similar to: Music on hold

Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call "off hold" And then the external line is disconnected. an exact log of events is

Hello When using LookupCIDName, Asterisk 1.4 says that it's deprecated, and we should use "${DB(cidname/${CALLERID(num)})}" instead, but I don't know how to use it: ;DEPRECATED exten => s,1,LookupCIDName ;ERROR exten => s,1,${DB(cidname/${CALLERID(num)})} I guess I should use this as a parameter to a function, but which one? Thank you.

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

Hi list! I'm trying to get PrivacyManager working but for some reason it always thinks that CallerID is present (when it isn't). I get this on the console: == Primary D-Channel on span 1 up -- Accepting voice call from '' to '0711234567' on channel 0/2, span 1 -- Executing Ringing("Zap/2-1", "") in new stack -- Executing

<font size="2">I've got an odd situation where callerid is only picked up every other call.&nbsp; Is there anything I can do so callerid works on all calls?<br><br>I'm seeing the channel hangs up during a call<br><br>&nbsp; == Starting post polarity CID detection on channel 4<br>&nbsp;&nbsp;&nbsp; -- Starting simple switch

I'm trying to use LookupCIDName to tag outgoing calls on my CDRs but it seems that application only tags incoming calls? Any sugestions? Leandro

Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090708/cccd4587/attachment.htm

Hello, I have a rather odd problem with Asterisk detecting faxes. I have two POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2 is fof fax. When I set them up with channel => 1-2 in zapata.conf, all is fine, but as soon as I have two channel => definitions, Asterisk is unable to detect faxes. The fax line is not supposed to ring local phones, so the most obvious

Hi All, I'm about to modify my menu to allow for certain callers to bypass my menu entirely, and just ring my phone. I don't need friends hearing my business menu all the time. So, my idea was to use my already enabled callerid asterisk database and lookup the callerid info from it, if it exists consider it a friend and send them to a direct dial context. If the callerid is new, not

1. It would be very helpful if the voicemail module included the caller-ID info (name and number) in the "Subject" line. Otherwise, looking at the list of messages is like looking at an INBOX without a "From" column. I patched app_voicemail.c quickly, but it would be even better if this were a configuration option (since some installations don't have caller-ID

Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. I've found quite a few examples of the reverse of this. To where a script is

I am trying out a new service from www.talkn.com. They use Sipura to terminate your service like most providers. They are looking at directly connecting into asterisk in the future. Right now my configuration is Talkn?Sipura?Asterisk/FXO Card. When someone calls in and get?s answered and when either party hangs up , * releases the port with out a problem. If someone calls in and * answers the

I'm having a problem with intermittent lookup of Caller ID Name info using LookupCIDName. The same problem occurs when doing: asterisk -rx "database show cidname" No data is returned on every fourth or fifth query. No errors are being logged. I'm currently running CVS-HEAD-07/07/04-17:04:31 and first noticed the problem a few weeks ago. Is anyone seeing a similar problem?

I have found that I can make the phones display any one word on this second line by adding a fromuser=<word> in sip.conf. This really isn't good enough though. When you look at the received calls or missed calls directory, each item has two lines, the first is the CID name, and the 2nd is supposed to be the CID number. However, if it is asterisk, or some other word, when you hit the

Release 0.68 of IPSwitchBoard is ready for download at: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA Release Notes: Import/Export extensions to "cidname" in Asterisk Server, integrate this in your dial plan to see who's calling by using the Asterisk app. "LookUpCidName()." Set "Do Not Disturb" for an Extension, and type a reason for DND

I'm using Asterisk 1.09 with bristuff 0.2.0-RC8n, one BRI line and a Sipura SPA-2000. I have a HP LaserJet 3330mfp all-in-one. I can receive faxes but not send them. The faxes start whistling to each other but the transmission is stopped with a "communication error" To receive a fax I have this in my dialplan: exten => 0000000000,1,Ringing exten => 0000000000,2,zapEC(off)

Hi Is it possible with Asterisk to tell the called party which number was dialled by the caller? Or in place of the number dialled have a description such as 'Sales' or 'Accounts'? Ideally, I would like to show a description corresponding to the number dialled followed by CIDName. How might this be set up? Currently my extensions.conf is: exten => xx,1,LookupCIDName

On one installation (* 1.2.13) Pickup doesn't work. This is what I have in extensions.conf exten => _**2X,1,Pickup(${EXTEN:2}&8${EXTEN:3}&t&uevents) exten => _**2X,n,Hangup This is what I get on CLI -- Executing NoOp("mISDN/3-1", "incoming-beronet 80 - dolazni poziv s broja 270248") in new stack -- Executing LookupCIDName("mISDN/3-1",

I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party