similar to: Call Pickup how to display CND of incoming number

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

G'day list, It appears I got ahead of myself in believing I'd solved my CND (caller id) problem. My callerid.c change got it working on one of the handsets I have, but not the other. (Recap: The AU CND standard has an increased delay between the end of the ring signal and the start of the CND FSK data -- it's 500ms in US GR-30-CORE, and around 700ms in AU DC.002/TS030). I've

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, > Does anyone of you has a working configuration with SNOM phones that are > able to pickup a call from a flasing LED? Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and therefore I don't think any config changes will fix it. We've been told to roll back to our previous 1.4.13 installation. It also seems to manifest itself in "ghost

I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J.

All, I have a Polycom 650 phone, when turned on displays "Checking application". Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/daadda2a/attachment.htm

Reason. I have a fax/ans phone with handset, that lets you monitor the caller, so if you wish, you can pickup the call. The asterisk is undergoing testing, it will then be online tested at the house so I can get more familiar in setting components up, e.g. sip phones, voicemail, transfers etc. But, I really need the monitoring of the Voicemall being left, with the ability to pick up the

Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. from my dial plan: [incoming] exten => s,1,Dial(SIP/somebody1|60|tTrR) [internal] include => outbound-local include => parkedcalls

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

The Telco in Canada is been real painful. I was wondering if anyone has installed a Digium TE1X0P card in Canada and if their Telco was so difficult. The Telco will not provide us a service until they see a FCC or DOC number for the equipment ware are connecting to their service. If have found "FCC Part 68, ANSI/ITA-968-A, Including Amendment A1 and A2 Industry Canada CS-03"

Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a brand new TE120P card that I have installed and asterisk is not starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown signalling method 'pri_cpe' It seems it does not matter what I change the vaule for signalling= to, it always returns it as invalid. I have tried the config from my other 2 servers running TE110P cards and the config from AusTechPartniships

Hi guys, I currently have about 10 Asterisk servers scattered around the place each hosting their own dynamic conference centre. Is there any way that when people join these conference centres on each server that somehow Asterisk bridges the conference centres on each server to form one large conference? Many Thanks David. -------------- next part -------------- An HTML attachment was

Hi All, I'm stuck and need some help. I have installed the Asterisk CDR Analyser Version 2.0.1. It mostly works except for the CDR Report. I get the following error even though it lists the CDR details. Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day, sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE UNIX_TIMESTAMP(calldate) >=

I'm not sure if I have a configuration problem or not. I am unable to dial out. When I try to dial in I can hear the phone ring on the dialling phone but Asterisk does not register anything. In zaptel.conf I have loadzone = au defaultzone=au fxsks=1 In zapata.conf language=au context=from-pstn When I do: zap show channels I get: Chan Extension Context Language

I have a simple dial plan to cascade calls when the first phone does not answer: exten => 100,1,Dial(SIP/1000,10,tr) exten => 100,2,Dial(SIP/1000&SIP/1001,10,tr) exten => 100,3,Dial(SIP/1000&SIP/1001&SIP/1002,10,tr) exten => 100,4,Voicemail(u100) Problem is that the once the call goes onto the second and subsequent steps exten 1000 cannot answer the call. When the user

I have a TDM400 in the server. I want to press **1XX to pickup a call. It is ok if I pickup a call dialled from 1XX to 1YY (internal network call). However, it is failed to pick up a call from PSTN thro' TDM400 card. It seems I can't guess the correct context of it. How can I know the context of the call using CLI? The default context of the TDM400 is from-pstn but pickup still