similar to: fax & asterisk 1.2

hehe, only for a test _________________________________________________________________ ???? Windows Live Mail? http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/mongrel-users/attachments/20070126/b72e725e/attachment.html

i have a question with queuing discipline " if i create many classs with sfq and cbq and ... i would like to know about qos . which queuing discipline will be chosen first. ( not use priority ) (queuing discipline) sfq---------> packet incoming packet outgoing cbq--------> Thank you .

Hi all, I have a question: how to configure Asterisk to support SIP "INFO" method? I encountered this problem when I find my UA don't send "INFO" message to another UA, actually it should. Asterisk was used as a SIP proxy in this scenario (I know that SIP is not a SIP proxy:-)). Then I captured all the packets and found that my UA send "REGISTER" to Asterisk,

Hello Trevor, I wonder how I can find out for sure what is the H/W version for a PROSODY ACULAB SS7 Card? I dought that I have a ver 1.1 which have may issues with recently made computers. I have a case opened for my problem with aculab but sysdiag shows that I have ver 1.1 and aculab says I have ver1.5! can sysdiag be inaccurate? I tried config summary and I had the same result! wish you

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 all, I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk SVN-trunk-r15187 to avail the PARKEDAT variable. - I park the call using ParkAndAnnounce - plays moh. - accept the call using ParkedCall The following errors are coming on the console and there is oneway audio - no audio after Music-On-Hold at caller's side. Please advice. I am testing using cisco 7902

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,

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?

hi scott! well here is my smb.conf ... : (now linux is my wins server) # Samba config file created using SWAT # from 0.0.0.0 (0.0.0.0) # Date: 2003/05/06 15:57:41 # Global parameters [global] workgroup = SCIENCEU netbios name = TT security = DOMAIN encrypt passwords = Yes password server = srv-win-2000 ; wins server = 11.1.9.100 wins support = yes winbind uid = 10000-20000 winbind

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, I'm trying to get Supreme command forged alliance to run on ubuntu 11.10. I've finally got the game installed and can run it. However whenever I try and play a game (skirmish, training, campaign etc) the game will crash during the in transit loading screen. Dell studio 1745 64bit ATI mobility Radeon 4650 4GB ram Here is the end of the terminal output: Code:

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,

After lots of trouble with an ATI Card (better say ATI Linux drivers...) I switched back to an nVidia Card (GeForce 7600GT) and tried to *finally* get NWN2 working with Wine (1.1.3 on debian Lenny AMD64, 2.6.26.1). Well - the Game starts, but first thing that doesn't work as it should is the "graphic options" menus. Some Lists are empty, e.g. for the "Texture Filters" or