similar to: Asterisk can't connect voice

Hi I have the last firmware for zplex, if you like i send it to you, about the second question 24s means 24 extensions so you can configurate as you wish as fxo or fxs. Att Yelson Vivas

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Hi Everybody I'm connecting a zplex 10B to a TE410P card on my asterisk server (i use a cross cable), the zplex doesn't show any alarm neither the server. I'm trying to make calls from 1 line from the channel bank, to another, but when I dial the ext number, the dialtone doesn't stop... every way i make the call and the called extension rings but when i pick up the phone i just

Hi Guys I am new working with lumenvox products, and unfortunately I had not been able to install it properly, I follow the steps in lumenvox site and it looks like it works I mean: ========================================================= [root at pbx-millenium examples]# ./example 127.0.0.1 Connecting to 127.0.0.1 Interpretation 1: 8587070707 count=0, decode returns 1 Interpretation 1:

Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)<====iax2 trunk====>(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

hello all, I still experience the random off-hook on my fxo cxhannels, i am using a zplex-10b channel bank. which does not allow me to call out. The situation still persists...this is what i have in the zapata.conf [channels] context=internal context=incoming context=default usecallerid=no usecallwaiting=no signalling=fxs_ks channel=1-8 signalling=fxo_ks channel=16-24 but i still have the the

I've got a Wildcard T100P along with a Zhone Zplex 10 24S/O which has been working fine for me now for a while. These have been pulled out of a working Asterisk installation (as they were no longer required) to use at home only to find that the fan noise is too loud. As such I'm looking to sell off this hardware and replace it with some combination of fanless hardware that will allow me to

Hey guys, I've had a working Asterisk setup going for a while now, but am having problems with the Zhone Zplex 10b thinking that during ringing, an extension has answered the call when infact it hasn't. This only seems to happen on some of the ports and doesn't appear to be specific to the handset. Does anyone have any suggestions as to how I could go about fixing this (aside from

I'm hoping that someone else has seen this problem. Running CVS version from 03/08/03 20:00 T100P -> Zplex-10 All internal call routing seems to work fine. Atempts to make calls on the FXO interface fail. Asterisk picks up the correct number dialed but there seems to be a problem with the Zplex transmitting that number on the POTS line. All attempts to dial out fail with the LEC giving

Mine has been working well, but the only problem is that it doesn't support callerid (from the POTS side). > -----Original Message----- > From: John Schmerold [mailto:john@katy.com] > Sent: Tuesday, 5 August 2003 12:37 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Zhone Zplex 10 units > > Thanks for the Zplex heads up. > > Steven

hello * users, i am using a zplex 10b(8fxosand 16fxs) connected to a T100 digium card, RH8.0. soon after running * i get this warning "file chan_zap.c,line 4155 (ss_thread):CallewrID returned with error on channel 'zap/7-1' then executes the other applications itself. --executing Wait("Zap/7-1","1") in new stack. -- executing..... it gives the same warning

This may be a long shot, but I suppose it can't hurt to ask! I have seen chatter regarding Zhone channel banks amongst the asterisk users primarily. In all of my searching I am guessing that the majority of people using the Zhone's are probably reading this, and with the support that I have seen from Zhone so far this is probably my best bet, I recently picked one up for a project

Hi Guys i've trying to set a mfcr2 systembut i can't find the working combination between -asterisk version -spandsp version -unicall version i can compile but loading the pbx shows [chan_unicall.so]Dec 5 23:04:42 WARNING[27865]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: dtmf_put Dec 5 23:04:42 WARNING[27865]: loader.c:554 load_modules:

Greetings, I have just purchased several Astra 390 phones ready for asterisk. I have placed a line with adsi=yes in the Zapata.conf file just before channel => 13 I have also added an extension exten => 6199,1,ADSIProg(asterisk.adsi) exten => 6199,2,Hangup in the extensions.conf file. When I try to program the phone I get the following: Asterisk CVS-03/28/04-12:02:10,

Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GS d? s:- a- C+++ UL++++ P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e

Is there a way to initiate a transfer using an analog handset? For instance I'm looking for a way to do something like the following: External call comes in and is answered by user A. After talking to the caller they determine that the caller really needs to speak to user B. Is there any way for user A to initiate a transfer to user B, using only their analog handset? Now to make

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'm traying to link 2 * servers using SIP and the following errors was show: "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13

Hello All, I'am a new Asterisk user, and i have the following question. The following is the Status of all open channels from my Asterisk system, which was received through the Asterisk Manager Interface ((AMI)). ==================================================================== action: Status actionid: 65066874_3# Response: Success ActionID: 65066874_3# Message: Channel status will