search for: sipdtmfmode

Greetings. I'm having a bit of a problem using the sipdtmfmode app. I have two incoming paths to * from pstn via FWD that use differing dtmfmode. IPKall wants rfc2833, libretel wants inband. If I set dtmfmode= in the fwd peer config in sip.conf each works seperately, and I'm trying to use gotoif and sipdtmfmode to switch based on the CID calling. Outpu...

...ll 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!). So, I did some digging around, and figured I might try SIPDtmfMode to change my DTMF mode when I'm calling out.. that resulted in a prompt crash, and the info included below out of gdb. Is it me? Am I misunderstanding the appropriate use of SIPDtmfMode? If so, that's fine, just bonk me on the head with a yellow pages book or something.. Also.. how can I...

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?

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=*****

...ions, 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. I have added SIPDtmfMode to my outgoing extensions, that still doesnt help. Does anyone out there have experiance or ideas with this setup?

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 online with the Xtra Broadband Channel http://xtra.co.nz/broadband

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 !!! Any clues???...

...arsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' And my device(s) won't register. I don't even see them attempt the registration...(from the CLI in vvvvery verbose.) Maybe I'm not using the right version of asterisk??? Is that possible and how would I know? My "show version" gives me this: *CLI> show vers...

...chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' Problem 3) Sometimes the program crashes during (at end of) the startup sequence. Warning about "flexibel rate not heavily tested". Is this just a codec I can conf...

...ec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not working. #######################extensions.conf####################################### [local] exten => _XX,1,Set(DYNAMIC_FEATURES=nway-start) exten => _XX,2,SIPDtmfMode(inband) exten=> 10,3,Dial(SIP/saad,,tT) exten=> 10,n,Hangup exten=> 11,3,Dial(SIP/riz,,tT) exten=> 11,n,Hangup exten=> 12,3,Dial(SIP/rehmat,,tT) exten=> 12,n,Hangup [dynamic-nway] exten => _XXX,1,Answer exten => _XXX,n,Set(CONFNO=${EXTEN}) exten => _XXX,n,Set(MEETME_E...

...all: 10100 [193.82.116.194:4569] USERNAME : vikki DATE TIME : 2007-10-29 19:46:36 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00004ms SCall: 10100 DCall: 00002 [193.82.116.194:4569] -- Executing [01905888007 at sip-default:1] SIPDtmfMode("SIP/213.166.5.134-086112f8", "inband") in new stack -- Executing [01905888007 at sip-default:2] NoOp("SIP/213.166.5.134-086112f8", "01905888007 ") in new stack -- Executing [01905888007 at sip-default:3] Dial("SIP/213.166.5.134-086112f8",...

...on 0.0.0.0 port 5036 [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver) [chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver) [chan_agent.so] => (Agent Proxy Channel) == Registered channel type 'Agent' (Call Agent Proxy Channel) == Registered application ...

...erisk/sip.conf': Found == SIP Listening on 192.168.0.10:5060 == Using TOS bits 0 Connected to database 'asterisk_config' on 'localhost' as 'asterisk_user' == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Apr 16 14:10:34 WARNING[1074449120]: chan_iax2.c:6218 load_module: Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Connected to database ...

Hello I try to set up voicemails for extension. When VoicemailMain gets called, it prompts for mailbox and password. It seems not able to read from the phone. So the authentication always fails. I desparately need help to understand what is wrong. Here is a part of my extensions.conf: exten => _8500, 1, Wait(2) exten => _8500, 2, VoicemailMain(${CALLERIDNUM}) exten => _8500, 3, Hangup

I'm having an issue sending DTMF to cisco dialing this extension I should hear the dtmf tone RTP playload 101 has been sent to the cisco phone, but no audio. in the dialplan exten => 8603,1,Answer(1) exten => 8603,n,sipdtmfmode(rfc2833) exten => 8603,n,SendDTMF(1|100) exten => 8603,n,hangup() sip.conf dtmfmode=rfc2833 SIPDefault.conf I did play with all possible settings for dtmf_outofband: avt, avt_always, none and 0,1 for dtmf_inband nothing happens cisco 7905g is working OK with this example cisco 7960 firmwa...

...to inband (is this even a proper way to handle 2 DTMF modes?). [sipprovider] type=friend host=xxx.xxx.xxx.xxx disallow=all allow=ulaw maxexpirey=15 dtmfmode=rfc2833 dtmfmode=inband nat=no insecure=very canreinvite=no I have searched and searched and the closest thing that I have found is "SIPDtmfMode" but from what it looks like it needs to be initiated before the call is placed. By the way - the reason inband is not being used is that digit accuracy is terrible with the inband setting. Any thoughts are appreciated.

...rt = 5070 ; Port to bind to disallow=all ; Disallow all codecs allow=ulaw allow=alaw allow=ilbc allow=gsm dtmfmode=rfc2833 register => username:secret@myprovider.com/myDID just to make sure everything is set properly, i even threw in exten => myDID,3,SIPDtmfMode(rfc2833) in extensions.conf to make sure that dtmf is coming over in rfc2833 and not inband. if it helps, the provider in question is nufone and jeremy from nufone told me to use only rfc2833. anyone else have this problem? any pointers or ways to solve this? it's making me insane...if anyon...

...o, I always get == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 64.1.16.172:5060 == Using TOS bits 4 == Parsing '/etc/asterisk/sip_notify.conf': Found == Registered application 'SIPDtmfMode' Is there any other way to make the port change work? Also I never got an answer about how to prevent unregistered sip phones from sending inbound SIP calls. I can send calls regardless if my softphone is registered or not, when autocreatepeer=no. This is flaw that makes Asterisk very insecur...

Hi, In my dialplan I'm using a WaitExten() command. It works only with Zap phones. When I dial this command with Sip phone asterisk do nothing. Should I put extra definition in sip.conf to make this work with Sip phones? Thanks in advance Cheers

When I fail to pick up a call from Asterisk to the PSTN to my cell phone and let it go to voicemail, the sound quality is always really bad. When I call my cell phone's voicemail a few minutes later, it's really garbledy and sounds clipped or something. I've tried using Monitor to record the sounds that are being played to my cell's voicemail, and the monitored sound sounds fine