similar to: PRI dialing IVR with inband DTMF

I've got a ShoreTel 210 MGCP phone drops calls. My packet capture indicates that the phone may be trying to renew its registration with *, but reports Restart Method of Disconnected (frame 2), then * seems to take that as a sign that it has lost the connection and closes things down. The phone, meanwhile, seems to think it can continue the conversation until a few ICMP "port

We have a Shortel system at out main site. We're putting Asterisk servers at several smaller remote sites. I know I'll be able to get the Asterisk servers to talk to each other via IAX, but can they talk to the Shoretel server? Basically, I'd like to be able to, from the main site with Shoretel, dial an extension, and reach that phone at a remote site, and vice-versa. Thank

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

Okay, so I'm a noob. Asterisk looks very promising, so I say "thanks" and "good job" to all who contribute. My * test box is up and running with soft phones using IAX and SIP, so now I'm on to testing hard phones. I borrowed a couple ShoreTel 210 phones from somebody who had them on hand but they only support MGCP. I see that there's an mgcp.conf in

SIP user => Asterisk 1.6 server => SIP Trunk => external destination: works AMI script => Asterisk 1.6 server => SIP Trunk => external destination: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@(ipaddr)>;tag=alphanumeric' I?ve tried doing things like ?include => contextwithtrunk" in various places, googling, re-reading relevant

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

Is it possible to make Asterisk do inband DTMF over IAX?

Hi can anyone help me on this error msg?? dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames thnx St

Hi all. Why don't Asterisk support inband DTMF with G729? Is there a way to do that!? Are you using RFC2833? Doesn't it a security hole? Thanks. Denis.

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

I know I wouldn't normally want this due to double tones, but my upstream provider has an issue where they negotiate rfc2833 but then send dtmf inband. I don't expect to get both at the same time, so is there a way to make asterisk turn on both inband or rfc2833? Auto doesn't work because it sees the rfc2833 in SDP then ignores inband for the remainder of the call. Thanks.

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:

Hi all, I wish to use analog interface card for the inband capturing media and use the Asterisk Open Source as a core software. I have tried the Sangoma card, and Digium card, and found that the inabnd can't capture from some PBX system. Why ?? Or any other hardware card has such features in order to work with all the inband capturing? Or may be some of your has such experience, and I am

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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

Hey, all. While ADA can still be downloaded, that's about all that I see. No development, no recent mention, and -- perhaps worst of all -- it appears not to work properly under 64-bit systems. So, assuming Digium's abandoned it, are there any suggestions of alternatives? Right now, I'm replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly fat client they

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

Hi, I'm getting unreliable dtmf recognition (it works fine for 4-5 digits, errors (duplicates) on more), when transferred inband from gsm gateway to NT port of quadbri under bristuffed Asterisk..... Since Asterisk is claimed to have good dtmf recognizer, I suspect there are some settings to workarouned... I've tried dtmf relax, but didn't help, so I suspect gain settings.... Is

Hello everybody. I have a problem with an integration between an Asterisk (1.4.24.1) on FreeBSD 7.0 and a Shoretel 7.5 server. To make a very long story short, when someone behind asterisk call an extension behing shoretel everything work as expected. When someone behing the shoretel server call someone behind asterisk the first 10 seconds of the call seems ok but then the line is dropped