similar to: Issue in calling mobiles....

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 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

Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. Ciao Mauro

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 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

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some "numbers" and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country,

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 just bought the E100P from digium. It has both keys: 3.3V and 5V, so it would fit both, in a 5V-PCI slot and in a 3.3V PCI slot. Is it true, that I can plug it without destroying it in an ordenary 5V PCI slot? Roger.

Hi, I searched a while about T.38 decoding, and learned about the bounty for T.38 support for asterisk and some softdecoders and some hardware de- and encoding T.38. Now I wonder, if there is already any (almost) ready to use solution for decoding of T.38 faxes? My szenario would be: - Receiving a SIP call (containing the T.38 fax) by my provider with my asterisk box. - asterisk would

I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to

Hello, Can anyone help with the output shown below? It?s running on RH9, recent CVS of Asterisk and with one X100P card (2 channels), a budget tone 102 and Xlite softphone. CLI> -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

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 all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any

Hi, the following logs are being generated while i test sip-to-sip windows software phones. Dec 7 17:05:16 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 40dedd1535853f17250b4d0854e35c17@200.75.243.237 for seqno 102 (Critical Request) == No one is available to answer at this time Dec 7 17:05:22 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum