On 3 Oct 2009, at 20:38, James Stocks wrote: > On 3 Oct 2009, at 16:37, Jonathan Thurman wrote: > >> On Sat, Oct 3, 2009 at 6:17 AM, James Stocks <stoc...@stocksy.co.uk> >> wrote: >>> Hi everyone, >>> >>> I hope someone can help me with a problem I'm having with Cisco 7940 >>> phones on the SIP 8.12 image. When I place a call from one of the >>> handsets, the call proceeds as normal for 20 seconds and is then >>> terminated by Asterisk (1.4.26.2): >>> >> >> We are runing 08-12-00 on 7940/60s just fine (Asterisk 1.6.1.1), and >> have been for a while. >> >>> >>> As far as I can tell, the 'a=silenceSupp:off - - - -' header is not >>> accepted by the 7940, which seems like a bug in the SIP image to me. >>> However, I can't find a way to report this problem to Cisco >>> without a >>> support contract (which I do not have). Reverting to version 7.5 >>> fixes the problem, but it is still present in 8.11. The problem is >>> not present if the PSTN initiates the call, nor is it present if I >>> allow the handsets to reinvite each other. Here's the sip.conf >>> snippet if it helps: >>> >> >> That all looks fine to me. What do your SIPDefault.cnf and >> SIP<MAC>.cnf files look like? >> >> -Jonathan > > Hi Jonathan, > > Thanks for your reply. Here's the two files, SIPDefault.cnf: > > > # Image Version > image_version: "P0S3-8-12-00" > > # Proxy Server > proxy1_address: "pabx.spruce" # IP address here alternatively > > # Proxy Registration (0-disable (default), 1-enable) > proxy_register: "1" > > # Setting for Message > messages_uri: "222" > > # Time Server > sntp_mode: "unicast" > sntp_server: "snakebite.spruce" # IP address here alternatively > time_zone: "GMT" > dst_offset: "1" > dst_start_month: "March" > dst_start_day: "" > dst_start_day_of_week: "Sun" > dst_start_week_of_month: "4" > dst_start_time: "02" > dst_stop_month: "Oct" > dst_stop_day: "" > dst_stop_day_of_week: "Sunday" > dst_stop_week_of_month: "4" > dst_stop_time: "2" > dst_auto_adjust: "1" > date_format: "D/M/Y" > > # XML file that specifies the dialplan desired > dial_template: "dialplan" > > #Time Format (0-12hr, 1-24hr [default]) > time_format_24hr: "1" > > # URL for external Phone Services > services_url: "http://pabx.spruce/openxmldir/PhoneUI/index.php" # IP > address here alternatively > > # URL for external Directory location > directory_url: "http://pabx.spruce/openxmldir/PhoneUI/index.php" # IP > address here alternatively > > # URL for branding logo > logo_url: "http://pabx.spruce/cisco/asterisk.bmp" # IP address here > alternatively > > > and SIP<mac>.cnf: > > > # Image Version > image_version: "P0S3-8-12-00" > phone_label: " " > > # Line 1 appearance > line1_displayname: "James" > line1_shortname:"200 James" > line1_name: 200 > line1_authname: "200" > line1_password: "*removed*" > > # Line 2 appearance > line2_displayname: "Work" > line2_shortname: "206 Work" > line2_name: 206 > line2_authname: "206" > line2_password: "*removed*" > > # Line 3 appearance > line3_displayname: "" > line3_shortname: "" > line3_name: UNPROVISIONED > line3_authname: "UNPROVISIONED" > line3_password: "UNPROVISIONED" > > # Line 4 appearance > line4_displayname: "" > line4_shortname: "" > line4_name: UNPROVISIONED > line4_authname: "UNPROVISIONED" > line4_password: "UNPROVISIONED" > > # Line 5 appearance > line5_displayname: "" > line5_shortname: "" > line5_name: UNPROVISIONED > line5_authname: "UNPROVISIONED" > line5_password: "UNPROVISIONED" > > # Line 6 appearance > line6_displayname: "" > line6_shortname: "" > line6_name: UNPROVISIONED > line6_authname: "UNPROVISIONED" > line6_password: "UNPROVISIONED" > > # Phone Prompt (The prompt that will be displayed on console and > telnet) > phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP > Phone) > > # Phone Password (Password to be used for console or telnet login) > phone_password: "*removed*" ; Limited to 31 characters (Default - > cisco) > > # User classifcation used when Registering [ none(default), phone, > ip ] > user_info: none
OK. For anyone finding this thread, the problem exists in Asterisk 1.4, but upgrading to Asterisk 1.6.1.6 appears to eliminate the problem. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users