similar to: AGI script

Dear All Can you please do me favor and let me know how can I stop my Asterisk ? Can you please confirm if the following procedure is correct to stop it ? #/etc/init.d/asterisk stop #cd /etc/init.d #chmod 0000 asterisk Let me thank you in advance -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm using asterisk in real time mode...My extensions.conf table contains: [default] switch => Realtime/default at extensions I have added the following to extensions.conf table; context:micho exten: _X. priority: 1 app:Dial appdata: SIP/00XXXXXX at PSTN GAteway Asterisk server is connected succeffully to database...As soon as i make a call i got the following error message:

Dear helpers please provide me some helpful answer to my problem while I m trying to run a program .I m attaching both the program and the data to which I have to obtain my estimation results. "Motives.dat" is the data file, and "OBTfile4.3" is the complete code of program. by Running this // rawdata<-matrix(scan(inputFile, n = nsubj*ncomp), nsubj, ncomp, byrow = TRUE) \\

Dear Members i'm trying to access different packages used for Bayesian analysis, but failed to integrate after making the likelihood of the model the model like this a= exp(b)/summation(exp(b)) where 'b' = half of the natural log of 'a' please If some one knows about this type of integration for posterior distribution then pleae inform me SYED ADIL HUSSAIN MPHIL SCHOLER QAU,

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Dear all, i am using DGP 301 hard phone with my asterisk server. 1 : real time support is enabled .......all sip_buddies are stored in mysql database... 2: when i register my phone for first time it works fine.receives 2 ,3 calls then no call received.... hangup cause is congestion....i don't know why. 3: when i unregister or shutdown my dgp 301 hard phone .it still visible as

Hi Sir I have made a data set having 23 stations of rainfall. when I use the attach function to approach indevidual stations then following error occurr. *>attach(data)* *>S.Sharif #S.Sharif is the station name which has 50 data values* *Error: object "S.Sharif" not found* Now how to solve this problem. Thank You Regards -- AMINA SHAHZADI Department of Statistics GC

Some of you have been following our progress on http://farfon.convergence.com.pk as we blundered our way through the development of a low-cost ethernet IP phone that does IAX and augments the client options currently available for the kick-assterisk server. With help from the denizens of #asterisk and kind words of advice from Mr. Spencer and the rest of the gang ... we're proud to have

Hi there Plz go through the following info. Everything is working pretty fine but very frequently, I get the following error: "THERE ARE NO LOGON SERVERS CURRENTLY AVAILABLE TO SERVICE YOUR REQUEST" where these configurations work fine occasionally. [global] ;add group script = /usr/sbin/groupadd %g ;add machine script = /usr/sbin/useradd -s /bin/false -d /var/lib/nobody %u ;add user

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a

Dear All, I have the following CPU info on my asterisk server: Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux I need to install G729 on the asterisk server just to pass through and not for encoding...Which G729 package do you advice me to install? I tried several packages with no luck Regards -------------- next part -------------- An

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

Hi all, I took my decision to use Asterisk server for handling my VOIP calls...My next step is to choose the best hardware that I should use i order to have the best performance...Here I faced 2 choices for my hardware (CPU)... 1- Using Intel CPU or AMD 2- Use 32 or 64 bits Can you help me please to choose between the above choices and what is the advantage and disadvantage of each of choices

Dear Sir, Can someone help me please to find a free ebook talking about AGI scripting through asterisk? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/a59fc299/attachment.htm

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

Do you have more than one domain controller? Is this just after rebooting a PC? On 05/25/15 14:39, Andrew Bartlett wrote: > On Tue, 2015-05-19 at 14:13 +0500, Yawar Aziz Bhatti wrote: >> Hi there >> >> Plz go through the following info. Everything is working pretty fine but >> very frequently, I get the following error: >> >> "THERE ARE NO LOGON

*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean

Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other

Hello I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .....Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas