similar to: Load Balancing over 2 E1 Lines

Hi, all My Asterisk uses a Digium TE120Pand I would like to add an Astribank zaptel_hardware sees is, but I cannot get it working pbx:~# zaptel_hardware Argument "IRQ" isn't numeric in numeric comparison (<=>) at /usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114. usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware pci:0000:04:00.0 wcte12xp+

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

I have spent some time this morning trying to add an Astribank to our current Asterisk, but it failed, so I just removed the hardware, restore the config files to the original setup and started asterisk.; I could see that no Zap channels are started so I did load chan_zap.so: pbx*CLI> module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an

Dear all, I have small question in sip.conf I added [service] type=friend ;username= ;secret= qualify=900 host=X.X.X.X dtmfmode = rfc2833 disallow=all ;allow=g729 allow=gsm allow=alaw allow=ulaw and I can proccess incoming call from soft phone only I calling on number that is used in extensions.conf(in example below it is 1) exten => 1,1,Answer; exten => 1,2,Playback(hello-world,skip);

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all I am working in a new set up with Grandstream GXP-2000 handsets. I like those phone, but they lack a feature I need: the phone cannot be locked by the user. What I actually want is a user to be able to avoid someone else making calls from his phone without giving him access to SIP configuration access to the phone. i.e. let say I want user

I am aware of how to match a particular caller-id or a caller-id pattern and do something with the call like this: exten => 15554441212/_888NXXXXXX,n,Playback(GoAway) What I am curious about, is the best way to block unknown, private and 000-000-0000 calls. I know I can do this for 000-000-0000 calls: exten => 15554441212/0000000000,n,Playback(GoAway) Is there a better way to catch

What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's

I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones

Hi, all I want to configure a few FXS ports in an Antribank-16 to be able to receive faxes sent throught a PRI: E1 ==>Zap * ==>FXS * ==>Fax machine My asterisk box has a Digium TE120P (for the PRI). Versions are *=> 1.4.17 | Zaptel=>1.4.8 | libpri=>1.4.5 The Astribank is not configured yet, because I am a little bit confused about how to do it. Let's say I configure

Hi all I use asterisk (1.2 brach) from debian official packages and it works fine. Now I need to install and configure a Digium TE120P card, but I cannot find any guide to install it using debian packages. I would like to know if anyone of you knows about packages that would include the necessary kernel modules or any other method that won't be broken when the asterisk packages are updated.

http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

https://bugs.freedesktop.org/show_bug.cgi?id=92961 Bug ID: 92961 Summary: Xorg freezes (only mouse and ssh are still working) Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: critical Priority: high Component: Driver/nouveau

Hi all. I've ordered a TDM400P with 4 FXO, but after using my X100P I'm thinking about returning the TDM400P because of bad echo issues. If I do get the echo issues I'll look at digital options. My question: Is anyone using ISDN (BRI) in the states? I've heard ISDN4LINUX devices suffer bad echo but chan_capi works great. All the chan_capi cards I find though are for overseas

Hi everyone, I'm very new to all this, so please forgive me if I have the terminology mixed-up. We are preparing to install an Asterisk IP PBX over the weekend and I have an issue with the Polycom SP 500 phones we are trying to use. My problem is regarding DHCP. Our DHCP server is our Cisco PIX 501 firewall. I've specified option 66 and the phones connect to the FTP server

Hil list! I'm trying to authenticate Active Directory Users via freeradius. I can do it in a general case (user and domain) without problem. Now I have to do it restricting the authentication to the members of a group. I can exect the script (as is put in radiusd.conf) correct from the command line: Deb:~# /usr/bin/ntlm_auth --username=javi2 --require-membership-of='AAMM\MyGroup'

Anyway to make hitting `0` during a voice mail dial an extension? The bosses used to have that feature and love it. Their VM prompt would say: "Hello, My name is blah blah. I am currently unavailable. If you would like to speak to an operator press 0 now, otherwise leave me a message". Extension 0 exists, but dialing it during a VM prompt does nothing. Thanks, -- Daniel Jimenez

I frequently get this error message, it repeats itself hundred/thousands of times and never stops. chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again... During this period, I can make no SIP calls what-so-ever. The only way I've been able to stop it is to killall -9 asterisk. Doing a restart now doesn't respond. Anyone know why? -- Daniel Jimenez

Registration of 'native routines' (entry points in compiled code loaded into R) has been available for over 14 years, but few packages make use of it (less than 10% of those on CRAN with compiled code). Registration has similar benefits to name spaces in R code: - it ensures that the routines used by .C, .Call etc are those in your package (without needing a PACKAGE argument). - it

Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. I'll apreciate all your suggestion. Greetings, txus. The asterisk output: -- Executing Hangup("Zap/1-1", "") in new stack == Spawn