similar to: Transferring a call, IAX2->SIP, DTMF/RFC2833 doesn't work?

Displaying 20 results from an estimated 8000 matches similar to: "Transferring a call, IAX2->SIP, DTMF/RFC2833 doesn't work?"

2006 May 19
4
PRI dialing IVR with inband DTMF
I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is
2006 Mar 16
1
RFC 2833 and SIP? DTMF? What am I not getting?
Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took
2005 Sep 21
1
oh323 driver and RFC2833
Hello, I have installed oh323 channel driver. Outgoing calls to H.323 world do not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that userInputMode=RFC2833 has already been set. Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel? Kind regards, Fernando Herrera _____ De: Fernando Herrera [mailto:fherrera@iplan.com.ar] Enviado el:
2004 Dec 21
0
SIP dtmf=rfc2833 not working
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
2010 Jun 29
1
Asterisk 1.6 (and 1.4) DTMF problems using RFC2833
We are experiencing intermittent DTMF problems here, with the following setup: ITSP -> PIX -> Asterisk (g729, RFC2833 for DTMF). I am running Ubuntu server 10.04, but Asterisk is compiled by us and not installed from the software repository. Essentially, DTMF works for some time, but at some point it simply stops and the point at which it stops appears to be random. Using RTP debug, I
2005 Mar 29
0
rfc2833 cisco 7960 DTMF issue
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
2009 Apr 13
3
duration of rfc2833 generated dtmf
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your
2003 Aug 21
1
Voicemail2 and RFC2833 DTMF
Hi, In testing the Budgetone we have noticed something strange with DTMF and Voicemail. When we set the Budgetone for RFC2833, and connect to voicemail, the detected DTMF digits do not correspond with what we pressed on the phone. For example user=1001, password=1001 is detected as: Incorrect password '1111000000111' for user '111000000111' (context = <any>) Any idea
2005 Jan 24
2
"Inband DTMF is not supported on codec G.711 u-law. Use RFC2833"
Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2006 Jan 19
1
DTMF Simultaneous Inband and RFC2833 performedby Asterisk => Duplicate tones
> I have seen the following effect in Asterisk, though: where > it converts > an inband DTMF (eg coming off a Zap channel) into an > indication, it mutes > the audio where that tone is. But sometimes it leaves a > teeny bit of the > tone behind. > > If you take such a call over say IAX to somewhere and then > back out a Zap > channel, you end up with the
2007 Mar 29
1
DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp
Hello mailing list, I have been porting one of my Asterisk boxes to 1.4 and I have encountered a nasty DTMF problem. What happens is someone might come in to my IVR and enter "12345" and what will actually come through could be along the lines of "12234445". Sometimes it works, sometimes it doesn't. I had this problem with 1.2 back in November but was able to solve it
2005 May 16
0
DTMF asterisk-2-asterisk using SIP w/ dtmfmode=rfc2833
Hi, I'm am getting doubled DTMF on some digits with one of my providers who also uses asterisk. We're using SIP, with dtmfmode set to rfc2833, and the codec G.711. Once out of every five or ten calls, there are no problems, but more often than not, the DTMF is getting doubled-up on at least one of the digits of the extension dialed. I've tested with a CVS-HEAD from Febuary, and just
2010 Aug 27
0
Asterisk DTMF RFC2833 issues
Hi all I have posted a question on the asterisk dev board about this issue but I want to see if any users have run up against this. This issue is that when calls are run through Broadvox and Level 3 the in-call rfc2833 dtmf is not reliable. This occured for me on asterisk version 1.6.1.18, 1.6.1.20 it appears to have been fixed when I went to 1.6.2.11 but broken again in 1.6.2.12-rc1. I have
2005 Jun 01
1
RFC2833 & firewall problems? (16-byte UDP packets)
We are tracking the following situation: SIP client connects to our Asterisk server, and then connects to another SIP user. Re-invite is OFF, so Asterisk is in the middle of the whole conversation. When one SIP client sends DTMF tones, the SIP client uses RFC2833 to send the tones to the server. (This is correct). The server then sends RFC2833 tones out to the other SIP client. The problem is,
2008 Jan 12
2
Asterisk RFC2833 to SIP INFO DTMF conversion erros.
Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk
2010 Jul 08
2
DTMF issues/redial tones with rfc2833
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
2009 Jan 24
3
Passing DTMF
Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the
2006 Jan 18
1
DTMF Simultaneous Inband and RFC2833 performed by Asterisk => Duplicate tones
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2003 Nov 19
0
SIP/IAX2 DTMF
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, When making a call like the one below, I get double DTMF tones on the PSTN side. DTMF tones sent from the PSTN arrives squelched on the SIP side. SIP > Asterisk2 > IAX2 > Asterisk1 > ZAP > PSTN SIP has been configured to use rfc2833 on both the SIP endpoint and the Asterisk. SIP endpoint also suggests a payload value of 101.
2007 Feb 09
1
RFC2833 SIP trunks and DTMF
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