similar to: Caller ID Routing using VoicePulseConnect

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

I want to set up a dial rule like this 9304752#w9#w+NXXXXXXXXX The point of this is. It will dial into a pbx with the account number 9304752, wait a second, dial 9 to get an outside line, wait a second for the outside line, and then dial the number to be called. When ever I save this in amp anything after the first # disappears. When I try to call out it doesn't do the adding of the

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

I have a data set with observations on 549 cities spanning an 18 year period. However, some of cities did not report in one or more of the 18 years. I would like to implement the procedure suggested by Wooldridge section 17.1.3 in his "Econometric analysis of cross section and panel data" to correct for attrition. For example the table below indicates that the 3rd and the 7th cities in

Hi all, a noob here, I am trying to get outbound calls through asterisk working with Broadvoice. I have consulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register =>

I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail.

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Hi, I thought I'd repost this to the -users list for some background on the jitter buffer and its workings and remaining issue.s I'll also pu a little executive summary here at the top: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) The jitter

I did a quick google search of the lists site and couldn't find a definitive answer, so if it's there, I apologize for asking again. Starting about noon yesterday, I am no longer able to send/receive calls via Broadvoice. When calling in, I get a fast busy, and when calling out I get the following error: -- Executing Dial("SIP/112-572a",

Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: "xxxxxxxxxx" <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as23a29r59To: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060>Contact: <sip:xxxxxxxxxx at

I've tried everything with the * box after this weekend. I have read every document on the problems people are having with them after this weekend as well, but none of them address my problem. I checked my settings in my sips which I have below as well, I have changed the host file a few times, but this was new to me and I never had modified it before. I have and the same results

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

I have downloaded and installed Asterisk@home and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from

Hi, I've setup samba 3.4.7 to use idmap_rid as per the documentation: idmap backend = rid:DOMAIN=500-100000000 idmap gid = 500-100000000 imap uid = 500-100000000 It seems to work for groups: wbinfo --group-info="domain admins" domain admins:x:100512 PsGetSid v1.43 - Translates SIDs to names and vice versa Copyright (C) 1999-2006 Mark Russinovich Sysinternals -

Hello list- As a FreeBSD shop we've used Samba 3.x quite well for a couple years. With version 3.6 due to expire in due time, we've been experimenting with version 4.1 using winbindd with very limited success. We find that if we use the TDB backend instead of either RID or AD, we are able to enumerate our AD users via getent. I cannot enumerate AD users via either the AD or the RID

Hi I am testing the migration from our actual Samba domain, based on Samba 3.3.8 and LDAP (389DS) to Samba 4. I have followed the Samba4 Howto, and I have successfully compiled it. Now I am running the classicupgrade command, but I am getting some errors. First of them is that the script is ignoring the "ldap group suffix" parameter in smb.conf, and is always searching in the

Hi all! I'm configuring a samba PDC with an LDAP sam. Everything is working great except that when I do pdbedit -Lv root (which is my "Domain Administrator" account) I see that it's getting a Primary Group SID value of S-1-5-21-XXXXXXXXXX-XXXXXXXXXX-XXXXXXXXX-513 instead of the S-1-5-21-XXXXXXXXXX-XXXXXXXXXX-XXXXXXXXX-512 that is stored in my LDAP tree. Does someone

Hello Everybody, we have authentication problems with the mentioned configuration. Current situation: We have two Windows 2008 R2 domains (currently on 2003 level) in separate forests. Recently we created a two-way trust with selective authentication between them. A Debian Squeeze LTS machine running SAMBA 3.6.6 (latest Backports version) is member of our primary domain and provides file

There are times when I need to call a client from my cell and I want a recording of the call. I was trying to put into an * a way of doing that. Below is what I'm using in my extensions.conf exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5) exten=> X,2,SayDigits(${num}) exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify) exten=>