search for: subclasses

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

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digium card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where

hallo, i am really confused how nativ briging is working with asterisk, i use a asterisk server as central server and register another asterisk and an iaxcomm client to the server, all three have public ips on the internet. somtimes, when i call from iaxcomm to my asterisk, the calls go peer to peer (i can see it with tcpdump) but sometimes the get routed through the central asterisk server

I have an issue where * tries to route a call over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip

Hi I've 3 iax connections to my provider , each of them have own DID , PH1<----| | \/ PH2<-->|-----| <---------------------------> |----|<-- DID1 | A1 | <---------------------------> |ISP |<-- DID2 PH3<-->|-----| <---------------------------> |----|<-- DID3 I had iax phone on each of this connection , but now I want to terminate all

Hi, I have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable

I was wondering if anyone had seen this problem before and/or could offer any insight into what the trouble might be: I have an Asterisk machine that it set up as a mutual friend with another one (in another state... about 150ms away). Calls between the two fail to get established approximately 50% of the time. When a call works, everything is fine. When one fails, however, I see a large

What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Any assistance? -----Original

I'm new to this list, and I apologize if this is an already answered question, but my Google-fu was not strong enough to find the answer if it was. I'm having a problem with DTMF on incoming IAX calls. For the first few seconds of the call (between maybe 1 and 15, it varies from call to call) everything works fine. After that I continue get DTMF_E messages from the remote IAX server

Can someone explain what the problem is when I use the generating function? And how to get debug() to stop in the Superclass initialize method? ---- source ----- setClass("Superclass", representation(id = "character"), contains = "VIRTUAL") setMethod("initialize", signature(.Object = "Superclass"),

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

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

Hi, My termination with sixtel stopped working, is it something I did or anybody else is having the same problem. I am attaching log: *CLI> -- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing

Hi, I've managed to get HYLAFAX---->T38MODEM----->ASTERISK---->CISCOAS5400 working, but when they are negotiating asterisk drops a message telling "Unknown RTP codec 96 received from gateway" Do somebody know how to fix it ? Thank you ! << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Control (4) SUBCLASS: Answer (4) ]

I had a working 1.0.9 asterisk installation and tried to get a Firefly IAX phone to register, but it was failing. I upgraded to asterisk 1.2.5 and the PBX is working fine, but the IAX phone still won't connect. Below is my iax.conf and the output from setting iax2 debug while the phone tries to connect. Could somebody please give me some pointers? This doesn't seem to be a normal

This may not strictly be an asterisk question, but not sure where else to post ... I have an Asterisk test server setup with two firefly clients, one on the local lan and one on an external ip address. Both clients are setup the same way and voice calls work fine. The asterisk console reports a "Registered" message for the external client at about one minute intervals but the

I''m trying to use the wxLC_VIRTUAL style of the wxListCtrl, which seems to require that I subclass the control (as opposed to being able to provide a data model a la Swing or Cocoa), and I''m running into trouble when loading the control from an XRC file. Apparently I need to specify the name of my subclass in the XRC as follows: <object class="wxListCtrl"

Hi, I have a duplicated DTMF issue with, it appears, bridged IAX channels. I have the following setup: PRI IAX <-------->* PSTN <------->* Dialplan I've configured a number on the dialplan server to make and outbound call to the pstn. This call then comes back into the dialplan server to SayDigits(). I'm seeing that a few of my digits are being duplicated

...or FR64 and VR128 (XMM0-15). The coalescer will join these classes as follows: RFP32 + RFP64 -> RFP64 FR64 + VR128 -> VR128 This seems perfectly reasonable - choose the larger spill size and avoid losing data. TableGen thinks these classes are unrelated - it currently defines register subclasses as follows: A is a subclass of B iff A != B and A.Regs is a subset of B.Regs (subset means improper subset) and A.SpillSize == B.SpillSize Since the spill sizes differ, FR64 and VR128 have no sub/super class relation. I propose that we change the definition to: A is a subclass of B iff A != B...

Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register => username:password@speex.dyndns.org and I cant get it to work. Maybe someone who already got this to work will help... When dialing my speex extension I see the next output from consol: IAX2 Debugging Enabled *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: