similar to: E1 signalling pridialplan

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

Hi all, I have a problem setting the MSN for outgoing calls. I'm using a HFC PCI card together with zaptel and bristuff. All my outgoing calls are using the same (first|default|main) MSN. In my zapata.conf I tried different values for pridialplan, prilocaldialplan, nationalprefix, etc but without any success. In my extension.conf I'm setting the MSN with: exten =>

Hi list! What exactly is the meaning / function of the pridialplan & prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks!

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Can someone explain what the parameters pridialplan and prilocaldialplan are? What do they do and do I need them? I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx. The pbx technican complains about the format of the nr asterisk sends. Asterisk sends all numbers in on piece the pbx expects the numbers devided into

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 -

Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva

Hi all, I've been batting my head against a brick wall for the best part of the day and still haven't got any further (apart from getting a big headache, that is). I've searched the Wiki and Googled the hours away, but I still can't find supportive documentation. I've just replaced my ISDN Fritz!/chan_capi setup with a HFC/Zap configuration and had the following problems: 1

Possible Mediatrix 1204 fxo sip gateway workaround Need some feedback from experienced * users relative to this workaround please please please. Problem: The mediatrix 4-port fxo gateway does not provide any mechanism for * to select which "port" an outbound pstn call will use. (See lots of previous posts over the past four days for more detail if needed.) Our reseller has been

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

Hi, Is it possible to set the pridialplan (or prilocaldialplan) on a call-by-call basis, or for a particular PRI channel-group? >From my experimentation so far, it appears to me that the pridialplan and prilocaldialplan are global in zapata.conf, unlike other options such as switchtype, signalling, and context. I currently have two PRIs from different providers (and different

I tried to get callerid working the normal way but the cid is never passed to the phone. It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in extensions.conf which I found in the wiki: http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc Is this intended behaviour, or still a bug? It does work but it only shows one zero even though I have nationalprefix = 0

Hello, I have an Asterisk system in the UK using ISDN service from BT. My problem is that the called number is always passed to the provider with the Type Of Number declared as ?national? despite pridialplan (and prilocaldialplan) is set to ?unknown?. My questions are: - How can I find out what the effective value is for the above two variables without actually making a call? (I have very

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

For whom which may be interested: Here in Italy we have GSM #numbers without leading zero PSTN instead has prefix starting with '0' to have '0' recognized by * i need to insert nationalprefix=0 as Jason Williams suggested me in irc; now, you cannot have: pridialplan=natonal otherwise * will not be able to call GSM phones you need to setup: pridialplan=local

After setting the pridialplan=unknown I seeing the Called Number TON change to Unknown Number Type but not the Calling Number TON. Should both be following this parameter or not. If not is their another option to change the Calling Number TON? > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) > Ext: 1 Progress

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

I am experimenting with my * server to use SIP with my long-distance providers instead of IAX, so that the media path is from the end user straight to the provider's gateway (hopefully reducing my bandwidth consumption). I have it working with VoicePulse Connect but SetCIDNum doesn't appear to work. Is this something with VoicePulse Connect only or is it generally difficult to set the

Hello, I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it on 1.4svn 56126 and it does not recognise this application. Any idea?... Thanks! __Yehavi:

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(),