similar to: Noob question on connection

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Hello everybody, I'm trying to receive and sending faxes with asterisk using spandsp. But while compiling the spandsp0.0.2pre11 (tried also spandsp0.0.1), I get following errormessage: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -Wp,-MD,.deps/t4.TPlo -fPIC -DPIC -o .libs/t4.o In file included from spandsp.h:42, from t4.c:71: spandsp/arctan2.h: In function

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

hi, we are using a voip h323 switch. the switch sends all caals to our Gatekeeper (gnugk). gnugk musst send all calls to asterisk and asterisk must do his choice (sip endpoint or out to PSTN) Making calls to our h323 switch works fine over asterisk. what must i configure to get inboung h323 calls from our gnugk to asterisk? any hints for me? thx -- Thomas K?pper 01063 Telecom GmbH &

Integrics is pleased to announce the release of ITSP version 1.6. This version has the following new features: - Comes in 2 editions: * Carrier edition, for 250 to tens of thousands of users on hosted systems. Integrics sells this edition directly and through partners. * Office edition, for 10 to 250 users. This edition is sold only through our partners, for them to sell as PBX systems at

This is a request to the website manager for asterisk.org. The build scripts for our ITSP product include the URLs to download the Asterisk files, such as: wget "http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz" However, if a new version is released, asterisk-1.2.5.tar.gz is moved to the "old" directory. This breaks our scripts until we can update them and send

I've been considering doing a web based database system, where you can post your termination offerings or wanted, then search by location, price, minimum volumes, etc. I'd probably make it free, supported by advertising my consulting company, or Google Adwords, or something like that. I've got the design written down, all ready to start coding. I could probably have a prototype

Integrics is pleased to announce version 2.0 of Enswitch, the most integrated platform available for offering commercial telephony services such as ITSP, hosted PBX, calling cards, call shops, number translation services, and much more. Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is effectively the same product as ITSP 1.7. The product has been rebranded as, although it

Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true?

I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the

I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, Dunc

Hi all, I'm currently looking for a VoIP platform to support the following features: Caller ID Call Waiting with caller ID Call Hold/Retrieve Three-way conference Calling Line Identity Presentation Call back last missed call Last called number redial User line locking/Call Barring (all current levels) Itemised bill Call Forward Call Forward on No Reply Call Forward on Busy Call Forward

We've been asked to quote for a large cluster running Asterisk and our ITSP in a box product. The system will be SIP throughout, with mixed codecs. We're considering using Dell blade servers, 1855 or similar, on the grounds that we normally use Dell machines and they work well, but we need higher rack density. Has anyone used these? Any feedback on whether they're

I have some simple questions and i need your help guys. I have ser server which working fine, between users. I am trying to add some more features to the ser. Most important is the IVR. I installed Asterisk and i am trying to register user in asterisk with no success. Part of ser.cfg file where i am trying to redirect the call to the asterisk.

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

Hi, Alistair Cunningham of Integrics was our guest yesterday. We talked about Integrics new product Geons, a suite of software for building large-scale distributed enterprise applications. The recorded session is now available here: http://www.voipusersconference.org/2010/geons/ The extremely rare John Todd was sighted (and heard) at this event. If you are developing a product or service

Hi, Does anybody has experience with high capacity PSTN voicemail and asterisk, running more then 5k mailboxes for PSTN users ? How many mailboxes can I serve with 4xE1 card if we assume that we have enough harddrive capacity. What would be server requirements. Would the CPU load be the same when storing voicemails in gsm format as compresing to gsm for ip calls ? Any hints would be greatly

Our company is at the point now where we're almost ready to switch over to an Asterisk server for a number of telephony applications. There is one final application I've been trying hard to find to replace something we already use with another provider. It's kind of an advanced "FollowMe" application. (freedomvoice.com) It works as follows: 1. An outside caller calls into

Hi, I have a meeting with an important customer in a couple of days and I am aware that most of their questions are going to be related about scability of Asterisk. We want to propose this customer to integrate Asterisk with SER, but I have a loot of complex doubts that I would like to known before this meeting. I would like to contact with a busines that has experience with large

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, As you know, we released Asterisk 1.0.4 earlier this week. However, there was a callerid bug in chan_zap that has caused us to go ahead and make another release. Asterisk 1.0.5 is available at all of the usual locations. I'm sorry for any inconvenience this may cause. Russell Bryant -----BEGIN PGP SIGNATURE----- Version: GnuPG