similar to: Caller ID question

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

Hiya, i've just noticed that chan_misdn writes the AOCD information into a logfile. Has someone done a patch that sends this information via sendtext() to the active channel? At least some phones (like Cisco with chan_sccp and the snom-phones with SIP) can show this information on the display, what is kinda the idea with AOCD ;-) Regards, Andeas

Hi all, I'm seeing slow transfers to/from my samba server. I just tried to copy a 270MB file from a samba share to an XP machine. At first, the estimated time displayed as 141 minutes :( The odd thing is that if I then flood-ping from the server (in this case, I pinged my dsl router), the transfer completes in about 10 seconds. Hitting F5 repeatedly when browsing a share on the server

Hi, I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When calls come in on our POTS lines, the caller id shows up like "555-555-1234 at 192.168.1.10" where 555-555-1234 is the correct phone number and 192.168.1.10 is my pbx server IP. This format does not work for redialing on outbound calls. While there may be an outbound dialing change that could be made, it

Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma <---> zaptel <---> asterisk <---> sip <---> SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Executing Dial("Zap/62-1",

I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!!

Hello, I've been testing out 3.0.25 before putting it into production and came across an issue. As a simple example.. in a share definition root preexec = echo "hey there, hi there, hello" > /tmp/hiya.txt works fine under 3.0.24 not so under 3.0.25, /tmp/hiya.txt isn't created. Cranking up the debugging and using some other similiar tests it seems to me that Samba is in

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -- 202-456-1414 is an actual White House number. Using SetCIDName("Flying

Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain1.org/1000 register => andy@sip.mydomain2.org/1000

Hiya, I sent the enclosed email to linux-kernel before realizing that it would, perhaps, be more proper to send it here, which I hereby do. I shall be happy to try out any and all patches. Best regards, Sune M?lgaard -------- Original Message -------- Subject: [bisected] nouveau failure with 3.9-rc1 and ongoing Date: Tue, 12 Mar 2013 19:57:55 +0100 From: Sune M?lgaard <sune at

Hiya, just wondering how easy this would be, I'm guessing through yum, but is there a preferable process from updating from fed core 3 to centos ideally keeping settings as is? (desktop rather than server, just want to migrate to centos, nothing too essential apart from raw data). Ian

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Excuse me if I recently posted on this, but I cannot find it, in my, or the list archives. Is it possible, when transferring a call, to set the user ID to that of the "outgoing number" instead of the incoming number? I believe the answer is (was) yes. New twist, does it matter what the destination "media" is? Meaning, the call would be coming in on a T1, going out on a T1,

Has anyone had success with the Snom 190 displaying caller ID name and number on the Snom 190 on for an inbound call from *? Right now our Snom's only show the caller id name, not number. I know the number is transmitted from the Telco and received by * since the number shows on the incoming call event at the * console. We are not setting the caller id in the extensions.conf, simply passing

All, Having trouble getting answer from Verizon. I believe Asterisk will let me specify a name and number that is sent to the PSTN (Verizon) of outgoing calls. For instance, if I have a client, First Bank, and their toll free number is 888-555-1234, I could send that name and number. Verizon is telling me that they will forward the number I send them, but the name will be my company's

hiya, i want the selector that would return a <tr> of a table if any td contains some text, so i can use it in click_link_within e.g. When I click the "show" link within the row containing "user at example.com" When /^I click the "(.+)" link within the row containing "(.+)"$/ do | link, text| selector = ?? click_link_within selector, link

Hiya, I would have expected this to show the channels in the bridge inside the anonymous function - it shows the bridge is empty though? var bridge = ari.Bridge(); bridge.create({ type: 'holding', name: event.application+" bridge" },

Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.

Hiya, I was trying to figure out how to display PNG files using wxWindows/wxRuby. Google turned up a few things, mainly discussing the wxImageHandler class and it''s children, etc, but there seems to be no mention of them in the wxRuby source. Are they there, and I missed them, or are they just not there? If there not there, is there a quick and dirty way to get them in? It seems like