similar to: Conecting to asterisk server through NAT using IAX

Hallo Did you allow udp outgoing on 4569 as well.. i found udp bit different than tcp when comming to firewalls liaan ----- Original Message ----- From: "Bartosz Wegrzyn - asterisk" <junk@lexon.ws> To: <timebandit001@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, February 21, 2005 12:29

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

I'm experiencing a strange problem. My inbound calls are hanging up right after Background() message even though response timeout is set to 10 sec. [voicepulseincoming] exten=>_X.,1,Answer exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)

The problem was that autofallthrough=yes was set in extensions.conf > I'm experiencing a strange problem. My inbound calls are hanging up > right after Background() message even though response timeout is set to > 10 sec. > > [voicepulseincoming] > > exten=>_X.,1,Answer > exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1) >

<P>I am running on Core 3 also with a voicepulse account.</P> <P>I found this document quite helpful....www.voip-info.org/tiki-print.php?page=Asterisk+Fedora+Core+3</P> <P>I did deviate in that I ran my make of Asterisk itself as follows</P> <P>cd /usr/src/asterisk</P> <P>make clean <BR>make linux26 <BR>make install <BR>make

Hi all, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see

This is BAAAAAAAD! Now even SIP get's "tainted"... http://slashdot.org/articles/04/09/16/1317247.shtml?tid=95 _________________________________________________________________ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream

Dear all, I have a question about "union". "union" handles two vectors' elements to get their union, but I have many vectors and I want to get the union of all of them. So I wrote a loop: all <-c(); for(var in ls(pattern="xyz")) # all of the vectors with pattern of "xyz" { all <- union(all, var); } but the result is like this:

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

G'Day List, Can someone help me out a bit please. I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying to install * I am following http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation After running: cd /usr/src/asterisk make clean make make install make samples The instructions says: Configuring Asterisk - Login to your server as user

Hi all, I have got some problem with samba and the browsing. I've got a small network with 4-5 computers wich are using windows xp home and pro. There isn't domain, just a workgroup. The samba serves the master browser and wins functions. The problem came forward when the clinents get ip address via dhcp. The situation is the following: the clients are dissapearing slowly from the browse

The documentation for agrep says it uses the Levenshtein edit distance, but it seems to get this wrong in certain cases when there is a combination of deletions and substitutions. For example: > agrep("abcd", "abcxyz", max.distance=1) [1] 1 That should've been a no-match. The edit distance between those strings is 3 (1 substitution, 2 deletions), but agrep matches

Hi All, We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one day it stop to response to the SIP Clinets so they cannot make call or register. But safe_asterisk not restarting it back because asterisk running without any response to the sip clients. When we try to do 'core show channels' using Manager it returns only Action: Command Command: show channels That time

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

I posted this yesterday to r-help and Ben Bolker suggested reposting it here... Dickison, Daniel <ddickison <at> carnegielearning.com> writes: > > The documentation for agrep says it uses the Levenshtein edit distance, > but it seems to get this wrong in certain cases when there is a > combination of deletions and substitutions. For example: > > >

Hi everyone, THis is my second thread regarding the issue.(before I was having problems with accessing my email, which slow down my responses, sorry for that) My setup looks like this Firewall | | Asterisk---Asterisk (two asterisk servers with the same setup for high avail) | | phones Ports 5060, 10000-20000, 4569, 5036 are forwared to 192.168.1.251 which is virtual ip address on one of the

Hi! As you can see this i a print of my configfile, that i made with SWAT. But i can't seem to be able to connect to it from windows, an i can't se what i've shared when i browse Microsoft Network. And if i shall map it i need to give the root username and password. What can i do to be able to login as a guest? Please help me. And please send the answer to my e-mail

Hi, today we had to hard-reboot one of our dovecot proxy backends and after boot up, lot's of accounts were showing this error and appeard empty: Error: Corrupted transaction log file ... After deleting the indexes (not uid file, problem wasn't in deleting indexes, i'm doing this often), e-mails showed up ok but e-mail clinets started to download them again. Why this

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is