hello everyone, Please excuse me for this rather long email as I wanted to explain the steps taken in details which is a complete mystery to me and desperately looking for solution.
We are experiencing some rather weird behaviors with the Polcyom 330 and SipX 3.10.3 (upgraded via yum from Centos ISO 3.8 installation) connected to Mediant 1000 Gateway. We were using SIP firmware 2.1.2 with Bootrom 4.0 with sipx 3.8 While we had sipx 3.8 we were using AudioCodes MP118 FXO and recently switched that with Mediant 1000 followed by yum update to latest stable sipx 3.10.3 After the sipx 3.10.3 (knowing that it support firmware Polycom firmware 3) update and restart, i logged into sipxconfig and gone through phone-groups (Select Polycom 330 which pops-up the configuration pages) and noticed that the dropdown that normally shows the configuration/firmware version (1.x or 2.x) is no longer visible and there are new sections like Licensing (which im sure it wasnt there in sipx 3.8). So i assumed the 3.10.3 not showing the version number of the Polycom 330 firmware in the Phone configuration screen, should be normal and gone through the settings to make sure everything is inline with how we configured Polycom 330s previously. Following that, as there were new config features in the UI, i uploaded the latest SIP firmware 3.1.1 and boorom 4.1.2 and published to all the phones. The phones picked the config but after several use start freezing out of blue. I suspected from bootrom and downgraded bootrom to 4.1.1. The freezes become much less frequent but still being freezing at the end of the day. I've decided to switch back to SIP firwamre 2.1.2 with BootRom 4.0 which i knew the phones get less or almost no freeze. The freezing of phones drop dramatically but then a new big mystery arise...... Also i would like to highlight, since we got a new ISDN-30 line, in each user's CallerID property and Alias, I've keyed in their direct number (DID) which allowed us to forward the incoming number to the correct person and also show the correct callerid for outgoing calls. Which worked as expected and didnt have problems with CallerID. Then i start receiving feedbacks of user A talking to customer B, and a new call comes in from customer C which automatically cuts the caller customer B off the line and becomes the active call of user A (talking to customer C). I have checked the Mediant Gateway, contacted their support and the only thing they can say is "Enable"/"Disable" Call Waiting in the Mediant to see if it makes any difference. Which it didnt. Disabling Call Waiting feature on the Mediant level didnt do any change to the situation. Then i went back to Sipx 3.10.3 Phone group, select a group's Polycom 330 template and under features disabled unnecessary stuff like Call Recording etc. and enabled Call Parking and published the configuration to the phones. Now the behaviour experience above take a different twist; while user A is in the call with customer B and customer C callsin to the same number, customer B will not be disconnected and can hear user A but user A cannot hear Customer B. Again i would like to highlight, the only changes done was to enable/disable so-called redundant features (i.e. call recording) and re-publish to the phones. When i go into sipx configuration phone group polycom 330 template, i still dont see the dropdown selector where you can select version 1.x, 2.x. Since i am running a SIP firmware (2.1.2) which does not freeze the Polycom 330 phones, the Polycom 330 Template file is i believe still the latest version 3.x (because i can see licences section in phone configuration page - which i didnt see in sipx 3.8.1) The main trouble is these two mystery issues and how should i go about solving them. Your help and guidance would be very much appreciated Case 1: user A talking to customer B, and a new call comes in from customer C which automatically cuts the caller customer B off the line and becomes the active call of user A (talking to customer C). Disabling call recording and few (not really related) features from sipxconfig polycom 330 phone template and pushing to phones gives the Case 2 as following: Case 2: while user A is in the call with customer B and customer C callsin to the same number, customer B will not be disconnected and can hear user A but user A cannot hear Customer B Thank you for your help in advance. _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
