Re: [asterisk-users] What don't I get about SIP?
Mike wrote: Here it is: dialplan dialplan.impossibleMatchHandling=1 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=[7]xx|[9]xxT|[9][1]xxT dialplan.digitmap.timeOut=3/ When I dial 845, I get fast busy. When I dial 9-555-555-, it dials without the need to press send. All good result. Actually, as soon as you hit 8 you will get the fast busy. Is that your full dialplan? What about an emergency (911) or other N11 calls? What about direct dial international calls (011...)? When I dial 9-555-5 and wait, nothing happens So, it looks like what you want is a global dialing timeout in the phone, which the Polycom phones don't appear to have once you break dialtone. But you may be able to kluge the digit timeout to give you that feature if you don't need it for what it is meant for. Right now you are using it to timeout when a digit other than 1 is pressed after the 9. That isn't really necessary (unless 91 followed by 9 digits is actually a valid number for whatever you are doing with it). Also, you are using the brackets unecessarily, since you only have one digit within them. An equivalent dialplan that doesn't use the digit timeout feature would be: digitmap dialplan.digitmap=7xx|9[2-9]x|91xx The digit timeout feature is typically used for direct dial international calls and calling the operator. If you don't need either of those then you could do something like this: digitmap dialplan.digitmap='7xx|9[2-9]x|91xx|[79]x.T dialplan.digitmap.timeOut=15/ which would timeout and send whatever sequence you had pressed after 15 seconds if you hadn't already matched one of the other patterns. Note that asterisk may possibly respond with error code 484 if the sequence pressed isn't complete, which would make the phone continue to ask for more digits. So, the other part of the solution is to add: exten = _X.,1,Congestion() to extensions.conf in the context you are using for your polycom phone(s). That will match anything that doesn't match one of your valid extensions as long as it is two digits or more. So you still will get the behaviour you don't like if someone just presses 7 or 9 and nothing else. But it will give you most of what you want, assuming I understand what you are looking for in the first place (you could try x.T in the digitmap and _. in extensions.conf, but _. is likely to cause other problems). Note: When the Polycom gets the congestion response from Asterisk it plays the congestion tone for only about 3-4 seconds, and then hangs up, which is different behaviour from when you press an 8 for instance. If you want the behaviours to be similar you could do something like this: exten = _X.,1,Answer() exten = _X.,2,Playtones(congestion) exten = _X.,3,Wait(30) exten = _X.,4,Hangup() John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
Actually, as soon as you hit 8 you will get the fast busy. Is that your full dialplan? What about an emergency (911) or other N11 calls? What about direct dial international calls (011...)? Its my full test dialplan for now. I do get fast busy as soon as I hit 8, so that part works. So, it looks like what you want is a global dialing timeout in the phone, which the Polycom phones don't appear to have once you break dialtone. But you may be able to kluge the digit timeout to give you that feature if you don't need it for what it is meant for. Right now you are using it to timeout when a digit other than 1 is pressed after the 9. That isn't really necessary (unless 91 followed by 9 digits is actually a valid number for whatever you are doing with it). Also, you are using the brackets unecessarily, since you only have one digit within them. An equivalent dialplan that doesn't use the digit timeout feature would be: digitmap dialplan.digitmap=7xx|9[2-9]x|91xx Fair enough, but that doesn't solve the original issue, but it makes my kludge a bit better [lots of good info removed] Note that asterisk may possibly respond with error code 484 if the sequence pressed isn't complete, which would make the phone continue to ask for more digits. So, the other part of the solution is to add: exten = _X.,1,Congestion() That will match anything that doesn't match one of your valid extensions as long as it is two digits or more. So you still will get the behaviour you don't like if someone just presses 7 or 9 and nothing else. But it will give you most of what you want, assuming I understand what you are looking for in the first place (you could try x.T in the digitmap and _. in extensions.conf, but _. is likely to cause other problems). Did I misread the Asterisk wiki pages, because I believed that when a pattern was present, the pattern takes precedence over any real extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)? Thanks John, I appreciate all the info. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Mike wrote: Did I misread the Asterisk wiki pages, because I believed that when a pattern was present, the pattern takes precedence over any real extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)? It's the opposite. Asterisk always uses the most specific match for an extension, i.e. anything that matches _1XXX will take precedence over _, but if it matches _12XX that will take precedence over _1XXX, etc. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
It certainly makes sense, and I tried it...it works, you are right. So what do you make of this page : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf +sorting Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Marvin Sent: September 9, 2006 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: Did I misread the Asterisk wiki pages, because I believed that when a pattern was present, the pattern takes precedence over any real extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)? It's the opposite. Asterisk always uses the most specific match for an extension, i.e. anything that matches _1XXX will take precedence over _, but if it matches _12XX that will take precedence over _1XXX, etc. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Mike wrote: It certainly makes sense, and I tried it...it works, you are right. So what do you make of this page : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf +sorting Interesting. I got my information from Asterisk: The Future of Telephony (in the dialplan chapter). Perhaps the wiki page refers to 1.0 behaviour, and 1.2 behaviour is what is defined in Asterisk: TFOT? My experimentation so far has shown the Asterisk: TFOT information to be correct. I haven't played around with #include, which the wiki says can change the dialplan extension sorting. I may have to experiment with that to see if it has any effect. I hope that what I said is correct regardless, because it makes the most sense and is less likely to cause weird issues when changing the order of #includes, etc. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What don't I get about SIP?
I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is "intelligent enough" to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this "intelligent" error handling? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpHvcCeRVwOA.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
Thanks Tim. I've been trying to find out what's happening. Basically, somehow, it seems that my Polycom 501 knows what extensions are valid and which aren't in my dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems to return it this info (sort of :valid, invalid or could be valid, need more digits to know) when I press send. I know it sounds mad, and I would love nothing more than being told I am an idiot because or x and y. Why do I feel that this info is passed from Asterisk to the 501? Well, take the following (very simple) dialplan [context_a] Exten = 1234,1,Noop(foo) Exten = _9,1,Noop(bar) Exten = i,1,Noop(invalid) What happens when I dial out is the following: 1) 1234: Noop(foo) ; good 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI doesn't show anything...no sent into invalid extension '4' in context 'context_a', but no invalid handler 3) 934 : It's invalid, but it could match the pattern is I added some digits. I expect an invalid extension message, but what actually happens is the phone tries the send something (I can see an icon moving on the phone) but the phone stays quiet (no stuttering tone or whatever). It waits, I can input more digits on the phone. Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre Sent: September 8, 2006 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike You want to look at the dialplan section in your sip config file or on the device's web config. Specifically the dialplan.digitmap and dialplan.impossibleMatchHandling settings. (These are explained in detail in the Admin Guide available on polycom's website). If you don't want the phone to do any dial plan matching you should make the digitmap accept any dial string (ie .) and set your timeouts appropriately. -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Now that is really odd. Try sip debug peer (peername of the polycom) This will let you see the sip packets go by when you do this, so you can see the responses it is, or isn't getting. I'll have to look up the SIP response codes, but I do know that there is one for not found which should correspond with an invalid extension. Because the call is not actually set up yet, asterisk will return a not found message rather than answer the call, only to direct it to an i extension. This is only used for calls already in progress. I don't know if there is a sip response for need more digits or something like that. Turning on the sip debug will tell you EXACTLY what the polycom is saying to asterisk, and vice versa. Note: I like to hit scroll lock after I hit call, before I hangup so that it doesn't fill my screen up with all the cancel messages - that will put you just below the important parts of the data. On September 8, 2006 15:21, Mike wrote: Thanks Tim. I've been trying to find out what's happening. Basically, somehow, it seems that my Polycom 501 knows what extensions are valid and which aren't in my dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems to return it this info (sort of :valid, invalid or could be valid, need more digits to know) when I press send. I know it sounds mad, and I would love nothing more than being told I am an idiot because or x and y. Why do I feel that this info is passed from Asterisk to the 501? Well, take the following (very simple) dialplan [context_a] Exten = 1234,1,Noop(foo) Exten = _9,1,Noop(bar) Exten = i,1,Noop(invalid) What happens when I dial out is the following: 1) 1234: Noop(foo) ; good 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI doesn't show anything...no sent into invalid extension '4' in context 'context_a', but no invalid handler 3) 934 : It's invalid, but it could match the pattern is I added some digits. I expect an invalid extension message, but what actually happens is the phone tries the send something (I can see an icon moving on the phone) but the phone stays quiet (no stuttering tone or whatever). It waits, I can input more digits on the phone. Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre Sent: September 8, 2006 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgp3K2UaXtL4T.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Mike wrote: Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Because the i extension is for IVRs and things like that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike wrote: Thanks Tim. I've been trying to find out what's happening. Basically, somehow, it seems that my Polycom 501 knows what extensions are valid and which aren't in my dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems to return it this info (sort of :valid, invalid or could be valid, need more digits to know) when I press send. I know it sounds mad, and I would love nothing more than being told I am an idiot because or x and y. Why do I feel that this info is passed from Asterisk to the 501? Well, take the following (very simple) dialplan [context_a] Exten = 1234,1,Noop(foo) Exten = _9,1,Noop(bar) Exten = i,1,Noop(invalid) What happens when I dial out is the following: 1) 1234: Noop(foo) ; good 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI doesn't show anything...no sent into invalid extension '4' in context 'context_a', but no invalid handler 3) 934 : It's invalid, but it could match the pattern is I added some digits. I expect an invalid extension message, but what actually happens is the phone tries the send something (I can see an icon moving on the phone) but the phone stays quiet (no stuttering tone or whatever). It waits, I can input more digits on the phone. Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre Sent: September 8, 2006 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Silly idea, why don't you sniff the packets being sent over port 5060? You'll be able to verify the conversation taking place. - -- S McGowan VoIP Consultant [EMAIL PROTECTED] - -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW
RE: [asterisk-users] What don't I get about SIP?
It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk Mike Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.1.200 : 5060 (NAT) Found user '000f42056d58-1' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.200:2228 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 965 in context_a (domain test.test.ca) Reliably Transmitting (NAT) to 45.67.312.45:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.1.200;branch=z9hG4bK93732511F5970F9E;received=45.67.312.45 From: CAP sip:[EMAIL PROTECTED];tag=DAD6C20C-68263D4F To: sip:[EMAIL PROTECTED];user=phone;tag=as4db2b55c Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rushowr Sent: September 8, 2006 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike wrote: Thanks Tim. I've been trying to find out what's happening. Basically, somehow, it seems that my Polycom 501 knows what extensions are valid and which aren't in my dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems to return it this info (sort of :valid, invalid or could be valid, need more digits to know) when I press send. I know it sounds mad, and I would love nothing more than being told I am an idiot because or x and y. Why do I feel that this info is passed from Asterisk to the 501? Well, take the following (very simple) dialplan [context_a] Exten = 1234,1,Noop(foo) Exten = _9,1,Noop(bar) Exten = i,1,Noop(invalid) What happens when I dial out is the following: 1) 1234: Noop(foo) ; good 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI doesn't show anything...no sent into invalid extension '4' in context 'context_a', but no invalid handler 3) 934 : It's invalid, but it could match the pattern is I added some digits. I expect an invalid extension message, but what actually happens is the phone tries the send something (I can see an icon moving on the phone) but the phone stays quiet (no stuttering tone or whatever). It waits, I can input more digits on the phone. Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre Sent: September 8, 2006 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do
Re: [asterisk-users] What don't I get about SIP?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk Mike Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.1.200 : 5060 (NAT) Found user '000f42056d58-1' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.200:2228 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 965 in context_a (domain test.test.ca) Reliably Transmitting (NAT) to 45.67.312.45:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.1.200;branch=z9hG4bK93732511F5970F9E;received=45.67.312.45 From: CAP sip:[EMAIL PROTECTED];tag=DAD6C20C-68263D4F To: sip:[EMAIL PROTECTED];user=phone;tag=as4db2b55c Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rushowr Sent: September 8, 2006 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: Thanks Tim. I've been trying to find out what's happening. Basically, somehow, it seems that my Polycom 501 knows what extensions are valid and which aren't in my dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems to return it this info (sort of :valid, invalid or could be valid, need more digits to know) when I press send. I know it sounds mad, and I would love nothing more than being told I am an idiot because or x and y. Why do I feel that this info is passed from Asterisk to the 501? Well, take the following (very simple) dialplan [context_a] Exten = 1234,1,Noop(foo) Exten = _9,1,Noop(bar) Exten = i,1,Noop(invalid) What happens when I dial out is the following: 1) 1234: Noop(foo) ; good 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI doesn't show anything...no sent into invalid extension '4' in context 'context_a', but no invalid handler 3) 934 : It's invalid, but it could match the pattern is I added some digits. I expect an invalid extension message, but what actually happens is the phone tries the send something (I can see an icon moving on the phone) but the phone stays quiet (no stuttering tone or whatever). It waits, I can input more digits on the phone. Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre Sent: September 8, 2006 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid
Re: [asterisk-users] What don't I get about SIP?
Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Or better yet, set dialplan.impossibleMatchHandling to 2. This should disable earlydial altogether. CP On Sep 8, 2006, at 2:49 PM, Eric ManxPower Wieling wrote: Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesnt allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesnt allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
Thanks Dave. Unfortunately I've been through this already. I understand that digitmap are used to automatically press send when a certain pattern is reached. Nowhere can I say if the pattern isn't fully match within x seconds then consider it a bad extension. That`s the only relevant thing I haven`t yet found how to do. Regards, Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: September 8, 2006 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike You want to look at the dialplan section in your sip config file or on the device's web config. Specifically the dialplan.digitmap and dialplan.impossibleMatchHandling settings. (These are explained in detail in the Admin Guide available on polycom's website). If you don't want the phone to do any dial plan matching you should make the digitmap accept any dial string (ie .) and set your timeouts appropriately. -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It's actually your phone's responsibility to respond to the 484 and/or a dial timeout. ;-) - -- S McGowan VoIP Consultant [EMAIL PROTECTED] - -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb
Re: [asterisk-users] What don't I get about SIP?
Then you are doing something else wrong. If the call gets to Asterisk then the exten = lines I gave should match if they are in that context. I use this all the time. Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
What IS your Polycom dialplan, and do you have the digit.impossiblematch set? Eric ManxPower Wieling wrote: Then you are doing something else wrong. If the call gets to Asterisk then the exten = lines I gave should match if they are in that context. I use this all the time. Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] What don't I get about SIP?
Here it is: dialplan dialplan.impossibleMatchHandling=1 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=[7]xx|[9]xxT|[9][1]xxT dialplan.digitmap.timeOut=3/ When I dial 845, I get fast busy. When I dial 9-555-555-, it dials without the need to press send. All good result. When I dial 9-555-5 and wait, nothing happens Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 7:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? What IS your Polycom dialplan, and do you have the digit.impossiblematch set? Eric ManxPower Wieling wrote: Then you are doing something else wrong. If the call gets to Asterisk then the exten = lines I gave should match if they are in that context. I use this all the time. Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesnt allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
I don't know if this helps, but this is how my 80+ Polycom phones are set up. dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=0 digitmap dialplan.digitmap=9,1[2-9]xx[2-9]xx|9,[2-9]xx|[2-8]xxx|9,[2-9]11|911|1x|9,011x.T|*xx dialplan.digitmap.timeOut=5/ Mike wrote: Here it is: dialplan dialplan.impossibleMatchHandling=1 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=[7]xx|[9]xxT|[9][1]xxT dialplan.digitmap.timeOut=3/ When I dial 845, I get fast busy. When I dial 9-555-555-, it dials without the need to press send. All good result. When I dial 9-555-5 and wait, nothing happens Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 7:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? What IS your Polycom dialplan, and do you have the digit.impossiblematch set? Eric ManxPower Wieling wrote: Then you are doing something else wrong. If the call gets to Asterisk then the exten = lines I gave should match if they are in that context. I use this all the time. Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth
