similar to: Re:TE110P EuroISDN dial out timing out

Hi dear group members, I have finally an Asterisk box working, capable of receiving and making calls. I have this issue while calling mobiles from our SIP softphones: -------------------------- linux*CLI> -- Executing NoOp("SIP/2000-6850", "3487024125") in new stack -- Executing Dial("SIP/2000-6850", "ZAP/g1/3487024125") in new stack -- Called

Hi everybody, I have corrected this line in extensions.conf by stripping spaces off and now it executes: exten => 22999,1,VoiceMailMain(s${CALLERIDNUM}) when it runs, the mail box number is asked and password too. I expected no question were made, because I inserted CALLERIDNUMBER and s in front of box number. Anybody knows why? Thank to you all, very kind members of this list! Ciao Mauro

Hi everybody, I'm in a middle of a Asterisk learning period. I am at a very good point except I'm not able to use VoiceMailMain. This Is my simple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display

Hi everybody I'm trying to make Asterisk 1.2.1 run under VMWARE and Suse 9.2. I use ZTDUMMY module for timing and ZTTEST gets an average precision of 98,4 %. Is there any way to improve it? Best regards Mauro Zanin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060117/945d544a/attachment.htm

Hi, Been asking google and browsing the lists but haven't found any answers for this. I've connected a TE110P E1 using EuroISDN to a PBX (for me at the time unknown model). All is fine _except_ when placing calls to mobile phones (which takes too long, more than 2 seconds it seems) asterisk seem to be impatient giving up saying the curciet beeing busy. So, what I'm looking for, is

Hi everybody, I'm strugling between two devices: the both TE but one was set up as a TN. I have no current on that interface. I have tried to find some circuit over the net to power the connection, both commercial and home made. Can anybody give some hint? Ciao Mauro

Hi everybody, I have this issue: I need to automatically park an incoming call, play a welcome prompt and then connect to some extension but under extension user's command. I was thinking to use a small database to comunicate between asterisk and the main application. Has anybody had this kind of experience? Best regards Mauro

Hi Everybody, I'm experiencing this kind of issue. One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel card. Everything seems to work but sometimes the third party caller when listening to MOH listens some "SSHHHHH!" instead of MOH, this is not continuos, MOH plays ok for, say, 20 seconds then the sound and then another 30 seconds of good MOH. We have some

Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voil? the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: "Chi lascia la via vecchia per la nuova, sa

Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP!

Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate("unknown", "unknown") does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa'

Hi everybody, I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm using Trixbox..). I must be as fast as a flash to press *2 and do an attended transfer. If I wait only a tenth of a second nothing happens. I think it is an issue. I have seen the source code and found nothing bad. Is this a known issue? Many thanks Best regards Mauro

Hi, I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP). The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss") which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1". Everything works fine except that I can not see the called number/MSN of incoming calls within Asterisk and because of this I can not route incoming calls

Hello out there, I'm running an Asterisk 1.8.15-cert1 with DAHDI. Today I noticed that Asterisk is signalling to the calling party the current internal CallerID whenever I put a call to another internal phone. Example: Customer calls 020212345-555 -> IVR answers and puts caller to the chosen queue -> Someone picks up the phone (Internal ext. 321) -> CallerID shown on customers

Hi, I successfully installed 2 avm card in my asterisk box but I'm unable to make call. My capi.conf is: msn=0721111,07211115 incomingmsn=* controller=1,2 softdtmf=1 context=default echocancel=yes callgroup=1 devices=2,2 my capi info : Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. my extensions.conf : exten =>

Hi List I have had a look at the various posts on this and seem to be more confused than ever - but then again that's not hard ;-) I am using Version 1.4.33.1 build from the Debian "lenny" distros I am trying to implement a simple screening [macro-screen] exten => s,1,Background(press1) exten => s,n,WaitExten(5) exten => 1,1,NoOp(accepted) ; Dont set a reply so dial

Hello List, is someone operating a DID /P2P / Anlagenanschluss with more than one HFC-Based ISDN-Card ??? I have now 12 hours of setup-troubles behind me with Colt-Telekom, where we did not get it working with two HFC-based cards. Here the setup: - 2 HFC-ISDN-Cards (the one from Conrad-Electronic) - bri-stuff.0.0.2 (with the asterisk-sources from the download.sh-skript) - two NTBAs from

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered

Hi to all, I'm trying to make outside call in this way : ignorepat => 0 exten => _0.,1,Dial(CAPI/xxxxxxx:b${exten}) But the first number 0 is not ignored. I'm doing something wrong ? Bye

Hi, I'm trying to configure a natted budgetone phone to a asterisk server as described in wiki using port forwarding. I successfully make call from the client but it seems it does not register the client ip address and when I try to recall it is not reacheable. Asterisk can manage natted sip client with dynamic ip address ? Bye -------------- next part -------------- An HTML attachment