similar to: GrandStream CallerID

Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/f220d3ae/attachment.htm

Thanks to everyone for their help. I sorted out my CallerID problem - I had a stray "fromuser=101" command in my sip.conf which was overwriting any CallerID info. It was a process of elimination (on my part) helped by all the comments I had back. Regards, George

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

Why is it that newcomers always feel like inserting 'Answer' is a necessary step in their extension.conf entries? >[voiptalk.org] >;forwards any calls starting with an "8" thru voiptalk.org >exten => _8.,1,Answer >exten => _8.,3,SetCIDNum(55555555) >exten => _8.,4,SetCIDName(My Name And Surname) >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G

Is it possible to send the CallerId to IconnectHere with Asterisk when making outbound calls? I read somewhere that it doesn't work. I have set up everything to send the correct CallerId info to IconnectHere but I get a "442-887-926267" caller id. In [globals] ICONNECT1=1713...(my number) MYNAME=My Name I set up the Caller Id in the dialing macro: [macro-iconnecthere] exten =>

Hello, We have 4 'Onramp-2' Telstra ISDN BRI services operating on Asterisk Server with Eicon 4BRI card. For most part the service is okay. However, we are are having problems with passing callerID to internal extensions. This is the set of command executed. exten => <pattern>,1,Answer ; Answer the line exten => <pattern>,2,NoOp(${DNIS}) ; debug statements exten =>

Hello, I've added a ZAPHFC card to my CAPI based system. Calls coming in via ZAPHFC do not forward the caller id to the SIP phones. Calls coming in via CAPI do forward the caller id to the SIP phones. Any and all help is greatly appreciated. The (hopefully relevant) conf file excerpts are: extensions.conf =============== exten => 807440,1,Answer exten => 807440,2,Noop exten =>

Try commenting out the line pritrustusercid = yes Or set it to 'no'. That worked for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jens Sent: Friday, January 21, 2005 7:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zaphfc no callerid

Hi everyone, My manager interface seems to be producing wrong CallerIDs when internal extensions call each other. Can anyone see anything wrong in the configuration snippets pasted below? The following instance has extension 101 call 103. The phone does show the right caller ID, but notice that the manager interface has the CallerID as the target number (103). Thanks a lot for your time.

Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp

Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp

I get this message at CLI. what does it mean? - shabanip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040925/dbcd8d80/attachment.htm

hi, i'm under the impression that this feature is not available in asterisk, consider this scenario: - you are the operator. you answer a call from outside and you want to transfer it to one of the extensions. after you transfer, if the person you transferred the call to, doesn't pick up or if his line is busy, the call is transfered back to you, you can speak to the caller and tell him,

Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm currently using chan_capi-cm-0.6, with the following capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [ISDN1] msn=8304490 incomingmsn=8304490 isdnmode=msn group=1 controller=1 softdtmf=1 context=demo echosquelch=1 echocancel=yes echotail=64 callgroup=1 devices=2 Each user has a different numer, e.g. 83044910, 83044911, 83044912 and so

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 -

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

Hi Asterisk-Users, We have a problem with queues.conf / extensions.conf queues.conf file reads like ... member => SIP/8399 extensions.conf reads like ... exten => 8399, 1, SetCIDNum(${AccountNumber}|a) exten => 8399, 2, Dial(SIP/8399,10,Ttrf) When somebody calls to the queue, we observed that it is not going through extensions.conf (previous two lines) That mean's it is not