similar to: Cut Through DTMF & caller ID on SIP phone

Hi Is there any way to stop Asterisk from sending Caller ID display on the softphones ? I;ve E1 PRIs and SIP extensions , i need to stop caller ID from appearing on the softphones ...but in CDRs caller Ids should show - so please dont suggest to set "blockcallerid=yes" in zapata.conf ;) Thanks Sriram -------------- next part -------------- An HTML attachment was scrubbed... URL:

My setup : Trixbox 2.6.1 & TE410P running well .: 1. I need to store the CallerId of the PSTN caller with his language preference so that next time he is played the prompt in his language that he chose the first time.What would be better - storing his number in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial plan and quering it everytime to see the callers

Hello Everybody I am using Trixbox 2.4 (with TE420P & PRI lines) .. my setup is like Calls -->Asterisk-->legacy pbx--->analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents

Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk

Hi; How can a user block his caller-id in Astersik? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041229/07ecf20f/attachment.htm

Hi I am trying to implement monitoring of asterisk (all 4 spans-i want to show them line by line Up or down) using nagios using below script, but i always get the status as down and red..can anyone let me know how to read an output from nagios plugin ? nagios etc is configured already and is working PATH=/bin:/sbin:/usr/bin:/usr/sbin FAILS="" SPANS=$(asterisk -rnx "pri show

Hi List I've a CID lookup hooked onto an inbound route (i m using trixbox) ...it runs well but it returns the value as "CIDNAME<CIDNUMBER>" ... if i just want to display the CIDNAME [leaving the quotes and <CIDNUMBER>] .. how can i do it ? do i have to edit some macro in extensions.conf ? rgds Sriram -------------- next part -------------- An HTML attachment was

Hi I;ve trixbox installed with 2 NICs. One NIC carries the MPLS-VPN traffic (only a 1 MB link without internet for carrying voice to another site) while the other NIC has a connection with public IP for internet services on that machine. the first NIC (eth0) has IP of 172.16.0.1 and is connected to router with WAN IP: 10.18.6.254 , the second IP is 203.234.82.98 (eth1). i want to have the

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1]

Hi , I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101 ) in a queue..When a caller arrives in queue , it lands on first 100 , 100 then does a blind transfer to 101 .. so that the caller can converse with 101 .. strangely enough the queue_log shows : 1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123

Hello, This patch aims to improve compile time performance by increasing the SCEV vector size in LoopStrengthReduce. It is observed that the BaseRegs vector size is 4 in most cases, and elements are frequently copied when it is initialized as SmallVector<const SCEV *, 2> BaseRegs. Our benchmark results show that the compilation time performance improved by ~0.5%. Patch by Wan Xiaofei.

Hi Daniel, I made some changes to the LNT perf reporting tool to make it more user friendly by adding some features: 1. Make the sidebar and the navigation bar stationary, so that it is easy to navigate the site 2. Have the pop-down menu for the items in the navigation bar, activate upon hovering the mouse, rather than clicking the item 3. Add a nav-link in the sidebar for the

Hi, I am trying to get a small loop to *not vectorize* for cases where it doesn't make sense. For instance, this loop: void foo(int a[4][8], int n) { int b[4][8]; for(int i = 0; i < 4; i++) { for(int j = 0; j < n; j++) { a[i][j] = b[i][j]; } } } * Has maximum of 8ints copy. LLVM tries to use Memcpy for the inner loop. It is not helpful to perform

Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan --

Hello there, I recently started working on the LLVM backend for a target that doesn't support x87 instructions. Currently, I am in the process of completely disabling some x87 instructions such as fcomi, fcompi,... for a specific sub-target. I also do not have SSE enabled for my sub-target, and llvm resorts to fcomi* instructions for FP compare instructions. Is there a way to bypass the

Hello SetCallerPres function seems to be removed from Asterisk 1.4. What function or application replaced it? Bit of a problem if you want to use CLIR on your PRI connections. Jon No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date: 22-05-2007 15:49 -------------- next part -------------- An HTML

Hi Nadav, Thanks for the response. I forgot to mention that there is an upper limit of 16 for the Trip Count check, TinyTripCountVectorThreshold = 16; if (TC > 0u && TC < TinyTripCountVectorThreshold). So right now, any loop with Trip Count as 0, or with value >=16, LV with unroll. With the change to the lower bound, it will also include the loop with 0 trip count. SCEV returns 0

I have an issue where a particular dialplan works but another doesn't. I'm not sure why. To me they look identical and it has me stumped. This works: [to-test] exten => _X., 1, SetCallerPres(allowed) exten => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb) exten => _X., 3, Ringing exten => _X., 4, Dial(SIP/9330 at a-test,20,ro) exten => _X., 5,

Wow, that sounds great! Thanks for working on this, and yes, please, send the patches! --renato On 30 April 2013 16:23, Murali, Sriram <sriram.murali at intel.com> wrote: > Hi Daniel,**** > > I made some changes to the LNT perf reporting tool to make it more user > friendly by adding some features:**** > > **1. **Make the sidebar and the navigation bar stationary,