFTP server. But I am going to try that. Regards and thanks, Christophorus Try naming the empty file: SEP0019E7D16CD6.tlv Not CTLSEP0019E7D16CD6.tlv -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus Laube Sent: Friday, January 04, 2008 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Thanks for the hint. I just tried that although I only see my worries coming true: the CTLSEPmac.tlv file is the first one the phone requests when booting, no possibility to set something different as the SEPmac.cnf.xml should be loaded after the successful load of the CTL file. And thus the phone is looping with Configuring IP and CTLFile failure. Can I set this option by ssh? Thanks a lot and in advance, Christophorus In your SEPmac.cnf.xml file look for the setting below and set it to 0: deviceSecurityMode0/deviceSecurityMode -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn Cobb Sent: Friday, January 04, 2008 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Here is a little more info... I hooked up the 7971G-GE to my pc and grabbed this with tera-term. Its the console output during the CTL update process. I am using SIP70.8-3-3. NOT 09:28:45.969295 DHCP: Restart - delay = 1 NOT 09:28:45.981198 DHCP: Sending Release... NOT 09:28:49.000449 DHCP: dhcpSendReq: status 0x12301000 NOT 09:28:49.001281 DHCP: Sending Request... NOT 09:28:49.015673 DHCP: ACK received NOT 09:28:49.016517 DHCP: Succeeded NOT 09:28:49.058273 DHCP: IP Address -- 10.10.10.247 NOT 09:28:49.059129 DHCP: Subnet Mask - 255.255.255.0 NOT 09:28:49.059960 DHCP: Default Gwy - NOT 09:28:49.073169 PAE: SIGIPCFG received... NOT 09:28:49.075897 ESP: send ADMIN, logging = 1, shell = 0, ipconfig = 1 WRN 09:28:49.120127 SECD: WARN:getCTLInfo: ** phone has no CTL WRN 09:28:49.127292 SECD: WARN:getCTLInfo: ** phone has no CTL NOT 09:28:49.140946 CDP-D: catchipcfg getdhcpinfo IP: a0a0af7 Chng:1 NOT 09:28:49.152532 tftpClient: request server 0 --- 10.10.10.10 NOT 09:28:49.178685 tftpClient: request server 1 --- NOT 09:28:49.201261 tftpClient: request server 0 --- 10.10.10.10 NOT 09:28:49.204518 ESP: server 0 = 10.10.10.10 NOT 09:28:49.228784 tftpClient: request server 1 --- NOT 09:28:49.233253 ESP: server 1 = NOT 09:28:49.319960 SECD: updateCTL: starting CTL update NOT 09:28:49.323284 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:28:49.324525 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:28:49.327942 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock
[asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEPmac.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Does anyone of you have experience in getting these phones to work or can point me to any information bringing me back in the game? Thanks in advance, Christophorus Laube ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
As I see it this matrix is only for the 79X0 generation, right? Every howto I found said that it would be no problem to have the CTLSEP file empty. I just tried to build up an empty file on Windows but that did not help. So my problem is that every howto is proposing that this will work with an empty file but in fact it does not work for me. I have SIP firmware 8.3.3-SR2. Do you have any experience with this image? Greetings, Christophorus I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without any problems. As far as the CTLSEP File (Straight from Cisco): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i pp7960/addprot/mgcp/frmwrup.htm#wp1047292 The CTLSEP MAC file is a certificate trust list, which if populated, contains information about the servers to which the phone is attempting to connect and whether the server connection will be secure or nonsecure. Based on the information above an empty file will work just fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Friday, January 04, 2008 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee On Fri, 2008-01-04 at 09:11 +0100, Christophorus Laube wrote: Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEPmac.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Does anyone of you have experience in getting these phones to work or can point me to any information bringing me back in the game? Thanks in advance, I don't remember if I had this same problem with a 7961G but I did figure out that you can not do an upgrade from factory default SCCP to the latest SIP 8.x.x firmware. In my case the phone just did not work properly. To make it work I downgraded the phone back to SIP 7.x firmware (iirc I used 7.5) and then upgraded to the latest SIP 8.x.x firmware. Hope this helps. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:28:58.205791 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:28:58.208403 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:28:58.209244 tftpClient: look up server - 0 WRN 09:28:58.215701 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:28:58.219254 tftpClient: secVal = 0xa NOT 09:28:58.220320 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:28:58.221096 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:28:58.221849 tftpClient: retval = 10 NOT 09:28:58.222629 tftpClient: Non secure file requested NOT 09:28:58.235315 TFTP: [16]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:28:58.238209 TFTP: [16]:Finished -- rcvd 1 bytes NOT 09:28:58.241145 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:28:58.242856 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:28:58.244754 SECD: updateCTL: finished CTL update ERR 09:28:58.245704 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:02.648053 SECD: updateCTL: starting CTL update NOT 09:29:02.651331 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:29:02.652499 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:29:02.658547 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:29:02.661503 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:29:02.662335 tftpClient: look up server - 0 WRN 09:29:02.665405 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:29:02.668874 tftpClient: secVal = 0xa NOT 09:29:02.669746 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:29:02.671475 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:29:02.672277 tftpClient: retval = 10 NOT 09:29:02.673060 tftpClient: Non secure file requested NOT 09:29:02.684870 TFTP: [25]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:29:02.687805 TFTP: [25]:Finished -- rcvd 1 bytes NOT 09:29:02.691794 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:29:02.693428 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:29:02.695315 SECD: updateCTL: finished CTL update ERR 09:29:02.696335 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:03.227508 DHCP: Restart - delay = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Friday, January 04, 2008 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without any problems. As far as the CTLSEP File (Straight from Cisco): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon /english/i pp7960/addprot/mgcp/frmwrup.htm#wp1047292 The CTLSEP MAC file is a certificate trust list, which if populated, contains information about the servers to which the phone is attempting to connect and whether the server connection will be secure or nonsecure. Based on the information above an empty file will work just fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Friday, January 04, 2008 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee On Fri, 2008-01-04 at 09:11 +0100, Christophorus Laube wrote: Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEPmac.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Does anyone of you have experience in getting these phones to work or can point me to any information bringing me back in the game? Thanks in advance, I don't remember if I had this same problem with a 7961G but I did figure out that you can not do an upgrade from factory default SCCP to the latest SIP 8.x.x firmware. In my case the phone just did not work properly. To make it work I downgraded the phone back to SIP 7.x firmware (iirc I used 7.5) and then upgraded to the latest SIP 8.x.x firmware. Hope this helps. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk
Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
This should result in the same problem. The CTLSEPmac file is the first that is requested on the TFTP server. But I am going to try that. Regards and thanks, Christophorus Try naming the empty file: SEP0019E7D16CD6.tlv Not CTLSEP0019E7D16CD6.tlv -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus Laube Sent: Friday, January 04, 2008 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Thanks for the hint. I just tried that although I only see my worries coming true: the CTLSEPmac.tlv file is the first one the phone requests when booting, no possibility to set something different as the SEPmac.cnf.xml should be loaded after the successful load of the CTL file. And thus the phone is looping with Configuring IP and CTLFile failure. Can I set this option by ssh? Thanks a lot and in advance, Christophorus In your SEPmac.cnf.xml file look for the setting below and set it to 0: deviceSecurityMode0/deviceSecurityMode -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn Cobb Sent: Friday, January 04, 2008 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Here is a little more info... I hooked up the 7971G-GE to my pc and grabbed this with tera-term. Its the console output during the CTL update process. I am using SIP70.8-3-3. NOT 09:28:45.969295 DHCP: Restart - delay = 1 NOT 09:28:45.981198 DHCP: Sending Release... NOT 09:28:49.000449 DHCP: dhcpSendReq: status 0x12301000 NOT 09:28:49.001281 DHCP: Sending Request... NOT 09:28:49.015673 DHCP: ACK received NOT 09:28:49.016517 DHCP: Succeeded NOT 09:28:49.058273 DHCP: IP Address -- 10.10.10.247 NOT 09:28:49.059129 DHCP: Subnet Mask - 255.255.255.0 NOT 09:28:49.059960 DHCP: Default Gwy - NOT 09:28:49.073169 PAE: SIGIPCFG received... NOT 09:28:49.075897 ESP: send ADMIN, logging = 1, shell = 0, ipconfig = 1 WRN 09:28:49.120127 SECD: WARN:getCTLInfo: ** phone has no CTL WRN 09:28:49.127292 SECD: WARN:getCTLInfo: ** phone has no CTL NOT 09:28:49.140946 CDP-D: catchipcfg getdhcpinfo IP: a0a0af7 Chng:1 NOT 09:28:49.152532 tftpClient: request server 0 --- 10.10.10.10 NOT 09:28:49.178685 tftpClient: request server 1 --- NOT 09:28:49.201261 tftpClient: request server 0 --- 10.10.10.10 NOT 09:28:49.204518 ESP: server 0 = 10.10.10.10 NOT 09:28:49.228784 tftpClient: request server 1 --- NOT 09:28:49.233253 ESP: server 1 = NOT 09:28:49.319960 SECD: updateCTL: starting CTL update NOT 09:28:49.323284 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:28:49.324525 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:28:49.327942 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:28:49.331598 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:28:49.332439 tftpClient: look up server - 0 WRN 09:28:49.335498 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:28:49.339140 tftpClient: secVal = 0xa NOT 09:28:49.340260 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:28:49.341141 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:28:49.341897 tftpClient: retval = 10 NOT 09:28:49.342678 tftpClient: Non secure file requested NOT 09:28:49.356155 TFTP: [26]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:28:49.359594 TFTP: [26]:Finished -- rcvd 1 bytes NOT 09:28:49.363943 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:28:49.365631 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:28:49.367522 SECD: updateCTL: finished CTL update ERR 09:28:49.368469 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:28:53.768028 SECD: updateCTL: starting CTL update NOT 09:28:53.772517 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:28:53.773673 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:28:53.776093 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:28:53.778770 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:28:53.779616 tftpClient: look up server - 0 WRN 09:28:53.782887 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:28:53.786443 tftpClient: secVal = 0xa NOT 09:28:53.787250 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:28:53.788022 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:28:53.788777 tftpClient: retval = 10 NOT 09:28:53.789616 tftpClient: Non secure file
[asterisk-users] SIP_INFO
Hi list, does anyone of you know wether asterisk can handle SIP_INFO on pure sip calls? Is that something I have to handle in the extensions? Does asterisk hand incoming SIP_INFO over to an already connected peer? Thanks and regards, Christophorus Laube ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AAI2UUI - how?
Hi list, on my asterisk machine I have an E1 (Beronet with chan_misdn) board and sip clients connected. I am getting some AAI (application-to-application-information, enriched SIP header, similar to the SipAddHeader application) from a sip client during the BYE method. I want to give this AAI to my ISDN line as UUI (user-to-user-information) during ISDN Hangup. Doing that with the SipGetHeader application is not possible as this is only allowed on incoming SIP calls. Is there a possibility I can customize my cdr in a manner that logs this AAI and I can strip that in the hangup extensions from the cdr to set the MISDN_USERUSER variable and write UUI? TIA and Regards, Christophorus ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel/mISDN and call transfer
Tzafrir Cohen schrieb: On Sat, Jul 14, 2007 at 01:23:35PM +0200, Christophorus Laube wrote: Hi list, I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and the channels are free again. Is it possible and with what card vendor (mISDN vs.zaptel) and how do I do that? Thanks in advance, I found an old feature-request bug in Zaptel which seems relevant: http://bugs.digium.com/3554 Not sure if this means that the feature is supported. Maybe ask Mathew Fredrikson or Digium support. by the way: Is this call deflection or ECT etc. only possible to be executed at ring time or can I redirect a yet running call? Thanks, Christophorus ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel/mISDN and call transfer
Hi list, I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and the channels are free again. Is it possible and with what card vendor (mISDN vs.zaptel) and how do I do that? Thanks in advance, Christophorus ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel/mISDN and call transfer
Philipp von Klitzing schrieb: Hi! I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). They keyword to search for is explicit call transfer (ECT). At least chan_capi-com (http://www.melware.org/ChanCapi) comes with support for that. Don't know about mISDN. Cheers, Philipp Thanks, but can I use chan_capi as frontend to mISDN or zaptel hardware? As I know I do have to choose between digium or beronet/junghanns hardware (E1) to use PRI with asterisk, right? Oh, I just caugh that I did not mention that before...sorry. Do I have to use chan_capi to access the zaptel hardware? Regards, Christophorus ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no special context for sip peer
Hi list, I want to set up special contexts for every sip user. But a context=XYZ does not help in the perr definition as I have to provide a context in the general section of sip.conf. This is my sip.conf: [general] port=5060 bindaddr=192.168.0.75 disallow=all allow=ulaw allow=alaw context=SIP maxexprirey=3600 defaultexpirey=120 language=de pritrustusercid=yes callerid=asreceived [bob] type=peer username=bob host=dynamic secret=nothing context=BOB_SIP qualify=yes canreinvite=yes callingpres=allowed_passed_screen So what am I doing wrong? What do I have to change in order to get my BOB_SIP extensions to work when I am doing a call from this peer? Now * always takes the default context SIP. Regards, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Servers
That depends on your distro. I have tested * with Beronet cards on OpenSuSE 10, Debain Sarge and Ubuntu Edgy. What has to be blacklisted is every remainder of old ISDN stuff and hotplug modules (*php = * pci hot plug). As far as I know these cards are not hotpluggable at all and who wants to have a hotplug telephony gateway? I also experienced asterisk crashes because of hotplug modules. But I have to admit that this was on an IBM machine. The responsible kernel module (I cannot remember which one that was) was polling for new pci hardware every now and then and that was something the card didn't like so it unloaded the mISDN kernel module and crashed asterisk. On ubuntu (2950) I did not have to blacklist anything. On OpenSuSE (2850) I had to remove the complete old ISDN stuff. Regards, Christophorus On Thu, 1 Feb 2007, Christophorus Laube wrote: We have a 2850 in a productive environment with a BNE1 performing well (OpenSuSE 10) and a 2950 with BNE1 and BN8S0 also performing OK (on Ubuntu Edgy). You only have to blacklist some hotplug kernel modules and yes, we do have very long pings (1 ping per week with a check rate of 10min per SNMP). But that does happen very rare and I never noticed any dropped calls or bad audio quality. The 2850 is running on SCSI, the 2950 on an SAS RAID. In general I like the Dell machines, also with asterisk on them. The only thing is that Openmanage ist quite bad to install but that's nothing asterisk specific but linux related. Does that help? Would you be willing to share your blacklist for the kernel modules? Thanks!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Servers
We have a 2850 in a productive environment with a BNE1 performing well (OpenSuSE 10) and a 2950 with BNE1 and BN8S0 also performing OK (on Ubuntu Edgy). You only have to blacklist some hotplug kernel modules and yes, we do have very long pings (1 ping per week with a check rate of 10min per SNMP). But that does happen very rare and I never noticed any dropped calls or bad audio quality. The 2850 is running on SCSI, the 2950 on an SAS RAID. In general I like the Dell machines, also with asterisk on them. The only thing is that Openmanage ist quite bad to install but that's nothing asterisk specific but linux related. Does that help? best regards, Christophorus Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? I bought a Dell 2850 as a pbx server and it just sucks IMHO The stupid thing has only 3 pci slots and even with only 3 pci slots Dell managed to have a shared irq on every slot, 1 for the scsi controller and one for each nic The result of this 'nice' piece of work is dreadfull irq hit/miss results in zttest, it barely meets the minimum requirement and i do get complaints of dropped calls on my pri I need to pass some options to the kernel at boot time to improve things, without extra options the results from zttest were unacceptable My spare pbx is a lowly Athlon XP 2600 with an Asus A7V8X-X mobo in it and it's scores with zttest are considerably better (but not full 100% hits) I know that everybody on the list will now start recommending me to buy Sangoma hardware but firstly I hate compiling extra modules and it doesn't make it right that the Dell hardware just sucks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long busy()
[Description]
Busy([timeout]): This application will indicate the busy condition to
the calling channel. If the optional timeout is specified, the calling
channel
will be hung up after the specified number of seconds. Otherwise, this
application will wait until the calling channel hangs up.
This is what I found when I typed show application busy in the CLI.
Did I interpret it wrong?
regards, Christophorus
Mailinglisten schrieb:
Christophorus Laube schrieb:
hi list,
I set up a new asterisk machine with asterisk 1.2.13 and misdn
0.3.1rc27.
I use an e1 card with sip clients. My extensions look like this:
[E1]
snip...snip
exten = 33006733,1,Set(CALLED=${EXTEN})
exten = 33006733,2,Dial(SIP/[EMAIL PROTECTED])
exten = 33006733-ANSWER,3,Answer()
[SIP]
exten = _X.,1,Noop()
exten = _X.,2,SetCallerPres(allowed_passed_screen)
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40)
exten = _X.-BUSY,4,Busy(1)
But whenever a sip client calls to an exten that is busy through e1 I
get busy tones for 10s before I get disconnected. But I want to have
it only for 1s.
Does anyone know how to fix that?
regards, Christophorus
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AFAIK the BUSY() command has nothing to do with the busy indication.
You can't pass anything to this command.
Check: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Busy
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[asterisk-users] long busy()
hi list,
I set up a new asterisk machine with asterisk 1.2.13 and misdn 0.3.1rc27.
I use an e1 card with sip clients. My extensions look like this:
[E1]
snip...snip
exten = 33006733,1,Set(CALLED=${EXTEN})
exten = 33006733,2,Dial(SIP/[EMAIL PROTECTED])
exten = 33006733-ANSWER,3,Answer()
[SIP]
exten = _X.,1,Noop()
exten = _X.,2,SetCallerPres(allowed_passed_screen)
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40)
exten = _X.-BUSY,4,Busy(1)
But whenever a sip client calls to an exten that is busy through e1 I get busy
tones for 10s before I get disconnected. But I want to have it only for 1s.
Does anyone know how to fix that?
regards, Christophorus
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[asterisk-users] lines usage statistics
Hi list, I want to make a statistics about the number of parallel calls on my * running a beronet E1 card. The easy variant would be to get a number of maximal parallel calls to my machine during a day. The extended would be a graph showing the load over the day. If noone knows a direct solution to my question I would have an idea how to make up the easy variant with extensions. The only thing I would be missing for that would be a way to read in the current date and time. The application DateTime does not what I first thought it would. So, does anyone of you know how to get it? Backticks is not working on that machine. I am running asterisk 1.2.7.1with chan_misdn 0.3.1rc17. I know this is not absolutely up to date but I cannot afford a longer downtime. TIA regards, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] read variable from shell script
Hi list, is it possible to call a shell script from * which returns a number or a string which can be read to an asterisk variable? Something like 'Set(VAR(System(/opt/scripts/something.script)))? Does anyone have an idea? Regards, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk logging per day
Hi list, I am searching for a possibility to let my * log per day. So that a new logfile is taken every night at midnight, with the date in the file name. Is there a way to do so? Does anyone of you has tried that before? Regards, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transfer call von D-channel
Hi list, how can I realize explicit call transfer? I want to transfer a call which I answered to another phone and it the other one answers I want to hang up so that my resources are freed. Is that possible with Zaptel or which channel can I use else? TIA, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 0000491...
Hi list, I had a call on my * from a mobile number that starts with the international dial prefix (0049), but with four leading zeroes. Does anyone know why this is happening and wether it indicates a certain situation? Regards, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] manager DBDel action
Hi list, is there a possibility to delete a key from the astdb through the manager interface? I managed to put and to get a key but I do not know how to delete an entry. The problem is that I want to use the manager interface because I can communicate remotely with my * this way. TIA, Christophorus begin:vcard fn:Christophorus Laube n:Laube;Christophorus org:SemanticEdge GmbH adr:;;Kaiserin-Augusta-Allee 10-11;Berlin;;10553;Deutschland email;internet:[EMAIL PROTECTED] title:Systemadministrator tel;work:+49-30-34507758 url:http://www.semanticedge.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remote setting - AGI or what?
Hi all, I want to do some stuff in astdb by remotely populating it. That means I want to make a kind of away-handling. A caller can specify a number on which he will be available until he resets this. I thought about doing that with the astdb by setting the the $EXTEN as key and the to-be-dialed number als value. If a call is placed the db is checked and the away number is called in place. But I want to set the number remotely and client initiated. Is AGI able to do such things or what can I use else? TIA, Christophorus begin:vcard fn:Christophorus Laube n:Laube;Christophorus org:SemanticEdge GmbH adr:;;Kaiserin-Augusta-Allee 10-11;Berlin;;10553;Deutschland email;internet:[EMAIL PROTECTED] title:Systemadministrator tel;work:+49-30-34507758 url:http://www.semanticedge.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] who is the mantainer ....
Have a look the sources of misdn... http://www.beronet.com and there should also be a link to their bug system. of chan_misdn ? I found a bug, and I don't know where to report it. Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Christophorus Laube n:Laube;Christophorus org:SemanticEdge GmbH adr:;;Kaiserin-Augusta-Allee 10-11;Berlin;;10553;Deutschland email;internet:[EMAIL PROTECTED] title:Systemadministrator tel;work:+49-30-34507758 url:http://www.semanticedge.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to delete a key from database in extensions.conf
show application DBdel on the CLI. OK this is deprecated but it still works. Maybe asterisk gives you hints what do use now. Doug Lytle schrieb: Shaun wrote: I can set a family/key=value just fine, but how can i delete it? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DBdel -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Christophorus Laube n:Laube;Christophorus org:SemanticEdge GmbH adr:;;Kaiserin-Augusta-Allee 10-11;Berlin;;10553;Deutschland email;internet:[EMAIL PROTECTED] title:Systemadministrator tel;work:+49-30-34507758 url:http://www.semanticedge.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BN8S0 problem - Extension can never match, so disconnecting
I do not see the problem exactly. Most of the messages tell about the layer 1 and 2 activation: P[ 5] MGMT: SSTATUS: L1_ACTIVATED ... P[ 5] MGMT: SSTATUS: L2_ESTABLISH First thing you should check is a misdn show stacks on the CLI. There should be a lot of UPs if the card has physical connection and ISDN recognized. Extension can never match, so disconnecting This is something I would spend more attention to. exten = 465670127,1,Dial(SIP/200) This is even too special for your first tests. Try something more lax: exten = _X.,1,Dial(SIP/200) This way every call coming to the card is directed to the SIP client. When this is working you see which number are dialed (the dad values in the CLI) and so you can adjust your extensions. By the way beginning a function test with a Dial is not so good. Try exten = _X.,1,Noop(nothing is happening here!) or even better exten = _X.,1,Playback(demo-thanks) Have a nice try. Cheers, Christophorus hi i've configured a Beronet BN8S0 Card with misdn beronet utility. the card is configured with all lines in TE and P2P mode, and it is connected with the special cable with an ISDN connection. i've turned on debugging to level 4, this is the output from the asterisk cli when i receive a call: P[ 5] MGMT: Short status dinfo 101 P[ 5] MGMT: SSTATUS: L1_ACTIVATED P[ 5] handle_frm: frm-addr:42000503 frm-prim:3f082 P[ 5] handle_frm: frm-addr:42000503 frm-prim:30582 P[ 5] set_channel: bc-channel:0 channel:1 P[ 5] I IND :SETUP oad:4656708 dad:465670127 P[ 5] -- mode:TE cause:16 ocause:16 rad: cad: P[ 5] -- facility:FAC_NONE out_facility:FAC_NONE P[ 5] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 P[ 5] -- screen:0 -- pres:0 P[ 5] -- channel:1 caps:Speech pi:0 keypad: P[ 5] -- urate:0 rate:16 mode:0 user1:0 P[ 5] -- pid:0 addr:0 l3id:a0001 P[ 5] -- b_stid:0 layer_id:0 P[ 5] -- bc:821d7ac h:0 sh:0 P[ 5] -- bc_state:BCHAN_CLEANED P[ 5] -- Bearer: Speech P[ 5] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:465670127 oad:4656708 P[ 5] -- CTON: Unknown P[ 5] * Queuing chan 0x8285bf0 Jun 8 19:32:29 WARNING[5019]: chan_misdn.c:4390 chan_misdn_log: Extension can never match, so disconnecting P[ 5] Tone Indicate: P[ 5] -- Busy P[ 5] misdn_write: zero write P[ 5] I SEND:RELEASE oad:04656708 dad:0465670127 P[ 5] -- bc_state:BCHAN_CLEANED P[ 5] -- mode:TE cause:16 ocause:1 rad: cad: P[ 5] -- facility:FAC_NONE out_facility:FAC_NONE P[ 5] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 P[ 5] -- screen:0 -- pres:0 P[ 5] -- channel:1 caps:Speech pi:0 keypad: P[ 5] -- urate:0 rate:16 mode:0 user1:0 P[ 5] -- pid:0 addr:0 l3id:a0001 P[ 5] -- b_stid:0 layer_id:0 P[ 5] -- bc:821d7ac h:0 sh:0 P[ 5] P[ 5] Sending msg, prim:34d80 addr:0 dinfo:a0001 GOT SETUP OK P[ 5] MGMT: Short status dinfo 201 P[ 5] MGMT: SSTATUS: L2_ESTABLISH P[ 5] handle_frm: frm-addr:42000503 frm-prim:35a82 P[ 5] I IND :RELEASE_COMPLETE oad:04656708 dad:0465670127 P[ 5] -- mode:TE cause:-1 ocause:1 rad: cad: P[ 5] -- facility:FAC_NONE out_facility:FAC_NONE P[ 5] -- info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0 P[ 5] -- screen:0 -- pres:0 P[ 5] -- channel:1 caps:Speech pi:0 keypad: P[ 5] -- urate:0 rate:16 mode:0 user1:0 P[ 5] -- pid:0 addr:0 l3id:a0001 P[ 5] -- b_stid:0 layer_id:0 P[ 5] -- bc:821d7ac h:0 sh:0 P[ 5] -- bc_state:BCHAN_CLEANED P[ 5] bchan_deactivate: called but not activated P[ 5] release_chan: bc with l3id: a0001 P[ 5] * RELEASING CHANNEL pid:0 ctx:from-pstn dad:0465670127 oad:04656708 state: EXTCANTMATCH P[ 5] -- * State Down P[ 5] -- Setting AST State to down P[ 5] -- * State Wait4dig | ExtCantMatch P[ 5] handle_frm: frm-addr:42000503 frm-prim:3f182 P[ 5] -- lib: RELEASE_CR Ind with l3id:a0001 P[ 5] -- lib: CLEANING UP l3id: a0001 P[ 5] -- empty chan 1 P[ 5] BC_STATE_CHANGE: from:BCHAN_CLEANED to:BCHAN_EMPTY P[ 5] BC_NEXT_STATE_CHANGE: from:BCHAN_EMPTY to:BCHAN_EMPTY P[ 0] $$$ CLEANUP CALLED P[ 5] $$$ CLEARING STACK P[ 5] BC_STATE_CHANGE: from:BCHAN_EMPTY to:BCHAN_CLEANED P[ 5] I IND :CLEAN_UP oad: dad: P[ 5] -- mode:TE cause:16 ocause:16 rad: cad: P[ 5] -- facility:FAC_NONE out_facility:FAC_NONE P[ 5] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 5] -- screen:0 -- pres:0 P[ 5] -- channel:0 caps:Speech pi:0 keypad: P[ 5] -- urate:0 rate:16 mode:0 user1:0 P[ 5] -- pid:0 addr:0 l3id:a0001 P[ 5] -- b_stid:0 layer_id:0 P[ 5] -- bc:821d7ac h:0 sh:0 P[ 5] -- bc_state:BCHAN_CLEANED P[ 0] MGMT: DELLAYER|CONFIRM Addr: 100 ! can you help me to guess the problem? in the file extension.conf i've got: exten = 465670127,1,Dial(SIP/200) where am i wrong? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialstatus
Hi, I use an E1-Board to hand the calls over to internal SIP-Clients. My Question is which Dialstatus is set when the SIP-client is unreachable. I tried with NOANSWER but does not seem to be suitable. Does anyone of you have a solution? In voip-info.org wiki there is a Dialstatus CHANUNAVAIL but this is explained by Channel unavailable. On SIP, peer may not be registered.. So this seems not to be right, or does it? TIA, Christophorus begin:vcard fn:Christophorus Laube n:Laube;Christophorus org:SemanticEdge GmbH adr:;;Kaiserin-Augusta-Allee 10-11;Berlin;;10553;Deutschland email;internet:[EMAIL PROTECTED] title:Systemadministrator tel;work:+49-30-34507758 url:http://www.semanticedge.de version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered
make an 'lsmod' and look for any old ISDN architecture modules such as hisax or isdn etc. There shall be no other modules loaded then hfcmulti and the misdn stuff. You don't need CAPI, maybe this is even the clue to your not working S0 card. Beronet provides all you need in order to get a BN8S0 card to run. Just download the install-misdn-mqueue script from the beronet website and the cardinstallation guide and everything will be fine. This is my experience. You do not need to patch the kernel an recompile it. Just install the kernel sources. Then execute the install-misdn-mqueue script. This is doing the rest. Another point of failure may be the hotplug system and drivers. If you got BN8S0 card to run you maybe have some asterisk crashes which depend on the machine you are running. I have an IBM xSeries and my Debian loads some hotplug drivers as default. This is also something you should get rid of. Just blacklist it in /etc/hotplug together with the old style isdn and capi. As far as I know chan_misdn does not support CAPI yet. hi i am experiencing some problems with the configuration of an BN8S0 Beronet card. I've downloaded last CVS of mISDN ans mISDNuser, i patch the kernel 2.6.16.18 and the enabled the following: * ISDN support x x Old ISDN4Linux --- x x --- CAPI subsystem x x M CAPI2.0 support x x [*] Verbose reason code reporting (kernel size +=7K) x x [*] CAPI2.0 Middleware support (EXPERIMENTAL) x x M CAPI2.0 /dev/capi support x x [*] CAPI2.0 filesystem support x x --- CAPI hardware drivers x x Active AVM cards --- x x Active Eicon DIVA Server cards --- x x Modular ISDN driver --- M Support modular ISDN driver x x [*] Enable memory leak debug for mISDN x x [*] Support for AVM Fritz!Cards x x [ ] Support for HFC PCI cards x x [*] Support for HFC multiport cards (HFC-4S/8S/E1) x x [*] HFC multiport driver with memory mapped IO after the kernel recompilation and the reboot i build and install mISDNuser with make and make install. The card is recognized by the system, this is the output of lspci: 00:0d.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] (rev 01) and i can load hfcmulti and mISDN_dsp without problems... all channels are configured in TE mode, i use this modprobe: /sbin/modprobe hfcmulti layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2 type=0x08 /sbin/modprobe mISDN_dsp and this is the output of dmesg: Modular ISDN Stack core $Revision: 1.34 $ mISDNd: kernel daemon started mISDNd: test event done mISDN: HFC-multi driver Rev. 1.41 0 devices registered mISDN_dsp: Audio DSP Rev. 1.17 (debug=0x0) EchoCancellor MG2 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies. why i get 0 devices registrered? where am i wrong? thanks in advance nik ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered
Oh sorry, chan_misdn supports CAPI. The other question is wether mISDN itself provides CAPI support.. hi i am experiencing some problems with the configuration of an BN8S0 Beronet card. I've downloaded last CVS of mISDN ans mISDNuser, i patch the kernel 2.6.16.18 and the enabled the following: * ISDN support x x Old ISDN4Linux --- x x --- CAPI subsystem x x M CAPI2.0 support x x [*] Verbose reason code reporting (kernel size +=7K) x x [*] CAPI2.0 Middleware support (EXPERIMENTAL) x x M CAPI2.0 /dev/capi support x x [*] CAPI2.0 filesystem support x x --- CAPI hardware drivers x x Active AVM cards --- x x Active Eicon DIVA Server cards --- x x Modular ISDN driver --- M Support modular ISDN driver x x [*] Enable memory leak debug for mISDN x x [*] Support for AVM Fritz!Cards x x [ ] Support for HFC PCI cards x x [*] Support for HFC multiport cards (HFC-4S/8S/E1) x x [*] HFC multiport driver with memory mapped IO after the kernel recompilation and the reboot i build and install mISDNuser with make and make install. The card is recognized by the system, this is the output of lspci: 00:0d.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] (rev 01) and i can load hfcmulti and mISDN_dsp without problems... all channels are configured in TE mode, i use this modprobe: /sbin/modprobe hfcmulti layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2 type=0x08 /sbin/modprobe mISDN_dsp and this is the output of dmesg: Modular ISDN Stack core $Revision: 1.34 $ mISDNd: kernel daemon started mISDNd: test event done mISDN: HFC-multi driver Rev. 1.41 0 devices registered mISDN_dsp: Audio DSP Rev. 1.17 (debug=0x0) EchoCancellor MG2 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies. why i get 0 devices registrered? where am i wrong? thanks in advance nik ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered
Do you still have the precompiled kernel installed? Try to boot to it. Install kernel sources and try to rerun the install-misdn-mqueue script. This is doing the needed stuff for you. Maybe the recompilation of your kernel caused the problem. I have installed three systems with misdn the last two months (two of them with the BN8S0) and I did not have problems in that with the vendor delivered precompiled kernel. To not need to recompile the kernel is one of the reasons they build the misdn in a kernel and a user space part. You can also contact beronet in order to get some support. They support their cards until they are correctly recognized in asterisk. Cheers thanks to your reply i've also tried to use install-misdn-mqueue but: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init scan [OK] found the following devices: card=1,0x8 [ii] run /etc/init.d/misdn-init config to store this information to /etc/misdn-init.conf as you can see, the scan finish succesfully, but if try to start misd i get: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init start FATAL: Module capi not found. FATAL: Error inserting mISDN_capi (/lib/modules/2.6.16.18/extra/mISDN_capi.ko): Unknown symbol in module, or unknown parameter (see dmesg) - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type=0x8 protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2 layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0 and dmesg: Modular ISDN Stack core $Revision: 1.34 $ mISDNd: kernel daemon started ISDN L1 driver version 1.16 ISDN L2 driver version 1.27 mISDN: DSS1 Rev. 1.38 mISDN_dsp: Audio DSP Rev. 1.17 (debug=0x0) EchoCancellor MG2 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies. mISDN_capi: Unknown symbol capi_cmd2str mISDN_capi: Unknown symbol capi_cmsg_header mISDN_capi: Unknown symbol detach_capi_ctr mISDN_capi: Unknown symbol capi_cmsg2message mISDN_capi: Unknown symbol capi_ctr_reseted mISDN_capi: Unknown symbol capi_ctr_ready mISDN_capi: Unknown symbol capi_message2cmsg mISDN_capi: Unknown symbol capi_ctr_handle_message mISDN_capi: Unknown symbol attach_capi_ctr mISDNd: test event done mISDN: HFC-multi driver Rev. 1.41 0 devices registered finally, if i try to start asterisk: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) mISDN_close: fid(18) isize(131072) inbuf(0xb7580008) irp(0xb7580008) iend(0xb7580008) Jun 3 08:42:31 ERROR[8296]: chan_misdn.c:3775 load_module: Unable to initialize mISDN Jun 3 08:42:31 WARNING[8296]: loader.c:414 __load_resource: chan_misdn.so: load_module failed, returning -1 Jun 3 08:42:31 WARNING[8296]: loader.c:554 load_modules: Loading module chan_misdn.so failed! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered
That's right, mISDN only supports kernels up from version 2.6.9. So I see you did have to compile a kernel yourself. Beronet has a telephone number where they offer support. This is german one. They also have a support mail address, just have a look at their site http://www.beronet.com and choose english language if you do not speak or better read german. I often phoned to Beronet and the support was great. If you can give them ssh access to your machine for a while and they will show you what to do. Another point of failure may be your jumper settings. The mISDN driver can only recognize wether the ports are jumpered TE or NT, not wether ptp or ptmp. This what you have to tell it. Again, just download the card_installation_guide.pdf from http://www.beronet.com/downloads. This helped me a lot. thanks to your reply using slackware the precompiled kernel is of the 2.4 series. I've also tried to remove all modules of my 2.6 kernel, download it , configure it and boot it. Then, using a new 2.6.16.18 kernel (and working) i've run the make install of the beronet utility, but i'm still getting the same error. :-( Beronet has a support forum or support mailing list? From their website it seems that the support is a payed service... hi and thanks On 6/3/06, Christophorus Laube [EMAIL PROTECTED] wrote: Do you still have the precompiled kernel installed? Try to boot to it. Install kernel sources and try to rerun the install-misdn-mqueue script. This is doing the needed stuff for you. Maybe the recompilation of your kernel caused the problem. I have installed three systems with misdn the last two months (two of them with the BN8S0) and I did not have problems in that with the vendor delivered precompiled kernel. To not need to recompile the kernel is one of the reasons they build the misdn in a kernel and a user space part. You can also contact beronet in order to get some support. They support their cards until they are correctly recognized in asterisk. Cheers thanks to your reply i've also tried to use install-misdn-mqueue but: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init scan [OK] found the following devices: card=1,0x8 [ii] run /etc/init.d/misdn-init config to store this information to /etc/misdn-init.conf as you can see, the scan finish succesfully, but if try to start misd i get: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init start FATAL: Module capi not found. FATAL: Error inserting mISDN_capi (/lib/modules/2.6.16.18/extra/mISDN_capi.ko): Unknown symbol in module, or unknown parameter (see dmesg) - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type=0x8 protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2 layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0 and dmesg: Modular ISDN Stack core $Revision: 1.34 $ mISDNd: kernel daemon started ISDN L1 driver version 1.16 ISDN L2 driver version 1.27 mISDN: DSS1 Rev. 1.38 mISDN_dsp: Audio DSP Rev. 1.17 (debug=0x0) EchoCancellor MG2 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies. mISDN_capi: Unknown symbol capi_cmd2str mISDN_capi: Unknown symbol capi_cmsg_header mISDN_capi: Unknown symbol detach_capi_ctr mISDN_capi: Unknown symbol capi_cmsg2message mISDN_capi: Unknown symbol capi_ctr_reseted mISDN_capi: Unknown symbol capi_ctr_ready mISDN_capi: Unknown symbol capi_message2cmsg mISDN_capi: Unknown symbol capi_ctr_handle_message mISDN_capi: Unknown symbol attach_capi_ctr mISDNd: test event done mISDN: HFC-multi driver Rev. 1.41 0 devices registered finally, if i try to start asterisk: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) mISDN_close: fid(18) isize(131072) inbuf(0xb7580008) irp(0xb7580008) iend(0xb7580008) Jun 3 08:42:31 ERROR[8296]: chan_misdn.c:3775 load_module: Unable to initialize mISDN Jun 3 08:42:31 WARNING[8296]: loader.c:414 __load_resource: chan_misdn.so: load_module failed, returning -1 Jun 3 08:42:31 WARNING[8296]: loader.c:554 load_modules: Loading module chan_misdn.so failed! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit
Re: [Asterisk-Users] BN8S0 Installation problem - 0 devices registrered
You may also have a look at http://www.voip-info.org/wiki/view/Asterisk+mISDN+channels thanks to your reply using slackware the precompiled kernel is of the 2.4 series. I've also tried to remove all modules of my 2.6 kernel, download it , configure it and boot it. Then, using a new 2.6.16.18 kernel (and working) i've run the make install of the beronet utility, but i'm still getting the same error. :-( Beronet has a support forum or support mailing list? From their website it seems that the support is a payed service... hi and thanks On 6/3/06, Christophorus Laube [EMAIL PROTECTED] wrote: Do you still have the precompiled kernel installed? Try to boot to it. Install kernel sources and try to rerun the install-misdn-mqueue script. This is doing the needed stuff for you. Maybe the recompilation of your kernel caused the problem. I have installed three systems with misdn the last two months (two of them with the BN8S0) and I did not have problems in that with the vendor delivered precompiled kernel. To not need to recompile the kernel is one of the reasons they build the misdn in a kernel and a user space part. You can also contact beronet in order to get some support. They support their cards until they are correctly recognized in asterisk. Cheers thanks to your reply i've also tried to use install-misdn-mqueue but: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init scan [OK] found the following devices: card=1,0x8 [ii] run /etc/init.d/misdn-init config to store this information to /etc/misdn-init.conf as you can see, the scan finish succesfully, but if try to start misd i get: [EMAIL PROTECTED]:/usr/src/install-misdn-mqueue# /etc/init.d/misdn-init start FATAL: Module capi not found. FATAL: Error inserting mISDN_capi (/lib/modules/2.6.16.18/extra/mISDN_capi.ko): Unknown symbol in module, or unknown parameter (see dmesg) - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type=0x8 protocol=0x2,0x2,0x2,0x2,0x2,0x2,0x2,0x2 layermask=0xf,0xf,0xf,0xf,0xf,0xf,0xf,0xf poll=128 debug=0 and dmesg: Modular ISDN Stack core $Revision: 1.34 $ mISDNd: kernel daemon started ISDN L1 driver version 1.16 ISDN L2 driver version 1.27 mISDN: DSS1 Rev. 1.38 mISDN_dsp: Audio DSP Rev. 1.17 (debug=0x0) EchoCancellor MG2 mISDN_dsp: DSP clocks every 64 samples. This equals 2 jiffies. mISDN_capi: Unknown symbol capi_cmd2str mISDN_capi: Unknown symbol capi_cmsg_header mISDN_capi: Unknown symbol detach_capi_ctr mISDN_capi: Unknown symbol capi_cmsg2message mISDN_capi: Unknown symbol capi_ctr_reseted mISDN_capi: Unknown symbol capi_ctr_ready mISDN_capi: Unknown symbol capi_message2cmsg mISDN_capi: Unknown symbol capi_ctr_handle_message mISDN_capi: Unknown symbol attach_capi_ctr mISDNd: test event done mISDN: HFC-multi driver Rev. 1.41 0 devices registered finally, if i try to start asterisk: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) mISDN_close: fid(18) isize(131072) inbuf(0xb7580008) irp(0xb7580008) iend(0xb7580008) Jun 3 08:42:31 ERROR[8296]: chan_misdn.c:3775 load_module: Unable to initialize mISDN Jun 3 08:42:31 WARNING[8296]: loader.c:414 __load_resource: chan_misdn.so: load_module failed, returning -1 Jun 3 08:42:31 WARNING[8296]: loader.c:554 load_modules: Loading module chan_misdn.so failed! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Misdn 0.2.1 BUSY tone
I have this problem on misdn 0.2.1: in extension.conf i have such a situation; [misdn_incoming] exten = 06786541,1,Dial(SIP/203) where SIP/203 is a GXP-2000. I want to make the 203 to answer just one call at the same time, so i've disabled the call waiting feature on the phone, but when I do this the caller does not hear the Busy tone, it receives the telco Network error tone. Don't you have to set up a context for the SIP-phone? In sip.conf where you defined the SIP peer you have to set something like context=my_sip. This is what you also have to set up in your extensions.conf [misdn_incoming] exten = 06786541,1,Dial(SIP/203) [my_sip] exten = _X.-Busy,1,Busy() just have a look at show application busy on the CLI. You also have to set up the right indications in indications.conf. Just set the country value to the right value. Maybe even that will just solve your problem. By the way there is a much newer version of misdn available. I want the caller to receive the busy tone when the called is busy . How can I do this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] behaviour depending on count of used lines
Hi there,
I want to set up an extension set that acts different depending on the count
of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer
10 lines. Therefore I set up a global variables LINES in the general section
of extensions.conf and instantiate it with 0. I a call is incoming I check
the LINES variable wether is 10 or more. If so I make a call transfer. If not
I increment the variable and direct the call to an internal SIP address.
After finishing the call I want to decrement the variable again, of course.
My extension set looks like this way:
[general]
static=yes
writeprotect=no
LINES = 0
[E1]
exten = 33006712,1,GotoIf($[${LINES} = 10]?101:201)
exten = 33006712,101,Dial(mISDN/g:E1/34507725)
exten = 33006712,201,SetGlobalVar(LINES=$[ ${LINES} +1 ])
exten = 33006712,202,Dial(SIP/192.168.0.65:5080SIP/192.168.0.65:5090)
exten = 33006712-ANSWER,203,Answer()
exten = 33006712-HANGUP,204,SetGlobalVar(LINES=$[ ${LINES} -1])
exten = 33006712,205,SetGlobalVar(LINES=$[ ${LINES} -1 ])
exten = 33006712,206,Hangup()
My problem is that the increment works perfectly, but the decrement is not
working. I added the last two extensions only because the hangup-extension
did not work.
Can anyone of you help me, please.
TIA, Christophorus Laube
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