similar to: IAX2 Trunk to PSTN (voicepulse) questions...

Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound calling. Our attempts at IAX/IAX2 connectivity with VoicePulse have been less than successful. We get "Registration Refused" errors from Asterisk whenever we launch the server. The front-line support folks at VoicePulse suggested that we are

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

Yes, and they are aware of the problem. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan Tew Sent: Monday, February 02, 2004 1:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoicePulse IAX2 lag Is anyone else noticing high lag on their voicepulse IAX2 connections? We're seeing

I am currently trying to transpose some large panel data set ie transposing multiple rows in into a single column. instead the transpose functionality transposes all rows into columns. my sample data set looks like below: ACCT_NUM ACCOUNT_NAME TRAN_AMT DATE EMPLOYER 101913 GK 7489 30-Apr-10 PENSION 101913 GK 7489 30-May-10 PENSION 101913

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]

Hello Everyone,   This is my first attempt to do something in R. As a precursor to a Willingness to Pay analysis, I want to conduct a Mixed Conditional Logit analysis but am unsure how to proceed because of some nesting within my data.   Below is some data and code that illustrate what I’m trying to do. The data are based on responses to a conjoint survey obtained during pilot testing. In the

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

Dear experts, I try not to trouble the list again after this question. I want to print this vector into a file > myvec [1] --Control --Control --Control --Control --Control HBA2 HBA1 [8] HBA1 --Control HBB --Control HBB HBA1 MBP [15] --Control HBA1 HBA2 HBB PTGDS GAPDH UBC [22] --Control GAPDH TPT1 HUWE1 PRM1 CKM

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl

I am wondering if there is a way to create a SIP/IAX group of outgoing lines like Zap groups. I am currently using the following method, but would like to use features such as ?g2? that would list all the accounts for a SIP or IAX connection. exten => _1NXXNXXXXXX,1,Dial(SIP/account_name:Password@gw5.voicepulse.com/${EXTEN }) exten =>

Hello- I am in the process of adding a new provider to my asterisk box (both for outbound termination as well as inbound DID). They are going to be delivering and receiving traffic via SIP only. Now, in IAX via Voicepulse or others I know that I can simply have one registration statement along with an inbound context, then in extension.conf map the outbound context. from iax.conf: register

Ever since upgrading to 1.6 I get messages like these. I want everything else that shows up, but is there a way to make all the dns messages go away? Ira > doing dnsmgr_lookup for 'gw5.telasip.com' > doing dnsmgr_lookup for 'sipconnect.ipcomms.net' > doing dnsmgr_lookup for 'proxy.ideasip.com' > ast_get_srv: SRV lookup for

Hi there, I'm trying to solve 2 nonlinear equations in 2 unknowns using the BB package. The first part of my program solves 3 ODEs using the deSolve package. This part works. The output is used as parameter values in the functions I need to solve. The second part is to solve 2 equations in 2 unknowns. This does not work. I get the error message "unexpected end of input". So what

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

I came across a problem when one of our clients was not able to access any of the servers on our network. This person has never connected to us before and now for this first time was trying to do it from his home is Houston, TX using earthlink cable service provided by Time Warner. All this information, I think, is important because when I started examining my shorewall logs I found out

This is a patch repairing a regression in code previously functional in 4.1.x. It appears that, during some refactoring work, calls to hvm_memory_event_cr3 and hvm_memory_event_cr4 were lost. These functions were originally called in mov_to_cr() of vmx.c, but the commit http://xenbits.xen.org/hg/xen-unstable.hg/rev/1276926e3795 abstracted the original code into generic functions up a level in

Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5