similar to: CIDlookup

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, I am setting up a hylafax server. From what I've read so far, hylafax supports CID numbers and names but currently does not support DID. I assume I can do something like this... [40faxDIDs] exten => _87[5-8]X,1,SetVar(CALLERIDNAME=${EXTEN}) exten => _87[5-8]X,2,Dial(Zap/g${hylafaxMODEMGROUP}) ...and use the CIDName variable in hylafax to route the faxes to the appropriate

I have all three running on the same box. I say OT because it appears asterisk is doing it's job just fine. It must be an IAXmodem or faxgetty (hylafax) problem When faxes work, they look great. I have ten IAXmodems setup with different ports and they register fine. I have ten faxgettys that startup fine. I start the IAXmodems and then faxgettys in inittab. They are setup as a roll

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

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:

Hi Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during "background(menu-filename)" method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting

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 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

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 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

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at

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

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: ---------------------------------------- rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"' rem rem Asterisk module

Hi Sriram, Thanks for performing this analysis. The problem here, both for memcpy and the vectorizer, is that we can’t predict the size of “n”, even though the only use of ’n’ is for the loop bound for the alloca [4 x [8 x i32]]. If you change the unroll condition to TC >= 0 then you will disable loop unrolling for all loops because getSmallConstantTripCount returns an unsigned number. You

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,

Hi All, I'm looking for a way to filter the AstDB cidname family to show only those entries with a specified area code in the Asterisk CLI. If this were a SQL database it would be something like: SELECT number, name FROM cidname WHERE number LIKE '1234%' I've tried "database show cidname 1234*" and substituted "%", "$", "-" for the wildcard

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

Admittedly this is probably pretty stupid of me, but there are just some things I can't understand by reading documentation. Any suggestions or recommendations about how to handle my problem are greatly appreciated. I'm trying to achieve the same functionality as my Nortel PBX, without rewriting much 'C' code. In my dialplan I'd like to compare two variables as a means of

Hi everyone, I am trying to get iaxmodem up and running. This is a very basic setup, which at this moment should only answer incoming faxes. What I did: zapata.conf (rest of it should be fine): faxdetect=incoming group = 1 channel => 1-2 context=from-pstn iax.conf: [200] username=200 type=friend callerid="Fax" <200> secret=dooo host=dynamic notransfer=yes allow=all

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with