[Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
If you can get an rtp debug while your pressing digits I can see if maybe your device is sending the digits incorrectly. /b On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote: my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
Other than below: Got RTP packet from x.y.z.sip_phone:10006 (type 18, seq 25407, ts 191360, len 40) Sent RTP packet to x.y.z.asterisk:10006 (type 18, seq 63928, ts 193744, len 20) I dont see any message while sending digits. -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:33:14 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain If you can get an rtp debug while your pressing digits I can see if maybe your device is sending the digits incorrectly. /b On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote: my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
