similar to: DTMF issue

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

Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions ram -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk -rvvvvv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891

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 all, I have a SPA 1001 with DTMF set to auto. A sip.conf peer with codec alaw, ulaw, g729. I have dtmf=inband as this peer was only supporting alaw/ulaw. They just add g729, it's a GW to landline phones. A call passing through this peer give me "WARNING Inband not supported with g729, use RFC2833 instead". The call is connecting and DTMF is working. Now I'm coming back

How can I know for sure if SIP Trunk Provider is sending DTMF 'inband' or 'rfc2833'? And more importantly if they could be sending both? If I specify 'inband' should they honor that? Thanks, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081002/3b34d38d/attachment.htm

Hello asterisk-users list. I have been scratching my head for almost a week. We are trying to set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not working. In our scenario the SP is sending call to our ser server and ser is forwarding the call to asterisk. In the asterisk debug I can see the DTMF keys are coming but ivr does not recognice those keys at all. I can see this in the

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

Hi folks, I'm running Asterisk 1.4.18 on Mac OS X. I'm using Broadvoice to connect external numbers with this system. So, what I'm dealing with here are inbound calls. The problem is that I'm not able to get DTMF in. I am behind NAT, but I've made sure that the IT folks here have opened and port-forwarded all the necessary ports. This is borne out by the fact that I

Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the

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

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

> I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in > routing calls to upstream carrier via SIP trunks out.? I spent a lot of time > in the lab testing 1.8 which included heavily testing DTMF with no issues > that came up.? It all just seemed to work fine.? But then again you can?t > reproduce every real work scenario in the lab. > > > > I?m

Topology: PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server When I make a call to a VoIP user from the PSTN, the call gets routed through the PBX, and Cisco. Because of that the DTMF tones are passed inband, which I can hear on the VoIP end of the call. However, I have one extension on asterisk set up so that I can check voice mail when away from my

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 should note that we are not running the Digium g729 implementation, but the intel one. Also, to not angry people, this ofcourse isn't used in our production environment, only for testing if we want g.729. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Moises Silva Sendt: 14. juni 2006 15:18 Til:

Hi, Anybody out there running Adit600s? I have in my care an Adit600 channel bank connected to an old (version 1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk (1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail. I have attempted to add the "slowsequence = yes" line to mgcp.conf. (It seemed to be the only likely candidate in the example files I found

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help