search for: 55i

Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4? Specifically with the Aastra 55i. Specifically, I am using the Aastra 55i with the expansion module. We want to see if other handsets are being used. (BLF) Getting BLF to work would be a great start. It sounds like setting up the hints properly will achieve this. right? Not totally sure how this should be configured. We als...

Hi Everyone, We use the top buttons on Aastra 55i to login and logout from Queues. This is the order: Button 1 = Login to English Queue Button 2 = Login to Spanish Queue Button 3 = Logout of English/Spanish Queues There are indicator LEDs on each of these buttons. Is there anyway we can send a SIP request or some other communication to get the...

Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP, V2.5.3.18, January 2010 , English , ZIP , 2,849 KB on the site: http://www.aastra....

...t; Nice To meet U and Good Morning. I got u r mail-Id from > http://www.asterisk.org/node/48325 > Recently i started the SLA configuration. But i didn't understand the > Flow of its Functionality > One of the My Client Ask to have do deploySLA feature > He Using the Aastra 55i, when users is busy , Aastra 55i will blink lamps > > in SLA.conf > > slatest] > type=trunk > device=SIP/1001 > autocontext=slatest > [slatest1] > type=trunk > device=SIP/1003 > autocontext=slatest1 > [slateststation] > type=station > device=SIP/1002 > a...

...ING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.215', "port" = '5060', "regseconds" = '1392692130', "defaultuser" = '4000', "useragent" = 'Aastra 55i/2.5.0.82', "lastms" = '-1', "fullcontact" = WHERE "name" = '4000'': near "4000": syntax error [2014-02-17 20:55:03] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = &...

...will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is "OK" but is discontinued hardware. Help? Thanks, Steve

We have several types of phones, cisco 7940/7960 aastra 55i/9113i/ grandstream gxp2010 I want to be able to implement hotdesking where an agent will logon to any phone. I got all of that working, without having to reboot phones, but then hit a brick wall. Voicemail. I still want each phone to use the BLF for voicemail indication, and to use the &quot...

Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones. Local outbound calling works fine, but ATT requires clients enter 7 digit code for long distance. All calls with 7 digit code are lost within 20 seconds of the call. This is the message I?m getting: Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on tr...

...INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "SIP Phone - Ext. 202" <sip:202 at 192.168.50.5:5060 ;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D25B72F>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.5.2.1500 Content-Type: application/sdp Content-Length: 594 Basically the phones should only send with FROM their local 192.168.100.0/24address and Asterisk should only send ANSWER and ACK back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24 (which is the openvpn client ip). Once abo...

Hello all, Our setup consists of an asterisk server (version 1.4.8 ) with a TDM400 card (4 FXO ports) and about 10 aastra 5Xi phones (2 phones are 57i and the rest 55i). The asterisk server is also configured with two SIP trunks. The problem that we're having is that if at any time there is an internet connection interruption to the asterisk server all the aastra phones freeze. If we disable the two SIP trunks in the asterisk configuration the phones are fin...

...Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 101-test SIP Options : 100rel gruu path replaces replace timer Codecs : 0x106 (gsm|ulaw|g729) Codec Order : (g729:20,ulaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : OK (34 ms) Useragent : Aastra 55i/2.6.0.1008 Reg. Contact : sip:101-test at 10.211.0.42:5060;transport=udp Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : parkinglot_test Use Reason : No Encryption : No...

Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 & 942's. Just curious about call quality, programability, and functionality with asterisk. I have read through the literature, but would like some real world feedback. Thanks

Hello all, I've been asked to look into my home dial plan to see if I can improve it by an important customer (my wife). What we would like to have happen is that an inbound call rings all the phones (This is done). Once one phone picks up, of course all the others stop ringing (Also done). Here's the gotcha. She doesnt like having to "transfer" calls to another phone;