Re: [asterisk-users] Redial dtmf tones randomly...asterisk 1.4.21.2
I would appreciate it if you didn't top-post. das sandesh sandesh...@gmail.com writes: Hi Benny... DTMF tones are heard at the SIP phones side and not the other party...'server side' means from the Asterisk side.from the wireshark captures it appeards that the dtmf digits were sent from the asterisk server ip to the phone ip randomly through Cisco(just passes the SIP packt) inbetween the conversation... How do you interface with the PSTN? A Digium card? Either way you may want relaxdtmf=no in dahdi.conf if you don't have that already. You can see the DTMF happening on the Asterisk console if you set verbosity high enough. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redial dtmf tones randomly...asterisk 1.4.21.2
We dont have any Digium cards, we just have a GrandStream FXS 8-port device with 2 analog phones and one Grand stream FXO 8-port device with one POTS line and both are connected to the netgear switchvery rarely the analog phones are used and its very rare that calls are made through POTS using FXO.we get this DTMF problem with the SIP phones(when called out or when we receive a call randomly)..I will try to capture the dtmf from the asterisk console with higher verbosity mode and also set the relaxeddtmf parameter. Thanks Sandesh On Thu, Jul 22, 2010 at 3:48 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: I would appreciate it if you didn't top-post. das sandesh sandesh...@gmail.com writes: Hi Benny... DTMF tones are heard at the SIP phones side and not the other party...'server side' means from the Asterisk side.from the wireshark captures it appeards that the dtmf digits were sent from the asterisk server ip to the phone ip randomly through Cisco(just passes the SIP packt) inbetween the conversation... How do you interface with the PSTN? A Digium card? Either way you may want relaxdtmf=no in dahdi.conf if you don't have that already. You can see the DTMF happening on the Asterisk console if you set verbosity high enough. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redial dtmf tones randomly...asterisk 1.4.21.2
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 pathAlso we have forced the dtmf of the fxs port to be rfc2833. In the wireshark capture attached we could see the random dtmf digits have been sent from the server side.can anyone share your thoughts in regards to this... Thank you Sandesh attachment: wireshark.bmp-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redial dtmf tones randomly...asterisk 1.4.21.2
das sandesh sandesh...@gmail.com writes: In the wireshark capture attached we could see the random dtmf digits have been sent from the server side.can anyone share your thoughts in regards to this... Which end hears the DTMF, the SIP phones or the phones on the PSTN? When you say sent from the server side, is the server side the Asterisk or the Cisco? /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redial dtmf tones randomly...asterisk 1.4.21.2
Hi Benny... DTMF tones are heard at the SIP phones side and not the other party...'server side' means from the Asterisk side.from the wireshark captures it appeards that the dtmf digits were sent from the asterisk server ip to the phone ip randomly through Cisco(just passes the SIP packt) inbetween the conversation... Thank you Sandesh On Wed, Jul 21, 2010 at 2:31 PM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: das sandesh sandesh...@gmail.com writes: In the wireshark capture attached we could see the random dtmf digits have been sent from the server side.can anyone share your thoughts in regards to this... Which end hears the DTMF, the SIP phones or the phones on the PSTN? When you say sent from the server side, is the server side the Asterisk or the Cisco? /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
