similar to: OH323: Dropping incompatible voice frame

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

Hi, When I call forward on PAP2, the incoming call will right the forwarded number. However, there is one-way voice problem. The caller can hear the destination(the forwarded number), but after the called party answers, the caller can't hear anything. Then the CLI> produce continuous errors as following: Oct 6 10:57:45 NOTICE[11026]: channel.c:1409 ast_read: Dropping incompatible vo ice

This is a serious problem! I have brought up this issue in four previous attempts to get some feedback. I find it hard to believe that no one else is having this same problem. Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible voice frame on Local/103@sip-00f3,2 of format alaw since our native format has changed to slin Apr 11 13:27:36 NOTICE[4446]: channel.c:1906

Hi, I'm having problems with the rxFax app. One of the messages that appear in my console is: Executing Set("SIP/something", "FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif") in new stack -- Executing RxFAX("SIP/something", "/var/spool/asterisk-fax/1137692307.5.tif") in new stack Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping

Having a problem here that I can't seem to find a fix for. PSTN call comes in, operator answers, transfers call to a phone behind an IAXy. Caller hears no sound after being transferred. IAXy can hear caller, but not vice versa. Client reads: NOTICE[11342]: channel.c:1950 ast_read: Dropping incompatible voice frame on IAX2/jim-7 of format gsm since our native format has changed to ulaw

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

Hello All I'm settup my asterisk as belows: sangoma card, connected with E1, CAS Signalling. I have two problem. 1. The asterisk don't received any DTMF when caller input to 2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error. Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', but no exception

Hello All i have big problem for unicall. my system work successful with sangoma card, E1 and CAS signalling (vietnam). when at the some time. i have trouble then my system is half (CPU instructions = 100) i tested for some case as belows: - When i dial, then my system became answer, the caller hangup. system error message show (loop without condition and half machine) Jun 11 12:15:45

Hi everyone, I am new to the list and new to Linux as well as to Samba. I've read quite a few howtos, man-pages and other docs on samba now. I configured my own SMB-server with smbd and nmbd. Of course I generated a smb.conf file and my server is accessible and running on the Linux mashine. I can connect with smbclient to the Linux-mashine and to the win2k-mashine (hostname and service

Hello Have a trouble. I have mashine with two interfaces. Samba running at both with the same name of computer. In wins-servers wins.dat i see next string: "GRAND#20" 1044171787 192.168.1.1 192.168.2.1 46R It's ok. But for ALL mashines in any interfaces with any IP (192.168.1.X or 192.168.2.X) returned first IP in list: ------------------ [2003/01/30 11:03:21, 3]

Hi there, first of all I want to thank everybody who gave me good advise so far ;-). Now it's the first time I can give advise, and I'm quite glad that I can start giving insted of only taking now :-) ! Hi Petr, Yes, I have solved my general problems as follows: First of all you have to check your firewall settings, because that was the major problem on my system. On RH 7.2 the

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hi When I make a call from the outside to asterisk and the call is asnwered, all is OK, but when I make a call from the outside to asterisk and hangup before the call is answered, you got this WARNING in the console: Jul 6 19:33:08 WARNING[10037]: channel.c:1521 ast_read: Exception flag set on 'UniCall/2-1', but no exception handler Jul 6 19:33:08 WARNING[10037]: channel.c:1521

Several times a day I get this meesage scrolling on one of our asterisk boxes: Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our

Hey geeks, I have been reading several faqs and howtos now and haven't come very far with what I really wanted to accomplish: I have installed samba from the RH 7.2 distro rpm. I have set it up so I can access my windows shares on the linux mashine. I can "ping athlon" (which is the netbios-name for the win2k mashine) I can "ping linux" (which is obviously my

hi, I'm currently trying to use Speex with Asterisk from my OpenH.323 client. It seems to mismatch the codecs, below is my log from Asterisk. My Openh323 client crashes in responding to a Speex request for bits per frame. I'm guessing it either isn't running the codec correctly or doesn't support the same subset of speex codecs as openh323. (I'm using speex-1.0.1 with

Hello! I trying to setup Domain Controller with Samba, but I have some trouble. As I understood, I ought to create "Trust Account" for every mashine, and if mashine name is "vasya" then account should be "vasya$" but FreeBSD is not allowed to create accounts with "$" characters, what should I do ? Thanks! /* spectre */