similar to: Cisco DTMF problem...

I've installed Asterisk from the Asterisk @home distribution. Ultimately I will be using Asterisk for voicemail for about 150 users. Calls are (mostly) handled by a legacy PBX although we do have a couple of Cisco 1760 routers that connect a remote office. I've setup a SIP trunk that routes calls from Asterisk to the 1760, and that works fine. I've also configured one of the 1760s to

Hi, When we use BudgeTone where the DTMF is set to "via RTP (RFC2833)" all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to "remote pick-up the call" through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Best

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

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

Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun

Hi list, I'm hoping that I'm being stupid, and someone can tell me what's going on, but for the life of me I can't figure it out. (it's been a long day, and I'm now in the last 3 weeks of organising my wedding, so I hope this makes sense ;) ) When at my desk, accessing (for example) my voicemail, the dtmf tones are passed perfectly, I can enter password, change

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

Hello, I am fairly new to asterisk. I am currently using asterisk as a more convenient sip side voicemail system. My problem: I have cisco 7960 phones whose out of band dtmf tones are recognized properly(when dtmfmode=rfc2833) by asterisk but whose in-band dtmf tones are recognised poorly(when dtmfmode=inband) . For example 7999 comes out as 799999, 4242 comes out as 442422 ... etc I

We're having an issue with connecting a Cisco ITS installation to * such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *. On the Cisco Side: dial-peer voice 8 voip destination-pattern 9999$ session protocol sipv2 session target ipv4:172.16.1.249 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad We have also

ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck.

Can anyone explain why this is happening? I have a server attached to a phone line that will play a .wav file, then play all the dtmf digits (after it answers the call). If I place a call from a SIP device (like a Cisco 7960 phone) through Asterisk and on to the test server, via PSTN, the .wav file sounds fine, but the DTMF digits are distorted ----->------------->--------------------audio

I have done some searching and not sure this is even possible, but here it goes... **Scenario** Let's say you have an asterisk server that you use to connect to a SIP provider that you push your PSTN-bound calls to using g711 and out-of-band DTMF. The SIP phones in use are Cisco 7960's and are set to also use out-of-band DTMF. For the most part, everything works great. However, a few

Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). The problem I'm having is that when I connect to voicemail the DTMF key presses dont seem to work

Hi, I have encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind

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>

Before I start ... I know there's been a lot of talk over the last couple of days about the list being slow etc... I understand there were problems and that mail is starting to get through now. Just wondering - should I be surprised if something I sent 48 hours ago hasn't turned up at all? And something I sent about 6 hours ago still hasn't turned up (yet). I'm just curious

Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

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 Dmitry thank you for your reply. Ok, you are right. i want to configure trunk h323 between asterisk 11.13.1 and 2800 cisco router. this is my scenario: PBX(100)--->cisco--->asterisk11.13.1---->PBX(200) when i call from 100 to 200, everything is ok but when i call from 200 to 100, phone rings but after i answer it, i have no voice and call terminates after 5 seconds. this is