similar to: dtmfmode type

I appreciate your report (below), but it's a strange and disturbing coincidence for me. DTMF out through Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? ---- Date: Tue, 22 Jul 2008 12:23:39 -0400 From: "Mark G. Thomas" <Mark at Misty.com> Subject: [asterisk-users]

Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001>

Hello everybody, I have set dtmfmode=auto in my sip.conf, but that does not work and I still got the following message: WARNING[4980]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 According to http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode: auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones

We have an issue with incoming calls from a provider in which DTMF tones are sometimes sent using 'inband' and sometimes using 'rfc2833'. All calls are G711 and the incoming SDP never indicates support for rfc2833. Is there a setting in sip.conf that allows asterisk to receive DTMF tones in either inband or rfc2833 formats? The option 'auto' does not work for us

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that

Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked "sip show peer" and saw that Vitelity for inbound was now reporting "DTMFmode : rfc2833" (it didn't used to), so switched my ountbound dtmfmode to

My WIP-5000 phone is working well with my Asterisk box now, except for DTMF. All DTMF key presses come across as clipped or just clicks on the remote side. I had this problem with my Sipura ATA as well, but fixed that by playing with the settings on the Sipura device. I've tried dtmfmode=inband and also rfc2833, but neither seem to work. I don't see any place in the settings on the

It appears Asterisk can handle DTMF inband on only a limited selection of formats, of which G.729 is not one. The issue appears to be something involving "short data" -- whatever that is. (I'm inferring all this from looking at dsp.c in the vicinity of the error message I was getting, which pointed to line 1424.) What *is* "short data"? Is this really a show-stopper for

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

YUP, this is the way that asterisk works. It is going to quelch all DTMF that goes out via a SIP gateway via asterisk. I spent a long time working this through and it has to do with the way that asterisk deals with DTMF and the DSP.c module that sits inband to the RTP/audio stream. There is a flag called DSP_DIGITMODE_NOQUELCH that is broken that might allow the inband DTMF after answer to work

Hello folks. I'm not sure if this is the right list for this question, but I'll start here. If I'm using a SIP provider and I have an entry in sip.conf that looks like: [8315551212] type => friend ... dtmfmode => inband ... When I pick up the phone, call someone through this provider, and press numeric digits to generate dtmf tones, who is actually generating the tones at the

We have an application which is primarily DTMF driven (automated on both sides), which we are trying to deploy over VOIP and Asterisk (using some Sipuras and some IAXY's). We are finding that in around half the cases, the Asterisk server can't decode the DTMF digits from the field office (or at least some of them). Though, when we place voice calls for testing, we can hear eachother quite

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

Hi, I have a setup where I have multiple lines to the same provider - in this case broadvoice. I have created the entries in sip.conf for both accounts - and independently they work fine. When they both are in use, incomming calls are placed to the one that's the last in the sip.conf file. On voip-info I found the following: *Quote:* When Asterisk receives an incoming SIP call, the SIP

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

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

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

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