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I looks like you have quite some cusom dialplan already. I think that you could use astdb for this, when the driver logges in selecting its route, you save it's details in the astdb. Than when dispatch needs them that dial the route number but the diaplan will lookup the last known extension...

If they are using a laptop is a softphone an idea, if its just for ocasional use it might. Richard

Thanks for your response, it is one that i expected, still hope they did build in support for the callerid update. Keeping my fingers crossed..... Richard

Anyone know if the Mitel SIP firmware supports the CallerID update on call transfer? I have a customer that is using these phones but when doing a attended transfer to another extension the callerid is not updated, only when doing a blind transfer and the destination has not yet pickedup the...

Hi TM1000, I know, but you have to leave the console again to look at the full log, if i'am not mistaken this log shows more DEBUG output that the console it self. Richard

Set your asterisk verbose and debug to 99 en look in the asterisk full log. asterisk -rx "core set verbose 99" asterisk -rx "core set debug 99" tail -f /var/log/asterisk/full Now dial the extension en let is go to voicemail and leave a message. There must be something in de log that tels...

Using tcpdump on the commandline should sufice. tcpdump -s 0 -w /tmp/mycapture.cap src or dst [ISPIP] and port 5060 or port portrange 10000-20000 You capture wil be in file /tmp/mycapture.cap readable by wireshark on you desktop machine. Richard

MM oke, Dual NAT absolutly NOT one of my thins to do, but sometimes....... The problem with DUAL NAT on one side is when the NAT soluctions are not perfecly writte, e.g. doe not both support the correct protocols you need, youre in al kinds of trouble. Any possebility of plugging in the phone...

Mmm, missed that one. Glad you found at least an answer. Richard.

Hi adkent, the problem is reported here : [2012-01-10 13:00:05] WARNING[2833] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 2 (Critical Response) -- See...

I breaks at the Dial command within a second, so iam not sure, but i think a SIP trace would be usefull here. I have to say i'am not familiar with polycom, but maybe i can help with the SIP trace. Also how are you transfering, are you using the InCall transfer(*2 or ##) of using a phone key...

Try looking at the log while opeming the main page. At the command prompt do : tail -f /var/log/httpd/error.log And see what error you get. Good luck. Richard

Mmm was afraid of availability for you. Phone is brand new, droped is off at customer site yesterday, will conigure it tomorrow if time allowes, but will let you know. Richard

I think you have a queue containing the number as an agent. Richard

Does this get anywhere close : http://www.doro.co.uk/products/Corded-phones/337ip/ Just deliverd one to a customer. Richard.

thanks for all the respone, the hard reset has worked and and i'am able to program the function keys now. i do have tho say the reset gave me a nice suprise.... i now have yahoo RSS feed on my Idle sreen as a present :D Indeed the BLF is only signally call status, and not ringing, it's a shame...

Whow, i love the reply speed in this forum..... Thanks for the link, phone is rebooting as i type..... I do have another question, could you or someone explain if setting the phones Advanced SIP features ON would benefit whe using it aith asterisk, i could only find references to the use of...

Hi Guys, Does anyone know how to Hard reset a Mitel 5220 Phone, i have a problem with some key programming witch i cannot change, looks like te config is messed up. i got a key that displayes '29' in the and no matter how i change it it stays that, it the second phones this happens on. I tryed...

The ETH0 interface will get the gateway and DNS from the firewall by DHCP so the machine will know that the internet is out via ETH0, when you put in antoher NIC this will be ETH1, then assign a static IP and no gateway, the machine knows its subnet is locally connected. So you have routing for...

Randy, in case you have missed it, i think you have a syntax error,the priority s,n(usercid) Is doubled, and when jused als a jump target asterisk will jump to the last priority label, effectively skiping the priority just above. Jusst remove the second (usercid) from the priority. Richard