similar to: Dropping Connections

I'm running chan_capi on a number of systems in France, France Telecom offer the possibility of having the caller's name, but say we must configure for EuroISDN+. Google doesn't show much and the best I could see was in Dutch. Any Europeans solved this one? Rgds -- Dave Cotton <dcotton@linuxautrement.com>

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

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

All, If I have video phones behind an asterisk server (with 2 network cards) and all the phones have extensions. Internally everything works great. Now for people that want to call my video phones external to my office is there a way to do that? On the extenal persons phone enter an IP/EXTEN where IP is my server and not the phone? Can that work? Would I have to have PUBLIC IP address for every

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

Hi Friends, I am planning to buy "IMate PDAL" mobile phone. This contains "Wi-Fi 802.11b/g" feature. So, Is it possible to get internet using my wireless router in my office? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Got a little couch potato? Check out fun summer activities for kids. -------------- next part

Hi group, I'm trying to configure the email notification when a user leave a voicemail, but don't work (send email notification). I configured esmtp in my linux box, if a try to use it with command line, it works fine. (echo "Hello" | sendmail a@b.com -f b@c.com). My voicemail.conf [general] format=wav49 attach=yes serveremail=anonymous@abc.com fromstring=Asterisk

Hi all, ive got 2 internet connections set up via the nano howto (which are working great) and we are running NAT. Was looking in to qos mainly to stop large http downloads/ftp downloads from hogging the line so that browsing for other users doesnt slow to a crawl, but if the line is free and no one is doing anything then for it to use the available bandwith. The wondershaper sounded exactly what

Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would

I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to voicemail or transfer it to

Does anyone know if X-Ten or SJPhone support multiple cordless handsets for multiple lines? I have an office with multiple roaming users(nurses) that are in and out. I'd like to provide them telephones, and my idea is to have a PC sitting in a corner somewhere running a softphone client. When a nurse comes in she just picks up any available handset(anywhere from 2-5 per office) and starts

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

I''m trying to build a QoS system that would divide the outgoing traffic into four categories, each one also subdivided into two more categories. For that I chose a htb root qdisc subdivided into four classes, each of these classes contains also a htb qdisc. Until these point everything goes well, but when I try to attach a new class to the leaf htb qdisc a problem with the sintaxis

Hi Ivo, --On Freitag, 23. Juni 2000 01:15 +0200 Ivo Timmermans <zarq@icicle.yi.org> wrote: > I'm not sure I fully understand your patch. This is not so important since I'm trying to get 1.0pre2 running. Although I had no luck so far, I'll point out what's going wrong at the end of this mail. > For instance, you force a > key exchange when the connection is made,

Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in

Hello, I’ve read this Article avout VOIP and HFSC http://automatthias.wordpress.com/2006/06/30/hfsc-and-voip/ I’ve got few questions ? Considering this tc class add dev $DEV parent 1:1 classid 1:2 hfsc \ rt m1 ${UPLINK}kbit d 50ms m2 $[1*$UPLINK/10]kbit \ ls m1 ${UPLINK}kbit d 50ms m2 $[3*$UPLINK/10]kbit \ ul rate ${UPLINK}kbit rt = realtime curve ls = linksharing curve but m1 = ? m2 = ? d =

Dear friends: I was analizing the HTB script at "The ultimate traffic conditioner", at the Cookbook, at Lartc. I have a very basic question: First, it is defined the root htb qdisc, as expected; then, it is defined the main (parent) htb class, with the total uplink as rate, also as expected; then, they are defined 2 child clases: first of them has also the total uplink as rate (!)

Hi I''m moving to hfcs (mostly succesfuly:) ), but I''ve encountered a strange problem: When I download something from this server using wget (one thread) via an idle uplink, I get no more than 130 kbit/s, and when I''m using prozilla (or start more wgets), I get about 200 kbit, but still it''s not all the link can do (240) - the rest remamins unused. Is this

tinc 1.0.19 worked fine on Raspberry Pi as long as my Pi was connected via ethernet cable. Today i changed the connection to wlan. Pi crashed. worse, it was no longer operational, even if connected via cable. recovery from backup before tinc was installed (yes, i had a backup ;) tried again. installed tinc: worked as long as connected via cable. crash after connecting via wlan. no longer