Re: [asterisk-users] (solved) Sangoma A101 problem
Marcin Kwiatkowski wrote: Hello, We bought Sangoma A101 card and after some troubles successfully compiled modules (Kernel 2.4.32, wanpipe 2.3.3-3 and zaptel 1.0.8 drivers). wanpipemon can reach w1g1 interface, but asterisk (1.2.8) not... When I'm trying to reload chan_zap so error occurs: christine*CLI reload chan_zap.so That is when administrator is after about 100 hrs. continuous work.. I forget to run ztcfg before starting asterisk... everything works excellent. -- Marcin Kwiatkowski System Administrator Mob: +48 663 617 664 Fix: +48 33 819 04 60 ext. 32 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A101 problem
Hello, We bought Sangoma A101 card and after some troubles successfully compiled modules (Kernel 2.4.32, wanpipe 2.3.3-3 and zaptel 1.0.8 drivers). wanpipemon can reach w1g1 interface, but asterisk (1.2.8) not... When I'm trying to reload chan_zap so error occurs: christine*CLI reload chan_zap.so -- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Aug 10 12:17:13 WARNING[9393]: chan_zap.c:10886 setup_zap: Ignoring switchtype Aug 10 12:17:13 WARNING[9393]: chan_zap.c:10886 setup_zap: Ignoring pridialplan Aug 10 12:17:13 WARNING[9393]: chan_zap.c:10886 setup_zap: Ignoring prilocaldialplan Aug 10 12:17:13 WARNING[9393]: chan_zap.c:10886 setup_zap: Ignoring signalling Aug 10 12:17:13 ERROR[9393]: chan_zap.c:10317 setup_zap: Unable to reconfigure channel '1-15' Aug 10 12:17:13 WARNING[9393]: chan_zap.c:11079 reload: Reload of chan_zap.so is unsuccessful! Any ideas about that? Here you have my zaptel.conf (yes, polish E1 provider uses extended super frame): span=1,1,0,esf,hdb3,crc4 #span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone = pl defaultzone=pl and here you have my zapata.conf: [channels] language=pl context=from-pstn switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-15 channel = 17-31 -- Marcin Kwiatkowski System Administrator Mob: +48 663 617 664 Fix: +48 33 819 04 60 ext. 32 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How much bandwidth needed?
Crazy Boy wrote: Hi Friends, I am implementing Asterisk PBX in our office with 180 extensions. In our office, we will make 3 calls to USA daily. We have 1 MBPS bandwidth from ISP and 100 MBPS bandwidth in our LAN. I have two doubts. 1) How much bandwidth should we allocate for making VOIP calls? What can be the projected use of bandwidth to make International VOIP calls? It depends on codec you are using. G711 needs 64kbps for raw voice stream (add protocol overhead). 2) Can I use Pentium IV system, 2.6 GHz processor speed with 512 MB RAM as dedicated Asterisk server? Of course. It will be enough. But if you want to transcode fe. between iLBC and Speex it will consume much more CPU power. 3) Now I am making calls to USA using Voipjet.com provider. How can I receive incoming calls through VoIP? How can I get my own incoming VoIP number? Look for DID service. Looking forward for your reply. ThanksRegards, Chandra. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Marcin Kwiatkowski System Administrator Mob: +48 663 617 664 Fix: +48 33 819 04 60 ext. 32 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G723.1 and G729 on Athlon 64
Ronald Wiplinger napisa(a): I would like to install G723.1 and G729 on an Athlon 64. I looked at http://readytechnology.co.uk but I could not get a clue how to compile / get all the things for an Athlon. It seems it is only for Intel architecture, ... Has anybody a clue how to get G723.1 and G729 on an Athlon 64 to work? It isn't possible, because this patch depends on Intel's proprietary code - IPP. Another platforms performs cold restart after loading codec. On the other hand even on Intel platform inband DTMF doesn't work. -- Marcin Kwiatkowski Senior IT Specialist Telebonus Sp. z o.o. Legionow 30 43-300 Bielsko-Biala pho/fax: +48 (33) 828 25 21 mob: +48 605 923 944 ___ 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] Registering to H323 Gatekeeper as client
oi geli wrote: Can I use the Asterisk to register to a H323 Gatekeeper as client? I have the GK IP address and the user id. I am using chan-h323 (from CVS). Please share the h323.conf if you have it working. I did not see any GK user id field in the h233.conf. Thanks We are using * as SIP, H323 and MGCP translator. You have sample h323.conf as attachment. [general] port = 1720 bindaddr = 0.0.0.0 ;allow=g729 allow=all ; turns on all installed codecs ;dtmfmode=rfc2833 dtmfmode=inband gatekeeper = 10.0.0.250 AllowGKRouted = yes context=from-h323 [asteriskgw1] type=h323 prefix=12 e164=110 ___ 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] ilbc problem
Hello, I'm going to use * as SIP-H.323 proxy (codecs doesn't matter - only pass through). I compile * (v1.0.0) without any problems as far as H.323 stack (pwlib, etc). But when I'm trying execute asterisk -vvv I'm getting error message: [codec_ilbc.so]Sep 25 15:15:43 WARNING[16384]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/codec_ilbc.so: undefined symbol: sqrt Sep 25 15:15:43 WARNING[16384]: loader.c:429 load_modules: Loading module codec_ilbc.so failed! Is it possible to compile Asterisk wo. any codecs, or what's the easiest way to solve this problem? (We are using AudioCodes hardware to terminate VoIP into PSTN). Distr. - Debian Woody 3.0, libc6 2.3.2, kernel 2.4.26 -- Marcin Kwiatkowski http://www.telebonus.pl/ Telebonus Sp. z o.o. 43-300 Bielsko-Biaa ul. Legionw 30 pho.: +48 (33) 819 49 66 mob.: +48 605 923 944 ___ 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
