Re: [Asterisk-Users] Problem with Vegastream 50 BRI
On Sat, 2004-03-20 at 16:36, Michael Devenijn wrote: Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... extensions.conf extract (from the contact [tlsgw]) : exten = 57228047,Dial(SIP/cs001,40,tr) The above line does not look like a valid extension line, the priority is missing. That _could_ prevent the context tlsgw from being loaded, which in turn might cause your installation to fallback to the default context. You may want to inspect the output of a 'show dialplan' to see if your tlswg context is loaded or not. Apart from that, you may want to increase logging in /etc/asterisk/logger.conf for a default installation: debug = debug console = notice,warning,error,debug messages = notice,warning,error This will cause debug messages to be show on the console where you are running asterisk, and to log them to a file /var/log/asterisk/debug, while notice, warning and errors will be logged to /var/log/asterisk/messages. These files will be your friends in debugging. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Vleutenseweg 86 [EMAIL PROTECTED] 3532 HM Utrecht tel: +31 (0)30 299 2109The Netherlands fax: +31 (0)30 299 2108 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] KPN BRI
On Wed, 2004-03-03 at 16:12, Mark wrote: The software configuration depends (of course) on your hardware I have 2 Eicon Diva cards which I am using chan_capi. I have chan_capi installed and configured and it detects the ports ok. I have the lines plugged in but when I dial the number associated with the line does not get picked up and I get a non-existant number tone. When I plug in a standard isdn telephone into the line it all works ok. This is where you would tell us what is in your /etc/asterisk/capi.conf, and what is in your extensions.conf hint: --capi.conf ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] amaflags = billing group = 1 msn=101234567 incomingmsn=101234567 controller=1 softdtmf=0 context=inbound-pstn ;prefix=0 ;echocancel=yes ;echotail=64 devices=2 msn=107654321 incomingmsn=107654321 controller=1 softdtmf=0 context=inbound-pstn devices=2 - I think I am using the wrong kind of signalling. I have found out that kpn use e164 as the signalling but I cannot find anywhere to configure this. Well, I'm using capi on KPN lines since ages, and have never knowingly configured any kind of signalling, so I doubt that that's the right path to search. Thanks for any advice you can give. Regards Mark ___ 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 -- Envida http://www.envida.net/ Armand A. Verstappen Vleutenseweg 86 [EMAIL PROTECTED] 3532 HM Utrecht tel: +31 (0)30 299 2109The Netherlands fax: +31 (0)30 299 2108 ___ 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] KPN BRI
Hi Mark, On Wed, 2004-03-03 at 15:08, Mark wrote: I want to configure my * box for my idsn 2 line which I ordered from KPN (Netherlands). Should be no problem. Does anyone have any configuration for this that can help me? Need more input. The software configuration depends (of course) on your hardware configuration. In this situation, what kind of hardware do you plan to use to connect your asstricks box to KPN? opinionYou are best of going with a capi capable isdn card, and using chan_capi (see http://www.junghanns.net/asterisk/page1.html)/opionion, your alternative would be to use chan_modem together with isdn4linux. I've done both, so my opinion / is based on personal experience. -- Envida http://www.envida.net/ Armand A. Verstappen Vleutenseweg 86 [EMAIL PROTECTED] 3532 HM Utrecht tel: +31 (0)30 299 2109The Netherlands fax: +31 (0)30 299 2108 ___ 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] Zaptel BRI and HFC-S cards in NT-Mode
On Fri, 2004-02-20 at 09:17, Klaus-Peter Junghanns wrote: to clear things up again, the problem is a wrong syntax for the Dial appplication exten = 74341423,1,Dial(Zap/g2/74341423,r) This will use r as the timeout value, so it will hang up immediately, actually too quick for the isdn phone to bring up a p2p layer 2 connection (it is on my todo list to handle this too). exten = 74341423,1,Dial(Zap/g2/74341423,,r) This is what you want (not the second ,). All true, but: Interesting fact is, that the ISDN-Phone on the NT line rings still, if the calling phone has dropped the call.. The same thing here. The ISDN-Phone continuing to ring forever after the calling party has dropped the call is not related to the configuration error that was introduced in this thread. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Vleutenseweg 86 [EMAIL PROTECTED] 3532 HM Utrecht tel: +31 (0)30 299 2109The Netherlands fax: +31 (0)30 299 2108 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Zaptel BRI and HFC-S cards in NT-Mode
Hi Ernst, On Thu, 2004-02-19 at 15:26, Ernst Lehmann wrote: use this: exten = 74341423,1,Dial(Zap/g2/74341423,r) snip Interesting fact is, that the ISDN-Phone on the NT line rings still, if the calling phone has dropped the call.. The same thing here. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Vleutenseweg 86 [EMAIL PROTECTED] 3532 HM Utrecht tel: +31 (0)30 299 2109The Netherlands fax: +31 (0)30 299 2108 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] suggested hardware especially sound cards
Hi, On Fri, 2003-10-03 at 15:52, mattf wrote: I've seen various suggestions thrown around for hardware when people ask, but can we all agree on some basic hardware recommendations for a few basic setups(and post them on a website) to make it easier for new people to avoid some of the hardware/software pitfalls when they are setting up their first systems. snip I think we should have these setups listed: - home user with 1-2 telco lines and 2-5 phones - small office with 4-8 telco lines and 8-16 phones - small office with a fractional E1/T1 and 12-24 phones - medium office with full E1/T1 and 24-48 phones - medium office with 2-4 E1/T1s and 48-100 phones - large office with 4-16 E1/T1s and 100-500 phones - multi-location corporate offices with 16-64 E1/T1s distributed and 500-2500 phones - ACD heavy office suggestions - IVR or Conference heavy suggestions You can add a section this to the wiki (http://www.voip-info.org/wiki-Asterisk), and fill out the suggestions you have information for, then invite others to complete the others. All you need to do is register, which is free. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
On Sat, 2003-10-04 at 18:53, Rich Adamson wrote: Why not add an Article to the www.voip-info.org site, and those that are interested with helping can list their FWD, IAXTEL, or other access number, probable hours of availability, any special focus skills, size of their current * environment, etc? I'm game. Sounds good. Wouldn't it be possible to login to the queue of an * server providing this servers from my extension through my * installation? That way calls could be routed to available volunteers. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] Help with GPL license of Asterisk
Hi, On Mon, 2003-09-29 at 16:40, Mark Spencer wrote: 1) if your application is not released to a 3rd party, you do not have to make the source available This is TRUE. 2) if you build your application as a module that loads into a stock asterisk server, you do not have to disclose your source This is FALSE. Even modules for Asterisk MUST be released under GPL, unless you obtain a license to release them outside of GPL from Digium. Maybe this should be re-thought? Allowing third parties to release modules under a non-GPL license (through a 'Mark exception' analogue to the 'Linus exception referenced below) could be intresting. A third party that really wants to release under a non-GPL license can do so by creating their application as an AGI script, or have it work using the management interface. Heck, they could release a wrapper to 'exec()' as GPL, and then use that application to call their non-gpl'ed code anyway, right? So, if 3rd parties are doing or going to do that, then why not allow them to do it in a way that doesn't require bypassing proper design? A third party could then for example start selling G.723 codecs, if they are prepared to pay the fee that allows them to do so. 3) if you need to make changes to the core in order for your application to work, you'll need to disclose source for your changes to the core, but not for your application. This sounds horrid, but it's not too bad, as your simply augmenting the core API and keeping your goodies in the binary only portion of the release. This is also FALSE. You MUST release both the module AND core changes unless you obtain license from Digium. I believe you are confusing the Linus exception which is an exception for the Linux kernel explicitly made by Linus Torvalds, allowing binary only modules to the kernel only. My suggestion above is based on my own egoistic view as a user of the software. I have no intention to create non-GPLed modules myself, but wouldn't mind to pay for some kind of third party module that does something for me thats not available in GPLed code. I prefer GPL, other forms of open source (payed for or not) is acceptable. I dislike closed source, but if it solves my problem against an acceptable rate with acceptable service and support, why not. With a 'mark exception', I'd be able to run GLP-ed asterisk with a channel driver from a third party. Win for me. Without the 'mark exception', I'll have to purchase a non-GPLed version of Asterisk, as well as the third parties' module. I'm not clear if that will lock me into paying upgrade fees to Digium, or if a non-GPL license will still allows me to follow CVS as I do now. I'll have the same question regarding the third party's module in the other case of course. I'm not sure how a 'mark/digium exception' would work out for the the Asterisk community. A third party would no longer be required to pay a fee for a non-GPLed Asterisk, and Digium would loose some revenue. Since Digium still is the primary sponsor of Asterisk development, this is a loss for the community. On the other hand, it is possible that under the suggested construction many more third party modules spring to live, causing Asterisk to be more usable for businesses, in turn generating more revenue for Digium. And, since third parties would benefit from a more stable Asterisk, there may be more parties be actively involved in maintaining and extending the core. I have no idea which way the balance would swing. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] Google newsgroup or Forum setup.
On Mon, 2003-09-29 at 19:16, Keith O'Brien wrote: I'll offer one better. Why don't we mirror all of the maillist posts to a forum. That way both parties are happy. Those that want a forum can use a forum interface and still post to the maillist and those that like the maillist can stay as is. It is a free world, so I won't opt against it. But if the forum posts would contain HTML formatted mail, I'd be very, very, upset (just as upset as I'm now when I get HTML formatted mail). wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Distinctive ringing
Hi Rich, On Fri, 2003-09-19 at 13:55, Rich Adamson wrote: ... I opened a problem with Digium late last week. One of their techs logged into this system, tested with real calls, and observed the problem. They made a source code change in chan_zap.c (and possibly others) and now callerid works fine with that distinctive ring. Since I don't have another copy of the cvs that was in use at the time, I don't know what they changed. I've asked multiple times, but never get a response from the support folks. Therefore, I'm not sure if they fixed a real bug or if they brute-forced this system to look for callerid elsewhere. (And, now I don't know what's going to happen if I apply a current cvs update either.) You can checkout an older cvs version using the -D option to the cvs checkout command, excerpt from 'man cvs': -D date_spec Use the most recent revision no later than date_spec (a single argu- ment, date description specifying a date in the past). A wide vari- ety of date formats are supported, in particular ISO (1972-09-24 20:05) or Internet (24 Sep 1972 20:05). The date_spec is inter- preted as being in the local timezone, unless a specific timezone is specified. The specification is ``sticky'' when you use it to make a private copy of a source file; that is, when you get a working file using -D, cvs records the date you specified, so that further updates in the same directory will use the same date (unless you explicitly override it; see the description of the update command). -D is available with the checkout, diff, history, export, rdiff, rtag, and update commands. Examples of valid date specifications include: 1 month ago 2 hours ago 40 seconds ago last year last Monday yesterday a fortnight ago 3/31/92 10:00:07 PST January 23, 1987 10:05pm 22:00 GMT Also, there's a mailinglist '[EMAIL PROTECTED]', that mails the diffs for every change made to cvs. You could browse through it's archive to see if you can find the relevant changes at http://lists.digium.com/pipermail/asterisk-cvs/ wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Aastra 390 w/ADSI - Doesn't automagically use Asterisk PBX script
Hi, On Fri, 2003-09-19 at 18:11, Eric Wieling wrote: I have an Aastra 390 ADSI phone. It's not locked. I can call ADSIProg without a problem and it programs my phone. Calling Voicemail2 also programs my phone. However, in order for the VMail option to appear on the screen I have to go into the Services menu, pick Asterisk PBX and pick Select. In the services menu, you'll see a few 'slots' into which adsi scripts can be loaded. One of these, typically the last one is the 'self-load' slot, suffixed with 'SL'. You will need to set the correct 'descriptor number' in the asterisk.adsi file: FDN 0x000f ; Descriptor number to get the asterisk script to load into the self-load slot. There are two ways to find the correct fdn: - get your supplier to give it to you - brute force, just try one by one untill you find it. The first option was bluntly refused by Aastra support staff in my case, and I haven't had the gusto to try the second option. If you manage to find the correct FDN for your phone, I'd be interested to hear about it. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] GSM player or plugin for XMMS
On Fri, 2003-09-19 at 19:52, Marcel Prisi wrote: One more : http://www.zipworld.com.au/~erikd/XMMS/ This one uses libsndfile : http://www.zip.com.au/~erikd/libsndfile/ which can play even more formats including gsm6.10, G721 G723 (quite impressive) The libsndfile library is LGPL'ed, yet from the page I seem to understand that it can encode and decode G.723. I may misinterpret this information, but if I don't, I wonder how that could be legal. And, if it is legal, if this library could be used to add G.723 support to asterisk. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Port problem
Hi, On Tue, 2003-09-23 at 19:43, Paulo Mannheimer wrote: I have an equipment loaded with 4 X100P (numbered 1-4)) and one T400P (numbered 5-8). Everything works fine except that I cannot use one of the FXS ports (number 5). If you can use the other ports, it may be that this port #5 fails to calibrate, or is toast. You need to look at dmesg output to see if such is the case. If it is a calibrating issue, check the archives for the compile options to work around it. If I configure zapata.conf to recognize it, the whole system voice quality suffers. I've tried already to switch PCI slots, with no results. Below is a snapshot of my /proc/interrupts, maybe this can shed some light on the problem. 0: 985385 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 3:9832048 XT-PIC wcfxo 4: 318730 XT-PIC serial 5: 0 XT-PIC usb-uhci, usb-uhci, usb-uhci 7:9832105 XT-PIC wcfxo 8: 1 XT-PIC rtc 9: 162893 XT-PIC eth0 10:9818599 XT-PIC wcfxs 11: 20891396 XT-PIC wcfxo, wcfxo 12: 36 XT-PIC PS/2 Mouse 14: 26399 XT-PIC ide0 NMI: 0 LOC: 0 ERR: 0 MIS: 0 Any ideas? The archives will also tell you that it is a bad thing to have the digium cards share interrupts with anything, and may cause trouble like line noise. If you remove serial support and/or usb support from your kernel, you should be able to free up a dedicate IRQ for the X100P's now sharing IRQ 11. Note that many motherboards have pci-slot 1 and 5 share irq allways. If this is the case with your equipment, I have no solution for you to get dedicated IRQ's on all digium boards. Note that the IRQ problem MAY be the cause of your problem, I'd first investigate the problem of the one port on the TDM40P, as I don't recall ever seeing a message indicating that IRQ overlap would cause only one of 4 port of a TDM40P to fail. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] ADSI Vista/Aastra 350
Hi, On Wed, 2003-09-10 at 15:50, Matthew M. Gamble wrote: I have ADSI working on my Aastra (Vista/Nortel) 350 phone and everything is working fine. However, I want the asterisk.adsi to load into the 'self-load' slot but can't figure out what the correct FDN for doing this is. Does anyone know the right FDN for the SL slot on these phones? I have hammered Aastra support for three weeks, and finally got the answer that it is impossible to load something in the self-load slot, unless I would buy a custom model. This is utter nonsens, as dialing the webconfig number will happily load a script into the self-load slot. I have yet to receive reply to that remark, and I'm confident by now that I never will. So, the only solution I see is trying different fdn's untill you hit the jackpot. I haven't found the motivation to do that. If you do, and do find the right FDN, please let us know. As I'm disgusted by Aastra's approach to this issue, I'm looking for other ADSI phones that will allow me to load the self-load slot. Suggestions, anyone? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] I need your help
Hi, On Thu, 2003-09-11 at 09:49, Steve Meyers wrote: P.S. Anyone want to take bets on how long it will take for Steven Critchfield to berate this guy for improper email usage? :) Please don't make it look as if Steven is being foolish. I fully agree with him on the improper mail usage. I just costs me less time to break something expensive than to reply and try to educate the culprit. I think top-posting and html mail are a clear sign that the sender thinks his own time and comfort more important than those of the people they're soliciting help from. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] ISDN
On Thu, 2003-09-04 at 01:35, Jay Tyndall wrote:
Stripmsd is commented out, problem still occurs.
Does this simply use ATDT to dialout ?
When I attempt to dialout using minicom it comes back with NO MSN/EAZ
Looks like I may need to issue another AT Command to the netjet to set the
MSN...
Has anyone encountered this before?
Yes, I did a long time ago. I don't use chan_modem anymore, I moved up
to chan_capi.
I _think_ I fixed it by putting:
group=1 ; group=1,2,3,9-12
msn=0
incomingmsn=123456789,123456780
device = /dev/ttyI0
device = /dev/ttyI1
in modem.conf, and using:
dial(Modem/g1/${EXTEN})
This way, chan_modem would figure out by itself which channel is
available, and use that.
good luck,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
signature.asc
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Re: [Asterisk-Users] telantek.adsi
On Wed, 2003-09-03 at 22:03, jerk face wrote:
I am working with the telantek.adsi file, and I was
wondering how I would create a softkey for Transfer.
I tried making a key definition and using SENDDTMF
#, but that didn't work. Is there another way I
could do this?
SENDDTMF #
has worked for me once I had enabled the appropriate transfer flag in
the dial statement ('t'). Another approach using flash worked as well:
KEY switch IS Switch OR Switch
FLASH
ENDKEY
Also, does anybody have any ADSI scripts for use with
Asterisk that they would like to share?
I have nothing much, just some trial and error stuff based on the adsi
scripts that come with the adsi source. Mail me if you want to take a
peek.
Something odd I noticed:
SHOWKEYS cwdisable UNLESS nocallwaiting
Does not work within a softkey definition nor do any flag operations. As
I have no access to the adsi specifications I can not tell if this is a
peculiarity of those specs, or a bug in asterisk's implementation of
adsi (adsiprog.c)
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
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Re: [Asterisk-Users] ADSI Programs
Hi Jerk, On Wed, 2003-08-27 at 18:31, jerk face wrote: I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? There is no adsi script, you need to set adsi = yes in /etc/asterisk/zapata.conf for the channel you want to enable adsi. app_voicemail will then program your phone automagically when you call for voicemail. Thank you for your time. Question: where did you get your unlocked phone, what type is it, and what did it cost? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Problem starting Asterisk after abnormalshutdown
On Tue, 2003-08-26 at 17:11, Lee Goodman wrote: While Linux comes up fine, Asterisk won't start because the drivers are loading in the wrong order. fixed by: 1) sh /usr/src/fix-asterisk-modules.sh 2) sh /etc/init.d/asterisk start Is this a known problem? Is there an existing bug on this or should I open one up? Anyone else seen this problem? It is nothing specific to asterisk. Depending on your distribution there are ways to manipulate the order in which modules load. If you can't find that info, why don't you just call /usr/src/fix-asterisk-modules.sh from the start section of /etc/init.d/asterisk. Mind you, you will be facing this same problem sooner or later with another set of modules, so it may well pay off to find out how to manipulate module load order in your distribution. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] sample configs / load module failure
Hi Ted, On Wed, 2003-08-27 at 16:27, [EMAIL PROTECTED] wrote: detailed docs on the config files. The distribution I compiled and installed doesn't have any config files, and the handbook is good but doesn't cover all of the configs. after 'make install' there is a message suggesting you should do 'make samples'. It will populate your /etc/asterisk directory with configuration examples. [res_parking.so]WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/lib/asterisk/modules/res_parking.so: undefined symbol: ast_moh_start WARNING[1024]: File loader.c, Line 368 (load_modules): Loading module res_parking.so failed! barf. fishy. What distribution you meant back up? If it's not CVS, try CVS. If this is merely a matter of not using the parking module, that's fine, but I can't find the docs on how to NOT use a specific module. in /etc/asterisk/modules.conf in the [modules] section: noload = res_parking.so noload = app_somethingelse.so wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Question About BRI Cards
On Wed, 2003-08-27 at 18:42, Gustavo Villaran wrote: Hi, im new in the list and i want to buy a BRI card that works with Asterisk PBX software for testing purpose, but i dont know which one works with that software. If someone knowns something that can help me, please write to me. http://www.junghanns.net/asterisk/page15.html wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Why doesnt anyone reply me ?
Hi, On Mon, 2003-08-25 at 13:33, kaku ustaad wrote: How can record a conversation with asterisk ? I tried to use Record() but dint work for me .. here is what i tried . http://www.loligo.com/asterisk/ has good example configurations. You'll find a working example for recording messages there. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Newbie Question / ISDN
On Thu, 2003-08-21 at 17:10, Peter Eckhardt wrote: I just found the draft of the handbook. The software is amazing Does anyone use Asterisk in Germany on a BRI (S2M) interface ? I'm in the Netherlands, but I use Asterisk on a BRI using a Fritz!Card ISDN adapter and the chan_capi software. This software is written by 'capejod', who lives in Germany. check: http://www.junghanns.net/asterisk/ I have had success with isdn4linux as well, but chan_capi is more actively maintained. You will need an ISDN card that has capi drivers under linux of course. wkr, -- A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing on usenet and in email? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] weird error message with zaptel
On Thu, 2003-08-21 at 20:08, Grzegorz Nosek wrote: After modprobe capi modprobe fcpci /proc/capi seems ok (shows one card with fcpci driver - sorry I don't post some real output but I had to revert to i4l to make it work as soon as possible) So far, no error messages of any kind, but chan_capi says that CAPI is not installed. My /etc/asterisk/capi.conf is empty (chan_capi demanded it and I didn't know what to put there ;) There is an example capi.conf in the source directory where you built chan_capi. -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Status of ISDN DTMF (AFAIK): Please addcorrections and comments
Hi Pedro, On Thu, 2003-08-21 at 21:34, pedro bulach gapski wrote: My setup is an Eicon Diva (HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2) running on standard debian woody. I'm not sure, but isn't there a linux capi driver available for that card? If you, I suggest you try to use the capi driver and chan_capi (http://www.junghanns.net/asterisk/) instead of chan_modem. After a standard instalation and setup, my DTMF detection was unreliable. I had similar problems with isdn4linux, but never had any problem using chan_capi. Current problems are: -outbound calls (h323-*-out) to analog phones have echo. This gets better with a better mic, but does not disappear. It seems that no solution to this problem is available. chan_capi solved my echo problem completely. -my callees complain of the silence supression. I have not looked at it yet. I can't comment on h323, as I don't use it. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Asterisk Newbie ...
On Mon, 2003-08-11 at 11:28, Julien wrote: Just a last question, if i configure G723 in my ATA, i can't call the voicemail for exemple. I've seen that messages were in GSM format. Is there a way to be able to acces to the voice mail in G723 (for remote users) and in G711 for local users ? In the short path no. G.723 is a patented codec and the Digium would have to prepay a f*ckload of mony to be able to include it in their software. By it's very nature it would not be in the GPL version of asterisk, but only available in a non-GPL module/addon, and you'd have to buy it from Digium. In the long run, check the list archives, and help raise the $30K up front fee for Digium to be allowed to implement G.723 codec support. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] FWD-gateway prefix
Hi,
On Thu, 2003-08-07 at 14:23, The Traveller wrote:
The correct way to dial Dutch toll-free numbers using the gateway-prefix is:
1010-666-0800-rest of number
I haven't tried the *31(800)... they mention on their site yet, but
I don't have special provisioning on the gateway for it and the first
time I heared about it, was in their newsletter. They're either using
a different gateway for it, or re-write the number to use the prefix
internally.
Well, they both work for the long 0800 numbers. I haven't tried 0402,
don't even know what that number is for. I tried 8051 ('postbus 51'),
because it get's me into an IVR after hours, and I don't annoy anyone in
my tests.
I'm currently only allowing 0800-0101 (KPN calling-card and
collect-calls, IVR-system) of the short 0800-numbers, as some of
them are service-numbers, tied to the phone-line calling them, on which
you can change service, request accumulated charges, etc. for the
line from which you're calling, without any other authentication. Not
a good thing to allow anyone to access for our lines. :-) I might
have a closer look at it in the future and allow the non-service
short numbers only.
Ah, well, that explains it. Thanks for enlightening me!
cheers,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
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Re: [Asterisk-Users] X-Lite - Snom200
On Thu, 2003-08-07 at 04:14, Jamie Carl wrote: Dunno what I'm doing wrong here but I just did an upgrade to the latest version and now I get no audio at all! I havn't changed a single thing. Is there anything special I need to do to get this to work again? I get a quick 'chirp' of audio, which you can tell is what I'm connecting to, (ie MOH), but then nothing. Try commenting allow=all in sip.conf. wkr, Armand. -- A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing on usenet and in email? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] ADSI and SoftKeys
On Thu, 2003-08-07 at 17:06, Jayson Vantuyl wrote: I've taken the liberty to edit your patch, to put back in the 'adsi_logo' and the values for adapp and adsec as they are in CVS. As far as I can tell those changes have no relation to problem this patch solves, they're just local changes to satisfy your local preferences, right? I've removed those to ease integration into CVS. Ooops. Those were actually the security code to unlock our Aastra phones. Please disregard that. ;-) will try to. BTW, Digium needs a disclaimer for your patch. It was asked for here on this list, and I'm not sure if you've reacted allready, but I'd really appreciate it I you could send the disclaimer in. I'm working on completing the options menu now, but it is not possible for me to disclaim code that builds on cod that was not disclaimed... wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] ADSI and SoftKeys
On Thu, 2003-08-07 at 22:17, Wade Weppler wrote: I've taken the liberty to edit your patch, to put back in the 'adsi_logo' and the values for adapp and adsec as they are in CVS. As far as I can tell those changes have no relation to problem this patch solves, they're just local changes to satisfy your local preferences, right? I've removed those to ease integration into CVS. Any idea if these fixes will get added to CVS? check http://bugs.digium.com/bug_view_advanced_page.php?bug_id=35 Armand. -- A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing on usenet and in email? signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] Iconnecthere
Hi Andrew, On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armand A. Verstappen Sent: Sunday, August 10, 2003 4:57 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Iconnecthere On Sun, 2003-08-10 at 10:18, Andrew Joakimsen wrote: On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote: Does anyone have Asterisk working with Iconnect here for incoming and/or outgoing calls? have a look at: http://www.loligo.com/asterisk/example-configs.2003-04-24/extensions.con f there's a section in there dealing with Iconnect That helps a lot. But now I get this message when I try to dial any number NOTICE[5126]: File pbx.c, Line 1089 (pbx_extension_helper): Cannot find extension context 'default' You get this notice doing what? dialing in, dialing out? At any rate, it looks like you've halfway implemented the examples, sip.conf having context=default, but no context [default] in extensions.conf. When I try to dial out, but there is a [default] section in my extensions.conf Hmm... okay. We're going to need a little more context here. What kind of device/software are you calling from? sip / h323 / zap / quicknet / ... ? What's the related config setup like (so sip.conf h323.conf/oh323.conf zap.conf phone.conf ...), and what are your extensions set up like (extensions.conf). I appreciate you not top-posting anymore, but now I must make it some harder: when replying do not include the '-- ' and anything under that in your reply, and most certainly don't put your reply below that. the '-- ' on a single line is the 'signature separator', and many mailclients automatically strip anything that is below there when replying. So in your case, I had to manually copy your answer back in. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Iconnecthere
Hi, On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote: Does anyone have Asterisk working with Iconnect here for incoming and/or outgoing calls? have a look at: http://www.loligo.com/asterisk/example-configs.2003-04-24/extensions.conf there's a section in there dealing with Iconnect wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] How to Asterisk
On Tue, 2003-08-12 at 19:50, Chris Hirsch wrote: Really? Thats awesome!! Thats why I wish there was a wiki available...is anybody opposed to one? What if I was to setup one at my site? Would anybody use it? One way to find out... set it up. Once It's set up I suggest offering it to appear as http://wiki.asterisk.org/ as well, and have it linked to from www.asterisk.org. There's a big bundle of asterisk documentation out there, the problem is mostly in finding it. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] Iconnecthere
Hi, I seem to have my configuration working except for outgoing and incoming calls for the rest of the world. For now I am concerned more about outgoing calls than anything else. Whenever I try to make an outgoing call I get these messages from the sip debug in the console s=session c=IN IP4 64.36.104.203 t=0 0 m=audio 6620 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 4.42.235.170:5060 -- Called [EMAIL PROTECTED] Sip read: SIP/2.0 404 Not Found 4 Via: SIP/2.0/UDP 64.36.104.203:5060;branch=z9hG4bK37d8c90a From: asterisk sip:[EMAIL PROTECTED];tag=as220b2c68 To: sip:[EMAIL PROTECTED];tag=1m6lkhivci11cjdooja30ex45 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 Notice in particular the From line. Now notice a working session from eStara softphone: v=0 o=eStara 22079953 22079953 IN IP4 64.36.104.202 s=eStara c=IN IP4 64.36.104.202 t=0 0 m=audio 8014 RTP/AVP 0 4 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.36.104.202:5060 From: Anonymous sip:[EMAIL PROTECTED];tag=1d436a9 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 22079953 INVITE Content-Length: 0 Notice how the from is different, my SIP service will not accept calls unless the proper from name is configured, how can I configure this? Here are the relevant sections from my sip.conf file [general] port = 5060 ; Port to bind to context = from-sip; Default for incoming calls maxexpirey=13600 ; Max length of incoming registration we allow defaultexpirey=3600 ; Default length of incoming/outoing registration register = 17862324057:[EMAIL PROTECTED]/5500 [packet8.net] type=friend username=17862324057 secret=xxx host=packet8.net context=demo Well, it's dialing out, that's one thing. Now to set the outgoing caller-id and name, you'll need to do something like: exten = s,1,SetCallerID(17862324057) ; your own callerid here exten = s,2,SetCIDName(Anonymous) ; your proper from name exten = s,3,Dial(SIP/[EMAIL PROTECTED]) Note that the 's' extension should almost certainly be set to something else for your configuration, I can't guess since you didn't volunteer the relevant parts of your extensions.conf. But the SetCallerID() and SetCIDName() functions will setup the from line accordingly (the original link I sent you has this for Iconnecthere and fwd, btw) wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] unsubscribe
On Wed, 2003-08-13 at 16:22, [EMAIL PROTECTED] wrote: unsubscribe no. you can't leave. Armand. signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] Iconnecthere
On Sun, 2003-08-10 at 10:18, Andrew Joakimsen wrote: On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote: Does anyone have Asterisk working with Iconnect here for incoming and/or outgoing calls? have a look at: http://www.loligo.com/asterisk/example-configs.2003-04-24/extensions.con f there's a section in there dealing with Iconnect That helps a lot. But now I get this message when I try to dial any number NOTICE[5126]: File pbx.c, Line 1089 (pbx_extension_helper): Cannot find extension context 'default' You get this notice doing what? dialing in, dialing out? At any rate, it looks like you've halfway implemented the examples, sip.conf having context=default, but no context [default] in extensions.conf. Could you btw, please not top-post. It makes my reading very hard when the reactions end up before the questions. wkr, Armand. -- A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing on usenet and in email? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID inUK?
On Mon, 2003-08-11 at 15:53, Mark Spencer wrote: I just hope that the price difference is small enough for Digium to consider this other chipset. From the lists it is obvious that there is a lot of interest for their hardware outside the US/Japan market. Same goes for the rumoured 4port fxo cards, of course. The FXO modules we've designed for the TDM400P's are definitely designed for worldwide operation and should be able to detect polarity reversal. Also there will be an option for hardware echo cancelation. The first rev of the FXO module is in layout now so hopefully we'll have some prototypes in the next few weeks. Clearly, the TDM400P is aspiring to be the logical platform for FXS/FXO in low density. We've done some very substantial improvements on the TDM400P to obtain extremely low noise on the Rev E boards, and to eliminate other problems people were reporting (e.g. bus master aborts, etc). That's good to hear. Will Digium end-of-life the X100P once the TDM FXO modules are proven and in production? Or will there be an international version of the X100P some day? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Newbie Issue
Hi, On Fri, 2003-08-08 at 00:24, [EMAIL PROTECTED] wrote: I recently purchased the Asterisk Developer's Kit (TDM) to try out Asterisk. After following the directions in the Digium's FAQ topic entitled Q. How do I configure my TDM40B and X100P?, I'm receiving the following error: ERROR[1074428608]: File chan_zap.c, Line 6692 (load_module): Unknown signalling method 'fxs_ks # X100P' looks like you need to remove the ' # X100P' bit from /etc/asterisk/zapata.conf. '#' is not a comment character, you where looking for ';'. ERROR[1074428608]: File chan_zap.c, Line 4793 (mkintf): Signalling requested is FXO Loopstart but line is in FXS Kewlstart signalling ... and because of that chan_zap is trying to send up the wrong kind of signalling. Any ideas? none. -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] FWD-gateway prefix
From FWD, you can now dial 1010-666, followed by the Dutch toll-free number or IAXTel-number you wish to reach, as you would have dialled it from the dial-tone at FWD-number 42442. I've tried dialing the following from my FWD-client (X-lite): 1010-666-800-0402 1010-666-0800-0402 1010-666-31800-0402 none of them worked. Am I doing something wrong? If you are, I'm making the same mistake. On the FWD website it says you can dial 31-800-0402 directly , but that doesn't work either It says *31-800-0402, actually. From my setup I get I'm sorry, that's not a valid extension, please try again, when I try *31-800-0402, 1010-666-800-0402 or 1010-666-31800-0402. The others just fail. The I'm sorry... bit does not come from my local asterisk installation, so I think there's something fishy on the remote iax2pstn gateway taking the 0800 calls. It does work on a longer 0800-number. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID inUK?
Hi Dave, On Tue, 2003-08-05 at 14:53, Dave Wilson wrote: I can't seem to find any info on this anywhere on the web, except that BT caller ID doesnt use the standard bellcore system in use in the US. So, if anyone here in the UK is onlist and using the x100p successfully, please let me know. I don't have the answer to your question, I just know that it doesn't work in the Netherlands. That does not mean it will not work in the UK, as UK uses a different standard. While searching for specs I found that in the UK there's different caller-id standards in use amongst the different telco's. I'd suggest you locate the exact protocol spec for BT caller ID. If it is a dialect of FSK, there's a fair chance that it could be made to work. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID inUK?
Hi Mark, On Sat, 2003-08-09 at 17:44, Mark Spencer wrote: The other thing is that we have to detect polarity reversal or we'll constantly be scanning for CID. Indeed. I'm not familiar with the internals of the hardware, could you give some hints on how this could be achieved? Then, looking into the future, I'd like to be able to send DTMF style CLIP information out over a TDM device, so that it can be used with locally available phones. To that end, it should be possible to create polarity reversal. Is that at all possible with current hardware? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] chan_capi: Hanging channels - again
Hi Klaus-Peter, On Wed, 2003-08-06 at 12:33, Klaus-Peter Junghanns wrote: always use latest chan_capi. the bug is fixed in 0.2.4a. today 0.2.4b is online which fixes some issues with sending dtmf and a small enhancement to capiECT. I checked the site, but can't find the 0.2.4b version. The sidebar menu offers 0.2.4a, and http://www.junghanns.net/asterisk/page1.html offers 0.2.2 (under download lates version _here_). wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] ADSI and SoftKeys
Hi, On Fri, 2003-08-08 at 22:46, Jayson Vantuyl wrote: On Fri, Aug 08, 2003 at 01:06:32AM +0200, Armand A. Verstappen wrote: On Thu, 2003-08-07 at 22:17, Wade Weppler wrote: Any idea if these fixes will get added to CVS? check http://bugs.digium.com/bug_view_advanced_page.php?bug_id=35 I've signed a release. See ticket. I saw, thank you for that. Also great news about the security code. Nice folk! cheers, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] X100P CallerID issue solved for my PSTNconnection
On Wed, 2003-08-06 at 20:00, Richard Alexander wrote: And further to Dan's message I will add that I was able to help because a colleague and I are working on identifying all callerid variants with a view to patching * to work with as many as possible. If anyone has specific examples of countries/networks which don't currently work or partially work with the X100P / X101P feel free to email: Be sure to include: Country Network Description of problem 1-3 examples of manufacturer/model of phones that *DO* work on the same connection. See: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9 The Netherlands, Denmark, Sweden, any other country using DTMF CLIP for caller-id presentation. The bug has links to the specifications for CLIP in those countries. I'm not sure if this falls within the context of your project. These countries use DTMF signalling before first ring, which is not a dialect of the BellCore FSK caller-id signalling at all. At any rate, I'm very interested in any pointer on how to make this work. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID inUK?
Hi Martin, On Fri, 2003-08-08 at 00:16, Martin Stubbs wrote: Unfortunately the present x100p driver code will not decode the callerid for 2 reasons 1) the UK protocol is different to the US system. I have downloaded the specs and coding it would not be too difficult. Maybe you could open a bug for it, and attach the specs / a link to those specs? Also, I suggest you reply to this message: http://lists.digium.com/pipermail/asterisk-users/2003-August/017459.html 2) in the US the callerid is sent between the first and second rings. In the UK the callerid is sent before the first ring. Same in the Netherlands. I have been unable to determine if we can get a zaptel event when a line reversal is received which happens before the UK callerid is sent. Without this function or continuously monitoring the line for the tones we don't know when to enter the callerid routine. I've been looking for an answer to that too, for quite some time. (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9) wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] AgentCallbackLogin
out. Is there some way to distunguish them in CDR? I also noticed the management interface maintains a Unique ID for each call and lets that call be traced throughout its life in the PBX. Can that data be added to CDR as well to allow for easier call tracking? It looks like if you define MYSQL_LOGUNIQUEID in the top of cdr/cdr_mysql.c and recompile, it will start logging the unique id you want. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] callwaiting in sip can't be disabled
On Mon, 2003-08-04 at 22:41, Steve Meyers wrote: What type of phones? Grandstream BudgeTones. Is it a function of the phones? Is there any way to limit them in sip.conf to one channel each? Looking at the source of channels/chan_sip.c, the threewaycalling parameter seems not to be honoured. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] 'System' application exit with error evenifitperforms the job as expected
On Fri, 2003-08-01 at 16:45, Dan wrote:
#include stdlib.h
#include stdio.h
int main() {
int ret;
ret = system(/bin/ls /dev/null);
printf(system(\/bin/ls /dev/null\) returned %d\n, ret);
return(ret);
}
gcc mysystem.c -o mysystem
./mysystem
what is the output?
On Fri, 2003-08-01 at 10:40, Dan wrote:
This is the result:
[EMAIL PROTECTED] temp]# ./mysystem
system(/bin/ls /dev/null) returned 0
[EMAIL PROTECTED] temp]#
Okay, that at least rules out the suggestion earlier in this thread that
your systems' system call is broken.
(line below moved down from top back into context)
Then?
What can it be?
Don't know really. I'd change app_system.c, to tell me what the exact
return code we got:
...
/* Do our thing here */
res = system((char *)data);
if (res 0) {
ast_log(LOG_WARNING, Unable to execute '%s' (result
%d)\n, (char *)data, res);
res = -1;
} else if (res == 127) {
ast_log(LOG_WARNING, Unable to execute '%s' (result
%d)\n, (char *)data);
res = -1;
}
...
and recompile and install app_system.so. It would help you to know what
the return code was that is causing you problems. I'm not sure how that
would help you, but it is the one thing I can think of that would get
you more information.
wkr,
--
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
A: Top-posting.
Q: What is the most annoying thing on usenet and in email?
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Re: [Asterisk-Users] 'System' application exit with error even ifit performs the job as expected
On Fri, 2003-08-01 at 09:14, Dan wrote:
Hi,
It is the same with any other application I try to run from the System()
application.
I don't think is a privilege problem. Even with 'ls' as command, it displays
the directory listing and then exit with error.
If you create mysystem.c containing:
#include stdlib.h
#include stdio.h
int main() {
int ret;
ret = system(/bin/ls /dev/null);
printf(system(\/bin/ls /dev/null\) returned %d\n, ret);
return(ret);
}
and compile it with:
gcc mysystem.c -o mysystem
and run:
./mysystem
what is the output?
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
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Re: [Asterisk-Users] 'System' application exit with error evenifit performs the job as expected
- Original Message -
From: Armand A. Verstappen [EMAIL PROTECTED]
If you create mysystem.c containing:
#include stdlib.h
#include stdio.h
int main() {
int ret;
ret = system(/bin/ls /dev/null);
printf(system(\/bin/ls /dev/null\) returned %d\n, ret);
return(ret);
}
gcc mysystem.c -o mysystem
./mysystem
what is the output?
On Fri, 2003-08-01 at 10:40, Dan wrote:
This is the result:
[EMAIL PROTECTED] temp]# ./mysystem
system(/bin/ls /dev/null) returned 0
[EMAIL PROTECTED] temp]#
Okay, that at least rules out the suggestion earlier in this thread that
your systems' system call is broken.
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
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RE: [Asterisk-Users] voicemail file access problems
On Wed, 2003-07-30 at 22:33, Patrick wrote: Did it work after you left a new voice mail message? I was looking into the source code to fix it so that the euid was set to nobody, create the file and then change it back to uid 0, but that didn't work. Or, maybe change the file mode was 770 with the group set so that the webserver could modify the file so I wouldn't have to run a suid .cgi script. If you create the _directories_ the files are going to be created in with group apache (or whatever group your webserver runs under), with the sgid bit set, doesn't that cause the file to be created with proper permission for the cgi? But the mask of the file is set to 0700. I don't think the sgid bit will make a difference if the file isn't written 0770. It's still on readable/writable/executable by the owner. ACK. I read over that. But I think that combining the sgid creation of directories and changing 0700 to 2770 for the mkdir calls, and 0700 to 0700 for the call to ast_writefile should get you where you want to be. I haven't tested it, it just _looks_ that way. (assumption is the mother of all fuckups) good luck, Armand. -- A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing on usenet and in email? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] ADSI and SoftKeys
On Thu, 2003-07-31 at 16:30, Jayson Vantuyl wrote:
On Wed, Jul 30, 2003 at 05:07:50PM +0200, Armand A. Verstappen wrote:
On Wed, 2003-07-30 at 16:40, John Congdon wrote:
Has anyone solved the problem on the ADSI phones
that when you hit one of the soft keys, the Number Pad
stops working?
It relates to not putting the phone back into voice mode when the
prompts are playing (after updating the screen).
Attached is my (partially incomplete but usable) voicemail2 patch.
Interestingly, it fixes an apparent typo in the Options softkey as well
(that one took me a week to track down).
Excellent!! Tested here, this solves the problem. Thank you.
I've taken the liberty to edit your patch, to put back in the
'adsi_logo' and the values for adapp and adsec as they are in CVS. As
far as I can tell those changes have no relation to problem this patch
solves, they're just local changes to satisfy your local preferences,
right? I've removed those to ease integration into CVS.
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
--- app_voicemail2.c.cvs2003-07-31 20:40:29.0 +0200
+++ app_voicemail2.c2003-07-31 20:43:38.0 +0200
@@ -1290,7 +1290,7 @@
bytes += adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 0, Listen,
Listen, 1, 1);
bytes += adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 1, Folder,
Folder, 2, 1);
bytes += adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 2, Advanced,
Advnced, 3, 1);
- bytes += adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 3, Options,
Options, 4, 1);
+ bytes += adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 3, Options,
Options, 0, 1);
bytes += adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 4, Help, Help,
*, 1);
bytes += adsi_load_soft_key(buf + bytes, ADSI_KEY_APPS + 5, Exit, Exit,
#, 1);
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DOWNLOAD);
@@ -1433,6 +1433,7 @@
bytes += adsi_input_format(buf + bytes, 1, ADSI_DIR_FROM_LEFT, 0, Password:
**, );
bytes += adsi_input_control(buf + bytes, ADSI_COMM_PAGE, 4, 0, 1,
ADSI_JUST_LEFT);
bytes += adsi_set_keys(buf + bytes, keys);
+ bytes += adsi_voice_mode(buf + bytes, 0);
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
}
@@ -1460,6 +1461,8 @@
bytes += adsi_display(buf + bytes, ADSI_COMM_PAGE, 2, ADSI_JUST_CENT, 0, ,
);
bytes += adsi_set_line(buf + bytes, ADSI_COMM_PAGE, 1);
bytes += adsi_set_keys(buf + bytes, keys);
+ bytes += adsi_voice_mode(buf + bytes, 0);
+
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
}
@@ -1546,6 +1549,8 @@
bytes += adsi_display(buf + bytes, ADSI_COMM_PAGE, 4, ADSI_JUST_LEFT, 0,
datetime, );
bytes += adsi_set_line(buf + bytes, ADSI_COMM_PAGE, 1);
bytes += adsi_set_keys(buf + bytes, keys);
+ bytes += adsi_voice_mode(buf + bytes, 0);
+
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
}
@@ -1589,6 +1594,8 @@
/* Except Exit */
keys[5] = ADSI_KEY_SKT | (ADSI_KEY_APPS + 5);
bytes += adsi_set_keys(buf + bytes, keys);
+ bytes += adsi_voice_mode(buf + bytes, 0);
+
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
}
@@ -1632,6 +1639,8 @@
keys[0] = 1;
bytes += adsi_set_keys(buf + bytes, keys);
+ bytes += adsi_voice_mode(buf + bytes, 0);
+
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
}
@@ -1670,6 +1679,8 @@
bytes += adsi_set_line(buf + bytes, ADSI_COMM_PAGE, 1);
bytes += adsi_set_keys(buf + bytes, keys);
+ bytes += adsi_voice_mode(buf + bytes, 0);
+
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
}
@@ -1681,6 +1692,8 @@
if (!adsi_available(chan))
return;
bytes += adsi_set_line(buf + bytes, ADSI_COMM_PAGE, 1);
+ bytes += adsi_voice_mode(buf + bytes, 0);
+
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
}
@@ -1695,6 +1708,8 @@
bytes += adsi_display(buf + bytes, ADSI_COMM_PAGE, 3, ADSI_JUST_LEFT, 0, ,
);
bytes += adsi_display(buf + bytes, ADSI_COMM_PAGE, 4, ADSI_JUST_CENT, 0,
Goodbye, );
bytes += adsi_set_line(buf + bytes, ADSI_COMM_PAGE, 1);
+ bytes += adsi_voice_mode(buf + bytes, 0);
+
adsi_transmit_message(chan, buf, bytes, ADSI_MSG_DISPLAY);
}
@@ -2036,6 +2051,18 @@
char newpassword[80] = ;
char newpassword2[80] = ;
char prefile[256]=;
+ char buf[256];
+ int bytes=0;
+
+ if (adsi_available(chan))
+ {
+ bytes += adsi_logo(buf + bytes);
+ bytes += adsi_display(buf + bytes, ADSI_COMM_PAGE, 3, ADSI_JUST_CENT,
0, Options Menu
Re: [Asterisk-Users] ADSI and SoftKeys
On Thu, 2003-07-31 at 20:59, Armand A. Verstappen wrote: Has anyone solved the problem on the ADSI phones that when you hit one of the soft keys, the Number Pad stops working? It relates to not putting the phone back into voice mode when the prompts are playing (after updating the screen). Attached is my (partially incomplete but usable) voicemail2 patch. Interestingly, it fixes an apparent typo in the Options softkey as well (that one took me a week to track down). Excellent!! Tested here, this solves the problem. Thank you. I've taken the liberty to edit your patch, to put back in the 'adsi_logo' and the values for adapp and adsec as they are in CVS. As far as I can tell those changes have no relation to problem this patch solves, they're just local changes to satisfy your local preferences, right? I've removed those to ease integration into CVS. and added to the bug-tracker: http://bugs.digium.com/bug_view_page.php?bug_id=035 wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Dummy account/extension
On Wed, 2003-07-30 at 15:55, Dan wrote:
It is possible to create a dummy account (SIP or IAX type) in order to be
used in a dummy extension?
I want to be able to use it as a normal extension (as an IP phone connected
to it), but without the need to answer or call from that extension.
I want that when I call that extension to hear the ring, and after the
defined period of time to enter in the Voicemail system.
I don't want to use a real phone (hardware or software) for this purpose.
It is possible to do this in a simple way?
doesn't:
[globals]
WAITTIME=10
MAILBOX=1234
[dummy]
exten = 1234,1,Wait(${WAITTIME}) ; give illusion we might pick up
exten = 1234,2,VoiceMail2(${MAILBOX}) ; then kick into voicemail
exten = 1234,3,Hangup
do the trick?
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
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Re: [Asterisk-Users] isdn4linux/Teles16.3
Hi, On Wed, 2003-07-30 at 16:15, [EMAIL PROTECTED] wrote: is it possible to use a Teles16.3 via isdn4linux for the external phone connections (phone provider net)? Yes, it is. I tested using an old card I had lying around. I quickly switched to a Fritz card and chan_capi however. This solved the big issue I had with echo for me. Since I had both available, I did not spend much time trying to solve the problems I had. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] ADSI and SoftKeys
On Wed, 2003-07-30 at 16:40, John Congdon wrote: Has anyone solved the problem on the ADSI phones that when you hit one of the soft keys, the Number Pad stops working? No, I haven't. Just confirming that I have the same problem here, using the VoiceMail2 app. Do you experience this outside VoiceMail2 as well? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Dummy account/extension
On Wed, 2003-07-30 at 16:44, Dan wrote: Thanks for the suggestion. I have change it like that: ;dummy extension exten = 199,1,Ringing exten = 199,2,Wait(60) ; give illusion we might pick up exten = 199,3,Hangup in order to hear the ring too. ..but now... how can I do to call this extension from a Dial command? Not sure what you are trying to do, but would the goto app be of any help? [other-ext] ... exten = 198,3,Goto(dummy,199,1) wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] %unsuscribe
On Wed, 2003-07-30 at 22:25, Carlos Crembil wrote:
%unsuscribe
variable subsitution on the mailinglist contents of asterisk is not
implemented.
If i were, the correct syntax probably would have been:
exten = _asterisk,1,Agi(mailinglist,%{unsubscribe})
There's a link on the bottom of this mail. You'll have better luck
there.
To the list admin, maybe it should say 'to unsubscribe, send mail to
...', just to be more foolproof?
--
'Just when you make something foolproof, the release a better fool.'
Armand.
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RE: [Asterisk-Users] voicemail file access problems
On Wed, 2003-07-30 at 22:33, Patrick wrote: Did it work after you left a new voice mail message? I was looking into the source code to fix it so that the euid was set to nobody, create the file and then change it back to uid 0, but that didn't work. Or, maybe change the file mode was 770 with the group set so that the webserver could modify the file so I wouldn't have to run a suid .cgi script. If you create the _directories_ the files are going to be created in with group apache (or whatever group your webserver runs under), with the sgid bit set, doesn't that cause the file to be created with proper permission for the cgi? -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Channel Language
On Mon, 2003-07-28 at 08:36, Peer Oliver schmidt wrote:
Ok, the first three things I did. Unfortunately, I am no c coder. But
the logic to say german numbers is identical to the english logic, ie.
21 = twenty one
11 = eleven
210 = two hundred ten ('and' between Hundred and ten is optional)
1200 = one thousand two hundred
2102 = two thousand one hundred two ('and' between hundred and two is
optional)
Are you sure the logic is the same? I thought german number logic was:
21 = ein und zwanzig (one and twenty)
34 = vier und dreizig (four and thirty)
At least I'm sure the dutch logic is like that.
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
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tel: +31 (0)30 298 2255Postbus 19127
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Re: [Asterisk-Users] Cisco ATA Advanced CallerID
On Thu, 2003-07-24 at 17:02, Pauline Middelink wrote: The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Price is 77 euro something and available with Telec. (NL) Thank you for the pointer. From the quick glance I got, it supports dutch (DTMF style) CLIP, and the newer CNIP to receive the name part. Do you happen to know if this phone is switchable to FSK signalling? I did not see the tell tale trace 'also works on UPC cable phone network' in the add. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Cisco ATA Advanced CallerID
Hi Pauline, On Thu, 2003-07-24 at 22:21, Pauline Middelink wrote: The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Thank you for the pointer. From the quick glance I got, it supports dutch (DTMF style) CLIP, and the newer CNIP to receive the name part. Do you happen to know if this phone is switchable to FSK signalling? I did not see the tell tale trace 'also works on UPC cable phone network' in the add. Well, since the ATA we are using is in default mode (US, hence FSK) I presume the phone can do FSK. makes sense. Looking at the bits in the ATA, I only see sizes for number and text on the FSK mode setting, on DTMF it only has a number length. Can I conclude from this limited data that DTMF can't do text? Yes, you are correct. CLIP only presents the number of the caller. it uses DTMF based signalling. I'm playing with a devkit, split up over two installations: (analog) PSTN---X100P---asterisk---Quicknetanalog phone (isdn) PSTN---Fritz!Card---asterisk---TDM40B---analog phone now, on the analog side, I don't get the caller-id, because it is sent as CLIP-DTMF, and asterisk is looking for FSK. on the isdn installation I do get the caller id information (again, numeric only) from PSTN, using either isdn4linux or chan_capi, but on the analog phone, I don't get this callerid passed on, because the phone is looking for CLIP-DTMF, but receives FSK from the TDM40B. Because UPC uses FSK-signalling on their cable phone service, there are some phones available in the Netherlands that are usable with asterisk. It's just limiting the number of phones that can be used. The problem with the X100P not being able to pick up caller-id can only be solved if the hardware and drivers have certain capabilities, and the software is capable of recognizing the CLIP information sent. I'm not sure if the X100P and TDM40B hardware is capable of detecting / sending polarity switch. This is what is used in DTMF-CLIP to signal that caller-id info is coming, it is sent _before_ the first ring. The Number info is then sent as DTMF Dnum-1num-2...num-nC. There's a number of countries that use basically the same protocol, but some use A and # for start and end signalling, and other variants. Now all i have to figure out is how to confince the ATA to sent the callerid BEFORE the first ring instead of between rings, because i'm so close to the phone, i pick it up in the first ring and than the display has no name.. :( That sounds like FSK indeed. DTMF CLIP sends it information before the first ring. Funny thing about the DTMF-CLIP spec is that the first ring will be sent _after_ sending the caller-id info, but that transmission of caller-id is to be aborted if a call is answered before sending is complete... ;-) Met vriendelijke groet, ...en aan eenieder die mij een warm hart toedraagt _geen_ stomp in de maagstreek ;-) -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
Hi Steven! Small world isn't it? On Mon, 2003-07-21 at 15:52, Steven J. Sobol wrote: Hello, * newbie here, I've been lurking on the list for a few months now. I'm looking at DynExtenDB (and have played with it). I love that it reads the dialplans out of a MySQL database - that is a critical issue for me. But it has some issues. I haven't found this DynExtenDB however. Could you provide me with some pointers to it? PS: We never finished the Aegir Addon stuff. Maybe we can do that over iaxtel sometime? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] anyone with X100P Callerid working outsideUS ?
On Mon, 2003-07-21 at 19:25, Martin Pycko wrote: I'm just curious if anyone has the X100P Callerid receiving working outside US. It does not work in the Netherlands. The Netherlands does not use FSK signalling, but DTMF signalling: 1) polarity reversal 2) DTMF: DNumberC 3) ring signal where D and C are the DTMF tones 'D' and 'C' respectively, signalling start and end of DTMF Caller-ID transfer. Exact specification (including length of tones and pauses) is in this document: http://www.kpn.com/common/downloads/01_Part2-PSTN_V32.pdf paragraph 6.2.3. At least Sweden and Denmark use very similar CLIP protocols, the difference being mainly in the start and end tones used. The different protocol also bites on the other end, as asterisk will send callerid information to a phone connected to a TDM40B for example using the FSK protocol. Dutch phones don't understand FSK, and hence don't pick up on the caller id. I'm very interested in solving this problem, as it makes asterisk only usable in the Netherlands using ISDN BRI or PRI on the PSTN side, and imported phones on the (analog) internal side. I just don't have any idea where in the source, and how... One possible solution for the 'inside' problem: There's one company in the Netherlands offering telefony over cable infrastructure, they use FSK signalling for Caller-ID presentation. I'll get my hands on a phone suited for their network soon, wich will allow me to verify if they work with asterisk. -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] conference problem without zapata interface
Hello, On Thu, 2003-07-17 at 09:00, Andrzej Radke wrote: In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n INSTALLED FOR CONFERENCING FUNCTIONALITY.\n I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? No. What can I do if I want make conference only between my sip phones using asterisk ?? Buy it ??? Yes. Alternatively, you get the zaptel drivers, edit the Makefile to build 'ztdummy' (remove the '#' before ztdummy on the line just after the line starting with MODULES), compile, install, and do modprobe ztdummy. Why this would help can be found in the archives (just as this answer) and is left as excercise for the reader. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] asterisk and modem
Hi, On Mon, 2003-07-14 at 15:58, Angelo Sampietro wrote: i have to do a demo with asterisk, unfortunately i don't have yet an x100p card, so i need to use a 56k voice modem on my motherboard... could someone tell me how i can configure asterisk to use this modem to call? Forget about it. If you'd ever get it to work, you would demo something that is below acceptable standards. Rather demo voip-asterisk-voip without any PSTN functionality. Or, if you have an ISDN BRI, get an ISDN card and use that (chan_modem_i4l or chan_capi) depending on the ISDN card. Or, just delay the demo until after the X100P has arrived. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] EZ-Install
On Mon, 2003-07-14 at 18:36, Steven Critchfield wrote: Sounds like you needed to start a new thread. One of these days I will either need to look up a good resource for mail list rules, or write it for all these newer users. http://www.freeradius.org/list/users.html comes a long way... wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
