similar to: Getting firmware

Hi, Anyone is using LG ip phone LIP-100 with Asterisk. I've two of this phones but seems to work only with net2phone, in the product page http://isupport.lge.co.kr/html/ibu_lgic_modelView.jsp?jgrcode=D2_IPTP&modelid=M_IP100C the features are showing SIP and H.323 support. Can be used with my asterisk box? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq.

When I try call to net2pohe sip service in my debug I look next: ---------------------------------------------------- We're at 192.0.0.0 port 27916 Answering with preferred capability 1 Answering with preferred capability 2 Answering with preferred capability 256 Answering with capability 4 Answering with capability 8 Answering with capability 16 Answering with capability 32 Answering with

Hi, I?m looking for settings to configure net2phone carrier in my asterisk. I found this configurations, but it?s not work. I don?t known if this configuration is for voice line or voice access account. Anybody can help me, with other configuration? Thanks. ---- *sip.conf* [general] useragent = X-Lite release 1103m register => PHONENUMBER:PASSWORD@sip.net2phone.com [net2phone] type = peer

I've been trying to use net2phone's sip service at sip.net2phone.com with * but keep getting SIP/2.0 401 Unauthorized. Do you know if this should be possible? So far: I can use an ata186 to connected directly to n2p through sip.net2phone.com without any special settings. I can connect from * to iconnecthere, but, whatever I try from * to n2p produces "SIP/2.0 401 Unauthorized"

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER

Hi! Net2Phone is getting a common SIP status code, "404 Not Found," when trying to place a call to our Asterisk server. We're hoping someone on the list can shed some light on why this is happening. We can process a call from Asterisk to Net2Phone without any problems. Net2Phone sends the INVITE but immediately gets the "404 Not Found." The "To:" field

Has anyone had any success running net2phone on wine. I tried it but I recieved an error message: ]$ wine net2phone.exe Invoking /opt/wine/bin/wine.bin net2phone.exe ... /usr/bin/wine: line 380: 6945 Terminated tail -f $log_name Wine failed with return code 2 Even if I can't make it work I'm really interested to understand what's happening. (though net2phone is basically the

Hi All, As Net2Phone don't permit more than one session per account, I configured about 10 sip trunks and configure multiple trunk routing but once the first trunk is used I cannot make additional calls, I also cofigure my dial plan in other way using the chanisavail command but still not working. The chanisavail command configuration is correct as I can make calls using other trunk than

I'm trying to register with net2phone. I've already changed chan_sip.c, User-Agent: string to say "User-Agent: Cisco ATA 186 v2.16 ata18x (030401a)". But still I'm getting the error msg. Here is the debug msg: IP Address is xxx.xxx.xxx.xxx 11 headers, 0 lines Reliably Transmitting: REGISTER sip:66.33.146.12 SIP/2.0 Via: SIP/2.0/UDP

Hello, My server is Mandrake 10.1 eth0 is WAN with static IP connected to 512k DSL eth1 is LAN. I am using squid proxy for internet with NSCA auth. I am able to send and recieve mails. One of the client system wants to be able to make net2phone calls. As of now he is not able to. Howto allow net2phone calls ? Thanks Varun

Dear, I have this scenario: - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10 - Behind a Cisco ASA firewall that connects to Internet - SIP trunk to Net2Phone with these parameters (nat=no): host=200.58.113.60 username=DOLLY secret=123456 port=5060 type=peer dtmfmode=rfc2833 disallow=all allow=alaw&ulaw nat=no canreinvite=no qualify=yes -Softphones Xlite The PBX can't register to

Dear group, I want to use the mclust package on large data, and therefore I want to use a subset in the initial clustering phase. From help(mclust): k: If `k' is specified, the hierarchical clustering phase will use a sample of size `k' of the data in the initial hierarchical clustering phase. The default is to use the entire data set. m2 is a

As a newbie, can anyone advise me if Asterisk can route international calls to a US based service such as Net2Phone so we can take advantage of the internet and save on calls? That would be my main reason for an Asterisk based PBX. Chris Mason masonc@masonc.com Box 340, The Valley, Anguilla, British West Indies Tel: 264 497 5670 Fax: 264 497 8463 Cell: 264 235 5670

Hello we are a small call center with only 8 lines we use max4 and the 2-2 port gateways from net2phone . There equipment is good but we are getting hit by lower cost competition. We need to be able to compete. We have a couple of providers who are 50% less in some cases even more. So it makes sense that we would like to be able to compete . Since we have spent quite a bit of money on existing

All: I'm trying to run the cygnus-port on Win2K Server. I have the dll and the ported code (exe). It seems to run without access violation or anything. Since Windoze doesn't have a good RSH, and I'm only going to use this between my two laptops, I decided to setup an rsync server on one of them...no problem. Seems to run, shows "Starting..." in the log...good conf file,

Hello, I've noticed that when I try to reencode VBR mp3s to broadcast @128k , they are not being reencoded at all. They stream as normal VBRs I am unable to reproduce this at lower bitrates (56-112k i've specifically tested and all work fine). It does reencode all non-VBR mp3s appropriately to 128k... Any ideas why VBR reencoding would suddenly stop working at 128k? I've attempted

I did some further investigating into this issue and here's what I found: It appears somewhere between ices0.1.0 and ices 0.2.0 is where the "bug" (for lack of a better term) was introduced that i'm noticing. When I compile ices0.2.0 w/lame 3.89beta, the VBR is *NOT* reencoded, but instead streamed out at full VBR bitrate(s). However, when using ices0.1.0 w/lame 3.89beta, it

Hi there, here some questions and experiences after playing for one day with 3 Swissvoice ip10s and the latest * CVS: QUESTIONS: - what is the user option "enter voice mail number" good for? It doesn't appear to be of any practical use - does anyone have some Swissvoice info that I cannot find on their web site like the guide to MGCP XML (.svd), guide to configuration file

Stealth Communications Announces Registry to Avoid Access Fees Posted on: 04/23/2004 Stealth Communications Inc. today announced the official launch of a registry that allows service providers routing calls over the Internet to avoid paying local phone companies access charges. The VPF ENUM Registry allows carriers to map telephone numbers to IP addresses for such things as SIP phones and

Hi, Is there a way to use swissvoice IP10 in MGCP mode behind NAT. I already use this fonctionnality with a PBX other then asterisk, and it work very well. In fact, I need to use this settings in the IP10 telnet session set xgcp rgw_name MAC_ADDR set tcid 0 notify_entity ca@[@IP PBX]:2427 In fact, the phone register with the MAC@ and not with the IP@, and the PBX reply the phone in the UDP or