similar to: dial (iax/X&sip/y) get y fraction earlier

Hello Does anybody who have experience in installing the h323 modules in asterisk. I try'd it many times an spend xxx hours to install it but didn't get lucky so far. I have asterisk 1.0.1 running with bristuffed 2.0 I'm willing to pay for this. Sjaak

Hello * Users Did somebody get managed to get AreskiCC work under mysql. If so is there anywhere to find the database structure for mysql. Thanks Sjaak

Hello I like to make calls to an h.323 device. I'm using Nuphone h323. Compiled everything okay "I Guess" When I make a connection * SIP > h323 device, the phone is ringing and then * tells me "No one available....." and disconnect Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing. I try many combination in the h323.conf like.

Hello everyone I'm using rsync over ssh rsync -ave ssh bla@bla.tld:/home /backup/server1 This works great for many years now Now I'm playing with Nagios and NSCA but how can I detect if rsync has done everything well. nsca works simple <hostname>[tab]<svc_description>[tab]<return_code>[tab]<plugin_output>[newline]. I have a text file named backup_okay with

Hello * users I've problems with sound quality on zaphfc Asterisk works fine good sound quality. If I do "make load" in the bristuf.xx zaphfc dir then sound quality drops directly. Even if I don't load the chan_zap in the modules.conf I use this config on more (even old 400Mhz machines) and works correctly. Looks like an hardware problem but I can't find it. I don't

Hello This Question maybe ask many times. I'm From the Netherlands and looking for an sip/aix > pstn provider in the US.I asked wipphone, vonage but they don't work for europe. I've visit many not finished website's about sip > pstn Can somebody recomed me any good sip/aix provider in the us. Thanks

Hi I am having a problem with extension matching in asterisk (I am using asterisk 1.6.0.6). Is there a difference between extensions matching in realtime architecture and extensions matching in extensions.conf file. For example when I have these two extensions: -- _0699[134]XXXXX -- _06[069]XXXXXXX that are in the database and number 0699123123 comes in asterisk will always choose exten

Hello I buyed a new server 2*XEON in a 2inch High 19"case. Now I have a problem that the riser card is 64bit so an ISDN PCI modem isn't possible. My question is can I use ISDN on com port /dev/ttyS0. If yes can I use it like the example in modem.conf as /dev/ttyI0 but use /dev/ttyS0 Does anybody have expirience with this ? Thanks Sjaak -- Dit bericht is gescand op virussen en

Hello Would it be possible to dail out to lett's say to 4 people with a .call file and put them directly into a free meetme room. Thanks Sjaak

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

Hi list! I am installing an * box that will be installed on a site with KPN BRI ISDN in The Netherlands. I am using bristuff fron Junghanns. Does anybody know the correct settings for this? I will not have internet access there which makes it harder to google around on location. switchtype = euroisdn is pretty obvious but what about these settings: signalling = bri_cpe_ptmp ; p2p TE mode

Is it possible to configure * so that if a caller reaches voicemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

Wondering, since they appear to be plentiful on eBay, whether I could get a Nortel 350 to use to learn my way around ADSI. The vendor claims that these are "generic," and looking through the archives I wonder if that means that they might be unlocked in the sense that the word is meaningful to asterisk. Of course I am green as could be on this topic, so this question may even be a

Hi all, I am having a problem on to different boxes and to different channel banks. I can't get dial tone out of either one. I can still send and receive calls but no DT. This is the error that I get: Jun 8 15:23:54 WARNING[5021]: chan_zap.c:6283 handle_init_event: Unable to play dialtone on channel 95 asterisk-1.2.9.1 zaptel-1.2.6 zapata.conf: signalling=fxo_ls context=gene

Thanks anyone, I found the problem in rhconfig.h. After the fix I successfully compiled zaptel. V. --- Ron Wellsted <ron@wellsted.org.uk> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Have you built your kernel on that machine? > > The errors suggest that while the kernel sources are > installed, the > kernel has not been built. > > Check on

When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507)