Re: [asterisk-users] Need to find small footprint asterisk platform
Christian Victor wrote: 2009/3/30 Peer Oliver Schmidt po...@theinternet.de mailto:po...@theinternet.de The Horst-Box Professional has a lot of problems in the ADSL area (like stopping transfers after a dozen or so megabytes for example), and I have had lots of needs to hard-reboot the box, after enabling VoIP functionality. Well - I never ran one in a professional enviroment and only use one single unit in my home. Until now and with the latest firmware it runs without bigger problems. But I agree that for professional use you neet to take a close look at reliability. Do you have a fast ADSL line? like 16mbit or higher? It always breaks in the middle of a transfer. If I limit the download rate, it progresses better, but still not perfect. The D-Link support is useless (The answer the support request and without taking the answer into account close the ticket). I agree - support is crap.It's basically here is the source so thats not our probolem anymore. The boxes would have been perfect for the german market, however, the way it was implemented, they are totally useless. And yes, I have tried all available firmware versions, and always made sure to follow the instructions to the letter, with regards to configuration reset. You found instructions??? Lucky bastard! :-D :D http://ip-phone-forum.de -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ -- 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] Need to find small footprint asterisk platform
Christian Victor wrote: Here in germany D-Link sells a device called the Horst-Box Professional wich is a ADSL modem/router with WiFi and an integrated embedded asterisk platform with 1xBRI in, 1xBRI out and 3xFXS if my mind serves me right. Size is about 180x250x50mm. Its been around for some years so maybe it is already EOL. The Horst-Box Professional has a lot of problems in the ADSL area (like stopping transfers after a dozen or so megabytes for example), and I have had lots of needs to hard-reboot the box, after enabling VoIP functionality. The D-Link support is useless (The answer the support request and without taking the answer into account close the ticket). The boxes would have been perfect for the german market, however, the way it was implemented, they are totally useless. And yes, I have tried all available firmware versions, and always made sure to follow the instructions to the letter, with regards to configuration reset. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ -- 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] Are .call files working with extensions.ael ?
Hello Olivier, With an extensions.ael enabled system, I keep getting whatever I change into my astup.call file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service: Invalid file contents in /var/spool/asterisk/outgoing/astup.call, deleting [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:505 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/astup.call' Do you by chance use bristuff? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ -- 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] Are .call files working with extensions.ael ? bristuff problem
Olivier wrote:
Do you by chance use bristuff?
Yes, I do.
bristuff patches pbx/pbx_spool.c
I have no knowledge of C, but there seems to be a problem around line
266.
The original line (pre-bristuff) looks like this:
if (ast_strlen_zero(o-tech) || ast_strlen_zero(o-dest) ||
(ast_strlen_zero(o-app) ast_strlen_zero(o-exten))) {
The patched line looks like this:
if (ast_strlen_zero(o-tech) || ast_strlen_zero(o-dest) ||
(ast_strlen_zero(o-app) ast_strlen_zero(o-exten)) ||
(ast_strlen_zero(o-message) ast_strlen_zero(o-pdu))) {
Try reverting that line, and see if that helps with your problem. And
maybe someone with a better understanding of C can take a look at the
above problem.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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Re: [asterisk-users] German SIP and/or IAX providers?
Hello Ken, Hi, all. My company is setting up a branch office in Germany, and I'm very interested in a VoIP provider over thataway. as I am living in Germany, let me advise you against using VoIP providers in Germany. Most of the time they do work, but they are not as reliable as a regular phone company. What I do is, use a regular phone line (ISDN BRI) for incoming traffic, and utilize the VoIP providers for dialing out. What is your reasoning for a VoIP provider? BTW: The language should not be the problem. The service is poor, no matter what language... -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] meetme conference using g729?
Mark, Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? Thanks! Tilghman wrote it out: You can not mix two compressed audio streams together. You first have to uncompress them. Even if both audio streams use the same codec, they are compressed thus have to be uncompressed for the mixing of the audio to happen. Better? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] iaxmodem, chan_capi, hylafax problem and faxing in general
Hello Faris, Only I've sidetracked and am currently trying to use capi4hylafax instead of iaxmodem which seems to work wonderfully but I'm having some issues with root verses uucp permissions which is spoiling my fun. Make sure not to run faxgetty together with capi4hylafax. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Siemens Openstage Asterisk ?
Hello Olivier, I don't have this answer but would be curious to know its price for reseller. Any clue ? the price for the OpenStage 20 seems to be in the range of the Snom 360. The Openstage 40 seems to be in the range of the Snom 370. The Openstage 60 is 50% higher than the 40, with the 80 about nearly twice the price of the 60. Hope that helps. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Hello Armin (and happy easter), thanks for you continuing support. Can you please try HEAD version of SVN trunk (443)? Did checkout the 443. It works without any verbosity. THANK YOU! I'll buy you a beer, if you ever happen to come to the northern part of Germany. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Good Morning Armin, tried the patch, but it did not work. It waits quite a long time before the chan-capi error message comes up, according to the time stamp it is about 12 seconds. It is kind of strange, that the whole startup process for asterisk usually takes only about 4-5 seconds. That's too long, normaly the confirmation message arrives within a few msecs. So it seems that the driver isn't responding. Do you need additional information? Which card/driver do you use? [EMAIL PROTECTED]:~# lspci -s 0:0e -v 00:0e.0 Network controller: AVM Audiovisuelles MKTG Computer System GmbH A1 ISDN [Fritz] (rev 02) Subsystem: AVM Audiovisuelles MKTG Computer System GmbH FRITZ!Card ISDN Controller Flags: medium devsel, IRQ 10 Memory at ff001400 (32-bit, non-prefetchable) [size=32] I/O ports at dcc0 [size=32] [EMAIL PROTECTED]:~# capiinit status 1 fcpci running fcpci-dcc0-10A1 3.11-07 0xdcc0 10 Driver from ubuntu edgy A debug log (capi trace) from the driver or kernelcapi helps to see what messages are wrong/missing. What is the best way to produce this? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Hello Armin, here are the results, after modprobe kernelcapi showcapimsgs=3 /var/log/kern.log Apr 3 15:06:14 server42 kernel: [255113.053170] CAPI Subsystem Rev 1.1.2.8 Apr 3 15:06:18 server42 kernel: [255116.814334] fcpci: AVM FRITZ!Card PCI driver, revision 0.7.2 Apr 3 15:06:18 server42 kernel: [255116.814356] fcpci: (fcpci built on Feb 27 2007 at 21:22:25) Apr 3 15:06:18 server42 kernel: [255116.814367] fcpci: -- 32 bit CAPI driver -- Apr 3 15:06:18 server42 kernel: [255116.817598] PCI: Found IRQ 10 for device :00:0e.0 Apr 3 15:06:18 server42 kernel: [255116.817642] fcpci: AVM FRITZ!Card PCI found: port 0xdcc0, irq 10 Apr 3 15:06:18 server42 kernel: [255116.817653] fcpci: Loading... Apr 3 15:06:18 server42 kernel: [255116.817665] fcpci: Driver 'fcpci' attached to fcpci-stack. (152) Apr 3 15:06:18 server42 kernel: [255117.049308] fcpci: Stack version 3.11-07 Apr 3 15:06:18 server42 kernel: [255117.050442] kcapi: Controller 1: fcpci-dcc0-10 attached Apr 3 15:06:18 server42 kernel: [255117.050456] kcapi: card 1 fcpci-dcc0-10 ready. Apr 3 15:06:18 server42 kernel: [255117.050480] kcapi: notify up contr 1 Apr 3 15:06:19 server42 kernel: [255117.051283] fcpci: Loaded. Apr 3 15:06:22 server42 kernel: [255120.102281] capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Apr 3 15:06:35 server42 kernel: [255133.725987] kcapi: appl 1 up Apr 3 15:06:35 server42 kernel: [255133.727235] kcapi: put [0x1] id#1 FACILITY_REQ len=18 Apr 3 15:06:35 server42 kernel: [255133.727712] kcapi: got [0x1] id#1 FACILITY_CONF len=26 Apr 3 15:06:35 server42 kernel: [255133.729933] kcapi: appl 1 down Apr 3 15:06:35 server42 kernel: [255133.730585] kcapi: appl 1 up Apr 3 15:06:35 server42 kernel: [255133.731478] kcapi: put [0x1] id#1 LISTEN_REQ len=26 Apr 3 15:06:52 server42 kernel: [255150.690252] kcapi: appl 1 down At 15:06:35 the loading stopped. Is this helpful, or do you need a higher verbosity? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Hello Armin, thanks a lot for your help. Can you please do the same with 'showcapimsgs=2'? It may give more info on the commands itself, maybe some parameters are wrong here. Here you go. 17:23:17 is the magic time. Apr 3 17:23:09 server42 kernel: [263323.308388] fcpci: AVM FRITZ!Card PCI driver, revision 0.7.2 Apr 3 17:23:09 server42 kernel: [263323.308411] fcpci: (fcpci built on Feb 27 2007 at 21:22:25) Apr 3 17:23:09 server42 kernel: [263323.308421] fcpci: -- 32 bit CAPI driver -- Apr 3 17:23:10 server42 kernel: [263323.311559] PCI: Found IRQ 10 for device :00:0e.0 Apr 3 17:23:10 server42 kernel: [263323.311602] fcpci: AVM FRITZ!Card PCI found: port 0xdcc0, irq 10 Apr 3 17:23:10 server42 kernel: [263323.311613] fcpci: Loading... Apr 3 17:23:10 server42 kernel: [263323.311625] fcpci: Driver 'fcpci' attached to fcpci-stack. (152) Apr 3 17:23:10 server42 kernel: [263323.539987] fcpci: Stack version 3.11-07 Apr 3 17:23:10 server42 kernel: [263323.541140] kcapi: Controller 1: fcpci-dcc0-10 attached Apr 3 17:23:10 server42 kernel: [263323.541154] kcapi: card 1 fcpci-dcc0-10 ready. Apr 3 17:23:10 server42 kernel: [263323.541833] fcpci: Loaded. Apr 3 17:23:12 server42 kernel: [263325.975634] capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Apr 3 17:23:17 server42 kernel: [263330.892916] kcapi: put [0x1] FACILITY_REQ ID=001 #0x0001 LEN=0018 Apr 3 17:23:17 server42 kernel: [263330.892926] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.892933] FacilitySelector = 0x3 Apr 3 17:23:17 server42 kernel: [263330.892939] FacilityRequestParameter= 00 00 00 Apr 3 17:23:17 server42 kernel: [263330.892946] Apr 3 17:23:17 server42 kernel: [263330.893153] kcapi: got [0x1] FACILITY_CONF ID=001 #0x0001 LEN=0026 Apr 3 17:23:17 server42 kernel: [263330.893163] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.893169] Info = 0x0 Apr 3 17:23:17 server42 kernel: [263330.893176] FacilitySelector = 0x3 Apr 3 17:23:17 server42 kernel: [263330.893182] FacilityConfirmationParameter = 00 00 06 00 00\37703 00 00 Apr 3 17:23:17 server42 kernel: [263330.893190] Apr 3 17:23:17 server42 kernel: [263330.900689] kcapi: put [0x1] LISTEN_REQ ID=001 #0x0002 LEN=0026 Apr 3 17:23:17 server42 kernel: [263330.900699] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.900706] InfoMask = 0x Apr 3 17:23:17 server42 kernel: [263330.900713] CIPmask = 0x1fff03ff Apr 3 17:23:17 server42 kernel: [263330.900720] CIPmask2 = 0x0 Apr 3 17:23:17 server42 kernel: [263330.900726] CallingPartyNumber = default Apr 3 17:23:17 server42 kernel: [263330.900733] CallingPartySubaddress = default Apr 3 17:23:17 server42 kernel: [263330.900739] -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Good evening Armin, This log below shows no error in parameters, but the problem is still the same: the fcpci driver doesn't respond and I cannot tell why. Ok. Thanks for your assistants anyhow. What strikes me as strange ist the fact, that turning on verbose helps to circumvent the problem. Thanks again, and have a good night. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Hi, I have installed the above two, and have two questions: * Is there a reason (or better=a fix), why the chan-capi-1.0.0 does not compile together with Asterisk 1.4.2? * Anyone else experiencing problems with chan-capi-HEAD not seeing the controller? If I run asterisk with verbose setting to 0, i.e. just asterisk -c chan-capi does not find the controller. Starting asterisk with verbosity turned up, most of the time the controller is found. Might this be a problem of missing CPU power (PII-400)? Any and all help is greatly appreciated. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Hello Armin, Actually chan-capi-1.0.0 does compile with Asterisk 1.4.2. [..]but you [..] just change the line if grep -q ASTERISK_VERSION_NUM 0104 $INCLUDEDIR/version.h; then to if grep -q ASTERISK_VERSION_NUM .*104 $INCLUDEDIR/version.h; then in create_config.sh thanks, will try. * Anyone else experiencing problems with chan-capi-HEAD not seeing the controller? If I run asterisk with verbose setting to 0, i.e. just asterisk -c chan-capi does not find the controller. Starting asterisk with verbosity turned up, most of the time the controller is found. Might this be a problem of missing CPU power (PII-400)? I don't know where this could happen. Do you have more info on what exactly is happening and when? [EMAIL PROTECTED]:~# grep capi /var/log/asterisk/messages [Mar 31 16:17:03] ERROR[3850] chan_capi.c: Unable to listen on contr1 (error=0x100f) Is this helpful, or do you need more information? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Hello Armin, [EMAIL PROTECTED]:~# grep capi /var/log/asterisk/messages [Mar 31 16:17:03] ERROR[3850] chan_capi.c: Unable to listen on contr1 (error=0x100f) Is this helpful, or do you need more information? Yes, at this state it might be possible that less CPU power causes problems. The 'listen' command expects an answer and maybe it is coming too late. Can you please try the patch below? Index: chan_capi.c === --- chan_capi.c (revision 436) +++ chan_capi.c (working copy) @@ -631,7 +631,7 @@ error = LISTEN_CONF_INFO(CMSG); break; } - usleep(2); + usleep(10); tried the patch, but it did not work. It waits quite a long time before the chan-capi error message comes up, according to the time stamp it is about 12 seconds. It is kind of strange, that the whole startup process for asterisk usually takes only about 4-5 seconds. Do you need additional information? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Can anyone recommend a large button sip phone for the elderley.
Chuck Bunn wrote: Can anyone recommend a large button/type sip phone (VOIP) that an older person could use. I have a client that needs to have large button phones for elderly residents in her facility. You might want to look into the original Grandstream Phone, the BT-101. I havn't found any phone with bigger buttons. And each time the phone rings, it lights up a dark room. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Asterisk installations in Germany
asterisk-robert wrote: I need to send some information to our German HQ regarding my experiences with VoIP. Asterisk is very prominent in those experiences. I would like to include information about installations of Asterisk at German companies/universities. We have installed Asterisk in a multi-site small business environment in Hamburg. We started out with HT-286 ATA, went to different ATAs, and ended up with SNOM 360s. Usage is light with a single queue. Connection between sites is via IAX. No central dial plan, just plain extensions, ie. Site 1 has extension 2x, Site 2 has extension 4x, and Site 3 has extensions 5x. Connection to the PSTN via chan_capi-cm (AVM C4) and bristuffed asterisk. Each site has its own PSTN connection, which can be used from the other sites via prefix. Make sure to get quality PCs, speed is not important, but I have found out some old Compaqs had weird problems, which went away by going to a new PC. Old Dell is working fine. HTH -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Asterisk with AVM B1 and HFC
Marco Dieckhoff wrote: I have two ISDN cards for my asterisk server, an AVM B1 (active card) and a HFC. I want to use the HFC card in NT mode, and the AVM B1 in TE. Afair bristuff and vISDN doesn't support the AVM B1, so mISDN should be my choice? I ran a AVM C4 and HFC together for quite some time. Do the bristuff thing, and install the CHAN_CAPI-CM driver afterwards. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Quad ISDN card
René Enskat [Teamware GmbH] wrote: Somebody know if the AVM C4 Quad ISDN card is supported by the current asterisk version? It is supported. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Receptionist Phones
Bob McDowell wrote: Can you chain these to get more that 42 buttons? I need about 60... Bob McDowell 42+12 is fairly near the 60 target. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] capiHOLD missing in BRIstuff 0.3.0
[EMAIL PROTECTED] wrote: Hi, I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more recent version, but I cannot find any working combination of Asterisk an chan_capi any more: On junghanns.net there is a chan_capi 0.3.6, but this won't compile against any recent Asterisk (missing channel_pvt.h). The production version of BRIstuff comes with an old asterisk (1.0), the experimental version 0.3.0-PRE-1 includes an asterisk 1.2.4 and compiles, but the module capiHOLD is missing. Did you try searching for the chan capi within the mailing list archive before posting? http://sourceforge.net/projects/chan-capi is the current way to go with a CAPI capable card. hth -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Grandstream BT-101 POS Error
Basically, I've setup the phone following the instructions at voip-info.org, and it registers for about 10 seconds, then after receiving the SIP NOTIFY from the * server, goes into flashing display mode, which indicates some sort of connectivity error. I've tried all The flashing dispay shows you have waiting messages in your voice mail... -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Application Faxing using SIP
J Poz schrieb: I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution. My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not receiving them. [..] my application -- QUESTION MARK??? VOIP Provider --- PSTN --- Fax Machine. don't use VoIP, but an external mail2fax provider. I am sure everythng else is asking for trouble. I am also positiv, a couple of businesses on this list would love to help you out with this. rgds posde ___ --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] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...
Francesco Peeters (Asterisk) schrieb: They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20 years. In germany you could move that number to a VoIP provider and use it from the main office direct. Then you won't need an asterisk in the remote location. My question is whether there are any tools better suited for this than an old banger (AMD 800 MHz) PC with a HFC-PCI card and (*) relaying (switch) the incoming calls to the central box. Should be plenty enough. I am running a PII-400 with a AVM C4 connected to two ISDN-ports and have another IAX connection to a customers site. Works fine. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Fritz card technology German *
Chris, Something that no one around me seemed to consider was what Legacy PBX is in place already. There is a PBX with analog phones going into it! Please re-read my post from January, 19th. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Routing Calls via chan_capi with AVM FritzCard
Christian Schmidt schrieb: [..] - asterisk 1.0.7.dfsg.1-2 - chan_capi 0.3.5-11 Do your self a favour and get chan-capi_cm of Sourceforge http://sourceforge.net/projects/chan-capi -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Re: Asterisk-Users Digest, Vol 18, Issue 134
Claudio Beffa schrieb: O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps telling me while startup: [ISDN2] isdnmode=ptp isdnmode=did might work ... msn=51 msn= is not needed anymore. use SetCallerID in the dialplan instead. incomingmsn=251 controller=4 softdtmf=1 accountcode= context=Amt callgroup=2 devices=2 [ISDN1] isdnmode=ptp same here. msn=3899231 incomingmsn=* controller=1,2,3 softdtmf=1 accountcode= context=Intern-ISDN callgroup=1 devices=6 /etc/capi.conf #SuSEconfig.isdn generated # card fileproto io irq mem cardnr options c4 c4.bin DSS1- - - 1 p2p c4 - DSS1- - - 2 p2p c4 - DSS1- - - 3 p2p c4 - DSS1- - - 4 p2p Versions: net-misc/asterisk-chan_capi-cm-0.5.4 get a later version -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Dect to SIP PCI card
Giordano Grandis wrote: I’m looking for a PCI card which i could install on asterisk box, with purpose to use 15-20 cordless dect phone in a very “dect cell”. Is there anyone that could help me pls ? You might find something by search for Com-On-Air. But, AFAIK, the product by Dosch Almand only works inside a Windows machine. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] AVM C4, asterisk-1.0.8, /etc/asterisk/capi.conf
Claudio Beffa schrieb: Has anyone a working /etc/asterisk/capi.conf example for Germany or Switzerland using the AVM C4 - ISDN Card. I try to connect asterisk to 3 wires BRI-ISDN (Swisscom). I appreciate your help and it would save me a lot of time, figuring it out by myself. This is my /etc/asterisk/capi.conf [general] nationalprefix=0 internationalprefix=00 rxgain=0.7 txgain=0.6 ;ulaw=yes;set this, if you live in u-law world instead of a-law ; interface sections ... [ISDN1] ;this example interface gets name 'ISDN1' and may be any ;name not starting with 'g' or 'contr'. isdnmode=msn;ptmp (point-to-multipoint) or ptp (point-to-point) ;when using NT-mode, ptp should be set in any case incomingmsn=* ;807440,807441;allow incoming calls to this list of MSNs, * == any controller=2 ;capi controller number to use group=1 ;dialout group ;prefix=0;set a prefix to calling number on incoming calls softdtmf=on ;enable/disable software dtmf detection, recommended ; for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf ; detection accountcode= ;Asterisk accountcode to use in CDRs context=default ;context for incoming calls hold=no ;when Asterisk put on hold, ISDN HOLD shall be used ;immediate=yes ;immediate start of pbx with extension 's' if no digits ; were received on incoming call (no destination number yet) ;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression ;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;echotail=64 ;echo cancel tail setting callgroup=3 ;Asterisk call group ;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels ; are busy devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) [ISDN3] ;this example interface gets name 'ISDN1' and may be any ;name not starting with 'g' or 'contr'. isdnmode=msn;ptmp (point-to-multipoint) or ptp (point-to-point) ;when using NT-mode, ptp should be set in any case incomingmsn=*;allow incoming calls to this list of MSNs, * == any controller=1 ;capi controller number to use group=3 ;dialout group ;prefix=0;set a prefix to calling number on incoming calls softdtmf=on ;enable/disable software dtmf detection, recommended ; for AVM cards relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf ; detection accountcode= ;Asterisk accountcode to use in CDRs context=default ;context for incoming calls hold=no ;when Asterisk put on hold, ISDN HOLD shall be used ;immediate=yes ;immediate start of pbx with extension 's' if no digits ; were ;received on incoming call (no destination number yet) ;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression ;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation ;echotail=64 ;echo cancel tail setting callgroup=1 ;Asterisk call group ;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels ; are busy devices=2;number of concurrent calls on this controller ;(2 makes sense for single BRI, 30 for PRI) And the corresponding /etc/isdn/capil.conf # card fileproto io irq mem cardnr options c4 /usr/lib/c4.bin DSS1- - - - c4 - DSS1- - - - c4 - DSS1- - - - c4 - DSS1- - - - The file /usr/lib/c4.bin had been extracted from the SuSE drivers on the AVM download page. I am using the just 2 of the 4 ports on the card right now. HTH and if you have questions, feel free. BTW: I am using chan_capi-cm of sourceforge. - Mit freundlichen Grüßen Peer Oliver Schmidt the internet company PGP Key ID: 0x83E1C2EA -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Fritz card technology German *
Chris Earle (CBL) wrote: So yes, we have 4 channels, so I'm going to need 2x Fritz cards -- but I would rather not have to apply patches just to get the two PCI cards to work in the same box Than don't use two Fritz! cards, but two hfc-s cards, or the 4-port cards mentioned below. The price difference between the cards you guys mentioned is interesting I have also heard about BERONET isdn cards? a single Beronet 4-channel card would suffice I think? It is not a 4-channel, but a 4-port card, i.e. 8-channels, just like the junghanns and sirrix cards. Thing is, whatever the legacy system in place already is (this is not a fresh operation) must have some sort of minor PBX in place, where all the phones are plugged in. So I would have to remove that and could use a TDM card to plug the phones in? These phones, isdn etc -- probably aren't analog -- probably don't work with a TDM card right? So I think what you were suggesting John is ISDN channel cards and a TDM in the same machine? with * just bridging calls between the two? An even better approach is the one outlined on the junghanns.net. Use the existing PBX and plug it into the other two ISDN ports of the 4-port beronet/junghanns/Sirrix card, and use asterisk as the middleman. Adding new phones would be done either as extensions of the old PBX, or as SIP-phones to asterisk -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] rcapi quality (was: Cisco 801 and rcapi)
Hi Armin, You can also use one Linux Server running CAPI cards with rcapid and have your Asterisk/OpenPBX with chan_capi on another maschine... Did you ever try something like that? What kind of implication had the remote CAPI with regards to sound quality? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Ominiis Asterisk TAPI driver
Tomislav Parcina schrieb: I have foloved instructions at this web pages http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call contacts from Outlook. Now I have few questions. When I place a call, my phone rings before * tries to dial out. Is it posible that * first dials out, and when other side picks up, at that moment that my phone rings? No. TAPI works this way. It only helps you to get rid of memorizing all kinds of phone number, but you first have to pick up the phone for the dialing to occur. Another question, when I recive a phone call, can that contact from Outlook pop-up? There is at least one third-party addon for Outlook which allows you to just that. Googleing for outlook incoming call popup tapi produces a couple of links. I myself tried ESTOS Procall once and seemed to work okay (mind you, that way before asterisk and is quite a few years ago). HTH -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Easiest way to use HFC-S?
Pisac wrote: I'm reading voip-info... and it's only confusing me: zaphfc, zapbri driver package, bristuff... So, what to download and install? If I install bristuff from junghanns.net, should I also install something else (patch)? What is (and where is) that zapbri driver package? Go to the junghanns.net page, get the latest bristuff. Unpack. You will find among other things a readme file explaining what to do. Do I need to use this download.sh script in bristuff? I already have working Asterisk (same version), so why to download and install again? Can I only patch sources and then compile? You can do a lot of things. Following the readme and doing it the way the developers envisioned installing the software will help you succeed much easier. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Easiest way to use HFC-S?
Pisac schrieb: I'm reading voip-info... and it's only confusing me: zaphfc, zapbri driver package, bristuff... So, what to download and install? If I install bristuff from junghanns.net, should I also install something else (patch)? What is (and where is) that zapbri driver package? Go to the junghanns.net page, get the latest bristuff. Unpack. You will find among other things a readme file explaining what to do. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Problems with current chan-capi-cm [SOLVED]
as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. This is just for the records. Ok. The above did not fix it, but Armin did do some more work and found a solution. If anyone else has a problem, I strongly suggest upgrading to the latest CVS, especially if you are using AVM C4, AVM C2 or AVM B1 cards. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Rolling dialplan... best practice?
Erik wrote
Create an waiting extension:
exten = _*XX*XX,1,wait(${EXTEN{1:2})
exten = _*XX*XX,1,dial($EXTEN{3:2})
Then dial using that waiting extension:
exten =
s,2,Dial(${E25}Local/*18*${E24}Local/*30*${E28}Local/*42*${E28}Local/*56*${E22})
This wil dial all the numbers at the same time, however eacht local number
waits a bit longer before executing the dial, hence it hunts :)
So ${E25} will ring instant, ${E24} starts ringing 18 seconds later, ${E28}
starts 12 seconds after ${E24} (timing is related to the 1st phone ringing)
This is bloody cool. Thanks for the idea.
--
Best regards
Peer Oliver Schmidt
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[Asterisk-Users] capi.conf - AVM C4 P2P or P2MP
Quick question, I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules load, but capiinfo says no CAPI installed). Any help is greatly appreciated. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] capi.conf - AVM C4 P2P or P2MP
Hello Armin, thanks for the quick response. I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? isdnmode=msn isdnmode=msn is in /etc/asterisk/capi.conf, but what about the /etc/isdn/capi.conf --- the configuration file for the capi modules? Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules load, but capiinfo says no CAPI installed). Is the card loaded with firmware? Correct permissions to /dev/capi20 ? Yes. As said, putting p2p into the /etc/isdn/capi.conf works. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Problems with current chan-capi-cm
Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. All against svn branch 1.2 as of Dec 9th. Anyone else experiencing problems with the chan-capi Here is an entry into the log file: Dec 12 09:53:39 ERROR[859] chan_capi.c: CAPI error sending DATA_B3_REQ ID=005 #0 x0232 LEN=0030 Controller/PLCI/NCCI= 0x10103 Data32 = 0x8164078 DataLength = 0xa0 DataHandle = 0x14f Flags = 0x0 Data64 = 0x0 (NCCI=0x10103) (error=0x1103) -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Problems with current chan-capi-cm
Armin Schindler schrieb: On Mon, 12 Dec 2005, Peer Oliver Schmidt wrote: Hi, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. [..] error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. Did you try latest CVS (11.12.)? I cvs checkouted today. Can you please provide a full log and one with the older, working version too? set verbose 50 enough? Or another type of log? The file is massive, and I don't want to waste everybodies bandwith. Thanks for your help. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Problems with current chan-capi-cm
Hello again, as of at least Dec 9, but also today, the cvs version of the chan-capi on sf.net gives problems dialing out. The call gets out, but no audio in any direction. Going back to a version from Dec 4th gives a working system again. [..] error 0x1103 is 'queue full', so the capi driver (isdn card) does not accept further voice packets. Did you try latest CVS (11.12.)? I cvs checkouted today. Can you please provide a full log and one with the older, working version too? set verbose 50 enough? Or another type of log? The file is massive, and I don't want to waste everybodies bandwith. After reading your notes regarding capi debug I did some more investigation. The solution was simple. One of the ISDN ports of my AVM C4 did not contain PTP in the capi.conf. It did not seem to matter before, but did now. Changed it, and everything is fine and dandy. Thanks for your help, and sorry for the bothering. Have a good week. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] [Asterrik-Users] Bristuff for Asterisk 1.2 error
asterisk183 schrieb: I have download the bristuff-0.3.0-PRE-1.tar.gz and I followed the instruction in INSTALL: [..] 5. modprobe zaptel 6. But when I doing insmod qozap.o and ztcfg don't start because in /qozap directory I don't have qozap.o files. Why? Try modprobe qozap instead of insmod qozap.o -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI cards, HFC, and bristuff - a general question to clear up my understanding.
Hamish Whittal wrote: I have an Asustek ISDNLink (P-IN100-ST-D) BRI card. [..] This is not a card compatible with the bristuff. I don't know about the availability of the hfc-cards in your part of the world, but they are very inexpensive in Germany (around 30 EUR ~ 30 USD) I am loathe to buy a AVM Fritz card as they are VERY expensive here and if I can get this card working - hey presto, since these thingies are very cheap. If you are serious about asterisk, you don't want to try the modem route with your card. Sound will be bad. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple zaphfc cards (for ISDN BRI) in a single machine
Chris Bagnall wrote: I've just returned from a visit to a client site where their existing incoming lines are in the form of 5 ISDN BRI connections (for 10 channels total). We have successfully deployed Asterisk boxes with 2 HFC-based cards in the past, but I've no idea how well a standard PC will handle 5 or 6 cards - i.e. every PCI slot has a BRI card in it. Any thoughts from folks who've tried this in the past? I would not use multiple HFC-cards, but would either use the QuadBRI or OctoBRI cards available (and pointed out by a previous poster), or use an external ISDN gateway. It seems INALP has come across with some new pricing, something like 500-600 EUR for a 4-Port ISDN gateway (Inalp SN4638 Gateway/Router) Let us know your final findings. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue scheduling...
Scott schrieb: Is it possible to schedule dymanic queues? Currently I have a queue that has dynamic members of which I would like to set a schedule for. From say 8am to 5pm the queue would ring the phones of queue members but after 5pm the caller get's VM. Do it in the dialplan by branching based on time, before entering the queue. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel omni.net USB ISDN works with Asterisk
Gabor Horvath wrote: Can you tell me is the Zyxel omni.net USB ISDN adapter works with Linux, and more specifically, with Asterisk chan_capi? I don't know, but I built an Asterisk PBX test environment on my laptop with Asterisk Management Portal, one hardphone, one ATA, and one softphone. I would connect the whole thing to an ISDN (Euro) line, but because of my laptop, I can use only USB or PCMCIA solutions. there are AVM PC-cards available which do work with CAPI. HTH -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan-capi_cm - 10sec silence before ringing
Hi, I am using asterisk and chan-capi_cm CVS as of yesterday, but the problem has been for a long time. After dialing a number via dial(CAPI/G1/0123-122) it takes roughly 10 seconds to hear the first ringing tone. Adding option b is not feasible, as it does not fix the dialout problem, but merely creates a ringing locally. Is this a known problem? console log: -- Executing Macro(SIP/25-0892, lcr|01729731418|807440|1) in new stack -- Executing NoOp(SIP/25-0892, ) in new stack -- Executing SetCallerID(SIP/25-0892, 807440) in new stack -- Executing Dial(SIP/25-0892, CAPI/g1/0101901729731418|60|bo) in new stack -- Called g1/0101901729731418 HERE WE HAVE A 10 SECONDS PAUSE (tried with or without option in dial string=) -- CAPI/ISDN1/0101901729731418-0 is proceeding passing it to SIP/25-0892 -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan-capi_cm - 10sec silence before ringing
Please ignore my message. Problem solved. Using a call-by-call vendor in Germany caused this long period of silence. Without it everything is working as expected. Have a great weekend. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI - displaying individual MSN
Stefan Günther schrieb:
With the above configuration the display always shows 8304490.
I've tried to change the number in the dialplan, but this doesn't change
anything:
exten = _90[23456789].,1,SetCIDNum(83044912)
Try to use SetCallerID instead of SetCIDNum and see if it helps.
exten = _90[23456789].,2,Dial(Capi/g1/${EXTEN:1},30,tr)
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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Re: [Asterisk-Users] [ISDN] Problem: Device '/dev/ttyI1' lacking dialtone
Patrick de Kok schrieb: What large number of answers? If I scroll through the lists no answers are present..and previous posts do not seem to help as well.. That is the point. No one seems to use ISDN together with chan_modem. Do yourself a favour and use chan_capi or chan_capi-cm -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbi stating question
Lorenzo wrote I'm working in an italian company who wan't start using asterisk. My problem: 1. What kind of hardware I need for make a PBX who speek with 3/4 ISDN BRI line and internal use a SIP VoIP telephone? 4 ISDN channels or lines. If 4 ISDN channels, get two HFC-S based cards, if 4 lines (i.e. 8 channels) get either an AVM C4 or an junghanns.net quadbri card. 2. I can install ASTERISK and HYLAFAX on the same machine ? Yes. Running (mostly) fine here, using an AVM C4 connected to two lines. Hylafax is receiving all faxes, using a ATA connected to Asterisk for faxing out. Take a look at voip-info.org and search for zaphfc and/or chan_capi. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM B1
Steve Foy schrieb: Hi, I'm trying to get Asterisk working with the AVM B1 card. I've tried every instruction set I can find, but to no avail. You should use the chan_capi or chan_capi-cm. I used to use an old B1 ISA card, which worked without much trouble, after I got the CAPI itself running. Asterisk ontop of CAPI worked just fine. You don't need to configure anything for the Zaptel, except if you want to use Asterisk functionality that needs a timing source. But that should be the second step. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] integrating asterisk smoothly
[EMAIL PROTECTED] schrieb: Hello List! I would like to integrate a Asterisk box in my current (german) telephone setup. Right now it looks like this: Provider -- DSL/ISDN Splitter -- Telephone-System(Box) or TK-Anlage ;) I have read that you can put Asterisk between my Splitter and the Telephone-System-Box, so that for the beginning asterisk will just forward the incoming and outgoing calls. The rest of the original Telephone Setup should stay the same. I think i will need a TDM400P for this. Depends on your current PBX (TK-Anlage). If it is connected to regular ISDN line (Mehrgeräteanschluss) you need a BRI-type ISDN card (Fritz or HFC-S, if you need more lines, quadbri or octobri come to mind). -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN ASTERISK Cabling...
Alainn wrote: 4) I need to connect 8 isdn phones However, what would I need in order to connect asterisk to ISDN phones within the office? Quad or Octobri cards from junghanns et al. I need not ask that we are operating on a shoestring... The quads will set you back ~600 EUR, the Octobri ~1000 EUR. Maybe you should think about VoIP hardphones ... -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Implementing a ISDN home PBX
Arik Funke wrote: Hi, I would like to implement a inexpensive home PBX with Asterisk. I have an internal ISDN bus with 6 ISDN phones. I now thought, I connect a Fritz card to my Mehrgerateanschluss (Point-to-Multipoint) supplied by my provider and a second Fritz card to the internal bus. Will this work? No. Use the Fritz!-card for the external connection, but an HFC-card for the internal bus. (see http://www.voip-info.org and search for HFC) -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Debian Bristuffed Asterisk work ignoreBeronet cards ?
Robert Rozman wrote: when loading qozap it says that no multibri card was found although lspci shows it... There were quite some rumours about bristuff not liking other than junghanns cards, but don't know if something happened http://www.beronet.com/download/card_installation_guide_en.pdf On page 37 you'll find that bristuff must be patched in order to recognize other cards. thanks for info. I've read that but on quadbri Beronet card we purchased 2 months ago everything worked without any changes... Do you know where to find those patches and if they are really necessary ? Did you try contacting beronet about this? I mean, you gave your business to beronet, might as well get something back ... -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc troubles
Nicolas Olivier wrote:
I'm trying to setup a small BRI ISDN - voip gateway.
The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
The versions i'm running:
kernel-2.4.27
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
zaptel modules 1.0.7
zaphfc is from bristuff-0.2.0-RC8e
When I'm doing the insmod on zaptel, zaphfc, zaprtc:
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:12.0
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
0xc2d58000(0x2d58000) IRQ 12 HZ 100
zaphfc: Card 0 configured for TE mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
Registered Span 2 ('ZTRTC/1') with 0 channels
Real Time Clock Driver v1.10e
I'm using zaprtc as the gateway is running on a VIA motherboard without USB controller.
[..]
Why are you running zaprtc? zaphfc provides your needed timing source.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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Re: [Asterisk-Users] zaphfc troubles
Nicolas Olivier wrote: Quoting from: http://www.voip-info.org/tiki-index.php?page=Asterisk%20Zaptel%20Installation As I haven't got a Digium card, I need a timer which can be provided by ztdummy, zaprtc or zaprai. But anyway the results are the same with or without zaprtc loaded. Irregardless of your problem, the ZAPHFC cards do provide the timer needed for MOH, IAX trunking etc. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I hide caller id on the fly (per each use setting) on Bristuffed * and quadbri
Robert Rozman wrote: I wonder if I can hide caller id for just certain users. Can I override caller id setting for show or hide on the fly from dialplan ? Did you try setcallerid()? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk connects to ISDN via Fritz!Box Fon 7050 anyone?
Good morning, did anyone try to use asterisk behind a AVM FRITZ!Box and utilize the Fritz!Box as the connection gateway to ISDN? Any and all information is greatly appreciated. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents CallBackLogin and HangUp to calling party on pick-up
Hello, we have setup a queue with a couple of agents, all of which are joining in via CallbackLogin. 1 out of 10 calls coming into the queue will get hung-up upon as soon as the agent picks up the phone. We are running 1.0.6 bristuff RC7k (single HFC-card). SIP phones, ATAs and outside mobile phones. Anyone else experience this kind of behaviour? Anything I can do to pinpoint this problem? Thanks for any pointers. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Odp: Re: [Asterisk-Users] capi problem with dialout
Pawe Staszewski wrote: Hello I live in poland and :) local numbers are: 752 (7 digits) zone prefix: 32 country prefix: 48 And i must add that i am behind a local PBX (Alcatel 4200E) Configured isdn port with msn 7523071 Why dial in is working but dial-out not ... ?? maybe your local PBX requires a 0 in front for an outside line? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi problem with dialout
Pawe Staszewski wrote: Hello i try with 0 and -- Executing Answer(SIP/478-c9a2, ) in new stack -- Executing Dial(SIP/478-c9a2, CAPI/7523071:07522333) in new stack -- data = 7523071:07522333 -- capi request omsn = 7523071 == found capi with omsn = 7523071 == CAPI Call CAPI[contr1/7523071]/7 -- Called 7523071:07522333 -- CONNECT_CONF ID=001 #0x0b3f LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- INFO_IND ID=001 #0x1987 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8 InfoElement = 81 81 -- DISCONNECT_IND ID=001 #0x1988 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3481 Before going any further with Asterisk, try to verify your CAPI setup is able to dial out. Install something like capi4hylafax, and see if you can dialout. If that works come back to asterisk and apply what you have learned. chan_capi works at least as reliable as the bristuff. I am running a AVM C4 plus a HFC-S based card in my asterisk server. The CAPI stuff got me started, bristuff came later. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debugging zaphfc + PBX integration
Gavin Hamill wrote: On Friday 15 April 2005 22:50, Olivier MONNET wrote: Hello, It can be that you need power on your ISDN bus. [..] I have just added an ISDN NT to power the bus: where does one go to find broken NT1 boxes?! We have three ISDN2e NT1s but they are all active and working, I don't think I would be very popular if I removed one 'to play with' :) Try it during off-hours. If it works, go to beronet oder junghanns and buy one. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hylafax and Asterisk
Kib Eki wrote: My question: How can I tell * to ignore special DIDs and let them through to Hylafax? Don't put them in /etc/asterisk/capi.conf as incomingmsn. You can still use them for outgoing if you want to though. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel and Fritz Card
Robson Ribeiro wrote: Does anyone has instructions on how to install the Fritz PCI Card with Zaptel? There is no clear instructions in Junghanns.net nor on the Fritz Card Do you want to install the Fritz! Card only, or in conjunktion with a Zaptel card? If you only want the Fritz! card, only chan_capi is need from junghanns.net If you want to install a zaptel card as well you will need an additional driver for the zaptel card you have. If it is a HFC ISDN card, you must download the bristuff of junghanns.net in addition to the chan_capi. HTH -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel and Fritz Card
Robson Ribeiro wrote: Hi Oliver, I am trying to install only the Fritz Card. But according to the instructions on: http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install it doesnt work. The directories, even the changes that they suggest on the makefile are not there!! I am really disappointed I have been on this for hours!! :-) Welcome to the club. Before doing anything with asterisk, make sure CAPI works. Ignore asterisk for the moment. Once you have CAPI working (for example with capi4hylafax or something), go back, get the latest *stable* asterisk release and built it, ignoring chan_capi for the moment. Make sure it works but using a soft-client to call-in to your system. After that, download chan_capi from junghanns.net and make make install it. If you receive errors in between, come back here (with a new thread) and report your problems as well as ways you have tried to solve your problem. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 x ISDN2 hardware...?
Marc wrote: Is it possible to use hylafax and asterisk with only the AVM C4? Or do I need a separete fax modem? Works fine and dandy with a single AVM C4 here. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI/Dialing out
Philip Hofstetter wrote:
capi.conf:
[..]
[interfaces]
msn=0442607572
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
devices=2
extension.conf:
[ch-fest-netz]
exten = _0[1-9].,1,Dial(CAPI/0442607572:b${EXTEN},30)
Are you sure 044260xxx is your MSN? In germany the MSN is your phone
number without the local area code.
rgds
pos
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Re: [Asterisk-Users] asterisk and outlook
Anton Krall wrote: I have outlook 2003 on my computer and wanted to check if there is a way of connecting outlook with asterisk so that caller id name could be set based [..] Go to http://www.voip-info.org and do a search for TAPI. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie uk questions...
Darrell Berry wrote: [..] - failing that, what my options for *-compatible, UK-legal interconnections between a *-based PBX and UK PSTN? I'm looking for more channels than I will get from ISDN-2e, but less than ISDN-33 (probably): enough for say 4-8 simultaneous incoming/outgoing calls. I admit this is the area I'm least clear on! You might want to look at the QuadBRI cards by junghanns.net They offer support for 4 ISDN-BRI ie. 8 channels. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 7, Issue 296
Aldo Bergamini wrote: I am using slackware 10.1 (kernel 2.4.29) and I am getting the following when I issue gcc -v while I never compiled chan_capi I thought you would need a 2.6 kernel to use it. You don't need a 2.6 kernel for chan_capi. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone with headset
Thibault Lamy wrote: Do anyone have any experience with SIP phone that support a headset ? We have Budgetone phones but we need headsets. I am using Snom 190's with Snom head sets and like them a lot. On my list of things to do is using the Snom with a Labtec PC headset and see if that works as well (would be cheaper). -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Capi channel - can I route call to another channel or back to PBX and free current channel ?
Robert Rozman wrote: I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. IIRC you can't do this. You must connect your ISDN PBX to a HFC card and route it thru there. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI not installed
A. Peverelli wrote: I own a ME600 EPIA Mini-ITX main board with the latest Debian distro (kernel 2.6.8) with isdnutils-base, libcapi20-dev, libcapi20-2, isdnactivecards installed. I have a QuadBRI module by Junghanns with bristuff-0.2.0-RC3a (with asterisk-1.0.3, zaptel-1.0.3 and libpri-1.0.3), and chan_capi-0.3.5. I followed all INSTALL instructions, but I have some strange behaviour. All modules seems to be correctly installed and actives, but on /dev I find only capi20. Anyway, starting Asterisk, I recevive a 'CAPI not installed!' error on chan_capi load and I can't find why. Anyone has some idea? quadBRI CAPI!!! The quadbri cards do not use/support CAPI. If you don't have another CAPI capable device in your system you can't/shouldn't use CAPI (I guess you could use CAPI via mISDN, but what is the point?) -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN zaphfc - What kernel are you using successfully?
Hi, some people report good success with the zaphfc cards, others, incl. myself have mixed results. I am using the debian stock kernel 2.4.27 with mixed results. Anyone care to tell what kernel(s) you on successful zaphfc integrations? Thanks. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN zaphfc - What kernel are you using successfully?
Thibault Lamy wrote: some people report good success with the zaphfc cards, others, incl. myself have mixed results. I am using the debian stock kernel 2.4.27 with mixed results. We are using 2.6.10 self-built kernel on debian unstable zapfhc works fine, we are able to send/receive calls and calelrid works What version asterisk and bristuff do you use? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
Anton Krall wrote: Guys.. which ATA is better for connecting analog phones (features, stability, experiences, etc)? Sipura 2000 or Handy Tone 286, etc? What are you experiences? In my experience the Sipura 2000 has three hardware advantages: * 2 independent phone ports * Mounting holes * The price for a single Sipura 2000 is less than the price for two Grandstreams. As far as software and compatibility with * goes, I only have experience in a LAN environment, where both worked (with the right firmware) without a problem. The Sipuras seem a little bit louder (or so the users tell me). -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's
Sascha E. Pollok wrote: Good evening, allow me to join in right here. Which ATA/TA would you suggest for connecting analogue fax machines to Asterisk? One of the ones named before or e.g. a ATA-186 made by Cisco? At the moment I am deploying Grandstream ATAs for faxing machines with out a problem so far. -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc NOTICE[6799]: chan_zap.c:7685 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
Hi, my success story with the zaphfc incl. florz patch has been to early. Allthough sound drop outs no longer happen, the following happens after a longer period (2 days) of inactivity on the asterisk box. Feb 13 22:30:15 NOTICE[6799]: chan_zap.c:7685 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Maybe this is helpful to find where the problem is. I will go and unload the drivers (and hope not to crash the box). -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
Remco Barende wrote: I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. [..] I have three different locations with HFC cards. I had the same stability problems on ALL of the installations. Since RC5 plus the florz patch *ALL* of the stability problems have vanished. No more seconds of silence, no more unavailability messages. It just works now. I won't touch the installations for a long time :-) If I ditch the HFC-S card and replace it with another card that will work with mISDN or chan_capi will this solve my problems? I have good results with an AVM C4 card using the CAPI drivers. I started out with an old ISA AVM B1 card which had echo problems, which got fixed with some later chan_capi driver releases. -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
Remco Barende wrote: Since RC5 plus the florz patch *ALL* of the stability problems have vanished. No more seconds of silence, no more unavailability messages. It just works now. I won't touch the installations for a long time :-) Thanks. I did look in the wiki and the webpage of florz but thought that the patch was only for multi card installations, therefore I never applied it. Will try it tonight. Just out of interest, why was that patch never incorporated in bristuff? I *assume* it has to do with the license. Junghanns wants to keep all commercial rights to the driver. But maybe Kapejod or Florz are able to shed some light on the issue. -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bri dropping calls
Altus Snyman wrote: Where do you get this new version of bristuff,I had a look on the webpage and there's only RC3 My first action every morning is to look at the top of this page: http://www.junghanns.net/asterisk/downloads/?C=M;O=D -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] announcement: astfax 1.0
Ken Jones wrote: astfax allows you to create an email to fax gateway. Are we going to see some integration of astfax with Courier-MTA/IMAP? -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bri dropping calls
Michael Bielicki wrote: conservation. (And yes, the latest to get is bristuff_0.0.2RC5 [RC6 seems to be for quadbri and octobri cards, only]) dIf you reread his email, he is stating that he has a quadbri Oops, you are right. Note to self, don't write ML messages before first coffee. Note to OP: The question whether the calls are getting really dropped or just silenced for a couple of seconds still stands. -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automated CallbackLogin
Hi, we want to provide our users with a Click To Login interface for the AgentCallbackLogin. Any sample.call or AGI anyone has developed out there? Any and all help is greatly appreciated. -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Germany specific settings for Grandstream ATA286 - Polarity reversal, impedence and onhook voltage
Hi, the new Grandstream release for the ATAs allows the setting of the FXS impedence, the Onhook Voltage and the Polarity Reversal. Anyone know how these should be set in Germany? -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC-5/S + Asterisk
Thomas Niesel wrote: [..] = Your Card should work with i4l (bad) and zaphfc from junghanns (good) Forget about Capi and mISDN, go for kernel 2.6.10 along with zaphfc, ztdummy together with uhci for timing and ask the wiki for more details. Pardon my ignorance, but isn't one of the reasons for zaphfc to provide a ZAP timing source? So, if you have zaphfc card together with the bristuff from http://www.junghanns.net you don't need ztdummy, do you? -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuff ZapHFC and Loosing D-Channel
Remco Barende wrote: Did you ever find the answer to your question? I am getting the same message on the console every second: == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up etc. etc. etc. I'm running Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 The error is only visible however if I run * with -v (but I guess I shouldn't see these messages nonetheless)? No, I still have these messages, and was hoping RC5 would fix them. Nice to know that upgrading won't help :-( Do you get calls which stop in the middle of the conversation as well? -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bristuff ZapHFC and Loosing D-Channel
Using the latest(?) bristuff (Asterisk 1.0.4-BRIstuffed-0.2.0-RC3a) I have problems with loosing the D-channel. Most of the time, after the message PRI D-channel down it only takes a second or so to come back up, noted by the message PRI D-channel up However, today most of the time the D-channel stays down. Calls come in, but can't be answered. Does anyone know of a fix for this, or might have some insights on how to circumvent this problem? Any and all help is greatly appreciated. -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile
Hello Stephan,
Another way is to set the callerid in your extensions.conf via exten
= 807440,2,SetCIDNum(0${CALLERIDNUM}).
Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
This did the trick. I will go and update the Wiki,,,
Thanks and have a good weekend.
--
Best regards
Peer Oliver Schmidt
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[Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile
Hello, I've added a ZAPHFC card to my CAPI based system. Calls coming in via ZAPHFC do not forward the caller id to the SIP phones. Calls coming in via CAPI do forward the caller id to the SIP phones. Any and all help is greatly appreciated. The (hopefully relevant) conf file excerpts are: extensions.conf === exten = 807440,1,Answer exten = 807440,2,Noop exten = 807440,3,Dial(SIP/26) zapata.conf === [channels] switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan=local prilocaldialplan=local pritrustusercid = yes echocancel=yes immediate=no group = 1 context=zaphfc-in channel = 1-2 zaptel.conf === loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 The log file (with SIP DEBUG) = -- Executing Answer(Zap/1-1, ) in new stack -- Accepting call from '1729731418' to '807440' on channel 0/1, span1 -- Executing NoOp(Zap/1-1, ) in new stack -- Executing Dial(Zap/1-1, SIP/26|20|t) in new stack We're at 10.1.3.111 port 19078 Answering with preferred capability 0x400 (ilbc) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5060;line=8vf8777b SIP/2.0 Via: SIP/2.0/UDP 10.1.3.111:5060;branch=z9hG4bK687aab06 From: asterisk sip:[EMAIL PROTECTED];tag=as3338f85b To: sip:[EMAIL PROTECTED]:5060;line=8vf8777b Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 21 Jan 2005 17:46:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 236 -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile
Jens, thanks for the feedback.
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
I think you didn't set usecallerid=yes in your zapata.conf?
Added it, rebooted, no change. (Before, I just had pritrustusercid =
yes, only.)
Another way is to set the callerid in your extensions.conf via
exten = 807440,2,SetCIDNum(0${CALLERIDNUM}).
Changed it, now the funny part comes:
extensions.conf
exten = 807440,1,Answer
exten = 807440,2,SetCIDNum(0${CALLERIDNUM})
exten = 807440,3,Dial(SIP/26,20,t)
exten = 807440,3,VoiceMail2(su25)
exten = 807440,103,VoiceMail2(sb25)
exten = 807440,104,Hangup
but the log says:
-- Accepting call from '1729731418' to '807440' on channel 0/1, span 1
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetCIDNum(Zap/1-1, 0) in new stack
It does not add the callerid it has two lines above ???
I know there have been some changes to the CID structure sometime within
Asterisk. But, this is using the bristuff download and install script.
The same problem happens using the debian packages (1.0.3) from marlow.dk.
Any and all help is greatly appreciated.
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Re: [Asterisk-Users] Fax and PRI
Lee Howard wrote: On 2005.01.19 01:39 tim panton wrote: My options include; 1) get a basic fax machine and plug it into a (iaxy/sipura?) ATA. Configure the ATA to do alaw so asterisk doesn't have to do any transcoding and hope it works. 2) use spandsp on my asterisk system and add suitable print queues 3) get an separate analog line and plug a fax into it [..] Andrew's experience seems to indicate that you could mix numbers 1 2 with success. In other words, send with #1, and receive with #2. I don't have any experience with #2. My experience with #1 is that sending works fine, but receiving is not fine. I'll second this. In my setup, asterisk has an AVM C4 BRI card. All incoming faxes are being dealt with by the server and stored within a database, with some basic web frontend running as an ActiveDesktop object within the users desktop. The majority of fax sending is done using a plain old fax machine connected to a GrandStream HT-286. This setup works just fine and dandy. It saves on paper, because not all faxes have to be printed, especially spam faxes. rgds pos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Execute dialplan command at startup
Bill Seddon wrote: How can Asterisk be configured to execute some number of dialplan commands when it is started or restarted? [..] In the meantime I'm hoping that it is possible to use the built-in database and be able to run some kind of autostart context. Does such a facility exist? Without getting into details, I would create a call file in the outgoing spool directory of asterisk within the asterisk startup script which calls a specific application. Haven't tried, but should easily work. Let me know, how it works. rgds pos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bristuffed Asterisk 1.0.3 hfc-s card doesn't work
Andrew Thrift wrote: Hi Remco, just wondering how you got Asterisk 1.0.3 BRI-Stuffed. On Junghanns.net I can only see 0.1.0-rc4 of bri-stuff and it uses something like asterisk 0.8 I got 1.0.2 using the bristuff v0.in the download section. It is not linked from the main page, but available in the download directory. http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-rc2b.tar.gz -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN, bristuff and hylafax
Hi, anyone has the above combination working? On one * system I use a capi supported device which allows me to run HylaFax in parallel with Asterisk. The other machine has a single ISDN BRI HFC card, only. Anyone who has a single HFC card and is using Asterisk and Hylafax together like to share his config? Thanks. -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax to email
Michael Welter wrote: How do I fax a .tiff file with asterisk? [..] Use Steve Underwood's spandsp library and TxFax function in Asterisk. See http://opencall.org It is my understanding that TxFAX does not allow to callout and send a fax, but to be used in Faxpickup only. Do you have it working in the manner you described, and if yes, could you update the Wiki, or share your setup here? Thanks. -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax to email
Steve, It is my understanding that TxFAX does not allow to callout and send a fax, but to be used in Faxpickup only. Why would anyone implement something as dumb as that? I didn't :-) Great :-) Do you have it working in the manner you described, and if yes, could you update the Wiki, or share your setup here? Many people have it working that way. Very few people use it for fax pickup. To change it from answerer to originator mode you just need to add the caller option when you call txfax. I asked my asterisk for the command and does not show anything regarding what to put into options. Is it save to assume, that to send a fax to 415-555-1234 the correct statement would be: TxFAX(myfax.tiff|caller=411234) ? Thanks -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
