similar to: Follow-Me Script

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Greetings, I have implemented the following command to allow CNAM to be delivered to my users. exten => 9969,1,Wait(1) This works great! However it has spawned a new problem. When this is implemented into a full dial plan. If a user is set to DND or sends a call to Voicemail by hitting deny the caller gets a busy. Below is a result of the calls. With the Wait(1) statement -- Executing

Hello All, I have Cisco 7960's, Cisco 2950 Switch. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone is disconnects at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone seen this? Thanks for the help,

Quick update on my issues, Voicemail doesn't pickup also. It just drops the line.. Thank you Chris Tuska ------------------------------ Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the

hello all, i have already configure sip.conf and dialplan. i done the follow me script. first problem: i want to call(with kphone) someone at my extension, i must dial the extension number. i can't dial their username. 20531603@192.168.8.125 (work) mustafa@192.168.8.125 (call fail) is it possible to do that?? second problem: if i want to call another number (not my

Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone it then disconnects the call at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone

Hello People, I was wondering if you could take a look at this script, exten => 505,1,dial(iax2/6311${EXTEN},t,25) exten => 505,2,playback(pls-wait-connect-call) exten => 505,3,set(NewCaller=${CALLERID(num)}) exten => 505,4,Set(CALLERID(num)=0${CALLERID(num)}) exten => 505,5,dial(Zap/g1/c/0296389675,20,r) exten => 505,6,Set(CALLERID(num)=${NewCaller}) exten

We have a small number of phones, when a call comes in we want all the phones that aren't in use to ring. Is there a simple way to test and see what phones are in use then ring the other phones? I tried some code like this: [zap] exten => s,1,Answer exten => s,2,ChanIsAvail(${DERRICK}) exten => s,3,SetVar,"EVERYONE=${DERRICK}" exten => s,4,ChanIsAvail(${DON}) exten

Hi, I'm trying to replace function call with call to wrapper(function_name, num_args, ...), where varargs hold args of original call. Function* launch = Function::Create( TypeBuilder<int(const char*, int, ...), false>::get(context), GlobalValue::ExternalLinkage, "kernelgen_launch_", m2); { CallInst* call = dyn_cast<CallInst>(cast<Value>(I)); if

We have just posted a review of the Sipura SPA-3000 with a complete setup guide for using the device as a trunk to Asterisk. It is easy to setup and works as good as any Zaptel setup and includes an ATA. It really is one cool device! http://www.geekgazette.com/index.php?option=com_content <http://www.geekgazette.com/index.php?option=com_content&task=view&id=28>

Hi Dimitry, This makes sense if you think about it from the perspective that the string you want passing must be passed at runtime, and so can't use a const char * from compile time. You need to make the string visible in the compiled image, and use that as the argument. A string is an array of 8-bit integers, so you need to create a ConstantArray. Value *v = ConstantArray::get(Context,

Dear useRs, Some time ago I queried the list as to an efficient way of building a function which acts as ls() but with a different default for all.names: http://tolstoy.newcastle.edu.au/R/e6/help/09/03/7588.html I have struck upon a solution which so far has performed admirably. In particular, it uses ls() and not its explicit source code, so only has a dependency on its name and the name of

Hi, I'm new in asterisk world. I have questions. For example I have my server with public IP address, but two customer with softphone in a private network. How can I do to make them work with the asterisk server? Best Regards -- Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm interested in developing an automated phone survey and am curious if Asterisk could be configured to run such a system.. My idea is to record a message and a series of sub-questions. The system would call each number on a list and play the message, Depending on the touch tone response another message would be played. Is it possible for asterisk to manage a survey like this?

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,

Hello, I'm trying to receive a fax to my inbound number from voipbuster. Asterisk receives the call and starts the rxfax application successful, but then nothing happens. The calling party is still hearing a ringing tone, or sometimes nothing. Voicecalls are working correct and without problems. For testing I've add a local number (300) to the dialplan. When I call this number

Hi, as noone out there seems to be able to maintain chan_sccp, i'm trying to switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly wrong/non functional. I see Redial NewCall CFwdAll more (more) CFwdBu... GPickUp Confrn more NewCall works, CFwdAll seems to toggle DnD, the rest of the buttons do notting. Any ideas how to fix this? Regards, Andreas

Dear list, following problem, i have some users, who are supressing their callerid. This setting is adjusted at the sip phone. So if these guys are calling internal persons nobody sees the callerid. I am looking for the following resolution: User has set his phone to anonymous, user calls somebody internal, Asterisk initials a lookup on the channel and generates a new callerid for the

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>

Anyone written an extension that will take a 12 digit number, convert it to an IP address so that you can make a sip call to it. Chris