similar to: question about IF

This is not an Asterisk specific question but doesn't anyone know if you can automatically prepend a 9 on the call lists so clients can return dial without having to repunch in the number? If you go to directories now it just shows the number without a 9 (obviously). Maybe on the Asterisk side?? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

192.168.1.107 199 6bd3fb49505 00102/00000 ulaw No Tx: ACK 192.168.0.100 110 5c5a4953-65 00101/00005 ulaw Yes Rx: ACK Those channels are stuck talking to each other. The phones are disconnected yet that connection remains. I can clear w/ a restart obviously, but is there any way to tear down a call like that from the CLI? Bill -------------- next

2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great!

I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows "chan_zap.c: DTMF digit: f on Zap/2-1". At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the

I have read that 1.4 has shared line appearances, which I assume will work with Polycom phones. Has anyone configured this and verified it working? I was going to start playing around with it but wanted to see if anyone else has tackled it yet. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've allowed ulaw and g729. I want to change the codec for outbond calls. Please help not able to find anything using search. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070501/af78de7a/attachment.htm

I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to

I've been playing with Asterisk for a few weeks and it's working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html What are people using for receptionist phones that show all the extensions in

I have a CentOS box that acts as a packet filter/firewall with iptables but the box itself isn't able to reach internet : here why : Internet ----- public IP|ISP router|private IP ----- private IP + public IP/32 + public IP subnet/29|my CentOS fw|private network/dmz As you can see my provider gave us a /29 public ip subnet but behind a private IP subnet (192.168.X.X/24 - used for the

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

Hello Some of our customers call with CID blocked. I'd like to save those numbers into a SQLite database using a command-line PHP script, so that I can... 1. Edit the CID name through a PHP web script which will just list all the customers in the database who have a phone number but no CID name set 2. Look up those customers' e-mail address from this database, and send them an e-mail

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

Hello, I have an E1 connected to an * server, which takes incoming calls and verifies the existance of the called number in our internal E164 tree. Now there is a number that exists on one of the servers, but the phone has registered itself, so the dial plan executes an hangup. This hangup however is not transmitted to the E1, the calling party hears no dial tone, but also no hangup or

Hi, We just upgraded from 1.2.7 to 1.2.12.1. Everything is fine, except that asterisk seems to just crash at random. Often I can make it crash by using the ChanSpy function (which we use to monitor agents). Sometimes it will just crash on its own. The reason we were initially running 1.2.7 was because of the stability it gave us (weeks without a restart). We upgraded to 1.2.12.1 because it

Hello, thank you very much for your previous answers about the C++ code. I am interested in the application of the Gibbs Sampler in the IRT models, so in the function MCMCirt1d and MCMCirtkd. I've found the C++ source codes, as you suggested, but I cannot find anything about the Gibbs Sampler. All the files are for the Metropolis algorithm. Maybe I am not able to read them very well, by the

Hi Folks, Compare print(1234567890,digits=4) # [1] 1.235e+09 print(1234567890,digits=5) # [1] 1234567890 Granted that digits: a non-null value for 'digits' specifies the minimum number of significant digits to be printed in values. how does R decide to switch from the "1.235e+09" (rounded to 4 digits, i.e. the minumum, in "e" notation) to

Relatively new to Asterisk and SIP and am trying to run a proof of concept using Asterisk to make an outbound call through an Audiocodes gateway via SIP using Asterisk version 1.6.1.12. The specific requirements of the gateway in the configuration I am trying to use specify that the Name part of the From header be blank with the outbound number that needs to be dialed in the number field of

Hi, Since we cannot join a W10 machine to NT3.x domain anymore, it is time to move on. We have a decade-old domain 'A1CWB' and will profit from the situation fixing the old S-1-5-21-1234567890-1234567890-1234567890 SID and implementing a new domain name: Old domain: A1CWB, SID S-1-5-21-1234567890-1234567890-1234567890 New domain: AD.A1.IND.BR, decent SID from net getdomainsid, two

Hi all, I install the last version of Asterisk and I tried to send faxes, but nothing works. Here is my configuration: Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA <----> Analog Fax 2 I tried Analog Fax 2 -> Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have