Re: [Asterisk-Users] messages of Mobile Operator
Il 12:38, martedì 20 dicembre 2005, Matteo Piazza ha scritto:
Hi,
I have this problem. When I call a GSM number with the IDSN line if the
GSM phone I not hear the messages of operator but I hear the ring.
I suppose that the problem is that asterisk waits a response that is yet
arrived.
Any idea?
This is my extension:
exten = _0XX.,1,Wait,1
exten = _0XX.,2,Dial(Zap/g1/${EXTEN:1},60,tT)
exten = _0XX.,3,Hangup
Matteo
I'm registering the same problem, I think is a problem due to bristuff
implementation, are you using bristuff? what version? what version of
asterisk? What version of kernel? libc? distribution?
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[Asterisk-Users] queue members with multiple devices (bug 4759)
Hello all, I've written the small patch to do what is said in the subject. kpfleming sais that the same logic can be accomplished in the dialplan but I could not find how (as I don't know how to verify if an agent is busy or not) anyone of you have an idea? Thank you ___ 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] Call inband progress indication and zaphfc
Hello all, I've a little clue with zaphfc used to connect to a BRI linethat probably can be a configuration issue (really I hope so) Here, telcos (expecially mobile operators) use to substitute the dialtone with some vocal indication without answer the line. (Indications like The customer is not reachable or wait because the customer is on the phone ecc..) For asterisk this condition is a normal dial tone and the message from the telco and it's not possible to listen theese indications. As I'm using zaphfc and with X100p and a normal analog line I can listen these indications, my question is Have you tryed with PRI cards? as I don't know if this is an issue of asterisk, zaphfc or my configuration. Thank you in advance Diego ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff-RC8b-CVS
for anyone is using RC8b-CVS: there are some major bugs in asterisk chan_sip and utils. It's convenient to download new asterisk/utils.c and asterisk/channels/chan_sip.c and reapply the kapejod patches to chan_sip.c ___ 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] Dialplan variables
Il 01:20, mercoledì 29 dicembre 2004, Norman Zhang ha scritto: Hi, May I ask what does exten = s,1,Answer exten = s,2,ResponseTimeout(5) exten = i,1,Playback(pbx-invalid) s, t, i stands for? It says it is someexten but I still don't get it. s: start is the extension invoked when there is the option immediate=yes in the channel t: timeout, is the extension where asterisk goes when a user doesn't respond in time to a directory request i: invalid, is the extension to go when it's digited a wrong extension. ___ 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] iaxComm to iaxComm
Il 02:23, venerdì 19 novembre 2004, Adam Fineberg ha scritto: Having some trouble with segfaults and sound quality all of a sudden (since I recompiled from the latest source) when 2 iaxComm clients connect. First off immediately after the server reports: -- Attempting native bridge of IAX2/[EMAIL PROTECTED]:4569/1 and IAX2/4589/5 One or both client may sometimes segfault. Additionally, when they do get properly connected, I'm seeing this message which I didn't before. Sound quality gets very poor as I'd expect from the message. Nov 18 15:56:23 NOTICE[-184759376]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/[EMAIL PROTECTED]/3 of format SPEEX since our native format has changed to ULAW Both have: disallow=all allow=ulaw in the iax.conf file. Anyone have any ideas how I messed this up? As I know, iaxcomm can't change the codec session-time, iaxcomm start the connection in one codec and then ignore change messages from asterisk. ___ 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] Service numbers
Il 12:20, venerdì 05 novembre 2004, amela cehajic ha scritto: Hi all, I was looking for the information of service numbers for the countries worldwide. I was using Google for couple of hours and managed to find phone numbers of Police, Fire, Amulance,Tourist Information and General Emergency services. I still need the phone numbers of Roadside Assistance and Directory Enquiry. I would be grateful if you could help me and send the information of those services for your country. Best Regards Amela Italy: Emergency: 112: Carabinieri 113: Soccorso pubblico di emergenza (Polizia) [Police] 115: Vigili del Fuoco [Fire dept] 118: Emergenza sanitaria [first aid station] 117: Guardia di Finanza Public Utility: 1515: Servizio Antincendi boschivi del Corpo Forestale dello Stato [forest ranger forest fire dept.] 1518: CCISS Viaggiare Informati [Traffic Informations] 1530: Capitaneria di Porto [Coast Guard] 187: Customer Care Residenziale Telecom Italia [Telecom Italia Phone Company Customer Care] 191: Customer Care Business Telecom Italia [Telecom Italia Phone company Customer Care] 803.116: Soccorso Stradale ACI [Road Assistance] 803.803: Soccorso Stradale EuropAssistance VAI please make a good web page! Hope this help Diego ___ 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] Got *80 working ... now some Blacklist questions
Il 05:41, mercoledì 08 settembre 2004, Steve Maroney ha scritto: On my default asterisk installation, *80 didn't work until I modified the source to move call pickup to *9. I wasn't sure what I was doing but *80 works now. Except I thought *80 would play some voice prompts that gave the option to add the last caller to the black list as well as other options. Instead I just get a studer dial tone after the last caller gets added to the database. I've opened a bug one month ago http://bugs.digium.com/bug_view_page.php?bug_id=0002247 that involve your problem. Diego ___ 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] PSTN tone simulation
Il 12:52, giovedì 29 luglio 2004, [EMAIL PROTECTED] ha scritto: Hi all is there any way how can I simulate PSTN tone on asterisk. I mean: I take up phone, select number '9' (so I want to call to PSTN) and asterisk change tone to something like . - . - . - exten = 9,1,dial(Zap/1/,60) in the context where you are with a zap channel configured (x101p) but I've tryied then I ear the call tone but I can't dial a number. ...so try your own and leave a comment if you succesfully configure it... bye ___ 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-0.3.4b and asterisk last cvs
Il 10:03, giovedì 22 luglio 2004, Jason Williams ha scritto: ensure you have the following in the [global] section [global] chan_modem.so=yes chan_capi.so=yes sorry, why do you need chan_modem? I don't understand as chan_modem is another channel as are chan_iax, chan_sip ___ 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] Zaptel Hung Up on x101p and cisco analog line
I'm having many troubles with x101p (orginal from Digium, wcfxo kernel module) on analog line simulated by a Cisco Router I'm experiency random hung up while zaptel doesn't recognize call progress (Italy signalling) Signalling is simple ignored as if someone hangs up on the other end of the analog line, on the asterisk end is it possible to listen the busy tone as generated from cisco router without hang up Zap channel On other calls a sort of hangup detect causes random hangup. (I've tryied with an analog phone and is not a cisco problem is related to asterisk) my question is: How is it possible to correctly detect call progress (callprogres=yes is for US signalling) or to ignore completely call progres to stop theese random hangup? Help please... ___ 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-0.3.4a
Hallo, due to everchanging CVS, chan_capi-0.3.4a doesn't compile anymore with new cvs my solution was to chande chan_capi.c the line 21 from #include asterisk/parking.h to #include asterisk/features.h now chan_capi compiles again and seems back on duty again. Hope this help. Diego ___ 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] Help needed regarding Grandstream phone
Il 15:54, venerdì 09 luglio 2004, Andrew Thompson ha scritto: Shanmuganathan Kumaravel wrote: If anyone knows it pls help it would be very helpful regarding my project work. Regards Shan I've got the same issue between ata-186 and grandstream, this was a codec issue to diagnose problems I suggest you to disallow=all allow=ulaw in sip.conf Diego ___ 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] Hyperthreading?
Il 10:34, martedì 01 giugno 2004, Chris Bond ha scritto: Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond I'm using asterisk on hypertreading processor, without any problem ___ 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] Can I do this ...
Il 06:52, mercoledì 26 maggio 2004, Shaun Ewing ha scritto: Try: exten = s,1,Playback(thanksforcalling) exten = s,2,Dial(SIP/SIP/1112|30|m) exten = s,3,Voicemail(uEXTEN) exten = s,4,Playback(vm-goodbye) That will answer and play back thanksforcalling.gsm, dial SIP/ and SIP/1112 with music. If not answered within 30 seconds, it will go to voicemail. You could also add a 103 line to be used if both extensions are busy (eg: voicemail(bEXTEN)). -Shaun Is it possible to answer with a message WHILE calling by dialling. in effect a sort of dial option A() [see show application dial] but for the calling party.. ___ 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] 11 instead of Star and how about #?
Il 10:34, mercoledì 26 maggio 2004, Peter Corlett ha scritto: On Tue, May 25, 2004 at 10:37:25AM -0500, Greg Blakely wrote: [...] + It's just as well that *8# isn't used for call pickup anymore. The # on the end really SHOULD mean end of dialing, and not have any other significance. Unfortunately, BT and GSM service codes give significance to # in the middle of the dialling sequence: *NN# - Enable service with code NN #NN# - Disable service *#NN# - Query status of service Or has this already been discussed to death? Possibly, but some of us are still arguing over the corpse :) Also here in italy. there is also another small problem, what happens if a called phone directory need to press the # to continue and # have a transfer mean for asterisk? It's possible to escape the # sequence ___ 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] (no subject) MGCP
Il 21:38, venerdì 30 aprile 2004, Philipp von Klitzing ha scritto: Hi! I try to connect an MGCP device(Terayon) to asterisk. I have found many example BUT the Terayon always return error 510 ! Verb:'510' Identifiers :'2' Endpoint: 'Error' Version'(null)' 1. Which version of Asterisk exactly (!) are you using? 2. Try CVS-03/05/04-00:50:56 instead and see if that solves your problem. For me recent CVS has made using MGCP completely impossible (with Swissvoice ip10 having been upgraded to newer firmware) 3. Look at the MGCP bugs on bugs.digium.com to find out if you find a related issue. Add your comment plus debugging info there, or create a new bug. Cheers, Philipp So do I, I've got cisco ata 186 with MGCP. Last versions of asterisk are very unusable with MGCP! ___ 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] Quality Problem
Il 17:28, martedì 13 aprile 2004, Robert Siedl ha scritto: Hi List, I have asterisk running on my server and work with 2 cisco ata und 1x snom device. I can intern call it´s fine. But wenn i make a extern call, I have many quality troubles. The extern user hear me good, but I hear him bad (robotics). I work with SIP an ALAW protocol. Where can i look this error? I am a new asterisk user. Thank you, best regards, Robert Siedl Hello, I've also had the same problem. As I know, this issue is related to chan_capi and the new lock features of asterisk (that are used by chan_mgcp) until new releases, the only solution (I've found) is to roll back to cvs version prior 12nd march 2004. Hope this help ___ 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] ANNOUNCE: Flash Operator Panel - keynames
Il 22:52, giovedì 01 aprile 2004, Nicolas Gudino ha scritto: http://sip.house.com.ar/operator Best regards, I've seen that keynames are very strictly. The problem is that for example CAPI channel, change the name every time with a serial number canal: SIP/GS1 canal: MGCP/[EMAIL PROTECTED] canal: MGCP/[EMAIL PROTECTED] canal: CAPI[CONTR1/0515871620]/40 canal: CAPI[CONTR1/0515871620]/40 canal: CAPI[CONTR1/0515871620]/41 canal: CAPI[CONTR1/0515871620]/41 canal: CAPI[CONTR1/0515871620]/41 canal: CAPI[CONTR1/0515871620]/41 canal: MGCP/[EMAIL PROTECTED] canal: MGCP/[EMAIL PROTECTED] canal: MGCP/[EMAIL PROTECTED] canal: CAPI[CONTR1/0515871620]/42 canal: CAPI[CONTR1/0515871620]/42 canal: CAPI[CONTR1/0515871620]/43 canal: CAPI[CONTR1/0515871621]/43 canal: CAPI[CONTR1/0515871621]/43 canal: CAPI[CONTR1/0515871621]/43 ___ 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] dialout with chan_capi
Il 23:56, giovedì 01 aprile 2004, Marc Dirix ha scritto: Hi, When I try to dialout over chan_capi everything works fine when I settle for msn=* in my capi.conf and use the primary msn of my ISDN-line. But trying to configure a different MSN the chan_capi doesn't dial and comes with: No one is available to answer at this time What can be the prob? I've specified: incomingmsn=* and msn=number1,number2, Number are to be specified with local prefix. ___ 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] Call pickup - phone continues to ring - still a problem?
Il 01:02, venerdì 02 aprile 2004, John Vogel ha scritto: I am still experiencing the problem where you pick up an incoming analog call ringing on SIP Phone A with SIP Phone B using *8 but Phone A continues to ring. This happens with Grandstreams and Snoms on the 0.7.1 code base. My theory is that Asterisk is not telling Phone A to stop ringing when the pickup occurs but I don't really know. The problem does not occur when it is purely a SIP-to-SIP phone call. Does anyone have a solution? How about callgroups and pickupgroup in sip.conf? ___ 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] CAPI - MGCP problem strange behaviour
Il 14:49, mercoledì 24 marzo 2004, Diego Ercolani ha scritto: [] As wrote, it seems that MGCP-CAPI together of lasts cvs releases are bad. Infact I've tryied the CVS version of 2004-03-12 and now I can phone via Cisco ATA-186 MGCP and CAPI-0.3.1 without the noising problems that I've found before. Hope to find more news. ___ 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] CAPI - MGCP problem strange behaviour
Very strange ISSUE. For an errouneus rm -rf command while exprerimenting softmodem, I've removed my source tree, so I've downloaded a new one and recompiled without softmodem but with CAPI. Now I've this problem: If I phone via CAPI from a SIP phone All goes right. If I phone via PSTN through a cisco ATA-186 using MGCP, and trhough CAPI, happens that my called party ear me cleary but I ear the other end in robotic manner with glitches I've forced the use of ulaw and alaw as voice codec but it continues. If I phone from ATA-186 device via a Zap Channel, I ear my partner and my partner ears me. Very strange. Could you address me? I've tryied to use head cvs, stable_1.0 cvs and even my previous (?) version (cvs of one week ago) but the same problem persists. I'm using a AVM B1 with AVM drivers Really, yesterday I was trying also zaphfc with another hfc card.. but today I've removed all. I repeat, CAPI semms working fine as I can phone from SIP phone. but also MGCP seems to work fine as I can phone from ATA to internal SIP PHONES and Vice Versa. I've also removed and reinstalled capi drivers. Thank You for your help. ___ 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] SoftFAX/spandsp: installing and results on Gentoo/SuSE
Il 14:27, martedì 23 marzo 2004, Stephen Davies ha scritto: Hi, I've installed spandsp library an RxFAX app on my Gentoo based * server - here's a little report on the process. [...] My conclusion - spandsp seems to have a compatibility issue with tiff-3.6.1. (Unless my install approach was broken). And - you may want to make the two missing header files available for more than just 3.5.7 of libtiff. I'm really impressed with the end-result - awesome to do that all in software! Regards, Steve Davies I've found the same unresolved issue with libtiff. I use SuSE 9.0 To address the problem, here there are two links to download my copy of libtiff and tiff 3.6.1 http://www.alleanzapopolare.net/asterisk/libtiff-3.6.1-307.i586.rpm http://www.alleanzapopolare.net/asterisk/tiff-3.6.1-307.src.rpm With this library, I compile just fine, I receive faxes, and don't fall in core dumps but the tiff file rxfax receive is readable but the image is only some black lines and dots not what I send. Thnak you ___ 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] Re: SoftFAX/spandsp: installing and results on Gentoo/SuSE
Il 23:08, martedì 23 marzo 2004, Reinhard Max ha scritto: Hi, On Tue, 23 Mar 2004 at 15:36, Diego Ercolani wrote: I've found the same unresolved issue with libtiff. I use SuSE 9.0 SuSE Linux 9.0 comes with libtiff-3.5.7 which allows to compile and run Steve's fax applications just fine with the additional headers provided by him. So which unresolved issue do you have with it? cu Reinhard The only thing I've found is that there is differen tpyedef declaration in asterisk/include/asterisk/md5.h (#typedef long int uint32) and in /usr/include/tiff.h (#typedef unsigned int uint32) that conflicts in asterisk build. I've resolved this issue using new tiff version but I don't know if this is correct or not. ___ 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] Q931 Message - Connect - Billing
Il 02:10, mercoledì 17 marzo 2004, Daniel Bichara ha scritto: Hi All, I have posted before asking for a Connect message sent from Zap (ISDN/PRI - by *) when receiving a call (incoming) and dialing to another extension. To clarify the situation, I will describe the problem: 1) My * box is connect to a Cisco (E1-ISDN/PRI) using a crosscable. my experience: - cisco is generally configured PtP and not PtMP this is a problem generally versus normal ISDN card - you can configure PTMP but you can not connect more than one device to the ISDN port hope this would help Diego Ercolani ___ 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] Newbie - Bashing head on wall! - RH9 - * - How do I install AVM C2 ISDN Pretty Please!
Il 18:24, martedì 16 marzo 2004, Nick Grindley ha scritto: Hi All, Sorry to be a pain nut I have spent three days with no joy at installing the AVM C2 ISDN card. I even tried to install it in a Suse box! So please (in child like terms for me please) what are the steps and what software package do I need to install the AVM C2 into the following config: - Linux Version 2.4.20-8 i686 I have also installed mpg123, zaptel, zapata (do I need this?), libpri and asterisk. All from CVS today. I am using grandstream phones and can call each phone, leave and receive voicemail etc. However I cannot do a pppd call isdn/avm out of the box - if I cannot even do this there is no chance with *! Any help you can give will make me very happy. But please keep it simple as I am now convinced that I have lost it Regards to all you good people here and thanks. Nick (FRUSTRATED OF SCARBOROUGH)!! Use yast2 from suse and configure isdn: 1. capi4linux - you have to install AVM CAPI4linux driver http://www.avm.de/de/Service/AVM_Service_Portale/Linux/index.php3 - you have to install capi_chan http://www.junghanns.net/asterisk/ - you have to configure asterisk capi.conf and extension.conf 2. i4l - you have to use hisax as driver - you have co tonfigure modem.conf to use i4l - you have to configure extension.conf Hope this would help you ___ 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] ISDN BRI with DDI support
Il 17:36, lunedì 15 marzo 2004, Maros RAJNOCH ha scritto: Hi all, I want to build in my small company little PBX with asterisk. I have one ISDN BRI link with DDI preselection and a couple of analog phones. I need somethink like this +--+--- phone 1 (extension 1) | LINUX with ASTERISK +--- phone 2 (extension 2) ISDN BRI DDI | +--- phone 3 (extension 3) ---+ | | +--- phone 4 (extension 4) +--+--- phone 5 (extension 5) So, my problem is, I'm lost in lots of abbreviations, and I have no idea which pci card to select. I know, I need one ISDN BRI DDI card for incoming isdn line and some pci cards with RJ11 for analog phones. 1) can anybody help me to select correct pci cards? 2) can i use more then one isdn cards (connected together) if I need more the 2 simultaneous phone calls? THANK you very much. This is what i've understood 1) Digium TDM40B offer 4 FXS to connect 4 analog phones two TDM cards are allowed on the samemachine but not more because of bus 2) No, althought ISDN is a bus, BRI connection can allow only two calls at the same time. You have to move to PRI if you want more usable lines. I'm not totally sure but this would be true enought. ___ 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
