similar to: Polycom 500...

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

Just got a Polycom 301 and I'm configuring. Examples given in wiki recommend using dtmfmode=inband, so that's what I set in sip.conf for this phone, as I have for various other IP phones on my network. But the telephone does not seem to send DTMF tones up thru the network (although I hear them in the handset when I bang the buttons). Also, I can't seem to find a corresponding

hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail

hi there I'm setting up asterisk@home and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some

I am having a problem with some Polycom 601 phones. If I dial without picking up the handset or selecting the speaker I can dial numbers that are any lenght. But if I pick up the handset or are using the speaker I can only dial numbers that are 8 digits. When I dial the 8th digit it dials immediately. Obviously this creates problems when I am dialing long distance numbers or anything that

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

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

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes. I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I configure dtmfmode=rfc2833 (I've tryied inband and info). Asterisk seems not to "see" the tones. Could somebody help me? Thanks

I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work?

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.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it

I have Polycom ip-300 phones that worked yesterday but dont seem to work today (at least dtmf signalling once connected to the asterisk box) The current configuration is: [general] port = 5060 bindaddr = 0.0.0.0 context = test srvlookup = yes dtmf = inband allow = all dtmfmode=inband progressinband=no disallow=all allow=ulaw pedantic=no [202] type=user secret=xxxx context=test mailbox=202

When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY

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

Hello, I have one problem and I need to disable dtmf (disable rfc2833, info and inband) on one (other peers must support dtmf) SIP peer . Is it possible? Workaround would be use g729 codec with dtmfmode=inband. Maybe there is better solution? Thanks for help. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

I understand there has been some issues sending DTMF tone through Broadvoice. Can some provide me with symptoms? --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the

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