similar to: Nwebie Config Problem

I am putting together my first *. I had it running with two other pc's running xlite and setup voicemail and a couple of menus and submenus and had that running well. I had order a couple of oem x100p cards from digitnetworks. I installed them as they said with their voicepet2.2.zip drivers and did the modprobe on zaptel and wcfxo and then ran ztcfg -vv and got this: Zaptel Configuration

I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found

Hi there I'm built samba-3.0.0-2 from the src rpm on a redhat 7.1 system. I had to make sure a few ldap and kerberos devel rpms were installed, but I managed to build and install it eventually. I added: --with-ads \ --with-krb5=/usr/kerberos \ --with-ldap to the SPEC file for the package. I followed mainly whats in here:

hi all, was wondering if someone could assist with a slight problem i'm having. I have asterisk setup with extensions 101 to 109 and am using xlite, grandstream budgetone, polycom ip500 and a couple of other phones. the problem is: 1. only the xlite extension (107) can receive calls. 2. all extensions can dial into voicemail and get mwi when msgs are received. 3. when dialing a non-xlite

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

I've been looking at how our budgetone's have been failing and have found the following: A quick layout -- Latest CVS as of tonight. Sip phone behind NAT. * server with public IP address. -------from sip.conf for my phone: [1747xxxxxxx] username=xxxxx secret=xxxxx host=dynamic type=friend nat=yes ------- -------from the * log messages Nov 6 01:50:07 DEBUG[4101]: File chan_sip.c,

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/949c1368/attachment.htm -------------- next part -------------- Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" Message on SNOM 200 LCD when ever I try

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031215/9327b656/attachment.htm -------------- next part -------------- Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" Message on SNOM 200 LCD when ever I try

I have a couple of Budgetones that I am playing with trying to get them to work with * from a remote network over the Internet (yes NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded (UDP) to my public address (through a Cisco PIX). Internally, I can setup my budgetone, it registers and works great. I then have a Linksys router connected to another Internet

I've just installed a Digitnetworks X100P clone in my * server and run the install script for the voicepet-single-x100p tarball. The install appeared to run OK with modprobe wcfxo successful and the ztcfg reporting "Channel 01: FXS Kewlstart (Default) (Slaves: 01)". When I try to start * though I get a segmentation fault after loading res_features.so. I discovered that the

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

Hello Folks, I'm trying to decide here between a few cards for connecting an Asterisk box to a single E1 channel (either PRI or R2 signaling): - Digium E100P: has been replaced by the TE110P below, but can still be had at places like digitnetworks.com for $475, and I guess there's always a place for good-olde-obsolete cards in the world as long as they work :-) - Digium TE110P:

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

While someone else is asking, has anybody any experience of http://www.digitnetworks.com/ Are there any other suppliers people would recommend for somebody importing to the UK? Also has anybody in the UK imported any IAXy or OEM X100P - FXO PCI Cards? I'm interested in knowing what sort of import duty they attracted, if anybody has the TARIC classification number that would be useful, I

Hi List, I've got * randomly hanging up on inbound or outbound calls on zap channels. I use a Digitnetworks X100P clone card. Any idea of what might be happening? Cheers, Jean-Michel.

I've a problem connecting uniVoice (http://voice.uni.it) from asterisk. Using my account data I can place a call smoothly using xlite or my budgetone phone directly, but I'm not able to use uniVoice as a peer from asterisk. Registration seems to work correctly, but when I try do dial, the sip authentication fails every time. Their tech people told me that they are unable to make

I'm using asterisk with XLite everything is working. But in the asterisk console I always receive some notice of Registration failed . What is the reason for this? How Can be fixed? message : Jul 1 16:18:29 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'damian <sip:damian@10.1.1.11>' failed for '10.1.1.11' Asterisk and Sip phones are all in one

Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced it with TDM04B. I changed the zaptel and zapata to include the four channels. When I run ztcfg, I get configuration errors: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: