similar to: Re: Vontage Problems

Hello all, I am trying to find out if anyone has a provider that is good with dtmf playback using a Sipura 2100? I've just dialed with voxee and the call goes through but when I press 1 my dialer does not " hear" it. My dialer is making the call using a Dialogic d/4PCI connected to the Sipura 2100 through voxee and I am calling my landline. When I pick up the landline

I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with

Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension from a sipura forced to ulaw. When the call goes out through Teliax IAX trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to transcode calls from/to a specific extension? I'm running asterisk 1.2.4 and that extension is for my home alarm/dish network and fax calls. Thanks -------------- next part

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

Hi, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is). We have a connection to and from providers via SIP and IAX. When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me. However, if I park the call,

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

I setup my samba sever to use ldap as a backend for authentication I can connect to the ldap directory using ldapAdmin from windows xp and diradmin in FC3 and administer the ldap directory but when I issue a command from the teminal window ( smbpasswd -a test I got the following error. ldap_initialized: time limit exceeded connetion to LDAP sever failed fot the 1 try ldap_initialized: time limit

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

Hello everyone. I can place outgoing calls no problem with my IP500 (using teliax as our provider). Thing is, when a call comes in, 90% of the time when I pick up the handset it drops the call immediately. I turned on SIP debug, and have listed my extension config from sip.conf. Any help is greatly appreciated.... sooo close.... TIA! -Ron [3004] type=friend username=3004 password=XXX

I saw the link someone posted to SL (Scientific Linux) yumex rpm's. I grabbed the SRPMS, changed some references from SL to CentOS in the spec files and rebuilt the RPMS. I managed to get yumex up and running by manually configuring yumex.conf. What is the use of the yum-utils and the dozen or so utilities? Are there any docs? (Bearing in mind my hack may have damaged the SL installation)

We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. Our asterisk setup has a 4-port

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time. As of today, I'm running the latest code from CVS. -- Called teliax/13143212222 -- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw) -- Format for call is

I have an account with a IAX service provider that I'm happy with but recently I started getting rather strange reports from users. They're saying that occasionally, they'll be on a call via the provider and the outbound audio appears to slowly fade out to nothing with a bit of static during the fade. Does this ring any bells with someone? I'm going to update * on the server

Apparently we've been having calls sporadically drop. We're using an IAX outbound trunk and SIP adapters on the inside. Below is a log excerpt detailing one of the calls which dropped, and it looks largely normal to me except for this: Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: IAX2/teliax-2 Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging

Hello, One of the Sipura 2k's I'm using has a dialtone that occasionally fades to static when the user picks up the line. Are there any settings that I can check that would affect this? Regards, Christopher