similar to: setting caller id number and using sip type=peer for incomming calles.

> > To get around this i updated CVS HEAD and changed the sip entity from > > type=user to type=peer (yes peer!) (type=friend works too but im making > > a point) the client now must register to set his outbound caller*ID > Number. > > Yes, that is normal. SIP has difficulty separating the remote party > identification from the authentication identification

Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I make a bridge call (using .call files in outgoing/) I always get 'billsec=12' in the cdr, both mysql and Master file even if the call lasted longer, watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked

> Yes, exactly (and there will be other settings as well, to identify the > type of peer (network, trunk, endpoint) for other reasons). > cool, I really should read the lists more :) > > That's coming too, but in a different way. Actually if your remote peer > can send you Remote-Party-ID headers now, you can set "trustrpid=yes" in > your peer definition

Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

Hi all, I can't seem to find any info on this anywhere on the web, except that BT caller ID doesnt use the standard bellcore system in use in the US. So, if anyone here in the UK is onlist and using the x100p successfully, please let me know. TIA, Dave

Any body used patch, http://bugs.digium.com/bug_view_page.php?bug_id=0001719 to get the callerid for BT Line. I applied the patch successfully but could not get it to work. Any help. Here are the logs: -- Starting simple switch on 'Zap/1-1' Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2 (Ring/Answered)... Jun 17 18:22:34 NOTICE[426000]: chan_zap.c:4811

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.

I've some code to manipulate incoming Caller-ID - so its suitable for replying to... [sipdef] exten => s,1,NoOp(FWD SIP: "${CALLERIDNAME}" <${CALLERIDNUM}>) ; Alter incoming calles from pulver - add a '87' exten => s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4) exten => s,3,SetCIDName(87${CALLERIDNUM}) exten => s,4,SetCIDNum(87${CALLERIDNUM}) exten

This is getting VERY annoying. Is there anyone in here that has access to the list administration to delete the user below??? >This is an automatically generated Delivery Status Notification. > >Delivery to the following recipients failed. > > asterisk-users@telappliant.com > > > >This is an automatically generated Delivery Status Notification. >

Running Asterisk Head 1.0.9. Below is a trace of a call delivered to my system which had no caller ID. For some reason, dialparties.agi shoves the incoming provider's IP address into the caller ID so you never have a call that is screened for PrivacyDirector. Is anyone else seeing this issue as well? Have I missed a patch? This call shows on the display with a name of "Unknown"

On incoming calls it seems that * is finding the callerid correctly but my BudgeTone is not showing it in the display. What am I doing wrong? The * console shows: -- Accepting call from '666666666' to '666666666' on channel 0/1, span 1 (numbers changed) but I guess that'c correct?

I have the following setup: Asterisk Server Sip Software Phones and a Wholesale connection At the moment, i can receive and send calls through the wholesale connection to and from the SIP phones. Now i want to collect the CallerID from the wholesale connection, and pass it on to the SIP phones when calls are routed through to them. I've tried a number of things including the line below,

Is it possible to reject all incoming calls that do not have a CID? Could I do something like that (modified version from the book): exten => 123,1,GotoIf($[${CALLERIDNUM} = ]?20:10) exten => 123,10,Dial(Zap/4) exten => 123,20,Playback(abandon-all-hope) exten => 123,21,Hangup( Alternatively, what's the privacy.conf file for? What does it mean for a user to have to chances to

Does anyone know if I can get Incoming caller id name and number on a sagnoma PRI.

I am aware of how to match a particular caller-id or a caller-id pattern and do something with the call like this: exten => 15554441212/_888NXXXXXX,n,Playback(GoAway) What I am curious about, is the best way to block unknown, private and 000-000-0000 calls. I know I can do this for 000-000-0000 calls: exten => 15554441212/0000000000,n,Playback(GoAway) Is there a better way to catch

I'm using QSIG between an NEC SV8300 and Asterisk (after giving up with CCIS). Things work pretty well with the exception of issues on stations on the SV8300. When I call from Asterisk to a SV8300 station and I send my extension as the caller ID number, it shows up on the SV8300 as "OPERATOR". I've tried a few different TON/NPI values with no difference. If I set callerid to a

Good Morning, I'm having an issue with callerid display when calles are placed _from_ an mgcp device (8x8 ata w/mgcp firmware). Internally, there are several different sip devices and one mgcp device. Calls from any of the sip devices to any other device (sip or mgcp) have name/number displayed properly by the called party's phone. Calls from the mgcp device to any other device display

I am not that good at regex and it's use in Asterisk. I am running Asterisk 1.4.13 Currently I have this in my extensions.conf for incoming calls on our house phone line: [housemenu] exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12); 815xxxxxxx is our home phone number, when caller id fails or is missing that is what is recorded. I want to expand this

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New