similar to: phone directory with asterisk

Dear all I have FAX machine connected with audiocode SIP device i am trying to send fax and when negosiation going on and i start send fax button then my after half page it got stuck in fax machine so is there any codec problem i am useing ulaw/alaw is it fine or not anybody have idea about sending fax with SIP connected device --------------------------------- Ready

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the

Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong? Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. Regards Mattias -- Mattias Andersson -------------------------------- Storskiftesv?gen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38

please can anyone help me knowing if i can install Linux and Asterisk on HP servers _________________________________________________________________ Put your friends on the big screen with Windows Vista? + Windows Live?. http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_012008 -------------- next part -------------- An HTML attachment was

Hello Now that I received an OpenVox PCI card (www.openvox.com.cn/products_detail.php?genre_id=9&id=28), I'm ready to try and set up a voice server with Asterisk. We need the following features: 1. When customers call in, they should hear a voice menu asking them which software they're calling about 2. Next, they should be able to leave a voice message to explain what their problem

Hi list, I bought two UTStarcom F1000 phones, pre-equipped with the latest firmware, including WPA support. Those are configured to register to an asterisk server on the internet (not LAN), and registration works. Calling and being called also, with transfer and all bells and whistles. After a few minutes up to 5 hours (varies widely), the display tells me that an Accesspoint is not available

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

Hi everybody, One of my customers wants to buy IP Phones and Asterisk solution, but his requirement is if he'll not be happy with Asterisk, his phones should be able to work with other IP PBX systems as well, so that he doesn't have to buy new phones again. After all IP Phones is the main investment. He'll most probably go with Nortel IP PBX system if he'll be not satisfied with

Hi List Members , its been about one months since I built my first Asterisk server. What I want to know is: are there ways to make Asterisk take recorded reminders. This is the scenario I have in mind. 1 You place a call to a specific extension say 350. 2 On recognizing the incoming extension the reminder application at extension 350 prompts you to enter a number say 1 to record a message to

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10

hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --------------------------------------- Marek

I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks _____________________ Kevin Savoy Business Unit Telecom Analyst 2218 4th

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

Hello, I have Snom, Cisco, Grandstream & YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101202/ea25cd7e/attachment.htm

Hello I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora 5 for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card, so no need for zaptel and libpri), but I'm stuck: The GrandStream BudgeTone phone fails registering with Asterisk :-/ Following the "Asterisk - The Future of Telephony.pdf", here's what I did: 1. Installed Fedora 5,

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

Hello, sorry for the bulk question, but I cannot find answer anywhere, I have model like Phonebook::Category, and i`m stuck on calling find() method on this model ?

Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I've used CISCO 79XX phones, and they're great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? ---------------------------------------- Rod Bacon