Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
You are lucky. I'm getting this: -- Incorrect password '1334' for user When I enter 1234. I'm using dtmfmode=rfc2833 and a GS Budgtone 100 phone. Why do I getr 4x while you get 2x ?? use dtmfmode=info (both in sip.conf and in your GS settings, of course) search the archives to find this mentioned often. ;-( Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
--- Darren Nickerson [EMAIL PROTECTED] wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context = any) You are lucky. I'm getting this: -- Incorrect password '1334' for user When I enter 1234. I'm using dtmfmode=rfc2833 and a GS Budgtone 100 phone. Why do I getr 4x while you get 2x ?? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
I think this is a problem on the Asterisk side. I'm seeing the same problem using a Grandstream Budgetone 100. And the GS does have setting for both in-band and RFC2833. My guess is asterisk is accepting the DTMF tone __both__ ways It is reading the RFC28833 stuff _and_ hearing the audio tones as well. --- Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 21 December 2003 00:29, Darren Nickerson wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context snip Changing the DTMF mode would indeed seem to be the logical solution. However, it appears that SJphone does not support that option (after a quick perusal of their PDF). You might want to file a bugtracker request on their website to implement that functionality. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
I had the same problem with Grandsteam phones and *. No other hard or soft phones have the 'double digit' problem with *. I don't think Asterisk can do both RFC2833 and in-band DTMF at the same time. It does, however, do RFC2833 and SIP Info at the same time (SIP Info method seems to be on all the time, even when RFC2833 is selected in the sip.conf file). Switching the Grandsteam to SIP Info allowed it to talk to Asterisk and fixed the double digits problem. - Jim Chris Albertson wrote: I think this is a problem on the Asterisk side. I'm seeing the same problem using a Grandstream Budgetone 100. And the GS does have setting for both in-band and RFC2833. My guess is asterisk is accepting the DTMF tone __both__ ways It is reading the RFC28833 stuff _and_ "hearing" the audio tones as well. --- Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 21 December 2003 00:29, Darren Nickerson wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context snip Changing the DTMF mode would indeed seem to be the logical solution. However, it appears that SJphone does not support that option (after a quick perusal of their PDF). You might want to file a bugtracker request on their website to implement that functionality. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- +---+ | Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC | +---+ | "I never let my schooling get in the way of my education." - Mark Twain | | "UNIX was never designed to keep people from doing stupid things, because | | that policy would also keep them from doing clever things." - Doug Gwyn | | "Cool is only three letters away from Fool" - Mike Muir, Suicyco | | "..Government in its best state is but a necessary evil; in its worst | | state an intolerable one.." - Thomas Paine, "Common Sense" (1776)| +---+ | Email: [EMAIL PROTECTED] ICQ UIN: 1695089 | +---+ | Reply problems ? Turn off the "sign" function in email prog. Blame MS. | +---+ smime.p7s Description: S/MIME Cryptographic Signature
RE: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=ilbc|gsm|ulaw|g723.1|g711 ;allow=all dtmfmode=inband ;dtmfmode=inband|rfc2833 good luck... Girish From: Darren Nickerson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SJphone, Asterisk and DTMF tones ... Date: Sun, 21 Dec 2003 01:29:16 -0500 Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context = any) This is with dtmfmode=inband in sip.conf. With either rfc2833 or info, DTMF tones don't seem to get 'seen' by Asterisk at all. I'm running CVS-12/17/03-02:39:14, in case it's relevant. Help? -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 office +1.215.243.8335 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _ Add glamour to your desktop. Let your screen sizzle. http://server1.msn.co.in/msnchannels/Entertainment/wallpaperhome.asp Download the hottest wallpapers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
So, it seems a new bug has been found, which may or may not be at the root of this problem. Let me describe it, and see if you agree with the synopsis: Asterisk, despite having dtmfmode= set to a particular value in sip.conf for a peer, will listen for SIP Info method transmissions even if RFC2833 is selected. In some phones (Grandstream, in particular) this causes double-transmission of digits, since the phone sends both types of DTMF transmissions without blocking the other. Asterisk should ignore the other two types of DTMF transmission when selected to do one type of reception to counter these types of equiment peculiarities which seem to prevent correct DTMF usage. If I have described this correctly (I don't know - I don't have visibility into this problem) then can someone else (preferably someone with the problem) open a ticket? JT I had the same problem with Grandsteam phones and *. No other hard or soft phones have the 'double digit' problem with *. I don't think Asterisk can do both RFC2833 and in-band DTMF at the same time. It does, however, do RFC2833 and SIP Info at the same time (SIP Info method seems to be on all the time, even when RFC2833 is selected in the sip.conf file). Switching the Grandsteam to SIP Info allowed it to talk to Asterisk and fixed the double digits problem. - Jim Chris Albertson wrote: I think this is a problem on the Asterisk side. I'm seeing the same problem using a Grandstream Budgetone 100. And the GS does have setting for both in-band and RFC2833. My guess is asterisk is accepting the DTMF tone __both__ ways It is reading the RFC28833 stuff _and_ hearing the audio tones as well. --- Tilghman Lesher mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: On Sunday 21 December 2003 00:29, Darren Nickerson wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context snip Changing the DTMF mode would indeed seem to be the logical solution. However, it appears that SJphone does not support that option (after a quick perusal of their PDF). You might want to file a bugtracker request on their website to implement that functionality. = Chris Albertson Home: 310-376-1029 mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 mailto:[EMAIL PROTECTED][EMAIL PROTECTED] KG6OMK -- +---+ | Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC | +---+ | I never let my schooling get in the way of my education. - Mark Twain | | UNIX was never designed to keep people from doing stupid things, because | | that policy would also keep them from doing clever things. - Doug Gwyn | | Cool is only three letters away from Fool - Mike Muir, Suicyco | | ..Government in its best state is but a necessary evil; in its worst | | state an intolerable one.. - Thomas Paine, Common Sense (1776)| +---+ | Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] ICQ UIN: 1695089 | +---+ | Reply problems ? Turn off the sign function in email prog. Blame MS. | +---+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
I have the same key bounce problem with a Budgetone 101. After using the ZapBarge application to monitor the audio channel I determined that the 101 is pulsing the DTMF tones as long as the key is depressed at a rate of 200mS or so. If you tap the key pad quickly only one cycle is transmitted. I don't understand this feature. Does anyone? Anyway to disable it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Sunday, December 21, 2003 9:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ... So, it seems a new bug has been found, which may or may not be at the root of this problem. Let me describe it, and see if you agree with the synopsis: Asterisk, despite having dtmfmode= set to a particular value in sip.conf for a peer, will listen for SIP Info method transmissions even if RFC2833 is selected. In some phones (Grandstream, in particular) this causes double-transmission of digits, since the phone sends both types of DTMF transmissions without blocking the other. Asterisk should ignore the other two types of DTMF transmission when selected to do one type of reception to counter these types of equiment peculiarities which seem to prevent correct DTMF usage. If I have described this correctly (I don't know - I don't have visibility into this problem) then can someone else (preferably someone with the problem) open a ticket? JT I had the same problem with Grandsteam phones and *. No other hard or soft phones have the 'double digit' problem with *. I don't think Asterisk can do both RFC2833 and in-band DTMF at the same time. It does, however, do RFC2833 and SIP Info at the same time (SIP Info method seems to be on all the time, even when RFC2833 is selected in the sip.conf file). Switching the Grandsteam to SIP Info allowed it to talk to Asterisk and fixed the double digits problem. - Jim Chris Albertson wrote: I think this is a problem on the Asterisk side. I'm seeing the same problem using a Grandstream Budgetone 100. And the GS does have setting for both in-band and RFC2833. My guess is asterisk is accepting the DTMF tone __both__ ways It is reading the RFC28833 stuff _and_ hearing the audio tones as well. --- Tilghman Lesher mailto:[EMAIL PROTECTED][EMAIL PROTECTED] om wrote: On Sunday 21 December 2003 00:29, Darren Nickerson wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context snip Changing the DTMF mode would indeed seem to be the logical solution. However, it appears that SJphone does not support that option (after a quick perusal of their PDF). You might want to file a bugtracker request on their website to implement that functionality. = Chris Albertson Home: 310-376-1029 mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 mailto:[EMAIL PROTECTED][EMAIL PROTECTED] KG6OMK -- +-- -+ | Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC | +-- -+ | I never let my schooling get in the way of my education. - Mark Twain | | UNIX was never designed to keep people from doing stupid things, because | | that policy would also keep them from doing clever things. - Doug Gwyn | | Cool is only three letters away from Fool - Mike Muir, Suicyco | | ..Government in its best state is but a necessary evil; in its worst | | state an intolerable one.. - Thomas Paine, Common Sense (1776) | +-- -+ | Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] ICQ UIN: 1695089 | +-- -+ | Reply problems ? Turn off the sign function in email prog. Blame MS. | +-- -+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
John, Having started this thread, I guess I should comment. While a bug may exist that (simply and only) doubles up DTMF digits (as others have reported in the case of Grandstream? phones), I cannot reproduce that exact behaviour with the soft-phone SJphone product I'm using. I'll try to clarify. Over a series of several logins to voicemail entering 1234 for username and password, here's what I see: -- Incorrect password '1f123f344' for user '11223344' (context = any) -- Incorrect password '11223f344' for user '11223f344' (context = any) -- Incorrect password '112f23f344' for user '1122334f4' (context = any) -- Incorrect password '1f123344' for user '1f12334f4' (context = any) -- Incorrect password '123f344' for user '12334f4' (context = any) As you can see, the digits are commonly doubled, but not always. And what's up with that f?? I'm happy to (and motivated to) look into this more deeply, but I'm relatively new to Asterisk and not quite certain how to go about troubleshooting/debugging this. I certainly don't feel I know enough now to point the finger at Asterisk and open a bug - I'm still thinking it's possible I've goofed up some config somewhere along the line. Does anyone have DTMF detection working over SIP with a softphone product running on Windows? -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 office +1.215.243.8335 fax - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 21, 2003 10:22 AM Subject: Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ... So, it seems a new bug has been found, which may or may not be at the root of this problem. Let me describe it, and see if you agree with the synopsis: Asterisk, despite having dtmfmode= set to a particular value in sip.conf for a peer, will listen for SIP Info method transmissions even if RFC2833 is selected. In some phones (Grandstream, in particular) this causes double-transmission of digits, since the phone sends both types of DTMF transmissions without blocking the other. Asterisk should ignore the other two types of DTMF transmission when selected to do one type of reception to counter these types of equiment peculiarities which seem to prevent correct DTMF usage. If I have described this correctly (I don't know - I don't have visibility into this problem) then can someone else (preferably someone with the problem) open a ticket? JT I had the same problem with Grandsteam phones and *. No other hard or soft phones have the 'double digit' problem with *. I don't think Asterisk can do both RFC2833 and in-band DTMF at the same time. It does, however, do RFC2833 and SIP Info at the same time (SIP Info method seems to be on all the time, even when RFC2833 is selected in the sip.conf file). Switching the Grandsteam to SIP Info allowed it to talk to Asterisk and fixed the double digits problem. - Jim Chris Albertson wrote: I think this is a problem on the Asterisk side. I'm seeing the same problem using a Grandstream Budgetone 100. And the GS does have setting for both in-band and RFC2833. My guess is asterisk is accepting the DTMF tone __both__ ways It is reading the RFC28833 stuff _and_ hearing the audio tones as well. --- Tilghman Lesher mailto:[EMAIL PROTECTED][EMAIL PROTECTED] om wrote: On Sunday 21 December 2003 00:29, Darren Nickerson wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context snip Changing the DTMF mode would indeed seem to be the logical solution. However, it appears that SJphone does not support that option (after a quick perusal of their PDF). You might want to file a bugtracker request on their website to implement that functionality. = Chris Albertson Home: 310-376-1029 mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 mailto:[EMAIL PROTECTED][EMAIL PROTECTED] KG6OMK -- +-- -+ | Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC | +-- -+ | I never let my schooling get in the way of my education. - Mark Twain | | UNIX was never designed to keep people from doing stupid things, because | | that policy would also keep them from doing clever things. - Doug Gwyn | | Cool is only three letters away from Fool - Mike Muir, Suicyco | | ..Government in its best state is but a necessary evil; in its worst | | state an intolerable one.. - Thomas Paine, Common Sense (1776
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
I'll try to clarify. Over a series of several logins to voicemail entering 1234 for username and password, here's what I see: -- Incorrect password '1f123f344' for user '11223344' (context = any) -- Incorrect password '11223f344' for user '11223f344' (context = any) -- Incorrect password '112f23f344' for user '1122334f4' (context = any) -- Incorrect password '1f123344' for user '1f12334f4' (context = any) -- Incorrect password '123f344' for user '12334f4' (context = any) I've never seen an f included, or had a double digit problem with SJPhone. Here's what I'm using. In general: disallow=all allow=ulaw [test] type=friend secret=test context=default host=dynamic callerid=blah 1234 ;mailbox=1234 dtmfmode=inband That's the only dtmfmode that SJPhone supports. Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
Thanks Kevin! My overall defaults in sip.conf are: allow=ulaw ; Allow codecs in order of preference allow=ilbc allow=gsm allow=a_mu So I would have thought that ulaw codec would have been chosen if available. Just in case, I added the following to my SJphone test entry in sip.conf: [darren] type=friend host=dynamic nat=yes dtmfmode=inband username=1234 secret=1234 disallow=all allow=ulaw I restarted Asrterisk, and re-registered SJPhone with it ... but I still see the same problem. -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 office +1.215.243.8335 fax - Original Message - From: Kevin Bockman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, December 21, 2003 2:05 PM Subject: Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ... I'll try to clarify. Over a series of several logins to voicemail entering 1234 for username and password, here's what I see: -- Incorrect password '1f123f344' for user '11223344' (context = any) -- Incorrect password '11223f344' for user '11223f344' (context = any) -- Incorrect password '112f23f344' for user '1122334f4' (context = any) -- Incorrect password '1f123344' for user '1f12334f4' (context = any) -- Incorrect password '123f344' for user '12334f4' (context = any) I've never seen an f included, or had a double digit problem with SJPhone. Here's what I'm using. In general: disallow=all allow=ulaw [test] type=friend secret=test context=default host=dynamic callerid=blah 1234 ;mailbox=1234 dtmfmode=inband That's the only dtmfmode that SJPhone supports. Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
On Sunday 21 December 2003 00:29, Darren Nickerson wrote: Folks, I can't seem to get DTMF signaling working properly using SJphone connecting to Asterisk via a SIP connection. Here's an example of a voicemail session where I entered 1234 for both the username and the password: -- Incorrect password '11223344' for user '11223f344' (context = any) This is with dtmfmode=inband in sip.conf. With either rfc2833 or info, DTMF tones don't seem to get 'seen' by Asterisk at all. Changing the DTMF mode would indeed seem to be the logical solution. However, it appears that SJphone does not support that option (after a quick perusal of their PDF). You might want to file a bugtracker request on their website to implement that functionality. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
