similar to: Readline & readline-devel installation on RH9

Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case

I have 2 Snom 200 and would like to get them to work properly with Asterisk. With the Firmware 2.02t I am able to use the phone. But only one line configured. With there newer firmware 2.03o it will not allow me to make calls. But I can get calls on the unit. Again the 2nd line is not able to be registered. Is this an issue with Asterisk or Snom? I could use some example configuration

Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options

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I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten =>

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

I just got 2 IpDialog phones for use with my Asterisk system. I have been able to get the phones to just dial local extensions but it is not able to register with my system correctly. I would like to know if someone has set these phones up before and how they did it! Is there any examples for use with Asterisk? They seem simple enough to config with there web interface. Thanks

Make sure you have libtermcap and libtermcap-devel loaded.. To check use #rpm -q libtermcap and #rpm -q libtermcap-devel I have an install guide for RH9 in PDF format if you are interested.. Later.. > Hello, I am trying to install asterisk and get this error. > > creating cache ./config.cache > checking for gcc... gcc > checking whether the C compiler (gcc ) works... yes

Hello, Great job on the Asterisk@home project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not

---------- Original Message ---------------------------------- From: Louis-David Mitterrand <vindex@apartia.org> Reply-To: asterisk-users@lists.digium.com Date: Mon, 22 Sep 2003 22:28:40 +0200 >On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote: >> My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of >> service. They were deployed for about 6

This message appeared in my inbox over the weekend. A useful idea although I have not reviewed the code. Might someone on the list (or the original author) be interested in running with it? Monty > Hello - > I hacked up an ogg winamp output plugin based on the vorbis > example_encoder and the out_raw winamp sdk sample. It requires the same > deal as the winamp input plugin

I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? Ariel Batista Kasi International - Computer Networking Ph: 305-574-6721 Fx: 305-574-0212 -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi. I got these compile errors while install asterisk. readline and openssl are compiled using gnu source, and kernel version is 2.4.17. Compile errors message is follows. Someone cleared this problem? Please, help! Regards. ---------------------------------------------------------------------------------- gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config. o channel.o

Hello, I hope this is the right place to post this. I added readline support to the sftp client. It adds optional --with-readline parameter for configure to enable this feature. You'll of course need to re-run autoheader and autoconf after applying this patch. It's patched against 3.4p1 but should work with any recent openssh. It works fine for me (GNU/Linux) - feel free to test it and

I would like to know if anyone has come up with a script for 911 dialing rules that put correct information on our locations. We have our office in 3 different building one being our production & shipping dock. It is almost 2 blocks away. We are connected with Ethernet Wireless between the buildings and have Sip phones setup in the other 2 locations. All the phones are working just fine.

hi, i am getting these errors while installing asterisk. i reconfigured kernel and i have all the modules installed. kernel-source readline readline-devel openssl openssl-devel this is the error: (at the last part of the installation) gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o

>Message: 11 >Date: Sun, 28 Aug 2005 11:46:29 +0800 >From: "chris" <chris@fivestartel.com> >Subject: [Asterisk-Users] error compiling on solaris 10 >To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> >Message-ID: <003a01c5ab83$149a9e30$650fa8c0@acid> >Content-Type: text/plain;

I tried to connect my * to IAXtel, but i always get this errors. chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected: Registration Refused On dial a iax number i get: chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority found chan_iax2.c:5528 socket_read: Immediately destroying 3, having received reject chan_iax2.c:2411 iax2_hangup: We're hanging