similar to: callerid

Is there a way to display the time of the 7960 running firmware 7.4? Im unable to find any information. Thanks, Ben Blakely -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/a7e575cc/attachment.htm

I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05. Whenever a call comes in with blank CIDName or CIDNum the phone reports the respective variable as "asterisk". I can manually set the variables to whatever I want: CIDName (alpha-numeric) & CIDNum (Numeric). But if I try to make them blank, or null, or maybe throw some alpha characters into CIDNum, they get reported

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I don't see the caller's number in the

How do I setup my T1 card as an E&M tie line? Any special configuration? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/8aac4dbf/attachment.htm

I am trying to receive both CID and DNIS from the telco through a non-pri T1. Currently I have the T1 setup and operational both outbound and inbound calls are completed as should be expected. The calls came in and were placed in the context specified in zapata.conf on exten => s,1. I have requested that the telco provide callerid (they call it ANI) along with 10 digit dnis for my 800

I currrently have Asterisk 1.2.8 with a TE110P Zaptel card tied to our Nortal DMS-10 switch via T1. This T1 trunk is configured for Feature Group D MF. The purpose of this is for LD delivery to a VoIP LD provider as a LD choice for our customers off of the DMS-10. When I sieze a channel on the zaptel, I get a spill of digits from the switch that is the called number, and I can see that in

I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. Prior to the resets I am getting similar notices to the following chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Telco claims the PRI's are fine on their end and that it is my unit. Is this timing? (google somewhat leads to this) I am running 1.08 asterisk zaptel

I went to put in the new sound files over the weekend, but forgot to backup the custom folder and lost my custom digital receptionist files. I then had to copy the old files back from a duplicate machine. The problem is now though that voicemail just hangs up when I dial it. Other apps work - *70 for example gives me 'call waiting...activated' so I know it's accessing the files

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, I am attempting to connect to the blacklist database using PHPAgi and it always seems to hang. The code snippet I am trying is: $r = $agi->get_variable("CALLERID(num)"); $cidnum = $r["data"]; if ($cidnum < 1000000000) # No valid callerid. { exit(0); } $r = $agi->database_get("blacklist", "$cidnum"); if ($r["result"] ==

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

hi all! I'm running an Asterisk-box with bristuff-....RC8n and 2 HFC-S cards. I m located in Vienna/Austria. I have the problem that on outgoing calls i hear my voice as a short echo (about half a second). This occurs not on every call. I tried some changes in my zapata.conf, with rxgain and txgain settings, but to me its hard to find a configuration which is good for every call i make. Is

Is it possible to control the dialing format used when making a call from asterisk separately from the inbound format? In my case I have a single robbed-bit T1 configured for featd (to receive ANI and DNIS) on a TE410P. This configuration works fine for inbound calls, however when an outbound call is made on this same T1 asterisk dials outbound in the featd format. I need asterisk to only dial

Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI> -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1

All; I am running Asterisk 11.6-cert16 and I have voicemail setup so voicemail messages are sent as email attachments. That works fine. However, the body of the email contains the CallerID(name), but is missing the CallerID(num). For example, the email body looks like this: Just wanted to let you know you were just left a 0:21 long message (number 13) in mailbox 101 from WIRELESS

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

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I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals? Regards L:ee ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ -------------- next part -------------- An HTML attachment was scrubbed...

I am trying to block/hide outgoing caller id on a PRI E1. It seems that it should be fairly simple, but it is defeating me. https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says: "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)" That doesn't seem to do it. The calls are originated from AMI and I have tried a blank "CallerId:" line and