similar to: [asterisk-dev] DTMF queuing

I have done some searching and not sure this is even possible, but here it goes... **Scenario** Let's say you have an asterisk server that you use to connect to a SIP provider that you push your PSTN-bound calls to using g711 and out-of-band DTMF. The SIP phones in use are Cisco 7960's and are set to also use out-of-band DTMF. For the most part, everything works great. However, a few

Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can

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!).

So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello all, I have a DID from nufone, transported via SIP to my * box, and even though i'm using rfc2833 DTMF i'm still getting double digits and all sorts of other stuff... sip.conf is as follows: [general] port = 5070 ; Port to bind to disallow=all ; Disallow all codecs allow=ulaw allow=alaw allow=ilbc allow=gsm dtmfmode=rfc2833 register =>

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

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

HELP! Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com

device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip

I have a telco providing DTMF inband, they say they can't provide it any other way. This is creating headaches for me. What is the common method for SIP DTMF? Kpml, or 2833 or inband? My handsets don't support inband so I'm tying up some expensive resources to convert the inband DTMF to out-of-band DTMF... Can you recommend a vendor in US that provides SIP with DTMF in RFC

Hi, we have an Asterisk server basically passing on calls using the Dial application. In the pjsip endpoint settings, the dtmf_mode is set to audio. This works with most calls. However, there is a scenario where DTMF tones don't get forwarded the way I would expect them to get forwarded. A: Caller without RfC4733 support B: our Asterisk, version 17.6.0 C: Another Asterisk, with RfC4733

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

Hi all, I have problem using DTMF on Snom Phones (300, 320 and 360) I read they use in preference out-of-band DTMF , and if the remote system does not support it they default back to inband. I would like to use DTMF as out of band , and I defined dtmfmode=rfc2833 in the peer configuration. Nope, I am no able to access any ouside services using DTMF; Another kind of phones, ATCOM AT320, can be

Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008. The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF. The issue I'm

I have a DID but for some reason is not working in asterisk-1.6 The same sip connection in asterisk-1.4 is working OK, but it doesn't work with asterisk-1.6 Here is my sip.conf section: ... [actio-out] type=friend secret=password user=48746612254 username=48746612254 fromuser=48746612254 authname=48746612254 callerpage=48746612254 fromdomain=sip.actio.pl host=sip.actio.pl insecure=very

The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. However, this presents another problem. When I'm using g729 to place a call, I get the warning "Unable to process inband DTMF" because inband is not supposed to work with g729 (although it does seem to work when I've tried it so far).

Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in

Are there any known problems converting dtmf from oob over iax2 to inband over rtp/ulaw? Obviously it works when converting to inband over pri/ulaw et al, but how about rtp? I've got packet traces that confirm that 2833 packets are properly generated when I have 2833 configured for the rtp link, but the other side seems to be ignoring those packets. So I tried inband on that link; nothing

Hi. I have sjphone installed on windows and working except for dtmf. I read the docs for sjphone and it uses inband dtmf. I configired dtmfmode=inband but it still does not recognize it. Someone on the lists said that inband only works using alaw or ulaw but i tried only allowing that too but still no go. Hmm.. any other ideas? I can't get any other client to work on windows :-/ I