similar to: Removing "-" (Dash) from Dialed Numbers

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

Hello, Anyone knows how can I do this in a cleaner way? mynumber = 1001 as.numeric(unlist(strsplit(as.character(mynumber),""))) [1] 1 0 0 1 Thanks in advance, Gustavo

On Jul 15, 2015, at 12:51 PM, William Dunlap wrote: > I think rapply() was changed to act like lapply() in this respect. > When I looked at the source of the difference, it was that typeof() returned 'language' in 3.2.1, while it returned 'list' in the earlier version of R. The first check in rapply's code in both version was: if (typeof(object) != "list")

In 3.1.2 eval does not store the result of the bquote-generated call in the given environment. Interestingly, in 3.2.1 eval does store the result of the bquote-generated call in the given environment. In other words if I run the given example with eval rather than evalq, on 3.1.2 "x" is never stored in "fenv," but it is when I run the same code on 3.2.1. However, the given

David, If you are referring to the solution that would be: rapply(list(test), eval, envir = fenv) I thought I explained in the question that the above code does not work. It does not throw an error, but the behavior is no different (at least in the output or result). Using the above code still results in the x object not being stored in fenv on 3.1.2. Dayne On Wed, Jul 15, 2015 at 4:40 PM,

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Are any versions of the Cisco 7940/7960 or 7940G/7960G phones compatible with the 802.4af Power over Ethernet Standard? Ok, I know the question has been asked before, but googling has turned up several contradictory results: 1/ No - not at all 2/ Maybe - 79XXG will work 3/ With a special cable/dongle (a la wikki) I am looking at getting

Dear all, May I first introduce myself. I'm a student of HAW Hamburg University currently working for my professor on a VOIP project. We have a Debian Linux system (server) on which Asterisk 1.2.6 has been successfully installed and running. Also the asterisk SIP server has been connected to the PSTN so users could make calls externally. We use Xlite softphone to make calls between users in

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

Okay, I've read as much as I can, and I think i've followed instructions, but I'm still having problems with * and firefly... I can get outgoing to other freshtel working, but not incoming (I get the "not available" voicemail), or outgoing to landline. I'm using the debian asterisk package (0.9.1-RC1-4) My iax.conf has in general (under my FWD register, which

Hi, all. I am building a SER+asterisk PBX airming at around 10k persons' usage. For authentication purpose I am in favor of ldap storage, while I am not sure the current ldap module for asterisk(0.9.9.2) is stable enough? sorry I do not master the proper testing mechanisms to find out myself. Thanks in advance.

Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI> Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '703XXXXXXX@147.135.8.129' timed out, trying again -- Got SIP response 404 "Not found"

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -

This puzzles me. If I start asterisk in the background, and then attach to it to perform some chores, is there a way to detach again without stopping the background process? Entering "stop now" kills both the console attachment as well as the background process. I need to attach to the running asterisk in order to do "init keys" but once I do that, it seems I cannot just

My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk

Hi, I use * release 1.0.9 with differents phones and softphone, i've got a problem with my Cisco 7940G (last SIP Firmware). Sometimes, when i but a call on hold, the caller has got the music, but when i "resume" the call, then the caller does not hear me (and nothing at all)... I must wait for 10, 20, sometimes 60 seconds before he could hear me again. Any body already had

Anyone have screen size + resolution for the cisco 7940g, 7960g? I'm having a hard time finding it anywhere. No vendors have the information and cisco doesn't list it. All I've been able to find is the rez of the 7940g : 100x145. -Dan

is there a way to support login and logout functionality in a phone? We are using Cisco 7940 and 7960 phones and have 2 shift. We want to be able to use the same phone using like 2 different extensions. The phone will then "remember" your settings if possible, if anyone has left you a voice mail etc. Is this possible? Regards, Paul -------------- next part -------------- An HTML

http://www.pcworld.com/article/id,143198-pg,1/article.html anyone know anything about it? Regards, Dean Collins Cognation Pty Ltd dean at cognation.net +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080311/e214b4d0/attachment.htm