similar to: sipdtmfmode problem

Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode: [fwd-rfc] type=friend secret=*****

All, Here's a cool one.. I was attempting to call a retarded conferencing service, and was having problems with it picking up my DTMF.. after trying all the settings my Sipura SPA2000 offers, I found inband actually works.. unfortunately, I can't get anything else to pick up my inband DTMF (including asterisk's builtin voicemail! It just times out and says I never entered a login!).

I am seeing some very peculiar things in the routines that REGISTER my * server with several accounts. I saw this on my console: . . . NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout):

Can't find a way to control the dtmf mode on a per session basis with pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any hints on how to do this?

hi list, is anyone succesfully using asterisk with libretel port-of-call (www.libretel.com)? If so, i would be grateful for configs..i set up libretel to forward to mynumber@myserver.com:5070 (asterisk is running on 5070 and SER on 5060) and when i call the number i see SIP messages with ngrep but the asterisk CLI doesn't seem to catch them. I assume i need to register...is this even possible

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

I have succeeded in getting two tinc1.0pre3 hosts to connect, but I can't ping one from the other. I can ping the local tap interfaces. tcpdump shows that icmp echo requests are received by the other host, but no replies are sent. The two hosts are North and South. It is the same pinging North to South and South to North. The hosts are configured as follows: South: Debian 2.2

Folks, from googling, I see that the dtmfmode parameter is not valid in the [general] context. My problem is that my overseas DID through Libretel seems to want to come into the [general] context! And, having done that, I get my welcome message, but then the DID does not accept the DTMF when I try to dial an extension! It plays the welcome message, waits, and then times out (and hangs up

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

Hi everyone, I have this code in Person.rb: def hashed_password(password) require ''digest/sha1'' require ''base64'' new_password = Digest::SHA1.new new_password << password new_password = "{SHA}" << Base64.b64encode(new_password.digest).chop end And here''s my test: def

Hi - Linux: Ubuntu 9.10 all updates applied as of 9/5/2010. Samba: 3.4.0 smb.conf global section: [global] netbios name=blahblahblah sever string=Domain Master workgroup=blahblahblah encrypt passwords=true wins support=yes domain master=yes local master=yes preferred master=yes os level=255 security=user domain logons=yes passdb backend=smbpasswd logon script=%U.bat add machine script=sudo

List, from the Dutch "Computable" paper ICT tabloid and daily ICT subscribable mailing-list and so-called verwittigingsbrief/ad hoc news letter, partly translated into English by me ;) Dutch readers only: http://www.computable.nl/nieuws.htm?id=524000 "Manage Windows clients from Linux servers" blahblahblah Zenworks 7, the latest release of Novell's sysadmin suite".

Hi Folks, This has been driving me nuts all day. I've been unable to google myself out of it. Maybe someone here can help? I followed the instructions from: http://www.tinc-vpn.org/examples/macbook-install/ Ubuntu server <-> Ubuntu server works just fine with the same config. Ubuntu server <-> Mac laptop not so much. The error I'm getting is: 2012-04-10 21:48:44

I am now at a total loss. Using Sipura spa-2000s connected to *, I get DTMF working just fine for internal extensions, voicemail, etc. If making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I get no dial tone. I am working unsuccessfully with Cisco right now on this, but they cant find anything wrong. I have tried all suggestions I can find from the list and elsewhere.

Brad, now I see that there is the Mount Point, but no audio is heard. Maybe the name of the Mount Point (66.228.49.182.ogg) is not correct ?? ... Try to rename the Mount Point. I think you should put blahblahblah.ogg, with only one point (between the extension "ogg" and the rest of the name): <mount> <mount-name>/blahblahblah.ogg</mount-name>

Hi, I've recently started using Samba4 (Sernet binary, 4.0.16 on Centos 6) as PDC. I would like to assign unix attributes using MMC console "Active Directory Users and Computers" however every time I'm trying to do this I get an error "Unable to modify the object property values. Check your credentials. There could be a network problem blahblahblah".. I've

I have a model for a task class. When I try to save a task, for some reason no INSERT statement is being generated. Here''s the pertinent part of the development.log: ------------------------------------ Processing TasksController#create_fromProject (for 70.247.24.238 at 2006-03-08 14:20:30) [POST] Parameters: {"commit"=>"Create",

Hi, can i somehow use the other buttons to execute some apps, *without* hanging up the call? Something like: exten => s,1,Dial/SIP(1234)|4,5,7,9 exten => 4,1,Monitor(wav) exten => 5,1,SIPDtmfMode(inband) exten => 7,1,AGI(turnoncoffeemachine.agi) exten => 9,1,System(smbnuke boss) Regards, AA _________________________________________________________________ Watch movie trailers

Hi I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also tried inband) and I get the following error: june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets

All, I run a bunch of (well 20+ actually) Asterisk boxes at home, work, friends and the lie with our own dialplan in the form 8EEXXXX where 'EE' is the exchange number and 'XXXX' is the extension number. This arrangement has been in for 2+ years and worked well with a central box (asterisk.thorcom.net) acting as the routing hub and SIP exchange point with various public