similar to: Oneway Audio

Hello everyone: I'm a new member of this group. I have a question about "oneway.test". When I use "anova(lm(....))" to analysis the ANOVA, I can get the information about Sum Sq and Mean Sq. (The R code and the results are as follows.)

Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the "fixups" disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the

Hi all, i have a little problem to understand this warning message, it's annoying and it cause a lot of spurious in the log files. Im working with this scenario: a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are always routed to this. a list of sip UAs that potentially can use any codec apart g729/g723. I setup the asterisk to do as mediaproxy so directmedia=no and

We are getting the following on 1.8.3 and 1.8.4-rc2, HELP! Why is Asterisk unable to transcode to/from ulaw and g722? [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722) [2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit frame type 256, while native formats

Hi list! I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?

Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console: WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write) We found that line in function "sip_write" inside "chan_sip.c". In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared

Hi What is special control 16? I am getting this error quite often -- special control 16, then for some reason it puts on hold and then logs is full of Asked to transmit frame type ulaw, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) Both peer and trunk have same codec priority (disallow=all then allow=alaw then alllow=ulaw) Any ideas how to fix this ? --

Dear all, I am getting the below errors when using asterisk. I am using sjphone for testing purpose. Below are the setting for sip.conf and extension.conf When i dial the number it rings on the remote telephone. but after ringing 1 time it will disconnect. Can anybody tell me what does this error means and the how to solve this issue. Thanking You, Joel sip.conf [general] context=default

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

Hi, For some time now, I've been fighting with RxFax and Asterisk. I had it working for some time, however, for some reason it just stopped working, I guess someone updated Asterisk or something, don't know exactly. At the moment I keep getting errors while entering the RxFax stage of a call. But due to the fact RxFax does not contain any code to directly interact with an RTP stream,

Good afternoon list. I'm experiencing a problem with my SIP channel's. When I have an external connection for one of my SIP carrier's, I can listen to the client and the client listens to me normally. The problem is when I will transfer this connection, the call is mute for the extension I have transfered. Only the client hears normally. In the console of Asterisk generates the

I'm trying to pick up a ringing SIP phone (203) across the office with exten => *9,1,Pickup(783743) where 783743 is the local part of the number that our ISDN works on. I tried all of these first: exten => *9,1,Pickup(203) exten => *9,1,Pickup(SIP/203) exten => *9,1,Pickup(203@internal) and got a "declined" message back from my phone (snom 300), so I then

Hi everyone, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange. Anyhow I was getting an error: Process_sdp: No compatible codecs! And from

Hi, sorry to bother u all, i have a trouble when I call a did number forward to my asterisk server, the server told me: [Jul 28 19:00:57] WARNING[28080]: chan_sip.c:3806 sip_write: Asked to transmit frame type 256, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 28 19:00:57] WARNING[28080]: chan_sip.c:3806 sip_write: Asked to transmit frame type 4, while

Hi all, I seem to only be getting (1) call to sip_write() in channels/chan_sip.c I have a very simple setup. one server (no cards) 2 polycom IP 330 phones. Server is 192.168.1.150 and phone is DHCP. Nothing else on the network. No firewall is enabled. I call into the dialplan with: exten => 112,1,Answer exten => 112,n,Playback(demo-congrats) exten => 112,n,Hangup I see this executing

Hi there, I'll need some help with this: Trying to establish an IAX2 link between two servers works in one direction (SIP client with ulaw), but not in the other (SIP client with GSM). The client used for this is X-Lite behind NAT while both servers have a public IP (I playback an anouncement before trying to connect to the second *). Error on the originating * server: