search for: calltyp

...al information -> octobri -> asterisk -> TE410E -> Internet Provider / Receiver for Capability Unrestricted digital information The question is: Is asterisk possible of transmit this digital call to the destination and what is needed to achieve this? I found out that a variable called CALLTYPE exists, but I could not find out, if the Dial applications set the capability for the outgoing call correct? Thanks in advance for our help, Andr?

...And i found something on the beginning of the session... Here's the session from OpenPhone: 2004/09/21 14:52:35.219 2 RasSrv.cxx(2224) GK Read from 216.119.135.3:2367 2004/09/21 14:52:35.223 3 RasSrv.cxx(2237) GK admissionRequest { requestSeqNum = 40386 callType = pointToPoint <<null>> endpointIdentifier = 9 characters { 0032 0031 0032 0033 005f 0065 006e 0064 2123_end 0070 p } destinationInfo = 1 entries { [0]=dialedDigits "835218329287863" } srcInfo = 2 e...

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

...ny different configs on the GK, but always come up with the same error. It appears to me that asterisk successfully registers with GK as I can see the aliases and the e.164 numbers, but when the h323 softphone tries to call my IAX softphone, I get this: admissionRequest { requestSeqNum = 2 callType = pointToPoint <<null>> endpointIdentifier = 9 characters { 0033 0032 0039 0037 005f 0065 006e 0064 3297_end 0070 p } destinationInfo = 1 entries { [0]=dialedDigits "100" <--------------------no...

Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten => 750,1,Dial(SIP/120,20) All this works fine. Now I have the need

Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Ganbaa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/cfe09e1e/attachment.htm

class Trunk < ActiveRecord::Base has_many :call_type_trunks has_many :call_types, :through => :call_type_trunks end class CallType < ActiveRecord::Base has_many :call_type_trunks has_many :trunks, :through => :call_type_trunks end class CallTypeTrunk < ActiveRecord::Base belongs_to :call_type belongs_to :trunk end The associaton class has a column named price. Each association trunk/call_type will have a...

...debug(" Call Loop Anaylsis for ".$v{callednum}." = NO LOOP"); $AGI->exec('Set',"__".$v{cfnum}."=".$v{callednum}) } return; } $AGI->exec('Set',"__".$callednumber."=1") if ($calltype !~/^Local/); if (callfwd_loop_check($callednumber,$callfwdtonum)) { return; } $AGI->exec('Dial',"Local/+".$callfwdtonum."\@localcontext\&SIP/+".$callfwdtonum."\@sipproxy|180"); I hope this all makes sense! :) Thanks, - Darren

Hello, I ordered the Devel lite kit, and installed it. I am just trying to get the FXO port to work, and am having trouble. To load the card I do the following. modprobe wcfxs modprobe wcfxo ztcfg -vv asterisk -vc My /var/log/asterisk/messages show Nov 3 11:03:39 WARNING[3317]: zt hook failed: Device or resource busy Here is my /etc/zaptel.conf fxoks=1 fxsks=4 loadzone=us defaultzone=us

...urn the correct file with matching md5sums. Under my module called "truth" I have the following: - files -> private -> domain.inter -> hostname -> truth_tags.yml ex: --- role: - base env: - dev - lib -> facter -> load_truth_tags.rb problem area: def apitruthtag(calltype) # set some client side variables to build on later sslbasedir = ''/etc/puppet/ssl'' sslprivdir = sslbasedir + ''/private_keys'' sslpubdir = sslbasedir + ''/certs'' sslcafile = sslpubdir + ''/ca.pem'' # this sets if we w...

......k } port = 1720 } activeMC = FALSE conferenceID = 16 octets { 80 d4 bf de d3 8a d9 11 8e 4c 00 01 02 3f c2 76 .........L...?.v } conferenceGoal = create <<null>> callType = pointToPoint <<null>> sourceCallSignalAddress = ipAddress { ip = 4 octets { c0 a8 01 97 .... } port = 32822 } callIdentifier = { guid = 16 octets {...

Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error "reason 24 (Call ended with Q.931 cause)" I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

...:842 'acadaeafb0b1b2b3b4b5b6b7b8b9babb'H 18:44:37:842 } 18:44:37:842 conferenceGoal = { 18:44:37:843 create = { 18:44:37:843 NULL 18:44:37:843 } 18:44:37:843 } 18:44:37:843 callType = { 18:44:37:843 pointToPoint = { 18:44:37:843 NULL 18:44:37:843 } 18:44:37:843 } 18:44:37:844 sourceCallSignalAddress = { 18:44:37:844 ipAddress = { 18:44:37:844 ip = { 18:44:37:844...