similar to: How to quickly replace ',' with '|' in dialplans?

Anyone had Realtime working with IAX2 and RSA authentication to connect two PBXs, please? It seems that inkeys/outkey fields are not read at all and the following warning is logged when dialing: Jun 2 02:41:36 WARNING[6299] chan_iax2.c: I don't know how to authenticate ******** to XXX.XXX.XXX.XXX Using iax.conf it perfectly works. Maybe a bug in Realtime? TIA, Alex

Hello, I try to compile the driver zaptel and they give the following error: linux01:/usr/src/zaptel# make install gcc -Iir/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -Iir/drivers/ l -I. -Wstrict-prototypes -fomit-frame-pointer -Iir/drivers/net/wan -Iir /net -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from zaptel.c:44: /usr/include/linux/module.h:21:

Hi all! So far I've always used plaintext passwords for SIP, but now I've decided to use MD5 encryption. For each client I edited its section as follows, then: auth=md5 md5secret=hashed_passwd ;secret=plaintext_passwd where hashed_passwd is the output of echo -n "user:realm:plaintext_passwd" | md5sum When the first SIP clients registers with Asterisk after a "sip

Dear All, I'm syncronizing a Win$$2k smb share to linux box (centos4). Rsync ver is 2.6.3 in my i18n file you can find: SYSFONT="latarcyrheb-sun16" LANG="it_IT" SUPPORTED="it_IT@euro:it_IT:it:en_US:en" when I try to make the syncronization I've go the followin error on file with special char Eg.: file has vanished:

Hello, they are successful to start asterisk, task that the error that I had previously had had to a configuration problem. Start asterisk in modality consol and when two softphone speaks is not felt well, and I have the following error: -- Registered SIP '1000' at 10.0.0.7 port 5060 expires 1800 -- Saved useragent "X-Lite release 1103m" for peer 1000 --

Dear all, Is it possible to print a dataframe without the row numbers? For example if I have a dataframe like that: >df <- data.frame(name1=sample(LETTERS,10),name2=sample(c(0,1),10,replace=TRUE)) after printing name1 name2 1 O 1 2 H 0 3 R 0 4 T 0 5 V 1 6 E 0 7 W 0 8 P 1 9 G 0 10 J 1

Dear all, I want to solve a score function by using Newton-Raphson algorithm. Is there such a fucntion in R? I know there's one called optim, but it seems only doing minimizing or maximizing. Thanks, Jimmy

Hello! I've a problem with the combn function and a set of vector. I would like to make a simple combination where, instead of scalars, i would like to combine vector, in order to form matrixes. In other words, i have nineteen 6-items vectors (for example coef1-coef19), that i would like to combine in n!/k!(n-k)! 6x6 matrixes. I tried with a code like this mma <-

Hi, May I ask what does exten => s,1,Answer exten => s,2,ResponseTimeout(5) exten => i,1,Playback(pbx-invalid) s, t, i stands for? It says it is someexten but I still don't get it. Regards, Norman Zhang

Hi, is there someone who speaks in Italian? I'll try to explain in english my problem, but if there is someone who speaks italian i think it would be better for me. I'd like to use asterisk only as IVR and call diverting. I have only one phone line, and no other phones, all the calls arrive at one number. I would like something that answare, and depending from the 'street'

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

Hi, I have 2 phones AASTRA 57i with Asterisk 1.6. When the internet connection for some reason fall down the 2 phones go to "NO SERVICE", searching on internet i found that this is due to DNS service. Has someone solve this problem? or suggestions? Thanks in advance Antonio Supera i limiti: raddoppia la velocit? da 10 a 20 Mega! Risparmia con Tutto Incluso: telefono + adsl 20

Goodmorning everybody, I need to run an R program via SSH. Usually I open R, I run the program and I stay logged-in, waiting for the output. As a matter of fact, if I close the connection with SSH I loose the calculations and the output of my R program. What command I have to use in order to preseve the calculations and the output without staying logged-in a SSH connection? thanks in advance

Hi, I'd like to use PSTN and analogic telephone with a Asterisk server which works on a LAN connected to the outside with fiber optic, on which voice stream passes too. Do I need some particular solution, some particular card to make it works or I just need a Digium or similar fxo card? Thanks, Bob __________________________________________________________________ Tiscali ADSL Senza Canone,

hi, i've a question. is it possible to buy digium x100p card from italy in some store (also online) without ordering it from USA? on more, did anyone buy a modem with intel chipset 537 or md3200 and where (in italy)? Thanks __________________________________________________________________ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27