search for: 011xxx

...these expressions lock up the new phone. Old Configs <digitmap dialplan.digitmap="[2-9]11|[*]xx|891xxx|[1-7]xxx|8[0-46-8]xx|8500|851xxx|9,1[2-9]xxxxxxxxx|9,xxxxxxxT" dialplan.digitmap.timeOut="3"/> Template from Polycom <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3|3|3|3|3|3"/> Anyone have any insight or suggestions on this issue, and on upgrading Polycom configs in general? -- Justin Phelps www.onitato.com 850.866.6864

...a series of six 3's seperated by a |. The 600 definition has a single 3. What is the proper way to define these timeouts as well? It also seems some of the definitions in the 600 group are redundant. Can you think of any reason as to why these shouldn't be combined? Default: [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT 600: [2-9]11|[*]xx|891xxx|[1-7]xxx|8[0-46-8]xx|8500|851xxx|9,1[2-9]xxxxxxxxx|9,xxxxxxxT 600 [2-9]11 Information/Emergency (411 911 etc) [*]xx Service Numbers (*69) 891xxx 891 then 3 digits [1-7]xxx 4 Digi...

Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel the lights come on and all seems fine, until I reboot that is... After a reboot I have to repeat the modprobe. I shouldn't have to do a modprobe every re-boot should I? How do you get the drivers to load

...2:06:54 dovecot: auth: Debug: client passdb out: CONT#0111 12:06:54 dovecot: auth: Debug: client passdb out: CONT#0111 12:06:54 dovecot: auth: Debug: client passdb out: CONT#0111 12:06:54 dovecot: auth: Debug: client in: CONT<hidden> 12:06:54 dovecot: auth: Debug: client passdb out: CONT#0111#011XXX 12:06:54 dovecot: auth: Debug: client in: CONT<hidden> 12:06:54 dovecot: auth: Debug: client passdb out; CONT#0111#011XXX 12:06:54 dovecot: auth: Debug: client passdb out: CONT#0111#011XXX 12:06:54 dovecot: auth: Debug: client in CONT<hidden> 12:06:54 dovecot: auth: winbind(?X.X.X.X,&lt...

I have four Polycom 330 phones connected to an asterisk system. There are other VoIP phones connected too. All of the extensions are four digits beginning with 11. From any of the phones, except the Polycom, picking up the handset to call extension 1103 for example works fine. With the Polycom 330, as I press the second 1 of 1103 it stops taking input and gives me an error. I tried

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not work. The Polycom does > not allow me to dial the extension of the person I want to transfer to after > I hit: > &...

...ix the problem I eventually did the following... On the ftp server I replaced the dialplan settings in the sip.cfg from <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1"> <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3"/> </dialplan> to this.. <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1"> <digitmap dialplan.digitmap=&q...

...ers with 10 or 11 at the > beginning. This is the Digit Map on the Polycom phone. By default it uses 10 and 11 as special cases (I don't know why). You can adjust the digit map in sip.cfg. Just take a look at the line that looks like this: <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT" dialplan.digitmap.timeOut="3"/> You can add in more patterns separated by '|' characters for any cases you want to cover. - Noah

...ing, for instance, and #505551212 to set the call forwarding number and turn it on. I have tftp set up and sip.cfg contains the following: <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1"> <digitmap dialplan.digitmap="#xx.T|[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3"/> <routing> <server dialplan.routing.server.1.address="" dialplan.routing.server.1.port="5060"/> <emergency dialplan.routing.emergency.1.value="911" dialplan....

...#39;default', on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 Even though the user can not hear the extension the call was parked on, the call can be retrieved by guessing. Which I am assumming means the call was successfully parked. Digit map: [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|1xxxT extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] ATTENDANT=SIP/1006&SIP/1002&SIP/1011&SIP/1009 OUTBOUNDTRUNK=ZAP/g1 [meetme-ext]...

...nc.alwaysReboot="0"/> <conference voIpProt.SIP.conference.address=""/> </SIP> </voIpProt> <dialplan dialplan.impossibleMatchHandling="2" dialplan.removeEndOfDial= "1"> <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxx xxx|[2-9]xxxT" dialplan.digitmap.timeOut="3"/> <routing> <server dialplan.routing.server.1.address="" dialplan.routing.server.1.port="506 0"/> <emergency dialplan.routing.emergency.1.value="9...

Hello, I have a mixture of Polycom SP IP 500 and 300 phones. I have been reading through the administration manual to try and solve this problem, but I do not seem to be able to find the answers to my question. I figured I would ask here and see if anyone has some suggestions. The problem is kind of annoying. After dialing 4 digits, the phone seems to pause and miss the 5th digit, often

<digitmap =20 dialplan.digitmap=3D"[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxx= x x|[2-9]xxxT" dialplan.digitmap.timeOut=3D"3|3|3|3|3|3"/> Don't think it's been modified from the original supplied. ...brig -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-...

...eboot="0"/> <conference voIpProt.SIP.conference.address=""/> </SIP> </voIpProt> <dialplan dialplan.impossibleMatchHandling="2" dialplan.removeEndOfDial= "1"> <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3"/> <routing> <server dialplan.routing.server.1.address="" dialplan.routing.server.1.port="5060"/> <emergency dialplan.routing.emergency.1.value=&...

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries

I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using