[Asterisk-Users] broken CDR (Master.csv) reports with HFC cards in Asterixk 1.2.x?
I use Asterisk with a HFC-S ISDN BRI card. This card needs bristuff patch from Junghanns.net. After upgrading to Asterisk 1.2.x, my CDR reports (located in /var/log/asterisk/cdr-csv/Master.csv*) are broken. Instead of telephone numbers, I get random characters like 'H? or $%. Sometimes, though, the telephone numbers are fine. The issue was also mentioned on Digium's asterisk forum: http://forums.digium.com/viewtopic.php?t=4400 http://forums.digium.com/viewtopic.php?t=4528 It was also mentioned and confirmed on [EMAIL PROTECTED] and AMP forums. Does anyone have an idea how to solve it? -- Tomasz Chmielewski WPKG - http://wpkg.org Software deployment with Samba ___ --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] [EMAIL PROTECTED] isdn
Alejandro Vargas schrieb: 2005/11/29, Tomasz Chmielewski <[EMAIL PROTECTED]>: you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card, not HiSax (well, technically, you could use HiSax too, but avoid that if possible). I prefered to use hisax because it is already included in asteriskathome (why bristuff is not included?) you can use hisax module, so isdn4linux, but it's not very well supported by asterisk. bristuff-0.3 is listed as experimental, should I use 0.2 (stable)? use 0.3 with asterisk 1.2, 0.2 version won't work. And then... I will obtain the module zaphfc, then how to configure asterisk to use it? normally, as a zapata interface :)) although it may seem as magic, it's not that hard; if you configure zaphfc, ask here at the mailing list, or me directly, as I use it with [EMAIL PROTECTED] 2.0 -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] [EMAIL PROTECTED] isdn
Alejandro Vargas schrieb: I'm testing asteriskathome with an ISDN card 00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) I found there is the module hisax and I loaded it: hisax 456177 0 crc_ccitt 2113 2 hisax,zaptel isdn 133409 1 hisax dmesg shows this: HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 2.46.2.5 HiSax: Layer2 Revision 2.30.2.4 HiSax: TeiMgr Revision 2.20.2.3 HiSax: Layer3 Revision 2.22.2.3 HiSax: LinkLayer Revision 2.59.2.4 I'm not sure if it is detecting the hardware, and I'm not sure what config I must do in asterisk. The documentation is confusing, because the references to hisax indicates to use cahan_modem_i4l but comments in modules.conf says "DON'T load the chan_modem.so, as they are obsolete in * 1.2". I tryed anyway but chan_mdem_i4l does not appear whan I type reload. you have to use bristuff-0.3.x (from www.junghanns.net) with a HFC card, not HiSax (well, technically, you could use HiSax too, but avoid that if possible). -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] setting variables in a .call file - how?
How can I set a variable in a .call file? I wanted to add a fax header with SpanDSP / txfax, and the information on soft-switch.org says: "If the variable LOCALHEADERINFO has been set when txfax is run, the value of that variable will be used as the user defined part of the header text". So I tried to set that variale in a .call file: Channel: $CHANNEL/$FAXNUM MaxRetries: 2 retryTime: 60 WaitTime: 20 SetVar: LOCALHEADERINFO=CompanyName Application: txfax Data: $DATADIR/$ATTNAME.tif|caller but it doesn't make any difference, fax header is not added. So perhaps I'm setting that variable in a wrong way? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] Small office with all employee's offsite
Jason Marshall schrieb: I'm sure these questions have been answered at some point, but I'm too new to this stuff to know the right words to plug into the search function to find what I need. I have never touched Asterisk before, but have wanted to for some time. Now I finally think I'm going to bite the bullet, as I have a real-world application for it! My office consists of two employees, neither of whom work in the office physically. Here is what I'd like to do. Hopefully someone can tell me what I need to do/buy/configure/install to make it work... I want all calls to come into the Asterisk box in the main office. I want all incoming calls to be recorded (not as concerned about outgoing calls). Both employees have regular POTS telephone lines (one fellow has a land line and a cell, the other has just a land-line). I'd like callers to be presented with a short menu of options, the behavior of which might change depending on the time of day (for instance, at night, I'd like both the "sales" and "support" calls to go to one employee, while during the day I'd like sales to go to one person, and support to go to another. I'd also like to have an answering machine (built into Asterisk?) pick up calls that go unanswered. what you're looking for is basically [EMAIL PROTECTED] - http://asteriskathome.sf.net It has all features you mentioned already integrated (and many more, too). -- Tomek http://wpkg.org/email2fax email2fax - email to fax gateway for Asterisk ___ --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] is it possible to force faxdetect / disable echo cancellation for a given extension?
I have the newest SpanDSP setup with asterisk 1.2. Generally, 99% of received faxes are OK, but only about 20% of faxes sent are delivered properly. In zapata.conf I have set faxdetect=both, but it doesn't seem to disable echo cancellation (I looked into asterisk logs and it says "Enabled echo cancellation on channel 1, Engaged echo training on channel 1" whenever I fax out). Why doesn't asterisk detect that it's faxing? So my idea was to disable echo cancellation whenever fax number is called: exten => 27229932,1,Answer exten => 27229932,2,DISABLE_ECHO_CANCELLATION exten => 27229932,3,Goto(in_fax,1) (...) And do the same when I sent faxes using .call files: OPTIONS: DISABLE_ECHO_CANCELLATION Channel: $CHANNEL/$FAXNUM MaxRetries: 1 WaitTime: 20 Application: txfax Data: $DATADIR/$ATTNAME.tif|caller Is it possible to do something like that? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] spandsp changelog
I have some issues with sending some faxes using spandsp (receiving faxes is generally OK). I noticed new versions of Spandsp come out every month or two, but they don't contain a changelog (they do, but it's outdated). Does anyone know if one can read anywhare what changed in Spandsp? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] curious bandwidth usage (incoming taking 3x more)
While we are in a process of moving our office, we use soft phones which connect over WAN/VPN to our Asterisk box in the old office. We use IAX2 softphones configured to use iLBC. When we call out using the softphone, the bandwidth usage is at about 3 KB/s (in and out), quality is fine. However, when someone calls us, the bandwidth usage is about 10 KB/s (in and out), audio quality is also fin. How can this difference be explained? It seems to me, that asterisk uses the ulaw codec when it calls our softphones, but after trying to play with iax.conf, I don't know how to make asterisk to try to negotiate iLBC first, then gsm, and ulaw at the very end. -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] is it possible to connect to Asterisk from an external application?
Is it possible to connect to Asterisk from an external application? What I mean, to connect and "execute" its own extensions, created by some other program: exten => 1234567,1,txfax(/home/steveu/testfax.tif|caller) or exten => $NUMBER_I_WANT,1,txfax($FILE_I_WANT|caller) and Asterisk will dial this number and "execute" these extensions. If it's possible, how do I do it or where can I read more about it? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] "GSM cards" / "mobile phone cards" for Asterisk?
Daniel Varella de Oliveira schrieb: > Tomasz, > > I'm from Brazil, and we are using here a solution that is based on a box where we can connect a GSM cellphone and use this directly to a phone or PBX extension. > I think that you can use some Digium's card (FXS or FXO) on your server, connect this GSM box there, and route your cellphone calls through this box. > > There are boxes with just one channel and others up to six channels. > They have a lot compatibilities with the most common cellphones. looks interesting. do you know by chance how much such a single-cell box cost (more or less)? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] "GSM cards" / "mobile phone cards" for Asterisk?
I was wondering if there is something like that on this Earth: Some of our users are "mobile users" - they are rarely in one place for longer than 15 minutes. They use mobile phones a lot. From our mobile operator we have an offer which allows us to call for free between our mobile phones. So the idea is to put a SIM card inside the Asterisk box, equipped with a special card, a card which would be a mobile phone really. This would allow all office users to reach our mobile users without the need of buying additional phones for the office users. Office users would call Asterisk over IAX, and asterisk would call "mobile users" using a free GSM/mobile. Does anyone have an idea if such cards exist, and if so, if they work with Asterisk? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] Help Installing Asterisk.....
Bharat M. Sarvan schrieb: Hello All, Are there any packages need to be installed before installing the Asterisk….? Cos I am facing problems compiling the zaptel for the Asterisk... Kindly please do let me know… If you're starting with asterisk, you might try [EMAIL PROTECTED] - http://asteriskathome.sf.net It's a distribution with a working asterisk + many useful addons, which you would want to add anyway (web interface etc.). -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] spandsp / txfax exit codes / logging?
Doug Lytle schrieb:
(...)
Is it possible to somehow read spandsp / txfax exit codes?
Run Asterisk in debug mode [asterisk -d] and use the -debug option on
the spandsp command line. Mine is as follows:
exten => s,3,rxfax(${FAXFILE}.tif,DEBUG)
After I get the debug output, I use cat and grep to break out the
app_rxfax.c to a fax log:
[EMAIL PROTECTED] asterisk]# cat full|grep -i app_rxfax.c >faxlog
With the output below:
Oct 27 15:36:10 DEBUG[21623] app_rxfax.c:
==
Oct 27 15:36:10 DEBUG[21623] app_rxfax.c: Fax successfully received.
Oct 27 15:36:10 DEBUG[21623] app_rxfax.c: Remote station id: 269xxx
Oct 27 15:36:10 DEBUG[21623] app_rxfax.c: Local station id:
Oct 27 15:36:10 DEBUG[21623] app_rxfax.c: Pages transferred: 5
Oct 27 15:36:10 DEBUG[21623] app_rxfax.c: Image resolution: 7700 x 7700
Oct 27 15:36:10 DEBUG[21623] app_rxfax.c: Transfer Rate: 9600
Oct 27 15:36:10 DEBUG[21623] app_rxfax.c:
==
but the debug mode means extremely big logs :))
anyone knows how does the log look like for a successful fax
transmission (so, sending fax from asterisk)?
I'm curious, as I have no problems with incoming faxes, but the outgoing
doesn't seem to work well for some reason.
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[Asterisk-Users] spandsp / txfax exit codes / logging?
Is it possible to somehow read spandsp / txfax exit codes? What I mean, I never know if the fax sent through the Asterisk box was sent successfully, or not (i.e., a real person picked up the phone instead of a fax machine). A possibility of reading an exit code, or a log file would allow to build some kind of "fax confirming" (via email/web page/etc.). Are exit codes (or logging, or something similar) possible with spandsp / txfax? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] Cisco 7905G Power over Ethernet - does it work?
rulle mus schrieb: Hello Tomasz, I got the 7905 working with an Dell POE switch without any modifications of cables, the 7960 also works on the Dell switch but you have to modify the cable. I also tried the Netgear FS108p and it does not work with the 7905, 7912 and 7960 as I have tested. Even with modified cables no go on the Netgear. I believe the Cisco uses the CDP protocol to get juice from the switch, and the Netgear doesn't understand that. thanks. could you tell me the model of the Dell POE switch you use? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] Cisco 7905G Power over Ethernet - does it work?
Sergio Chersovani schrieb: Sergio Chersovani ha scritto: No way to power up the phone is the the switch can be forced to send power in any case. I meant that the phone can power up with a custom poe injector that does not care about 802.3af does poe injector = poe switch (is poe switch and poe injector the same thing but a different name)? if so, it means my switch is not "dumb enough" or what? anyone knows if it can be "dumbified" (some special cable, adapter etc.)? ___ --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] Cisco 7905G Power over Ethernet - does it work?
Perhaps this question should be directed to Cisco support, but since
these guys made me nuts ("please check that your cable is plugged in
correctly" etc.), I thought I'd ask here.
We bought a Cisco 7905G phone, which boasts to have PoE (Power over
Ethernet) support.
We have a Netgear FS108P PoE switch, which works with other PoE devices,
but not with this Cisco phone.
I searched the voip wiki - http://www.voip-info.org/wiki-Cisco+POE - and
found a suggestions to reverse some cables in the ethernet wire.
So I did, but Cisco 7905G phone still doesn't power up.
Does anyone have any suggestions on how to make this phone work with a
PoE switch?
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Re: [Asterisk-Users] configuring Cisco 7905G for SIP - how?
Sergio Chersovani schrieb: Tomasz Chmielewski ha scritto: I searched the whole "Cisco IP Phone 7905 Series Administration Guide", but besides the copyright notes, logo is not mentioned. lol http://www.google.com/search?q=site%3Acisco.com+7905+bmp2logo I see, it's called bmp2logo.exe and it's for Windows only :( anything like that that works with Linux? BTW, I searched through 7905 Admin Guide for h323 (as it's the first link in google for "cisco 7905 admin guide"), assuming it's the same, and neither logo nor bmp2logo are mentioned there :) -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] configuring Cisco 7905G for SIP - how?
Erik schrieb: Logo file for a 7905 isn't a BMP file, it has to be converted with a cisco util aah, now I see. and what tool is that and where can I get this? in my firmware package I have only two tools: - cfgfmt.linux (a tool for converting text configuration into cisco format, which doesn't recognize 80% options) - prserv.linux I searched the whole "Cisco IP Phone 7905 Series Administration Guide", but besides the copyright notes, logo is not mentioned. -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] configuring Cisco 7905G for SIP - how?
Sergio Chersovani schrieb: Tomasz Chmielewski ha scritto: But I have (had?) a problem with 7905 phone (still minor problems with that, like a wrong timezone). You can easy change it with the phone web page. yup, I just figured that out :) one more issue though. any idea why a custom logo isn't displayed on a 7905G phone? I see in tftp server logs that the logo file is downloaded, but it isn't there on a telephone display. This same logo is displayed fine on a 7960 Cisco phone. -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] configuring Cisco 7905G for SIP - how?
tijmen van den brink schrieb: > I set up a Cisco 7960 in about 20 minutes with this document. I hope it works for you. > > http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html I too managed to set up a 7960 phone. But I have (had?) a problem with 7905 phone (still minor problems with that, like a wrong timezone). BTW, I managed to solve it - the contents of the SEP0014690620AA.cnf.xml file have to be like this (with the right asterisk box IP address), and then it downloads the other files: 2000 192.168.11.15 Too bad Cisco 7905 documentation doesn't even mention *.cnf.xml files, their contents, etc. Too bad Cisco binaries attached to 7905 firmware complain "option not recognized" when parsing even default config files (you need to "convert" the text files to some other mysterious format)... -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --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] configuring Cisco 7905G for SIP - how?
After reading the specifications of Cisco 7905G phone ("supports SIP,
easy to manage" etc.), we were so foolish and bought it.
Now we learned the hard way that we have to pay additionally for SIP
firmware.
So two months after purchase, after much struggle with Cisco
the-so-called "support" we have a shiny Cisco 7905G phone, support
contract, and a newly downloaded SIP firmware.
Unfortunately, the instructions attached to the SIP firmware seem to be
for a different phone, as they state that the 7905G phone should
download "lddefault.cfg" config file (which took some time to configure,
as it's 50 kilo big). In our case, the 7905G phone tries to download
SEP0014690620AA.cnf.xml, and XMLDefault.cnf.xml.
Does anyone have a good, step-by-step SIP upgrade instruction for this
phone?
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[Asterisk-Users] add 0 (zero) to incoming callerID - how?
I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial feature in the telephone (because the phone doesn't know that it should add 0 before the number). So the idea is to manipulate the incoming callerID number, and to add a 0 before it. This way the telephone user will be able to callback/redial. How can I manipulate the incoming callerID number (and add 0 before it)? -- Tomek http://wpkg.org ___ --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] spandsp + HFC poor fax quality?
Tomasz Chmielewski wrote: I've been trying to set up incoming faxes using spandsp with a HFC card. Unfortunately, incoming faxes are of very poor quality, the pages are not transferred wholly (sometimes only a bit of a page is transferred etc.). I tried sending faxes from different fax devices, always the same issue. So the last thing that comes to my mind is that my timing is still *not* fixed. Anyone has an idea how to fix timing issues with a HFC card? updating from 2.4.x to 2.6.x kernel seemed to help. I also compiled newer bristuff (0.2.0-RC8e). So either of the two helped. -- Tomek -- Znajdz swoja milosc na wiosne... >>> http://link.interia.pl/f187a ___ 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] spandsp + HFC poor fax quality?
I've been trying to set up incoming faxes using spandsp with a HFC card.
Unfortunately, incoming faxes are of very poor quality, the pages are
not transferred wholly (sometimes only a bit of a page is transferred etc.).
From what I've read, I may have troubles with correct "timing" (and
need to set up ztdummy etc.).
On the other hand, it is impossible to set ztdummy if one uses zaphfc
module (HFC card) - because with HFC cards, ztdummy/timing is not needed.
I tried different libtiff versions with spandsp (ending with 3.7.2 which
seems to be the latest), but I always get this issue (especially, when
the fax contains images etc. more complicated stuff).
A typical fax session looks like that:
DCS with final frame tag
In state 9
Coarse carrier frequency 1699.82 (66)
Training error 0.406683
Training succeeded (constellation mismatch 0.417317)
Start rx document
Start rx page - compression 2
Coarse carrier frequency 1738.28 (6)
Training error 692.017653
Training failed (convergence failed)
Coarse carrier frequency 1699.86 (66)
Training error 0.394675
Training succeeded (constellation mismatch 0.417396)
DCS with final frame tag
In state 5
Coarse carrier frequency 1699.93 (66)
Training error 0.489383
Training succeeded (constellation mismatch 0.576316)
Start rx page - compression 2
Coarse carrier frequency 1699.84 (66)
Training error 0.339757
Training succeeded (constellation mismatch 0.509557)
EOP with final frame tag
In state 5
DCN with final frame tag
In state 8
Sometimes it ends with ghostscript errors:
-- Executing System("Zap/1-1", "tiff2ps -2eaz -w 8.5 -h 11
/var/spool/asterisk/fax/asterisk-3834-1116329874.0.tif | ps2pdf -
/var/spool/asterisk/fax/asterisk-3834-1116329874.0.tif.pdf") in new stack
Error: /limitcheck in --setpagedevice--
Operand stack:
--dict:1/1(L)--
Execution stack:
%interp_exit .runexec2 --nostringval-- --nostringval--
--nostringval-- 2 %stopped_push --nostringval-- --nostringval--
--nostringval-- false 1 %stopped_push 1 3 %oparray_pop
1 3 %oparray_pop .runexec2 --nostringval-- --nostringval--
--nostringval-- 2 %stopped_push --nostringval-- 1 3
%oparray_pop --nostringval-- --nostringval-- --nostringval--
Dictionary stack:
--dict:1052/1123(ro)(G)-- --dict:0/20(G)-- --dict:88/200(L)--
Current allocation mode is local
Last OS error: 22
GNU Ghostscript 7.05: Unrecoverable error, exit code 1
I tried sending faxes from different fax devices, always the same issue.
So the last thing that comes to my mind is that my timing is still *not*
fixed.
Anyone has an idea how to fix timing issues with a HFC card?
--
Tomek
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[Asterisk-Users] Atcom AT-320 call forwarding - how?
Hi, I just wanted to know if anyone managed to do call forwarding with AT-320 phone from Atcom (not by reconfiguring Asterisk)? For example, when I take my lunch break, I would like to forward all calls to my mobile number. Is it possible with Atcom AT320? -- Tomek -- Znajdz swoja milosc na wiosne... >>> http://link.interia.pl/f187a ___ 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] extension based on a dialed number?
C F wrote: exten => 1234560,1,Dial(phonea) exten => 1234561,1,Dial(phoneb) and so on... It's called DID OK, I had that, but it didn't work. What I missed: 1) I have a "zapata" device - HFC-based ISDN card, so I needed to make: immediate=no in /etc/asterisk/zapata.conf (instead of immediate=yes I had previously). 2) I use [EMAIL PROTECTED] / AMP, and I needed: [default] include => ext-local include => from-pstn exten => s,1,Answer ; end of [default] in /etc/asterisk/extensions.conf 3) configure "DID Routes" in AMP web gui: assign a number to an extension. Hope that helps if someone looks through the archives! :) -- Tomek -- Znajdz swoja milosc na wiosne... >>> http://link.interia.pl/f187a ___ 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] extension based on a dialed number?
I have an ISDN line with 10 numbers. The line is then connected to * with one HFC-based card. The format of the numbers is like below: 123456-0 123456-1 ... 123456-9 Now I would like to connect those numbers to different telephones, i.e. when someone dials 123456-0, he/she is connected to the digital receptionist. If someone dials 123456-2, the connection goes to SIP/202 If someone dials 123456-3, the connection goes to SIP/203 etc. What should I look for? -- Tomek -- Startuj z INTERIA.PL! >>> http://link.interia.pl/f186c ___ 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] Web GUI
Marc Khayat wrote: Hello all, I just installed Asterisk 1.0.7 and astguiclient… so you can say I’m very new at this. How can I manage my Asterisk using the web or somehow, since there are too many configuration files and too many variables… You may take a look at [EMAIL PROTECTED] - it includes AMP (Asterisk Management Portal) and other tools. But it will be hard at the beginning, anyway :) Tomek -- Znajdz swoja milosc na wiosne... >>> http://link.interia.pl/f187a ___ 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] Is such a thing as a analog (or even IP) video door entry system available?
Angus Comber wrote: I want to setup a video door entry system. I understand a lot of the systems on the market use proprietary technology. But ideally if the system could connect into a normal analog port or even use IP to my Asteirsk that would be a lot better. Then I could have video phones on users desks so anyone can see who is at the door. Anyone aware of any suitable products. I don't know if that's what you mean, but you may take a look at ZoneMinder? http://www.zoneminder.com/ Tomek -- Znajdz swoja milosc na wiosne... >>> http://link.interia.pl/f187a ___ 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't create Zap channel
Matthew Boehm wrote: Tomasz Chmielewski wrote: Matthew Boehm wrote: first, let me know, if you can dial yourself? (i.e. PSTN -> * - that (zap) card) Yep. I sure can. so everything seems OK. I guess we would need: - /etc/zaptel.conf - /etc/asterisk/zapata.conf - the construction of the extension you are using - maybe "zap show channels", "zap show channel X" etc. Tomek -- NAJTANSZY KREDYT mieszkaniowy w Polsce! 100 tys. zl juz od 333 zl miesiecznie! Kliknij! http://clk.tradedoubler.com/click?p=20565&a=1029167&g=1130041 ___ 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 dialout problems
Tomasz Chmielewski wrote:
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel. Zap channels start at 1.
sorry, it was meant to be g0.
I tried Zap/1 and Zap/2 earlier - with that effect:
-- Executing Dial("SIP/201-152d", "Zap/1/98") in new stack
-- Called 1/98
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Congestion("SIP/201-152d", "") in new stack
OK, I found the problem.
I have two ISDN lines: one from a telco, and the second is a company
internal ISDN system.
Both dialin and dialout work with HFC/zap and telco.
Only dialin work with HFC/zap and internal ISDN system made by alcatel.
With i4l, dialin and dialout works both with a telco and internal ISDN
system.
So I guess an internal ISDN uses some exotic signalling?
Tomek
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Re: [Asterisk-Users] zaphfc dialout problems
Tomasz Chmielewski wrote:
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel. Zap channels start at 1.
sorry, it was meant to be g0.
I tried Zap/1 and Zap/2 earlier - with that effect:
-- Executing Dial("SIP/201-152d", "Zap/1/98") in new stack
-- Called 1/98
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Congestion("SIP/201-152d", "") in new stack
OK, I found the problem.
I have two ISDN lines: one from a telco, and the second is a company
internal ISDN system.
Both dialin and dialout work with HFC/zap and telco.
Only dialin work with HFC/zap and internal ISDN system made by alcatel.
With i4l, dialin and dialout works both with a telco and internal ISDN
system.
So I guess an internal ISDN uses some exotic signalling?
Tomek
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Re: [Asterisk-Users] can't create Zap channel
Matthew Boehm wrote:
Before you jump ahead, yes I do have chan_zap.so loaded..
Call Flow: Asterisk 1 --IAX2--> Asterisk 2 ---> PRI
-- Accepting AUTHENTICATED call from 22.22.22.22:
> requested format = ulaw,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (ulaw|alaw|gsm),
> priority = mine
-- Executing Dial("IAX2/[EMAIL PROTECTED]",
"Zap/R1d/18005551212|60") in new stack
May 5 15:21:37 NOTICE[16153]: app_dial.c:968 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0)
== Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'IAX2/[EMAIL PROTECTED]'
pri debug is irrelevent because the call never makes it to pri. What is
cause 0? Its not listed in the header files.
Nothing is busy on that span.
Any ideas?
yes, I was struggling with that for a long time recently, too, so maybe
I could help.
first, let me know, if you can dial yourself? (i.e. PSTN -> * - that
(zap) card)
Tomek
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[Asterisk-Users] HFC: zapata + bristuff - how to set an outgoing number
I have a HFC-PCI based ISDN card. How should an extension be constructed, when I want to set up a specific outgoing number (I have 10 or so MSN numbers)? For example, when I call 6546 from my SIP phone, I would like to call "100" with an outgoing number of "555" - how should I do this? exten => 5646,1,Dial(Zap/g0/98) Tomek -- Startuj z INTERIA.PL! >>> http://link.interia.pl/f186c ___ 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] Difference between Asterisk and Asterisk@home?
Kib Eki wrote: Hi, can one summarize the main differences between Asterisk and [EMAIL PROTECTED] or point me to a location where i can find such a list? See [EMAIL PROTECTED] site - http://asteriskathome.sf.net - and asterisk site - www.asterisk.org. Basically, asterisk is a program, and [EMAIL PROTECTED] is a distribution with running (and partially configured) asterisk, AMP, etc. and other additional stuff. Of course you have to configure your asterisk hardware yourself. It's like a question: what's the difference between KDE and Debian.. :) Tomek -- Znajdz swoja milosc na wiosne... >>> http://link.interia.pl/f187a ___ 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] Asterisk GUI
pinchien wrote: > What is Asterisk GUI architecture acturally? I could not get it... > hmm? check [EMAIL PROTECTED] - it contains AMP - http://asteriskathome.sf.net Tomek -- Startuj z INTERIA.PL! >>> http://link.interia.pl/f186c ___ 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 1.0.7 problems (again)
Remco Barende wrote: Then when I try to start asterisk I get this error: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' May 2 20:41:58 WARNING[8663]: loader.c:440 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] zaptel]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe I got *exactly* the same warnings with a wrongly configured HFC-based card. Fixing zaptel.conf and zapata.conf fixed this warning/asterisk not starting - so you might look there. (I'm using [EMAIL PROTECTED] / AMP on CentOS 3.4). I'm still unable to dial out, though, can only dial in. I wish there was a "card-setup wizard for asterisk"... Tomek -- Znajdz swoja milosc na wiosne... >>> http://link.interia.pl/f187a ___ 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 dialout problems
Eric Wieling aka ManxPower wrote:
Tomasz Chmielewski wrote:
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I
can call myself and talk to asterisk :)
Zap/0 is not a valid Zap channel. Zap channels start at 1.
sorry, it was meant to be g0.
I tried Zap/1 and Zap/2 earlier - with that effect:
-- Executing Dial("SIP/201-152d", "Zap/1/98") in new stack
-- Called 1/98
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
-- Executing Congestion("SIP/201-152d", "") in new stack
Tomek
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Re: [Asterisk-Users] zaphfc dialout problems
Deti Fliegl wrote:
Tomasz Chmielewski wrote:
exten => _0.,1,Dial(Zap/0/${EXTEN:1})
set g0 instead of 0:
exten => _0.,1,Dial(Zap/g0/${EXTEN:1})
Yes, it changed something, now I get an immediate hangup:
-- Executing Dial("SIP/201-e124", "Zap/g0/98") in new stack
-- Called g0/98
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
When I disconnect an ISDN cable, I don't get an immediate hangup, so it
was a push in a right direction... But I'm still not able to dial out.
Any more ideas?
Tomek
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[Asterisk-Users] zaphfc dialout problems
I just installed a HFC-based ISDN card, and I'm having problems with
making dialouts using that card. Dial-ins are working fine - i.e. I can
call myself and talk to asterisk :)
I have defined an extension:
exten => _0.,1,Dial(Zap/0/${EXTEN:1})
exten => _0.,2,Congestion
exten => _0.,3,Hangup
So when I dial 0500, I should be connected to number 500.
This is not the case: a telephone with number 500 never rings.
This is what asterisk says when run with -cvvv:
-- Executing Dial("SIP/201-f853", "Zap/0/500") in new stack
-- Called 0/500
-- Zap/pseudo-164837434 answered SIP/201-f853
It behaves the same even if I call an non-existing number, or I
disconnect an ISDN cable.
Dialing in works fine, so it's not a problem with a card.
Any clue?
If it's any help, I am using [EMAIL PROTECTED] 1.0.
# cat /etc/asterisk/zapata.conf
[channels]
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = dynamic
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
echocancel=yes
echotraining = 100
echocancelwhenbridged=yes
immediate=yes
group = 0
context=default
channel => 1-2
# cat /etc/zaptel.conf
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
# lspci
(...)
00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
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Re: [Asterisk-Users] Help to configure asterisk to dial to an PSTN line
Amit Singla wrote:
Hi Everyone,
I have bought a Digium TDM400P and am using asterisk on RedHat 9.0. I
was able to configure Asterisk and SJPhone, so I have been able to call
from IP to IP and also from IP to a analog phone which is attached to
the digium card.
My problem now is to dial from an IP phone to an PSTN line or any
telecom line and reverse. I don't know what changes or addition I have
to make in which files (sip.conf,extension.conf,zapata.conf etc). I have
tried to search for it but all in vain.
It depends on the hardware you have.
This is for an ISDN line.
This means that if you dial 01234567 on your sip phone (_0.), Asterisk
will dial Modem/group1 (which is configured in modem.conf), and dial
your extension (${EXTEN) 01234567, without the first number (:1})
exten => _0.,1,Dial(Modem/g1:${EXTEN:1})
; can be also Modem/ttyI0; will call through
; first available /dev/ttyI though
exten => _0.,2,Congestion
Dialing 6712 on your sip phone will call 12.
exten => 6712,1,Dial(Modem/ttyI0:12)
Try reading Asterisk Handbook Project, it should be explained there for
your configuration I think.
Tomek
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[Asterisk-Users] can asterisk send AT commands to a modem?
Can asterisk send AT commands to a modem? If so, how? I have two ISDN cards (with i4l - capi4linux doesn't work with them), and would like to specify which card to choose for dialing out (without it, i4l uses first free /dev/ttyI device). Tomek ___ 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] proper 2-card ISDN modem.conf configuration?
I'm trying to configure an asterisk box with two cards.
Incoming calls are working fine with two ISDN cards, however, I am able
to make outgoing calls only through the first card.
exten => _0.,1,Dial(Modem/g1:${EXTEN:1})
exten => _9.,1,Dial(Modem/g2:${EXTEN:1})
If I try to use the second card, asterisk says that the line is busy
(which isn't true).
So I thought that maybe my modem.conf is wrong?
Could you paste your modem.conf here, if you are using more than one
ISDN card?
Below my modem.conf:
[interfaces]
context=remote
driver=i4l
language=de
type=i4l
dialtype=tone
mode=immediate
dtmfmode=both
group=1
msn=27229933
incomingmsn=*
device => /dev/ttyI0
device => /dev/ttyI1
group=2
msn=624
incomingmsn=*
device => /dev/ttyI2
device => /dev/ttyI3
Tomek
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Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (notserver) workwithasterisk!
Gregory Wiktor - ADCom Corp. wrote:
Hello Tomek,
When I call my second msdn, I get the following:
== Starting Modem[i4l]/ttyI1 at incoming-isdn,2781984,1 failed so
falling back to exten 's'
-- Executing Answer("Modem[i4l]/ttyI1", "") in new stack
somersvoip*CLI> Apr 28 04:04:33 localhost kernel: isdn_net: call from
8005966511,1,0 -> 2781984
Apr 28 04:04:33 localhost kernel: isdn_net: call from 8005966511 -> 0
2781984 ignored
Apr 28 04:04:33 localhost kernel: isdn_tty: call from 8005966511, ->
RING on ttyI1
Apr 28 04:04:33 localhost kernel: isdn: HiSax,ch0 cause: E0260
Apr 28 04:04:43 WARNING[4476]: chan_modem_i4l.c:555 i4l_answer: Unable
to answer: NO CARRIER
== Spawn extension (incoming-isdn, s, 1) exited non-zero on
'Modem[i4l]/ttyI1'
-- Hungup 'Modem[i4l]/ttyI1'
I will try the defaults at some point this week, it's at the other
office. Hopefully I'll make out ok...
yeah try the defaults first.
I would look what "Unable to answer: NO CARRIER" means if the defaults
won't work.
Tomek
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Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) workwithasterisk!
Gregory Wiktor - ADCom Corp. wrote: Hello Tomek, Previously I did get asterisk to see the call, but not currently. This is in the usa, so my msn is a 7 digit number. The kernel is saying the following: Apr 28 03:46:17 localhost kernel: isdn_net: call from 8005966511,1,0 -> 2781980 so this is your MSN: 2781980 try loading the default asterisk config files, and you should be able to use your card. or try [EMAIL PROTECTED] - asteriskathome.sf.net - it's asterisk made easy (well, sort of) - if you decide to use it, let me know, because Eicon cards won't work with it right after installation (you have to "yum install kernel-unsupported" etc., then load hisax module etc.) Tomek ___ 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] Eicon DIVA PCI ISDN cards (not server) work withasterisk!
Gregory Wiktor - ADCom Corp. wrote: Hello Tomek, I also got a diva pci 2.02 card, but although the kernel sees the incoming calls, asterisk refuses to answer. Did you have this issue at all? The kernel seems to be denying the call... if you see the calling in the systlog, that's 98% of success :) you have to set up in modem.conf something like: driver=i4l ; your msn - without it (or if it's wrong) it won't work msn=4235 device => /dev/ttyI0 device => /dev/ttyI1 restart asterisk, and it should pick up the phone now (or, you don't have it configured in asterisk, but the default configuration should pick up the phone and play a demo). check asterisk logs if it sees an incoming call. Tomek ___ 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] Asterisk doesn't disconnect when I hang up SIP (SIP -> PSTN call)
Umair Bari wrote:
try putting
exten => _0.,4,Hangup
like
[ext-local-custom]
exten => _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten => _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten => _0.,3,Congestion
exten => _0.,4,Hangup
no, still does not hang up :(
I have to pick up the phone and hang up manually (or kill asterisk).
Tomek
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[Asterisk-Users] Asterisk doesn't disconnect when I hang up SIP (SIP -> PSTN call)
I'm trying to learn Asterisk.
So far I'm using kphone and an ISDN line (with Eicon DIVA 2.01 PCI card).
I have created that extension following The Asterisk Handbook (page 36):
[ext-local-custom]
exten => _0.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten => _0.,2,Dial(Modem/ttyI1:${EXTEN:1})
exten => _0.,3,Congestion
So whenever I call 055 from kphone, Asterisk connects me to an internal
55 number, and I can talk to myself (wohoo!) when I pick up the phone.
However, when I call 055 from kphone, and *don't* pick up the phone on
the other side, and then disconnect kphone (or even quit it), asterisk
keeps ringing 55.
I'd like to add, that Asterisk detects kphone disconnecting when the
phone is already established.
Any clue?
Tomek
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[Asterisk-Users] do I configure ISDN in zapata.conf?
I'm new to asterisk and still learning it. I wanted to ease my efforts a bit and use AMP (Asterisk Management Portal), and see what changed in the config files when I use it. However, I realized that I can only add SIP, IAX2 and ZAP extensions - I didn't see an option to configure an ISDN extension etc. So my conclusion was, that ZAP (zapata.conf) allows configuring ISDN extensions / numbers, too? Or am I totally wrong? If someone could make sme clarification about this, I'd be glad. Searching this list, wiki and google didn't bring me a definite answer. I have an Eicon DIVA 2.01 PCI ISDN card. Tomek ___ 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] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
I just wanted to let you know that it's possible to use Eicon DIVA PCI 2.01 ISDN cards (not "server" divas) with asterisk. First thing to do is to load the module. If you have two of these cards, you should do it like that: modprobe -v hisax protocol=2,2 type=11,11 And now you can have up to 4 incoming calls with two cards (try calling yourself and see if anything gets into your syslog - you should have "ignored" calls even if asterisk isn't running). Then configure your asterisk to use i4l (don't use chan_capi) - do it in modem.conf: (...) driver=i4l (...) msn=your_msn_number and that's it (you still need to configure your ISDN devices to allow incoming calls, for example, using conf-isdn-account - don't forget to set SECURE="off" etc. ISDN settings). Tomek ___ 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 kernel 2.6 problems chapi isdn4lin
Corvin wrote: Hello! I've encauntered some serious problems with asterisk. I have to install it on system: 1. Mandrake 10.1 2. kernel 2.8.1 3. four ISDN cards. And I am in big trouble, isdn4linux is no longer supported for kernels 2.6 (on this system there are not any /dev/ttyI0 and similar devices)/ msidn - is unstable and for brave people chapi - I can't compile (lot of errors and I don't know why) i tired to patch it but it didn't help :(. I don't know what to do and I need solution very fast. I have exactly the same problem. Tried compiling chan_capi on Mandrake 10.1 and SuSE 9.1, but it failed with lots of weird errors... I posted it to the group a couple of days ago, got no reply :( Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] software phones for Asterisk - is there a list?
Roger Hanson wrote: - Original Message - From: "Tomasz Chmielewski" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Wednesday, December 01, 2004 4:42 AM Subject: [Asterisk-Users] software phones for Asterisk - is there a list? Hello, Is there a list of software phones which will work with Asterisk? For Linux and Windows? I don't have any hardware yet, and before I buy anything I would like to know how Asterisk really works (with software "phones" for example). I'm sure you didn't search the wiki, did you? There's tons of information there on soft phones. http://voip-info.org/wiki-VOIP+Phones nopez, didn't really know there is one :) OK, so I found kphone, installed on two linuxes, configured the way the wiki says, registered with asterisk, but they won't connect to each other... will start a new post if I don't find anything useful in the wiki and lists.digium.com... :) Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to open IAX timing interface: No such file or directory
Dave Cotton wrote: On Wed, 2004-12-01 at 13:37 +0100, Dave Cotton wrote: On Wed, 2004-12-01 at 13:01 +0100, Tomasz Chmielewski wrote: So the only issue left I have is with this skinny not found when 0.0.0.0 is set in skinny.conf in modules.conf noload=chan_skinny.so Oops noload => chan_skinny.so what's this skinny anyway? Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to open IAX timing interface: No such file or directory
Peter Svensson wrote: On Wed, 1 Dec 2004, Dave Cotton wrote: On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote: What I found on voip-info.org was that I didn't have a working timer - and I had to load ztdummy module. So I did (modprobe ztdummy), started asterisk again, but I'm still getting the same error. Had you actually compiled zaptel? Had you un-commented ztdummy? Had you done lsmod to see if zaptel and ztdummy where loaded? Have you checked the permissions on /dev/zap/* ? Are the device nodes created properly? OK, this was the issue. As I had no /dev/zap directory, I guess the nodes were not created during "make install" of zaptel. Isn't it zaptel issue that should be corrected? So I had to mkdir /dev/zap and then: mknod /dev/zap/ctl c 196 0 mknod /dev/zap/channel c 196 254 mknod /dev/zap/pseudo c 196 255 corrected the situation. Thanks for the hint! :) So the only issue left I have is with this skinny not found when 0.0.0.0 is set in skinny.conf Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to open IAX timing interface: No such file or directory
Dave Cotton wrote: On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote: What I found on voip-info.org was that I didn't have a working timer - and I had to load ztdummy module. So I did (modprobe ztdummy), started asterisk again, but I'm still getting the same error. Had you actually compiled zaptel? yes. I compiled asterisk like that: # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login- the password is anoncvs. # cvs checkout zaptel libpri asterisk # cd zaptel # make clean ; make install # cd ../libpri # make clean ; make install # cd ../asterisk # make clean ; make install # make samples Had you un-commented ztdummy? What do you mean by uncommenting ztdummy? # pwd /etc/asterisk # grep -r dummy ./* # It's doeasn't exist in any file with asterisk... Or rather I had to uncomment it in Asterisk's Makefile (/me goes check) so I have to compile Asterisk once again now? Had you done lsmod to see if zaptel and ztdummy where loaded? yes, they are. # lsmod|grep zaptel zaptel183076 1 ztdummy crc-ccitt 1664 1 zaptel # lsmod|grep ztdummy ztdummy 2372 0 zaptel183076 1 ztdummy Any clue? Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to open IAX timing interface: No such file or directory
Patrick wrote: On Wed, 2004-12-01 at 11:26 +0100, Tomasz Chmielewski wrote: [snip] chan_iax.c:7507 load_module: Unable to open IAX timing interface: No such file or directory What does it mean? Is it something to worry about? How to get rid of it? For these and many other basic questions first search google: http://www.google.com/search?q=site%3Alists.digium.com+Unable+to+open +IAX+timing then search voip-info.org and maybe then ask on the mailing list. yes I did. What I found on voip-info.org was that I didn't have a working timer - and I had to load ztdummy module. So I did (modprobe ztdummy), started asterisk again, but I'm still getting the same error. On lists.digium.com most of posts with the same warnings don't have any answers (on two google pages when searching for "Unable to open IAX timing interface" site:lists.digium.com). Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] software phones for Asterisk - is there a list?
Hello, Is there a list of software phones which will work with Asterisk? For Linux and Windows? I don't have any hardware yet, and before I buy anything I would like to know how Asterisk really works (with software "phones" for example). Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to open pseudo channel for timing... Sound may be choppy
Hello, I just sent it with a wrong title... so once again: I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc and I got this warning (this was also before I changed from oss to alsa): res_musiconhold.c:564 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. What does it mean? Is it something to worry about? How to get rid of it? Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to get our IP address, Skinny disabled
Hello, Yet another warning I have. I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc and I got this warning (this was also before I changed from oss to alsa): chan_skinny.c:2602 reload_config: Unable to get our IP address, Skinny disabled What does it mean? Is it something to worry about? I temporarily got rid of it by putting my IP address (192.168.0.234) instead of 0.0.0.0 into skinny.conf file, but I don't think this is a good solution (what if I have more interfaces). Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to open IAX timing interface: No such file or directory
Hello, Another warning I have. I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc and I got this warning (this was also before I changed from oss to alsa): chan_iax.c:7507 load_module: Unable to open IAX timing interface: No such file or directory What does it mean? Is it something to worry about? How to get rid of it? Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to open IAX timing interface: No such file or directory
Hello, I just compiled and started Asterisk 1.0.2 following "Getting Started With Asterisk Version 0.1a" from http://www.automated.it/guidetoasterisk.htm I made only one change to default config files - I changed from using oss to alsa. I don't have any devices so far. I started asterisk from the command line: # asterisk -vc and I got this warning (this was also before I changed from oss to alsa): res_musiconhold.c:564 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. What does it mean? Is it something to worry about? How to get rid of it? Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi on 2.6 - impossible?
Derek Conniffe wrote: Chan_capi works fine on a 2.6 kernel for me (2.6.8 and SuSE 9.1) but I'm using a AVM Fritz PCI V2 BRI card and I firstly installed the AVM fritz PCI capi driver and then I installed chan_capi - everything went very smoothly and I've installed with different AVM Fritz cards a few times now. Yeah I figured out I can't compile it on 2.4 as well... How did you compile chan_capi on SuSE 9.1 though? I tried on that distro but I couldn't (see my "chan_capi compilation problems" post). Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi compilation problems
Hello, I can't compile chan_capi-0.3.5 (also tried with 0.3.4b). I tried compiling it on two systems with very similar (unsuccessful) results: 1) SuSE 9.1 on 2.6.5 kernel, 2) Mandrake 10.1 with kernels 2.6.8.1 and 2.4.27, using gcc 3.3.4 and gcc 3.4.1 I'm getting the following errors (similar on SuSE and Mandrake): # make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c In file included from /usr/include/time.h:38, from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/lib/gcc/i586-mandrake-linux-gnu/3.4.1/include/stddef.h:213: error: syntax error before "typedef" In file included from /usr/include/pthread.h:21, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/time.h:60: error: syntax error before "typedef" /usr/include/time.h:74: error: syntax error before "__BEGIN_NAMESPACE_STD" /usr/include/time.h:76: error: syntax error before "typedef" /usr/include/time.h:129: error: syntax error before "__BEGIN_NAMESPACE_STD" /usr/include/time.h:131: error: syntax error before "struct" /usr/include/time.h:178: error: syntax error before "__BEGIN_NAMESPACE_STD" /usr/include/time.h:181: error: syntax error before "extern" /usr/include/time.h:181: error: syntax error before "__THROW" /usr/include/time.h:184: error: syntax error before "__THROW" /usr/include/time.h:188: error: syntax error before "__THROW" /usr/include/time.h:191: error: syntax error before "__THROW" /usr/include/time.h:199: error: syntax error before "__THROW" /usr/include/time.h:226: error: syntax error before "__BEGIN_NAMESPACE_STD" /usr/include/time.h:229: error: syntax error before "extern" /usr/include/time.h:229: error: syntax error before "__THROW" /usr/include/time.h:233: error: syntax error before "__THROW" /usr/include/time.h:248: error: syntax error before "__BEGIN_NAMESPACE_STD" /usr/include/time.h:251: error: syntax error before "extern" /usr/include/time.h:251: error: syntax error before "__THROW" /usr/include/time.h:254: error: syntax error before "__THROW" /usr/include/time.h:272: error: syntax error before "extern" In file included from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/signal.h:31: error: syntax error before "__BEGIN_DECLS" In file included from /usr/include/signal.h:33, from /usr/include/pthread.h:24, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sigset.h:23: error: syntax error before "typedef" In file included from /usr/include/bits/pthreadtypes.h:23, from /usr/include/pthread.h:25, from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/bits/sched.h:83: error: syntax error before "struct" In file included from /usr/include/asterisk/lock.h:17, from chan_capi.c:14: /usr/include/pthread.h:59: error: syntax error before "enum" /usr/include/pthread.h:166: error: syntax error before "__THROW" /usr/include/pthread.h:169: error: syntax error before "__THROW" /usr/include/pthread.h:172: error: syntax error before "__THROW" /usr/include/pthread.h:186: error: syntax error before "__THROW" /usr/include/pthread.h:194: error: syntax error before "__THROW" /usr/include/pthread.h:197: error: syntax error before "__THROW" /usr/include/pthread.h:201: error: syntax error before "__THROW" /usr/include/pthread.h:205: error: syntax error before "__THROW" /usr/include/pthread.h:210: error: syntax error before "__THROW" /usr/include/pthread.h:216: error: syntax error before "__THROW" /usr/include/pthread.h:220: error: syntax error before "__THROW" /usr/include/pthread.h:225: error: syntax error before "__THROW" /usr/include/pthread.h:229: error: syntax error before "__THROW" /usr/include/pthread.h:234: error: syntax error before "__THROW" /usr/include/pthread.h:238: error: syntax error before "__THROW" /usr/include/pthread.h:242: error: syntax error before "__THROW" /usr/include/pthread.h:260: error: syntax error before "__THROW" /usr/include/pthread.h:265: error: syntax error before "__THROW" /usr/include/pthread.h:284: error: syntax error before "__THROW" /usr/include/pthread.h:289: error: syntax error before "__THROW" /usr/include/pthread.h:304: error: syntax error before "__THROW" /usr/include/pthread.h:310: error: syntax error before "__THROW" /usr/include/pthread.h:334: error: syntax error before "__THROW" /usr/include/pthread.h:337: error: syntax error before "__THROW" /usr/include/pthread.h:340: error: syntax error before "__THROW" /usr/include/pthread.h:343: error: syntax error before "__THROW" /usr/include/pthread.h
[Asterisk-Users] chan_capi on 2.6 - impossible?
Hello, I'm trying to get my Eicon Diva 2.01 PCI ISDN card working with Asterisk and was told that I still have some chance if I tried using chan_capi. I tried compiling it (chan_capi) on two systems running 2.6 kernels, but got lots of errors. Before I go investigating - is it possible to compile chan_capi on 2.6 kernels? Or does it work with 2.4 ones only? Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Patrick wrote: On Mon, 2004-11-29 at 13:51 +0100, Tomasz Chmielewski wrote: [snip] It appears they have been wrong. I just checked on www.capi.org, and Eicon Diva 2.0 PCI cards offers CAPI 2.0... This same page says this card is is not Linux compatibile, though, but it is. Capi.org is not the same as chan_capi or Asterisk. I have never heard of a cheap (non "Server" or active) Eicon Diva card working with chan_capi & Asterisk. If you want an ISDN based chan_capi/Asterisk solution either buy an Eicon Diva Server, any of the active AVM cards (B1 or C4 iirc) or the cheapest solution: an AVM Fritz! card. And I was so happy today because I thought I won't have to buy anything :) So because I have these cheap Eicon Diva cards - does this mean they won't work at all? Or rather that some features will be missing only? I need only incoming and outgoing calls through these cards. Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Jean-Michel Hiver wrote: Tomasz Chmielewski wrote: Hello, I'm thinking of deploying Asterisk. I already have a handful of EICON Diva 2.01 PCI ISDN cards. I was thinking if it's possible to insert 4 such cards to my PC-Asterisk server (which I yet have to install) and use them as 4 lines in case anyone has to call me in / I have to call out using ISDN line(s)? From what I have been told on this very list you can only use Diva Server cards with asterisk because the 'cheaper' diva cards do not support some stuff called 'capi'. It appears they have been wrong. I just checked on www.capi.org, and Eicon Diva 2.0 PCI cards offers CAPI 2.0... This same page says this card is is not Linux compatibile, though, but it is. Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Jean-Michel Hiver wrote: Tomasz Chmielewski wrote: Hello, I'm thinking of deploying Asterisk. I already have a handful of EICON Diva 2.01 PCI ISDN cards. I was thinking if it's possible to insert 4 such cards to my PC-Asterisk server (which I yet have to install) and use them as 4 lines in case anyone has to call me in / I have to call out using ISDN line(s)? From what I have been told on this very list you can only use Diva Server cards with asterisk because the 'cheaper' diva cards do not support some stuff called 'capi'. too bad. I have dozens of these EICON Diva cards, I thought I could use them and not buy any additional hardware :( Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Hello, I'm thinking of deploying Asterisk. I already have a handful of EICON Diva 2.01 PCI ISDN cards. I was thinking if it's possible to insert 4 such cards to my PC-Asterisk server (which I yet have to install) and use them as 4 lines in case anyone has to call me in / I have to call out using ISDN line(s)? Any reply appreciated. Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
