similar to: Stutter dialtone on TDM31B (TDM400P)

I have a brand-spanking new TDM31B (3 FXS, 1 FXO) and when I start wcfxs (the only module that recognizes the card) from Zaptel 0.9.1 I get: Zapata Telephony Interface Registered on major 196 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXS Module 1: Installed -- AUTO FXS Module 2: Installed -- AUTO FXS

I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack

When a user dials 999 he is always asked for the mailbox and has to enter his mailbox number and password. As I understand this shouldn't happen because the CALLERIDNUM is passed over to VoicemailMain. It's annoying to have to enter the number everytime ... The voice mail configuration is read from MySQL. We are using the CVS version from a few days ago. Extract from extensions.conf:

Hi All trying to compile asterisk under linux kernel 2.6.6. Currently under zaptel get the following error make linux26 Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 as going from the readme. is 2.6 not compatiable with asterisk and should I go back to 2.4.26. Also has anyone got the sipura 3000 working with asterisk, both fxo and the fxs ports on the

Hello, I have modified the VoiceMailMain application to satisfy the request of the "local users", i.e., my wife. She lost patience with too many options (we have one mailbox, so we don't need to forward messages, or reply to messages, or file them in 6 different folders...) So the modified app says "Message 1", reads the message, "Message 2", reads...

I've done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I don't use a "throw away" digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On

Hi: I really want to know how rsync works. Once it synchronize a file. Does rscync first create a temporary in the remote machine first and then rename it? Or it direct write the difference into the dest-file? Could you please tell me what will happen to the dest-file when a rsync process interrupted by some problems(network problem etc ...)? Thanks for your help Best Regards

Hy, I'm working with asterisk 1.2.4 and zaptel 1.2.4 With these version an the options in zapata.conf: answeronpolarityswitch=yes hanguponpolarityswitch=yes I don't detect polarityswitch. When asterisk reloads I see in CLI: May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring answeronpolarityswitch May 10 00:44:27 WARNING[14639]: chan_zap.c:10876 setup_zap: Ignoring

Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel will result in hearing a loud

How can I stop the following warning from occuring? invalid factor level, NAs generated in: "[<-.factor"(`*tmp*`, iseq, value = structure(1, .Label = "12", class = "factor")) The Label messages are for "5", "8", "12" and "46". I want the NAs to be generated as needed. Is this causing R to slow down by generating the warning

A question: is there any way to get * to answer certain DTMF sequences entered on an extension with a stutter tone? Long version: I would like to add features to my dialplan like "Caller ID Unblock" which work in the same way that the PSTN works: I pick up the phone, get a regular dialtone, press *82, and get a short stutter dialtone which confirms acceptance of the request, and then

I have the following entry in zapata.conf, but I don't get stutter dialtone when there is a message waiting. Suggestions? Please? callgroup=1 pickupgroup=1 callerid="Paul mahler" <100> context=inside mailbox=100 channel => 1 Thanks, Paul

I'm having problems enabling stutter dialtone for users connected to channel banks. Half of our users are on iaxy's and the other half are connecting to channel banks. The users on ixay's are getting the stutter dialtone on new voicemails, but the ones on the channel banks are not. Currently, all users are in the default context in the voicemail.conf file. I've tried the

Hey all..I'm looking to start with a single FXS card but with 3 extensions for VM purposes only. I'd like to know if there is a way that you can have different stuttering dialtones depending on which extension has a VM. For example If x103 and x104 have VM can there be a distinctly different stutter for each mail box and have them play back at once or back to back so that when you

NENA i2 The NENA i2 architecture was designed to support the interconnection of Voice over Internet Protocol (VoIP) domains with the existing Emergency Services Network infrastructure. This overview will describe the functional elements and call flow of a VoIP 9-1-1 call over the i2 architecture. The NENA i2 architecture was also designed to utilize existing 9-1-1 voice and data links to all

Hi I'm having a problem with ipfw, I think. For some reason it denies packets randomly for example: PING 196.14.239.2 (196.14.239.2): 56 data bytes ping: sendto: Permission denied ping: sendto: Permission denied 64 bytes from 196.14.239.2: icmp_seq=2 ttl=63 time=0.258 ms 64 bytes from 196.14.239.2: icmp_seq=3 ttl=63 time=0.233 ms 64 bytes from 196.14.239.2: icmp_seq=4 ttl=63

I've noticed on a few installations that the very first audio played after a call in answered (eg: Greeting), the first part of the audio is cutoff/stuttered. Is this because Asterisk needs some RTP to create a sync for audio - and the first 1 second is lost? Should one play 1 sec of silence first? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi to all, I am sending a SIP REFER message to Asterisk from a VoiceXML application using the <Transfer> element to do a Transfer through Asterisk. I need to know if Asterisk supports the full features of the SIP REFER message because if i set 'bridge=true' in the <transfer> element of the VoiceXML application to supervise the call, Asterisk sends a NOTIFY message with

I have asterisk running on my personal computer and am using Kphone to connect to it. My provider is broadvoice which is Ulaw and I had kphone connected as GSM. The lag was terrible coming from Pots-->--Broadvoice-->Kphone. About 2.5 seconds! Going the other direction seemed fine. I did a: show translation recalc 200 and see that the translation time should be about 2 ms. When I do the

Hello, does somebody have a tool to convert a /etc/passwd to a smbpasswd with getting a valid Lan Manager and NT hash. or does anybody have a trick, how I can synchronise the /etc/passwd with the smbpasswd without changing a unix passwd twice (passwd,smbpasswd). Thanks, Martin *********************************************************************** ** Martin Schuster ** Nortel DASA Network