similar to: Using CallingPres to set up CallerID blocking

I've just upgraded to the latest CVS head, and my outbound calls stopped working. I traced it back to the line exten => s,9,CallingPres(${ARG2}) It seems as if this application is now missing. I tracked back the changes and found in 1.415 of chan_zap.c the code was removed because it was "duplicated". However, it does not exist anywhere ! Am I being stupid, missed

Dear Everybody, I have to store variable from ${CALLINGPRES} and get birth date of our client and get back to him his birth prediction as numerology (numerology digit value is between 1-9). I have also mentioned below example here suppose client's birth date is 27-01-2000 then 2+7+0+1+2+0+0+0 = 12 and then 1+2 = 3. 3 is result so client can get his prediction as this 3 digit. Please

Alright, folks. I just deployed * into full production at my office. We have around 50 7905's, 5 7940's, and a handful of soft clients. We run a call center with around 15 agents. I also have a queue set up for the receptionists so that they don't get bombarded with calls. Everything seems to be working with a very few minor glitches. I firmly believe that the few problems we are

Hello all, Recently I purchased an QuadBRI card from junghanns.net after some playing around, reconfiguring dialplans etc with the exception of 1 thing everything seems to work: I seem to be unable to set the outbound callerid. The dutch telecom operator (KPN) provided me with 4 MSN's on 1 BRI interface. In the past years I'm more then used to setting the MSN without the leading 0, this

Hi All, In my dialplan and standard asterisk CLI logging i see that i am able to restrict the callerid when dialing out with asterisk. however, on the receiving phone, the callerid is still displayed. When i increment the logging of the pri with "pri set debug on span 1" on the CLI i also get the lower level debugging info from the pri.

Hello, we had a running configruation where asterisk passed the phone number and the ddi to the pstn (i.e. 595-431) Now only the rootnumber arrives: 5950 I do not know, what to do. I tried to use callingpres (now i am just hiding every number, because 595-0 is no valid extension..) but that did not worked. > Protocol Discriminator: Q.931 (8) len=44 > Call Ref: len= 2 (reference

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

Hi, Ive been playing on a asterisk to orion gsm box E1 pri setup. I have achieved incoming calls to be passed to my asterisk box successfully but outgoing calls will just I have tried playing with various pridialplan and overlapdial settings and with no success. If anyone can make more sense from the log, I'd certainly appreciate it. I am sending a 10 digit number to be dialed. I guessed

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)

On 2/20/15 6:15 AM, thufir wrote: > What's the difference between user "123" and "devries"? Based on the > output here, they seem the same..? > > tleilax*CLI> > tleilax*CLI> sip show users > Username Secret Accountcode > Def.Context ACL Forcerport > 201 password 201 > default

Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup

I'm trying to get sipusers working with a realtime odbc database on Asterisk 1.6. We have sippeers working from the database, but need sipusers to be in a separate table for other implementation reasons. sip show user test load returns results from the database. CLI> sip show user test load * Name : test Secret : <Set> MD5Secret : <Not set>

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

I've got a bit trouble with callerid and zaphfc cards. Basically zaphfc doesn't add the "0" in front of national numbers (haven't tried a international call yet). With chan_capi that allways worked fine, however i had to define the national and international prefixes in capi.conf. Is there something similar in zapata.conf ? Here is my zapata.conf: [channels]

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME