Vodia PBX Posted September 21, 2007 Report Share Posted September 21, 2007 See: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2108.exe http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2108 It seems that the RTP pass-through problems are fixed. Also, this version now always sets the processor affinity to a value (1 is default now). Looks like this version solves a lot of problems. Quote Link to comment Share on other sites More sharing options...
hosted Posted September 22, 2007 Report Share Posted September 22, 2007 Logo Link: under the Appearance tab still doesn't work. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 22, 2007 Author Report Share Posted September 22, 2007 Now you need to put a file path there, e.g. img/logo_custom.gif located in the html/img/logo_custom.gif file relative to the working directory. Quote Link to comment Share on other sites More sharing options...
kcnd Posted September 23, 2007 Report Share Posted September 23, 2007 See: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2108.exe http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2108 http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2108 It seems that the RTP pass-through problems are fixed. Also, this version now always sets the processor affinity to a value (1 is default now). Looks like this version solves a lot of problems. Paging is now working with relaease 2108 - I'm not sure what changed, but all is well now. I did not have to apply a new ringtones.xml file. No issues came up from using 2108 under a normal business day on Friday, so it appears to be stable at this point. Have not had any more issues with DTMF and some of the echo issues we were experiencing have gone away (still have other echo issues related ot he phone devices, but pbxnsip appears to be clean). ALso, some feedback on performance: we are using a server with two dual-core Xeon 3 Ghz processors, 4 GB RAM and 1GB NIC with Windows Server 2003 R2. One of the eight virutal processors is dedicated to pbxnsip via affinity control. Under typicall business office call loading, we see no more than 5% CPU utilization and 15% network utilization (several hundered calls per day with 25 key uesrs and 35 total users; 7 PSTN lines via a gateway with as many as 5 in use at any one time). Even under unicast paging of 25 phones, the CPU utilization doesn't spike higher than 15%. Seems release 2108 is quite efficient. Quote Link to comment Share on other sites More sharing options...
hosted Posted September 23, 2007 Report Share Posted September 23, 2007 Now you need to put a file path there, e.g. img/logo_custom.gif located in the html/img/logo_custom.gif file relative to the working directory. oh you cant have a URL to an external site anymore? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 24, 2007 Author Report Share Posted September 24, 2007 Nope, the file must be on the local file system now. Should not be a too hard requirement IMHO. Quote Link to comment Share on other sites More sharing options...
David McGowan Posted September 27, 2007 Report Share Posted September 27, 2007 I am running 2.1.0.2108 and everything seems to be working great... One of the reasons I upgraded was to try out TAPI... I have downloaded pbxtsp10.exe from the site and have it installed and properly setup in Control Panel... When I go to Dial in Outlook it just stays at Dialing and nothing happens... The Click To Dial with the link works, but I'm trying to get the Outlook dialing working... Is there anything else I should setup? Are there any other ports that should be open in the firewall? I left all the ports default... thanks! david Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 27, 2007 Author Report Share Posted September 27, 2007 Do yo usee SUBSCRIBE with the Event x-tapi showing up on the PBX? If not there must be something with the TSP setup. And you have to reboot your PC after installing the TSP and setting it up. Quote Link to comment Share on other sites More sharing options...
David McGowan Posted September 27, 2007 Report Share Posted September 27, 2007 It works now... I rebooted both the PC and the Server and it works... I feel stupid... thanks! david Quote Link to comment Share on other sites More sharing options...
phsean Posted September 27, 2007 Report Share Posted September 27, 2007 This appears to be a Unified Messaging issue with build 2108 (unless it's also due to the lang_en.xml that we removed): before this build, missed call notifications and voicemail notifications had the correct caller/caller ID information attached to it. 'Missed call from Jack', or 'voicemail from 3015555555'. Now, all missed calls and voicemails appear to come from the user themselves, for example, all my voicemails say: 'Voicemail from Sean'. We went straight from 2.0.3.1715 to build 2108 (and removing the lang_en.xml), so if this is a 2.1 bug I don't know how long it's existed. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 27, 2007 Author Report Share Posted September 27, 2007 2.1.0.2111 is out: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2111.exe, only Windows so far. It fixes problems with certain versions of Polycom phones putting callers on hold (using a=inactive) and another Problem with Linksys phones that believe that codec 18 is G729a. Quote Link to comment Share on other sites More sharing options...
hosted Posted September 29, 2007 Report Share Posted September 29, 2007 Nope, the file must be on the local file system now. Should not be a too hard requirement IMHO. Still doesnt work. The XML is ok I can manually put in the URL and pbxnsip shows me the logo "http://pbx.com/img/logo.gif" However pbxnsip will always show: http://pbx.com/img/main_logo.gif i update it and update it and it never shows anything different. db Quote Link to comment Share on other sites More sharing options...
hosted Posted September 29, 2007 Report Share Posted September 29, 2007 did you say earlier G722 was supported in this version? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 29, 2007 Author Report Share Posted September 29, 2007 G722: Yes. Also known as codec 9. But you must list it in the codec_preference setting. Image: I attached a screenshot of something that works. Quote Link to comment Share on other sites More sharing options...
hosted Posted September 30, 2007 Report Share Posted September 30, 2007 aparently i dont have permissions to view the attached file. What about the logo? am i crazy or is it broken? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 1, 2007 Author Report Share Posted October 1, 2007 Permission problem in the forum... Try again. Quote Link to comment Share on other sites More sharing options...
phsean Posted October 2, 2007 Report Share Posted October 2, 2007 This appears to be a Unified Messaging issue with build 2108 (unless it's also due to the lang_en.xml that we removed): before this build, missed call notifications and voicemail notifications had the correct caller/caller ID information attached to it. 'Missed call from Jack', or 'voicemail from 3015555555'. Now, all missed calls and voicemails appear to come from the user themselves, for example, all my voicemails say: 'Voicemail from Sean'. We went straight from 2.0.3.1715 to build 2108 (and removing the lang_en.xml), so if this is a 2.1 bug I don't know how long it's existed. Is anyone else running UM alongside a PBXnSIP 2.1 RC version that can either confirm or deny that this is a problem with 2.1? That's when we started to see it. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 2, 2007 Author Report Share Posted October 2, 2007 The missed call emails that I am getting are okay. Are you using your own files in the html directory? Quote Link to comment Share on other sites More sharing options...
gotvoip Posted October 2, 2007 Report Share Posted October 2, 2007 2111 is still having RTP issues with the linksys phones. Quote Link to comment Share on other sites More sharing options...
Detlef Posted October 3, 2007 Report Share Posted October 3, 2007 Just installed the .2111 and still have so far only two smaller issues: - Music on hold is not working with the pre-configured moh.wav file. It's silent to the caller if a call is on hold or parked. - The auto attendant does not announce "please enter the extension" after the customer recording ends. Those worked before with 2.0.3.1715. The .2111 release was a fresh clean install from 2.1.0.2093 and upgraded to .2111 Detlef Quote Link to comment Share on other sites More sharing options...
phsean Posted October 3, 2007 Report Share Posted October 3, 2007 The missed call emails that I am getting are okay. Are you using your own files in the html directory? Thanks, yes there are a few -- we've been working alongside Model12 on some Polycom issues and so right now what I have in the html directory are four files: polycom_adrbook.xml, polycom_master.xml, polycom_phone.xml, and polycom_sip.xml. The modifications that have been made by us to those files are all in the polycom_sip.xml file, and they are: - a hardcoded dialplan: <digitmap dialplan.digitmap="[2-6]xxx|7xxxx|[2-9]11|91xxxxxxxxxx|9[2-9]xxxxxxxxx|*xx[2-6]xxx|xx.#" dialplan.digitmap.timeOut="3"/> - a modded <CALL_WAITING tone.chord.callProg.6.onDur value - switched the <volume voice.volume.persist.handset value to 1 Thanks for the help... Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 3, 2007 Author Report Share Posted October 3, 2007 Here we go with 2.1.0.2113: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2113.exe. Changes: Some Email servers had a problem with the base64-encoded format, that was changed. Also the SOAP CC authentication (PIN) was not working. Credentials based on P-Preferred-Identity (inbound) were not working. The ACD IM messages could "stall". Linksys provisioning files updated. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 4, 2007 Author Report Share Posted October 4, 2007 Here we go with 2.1.0.2114: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2114.exe. Changes: Deal with the SDP bug in Linksys and this image also contains automatic provisioning for Linksys devices. Quote Link to comment Share on other sites More sharing options...
David McGowan Posted October 4, 2007 Report Share Posted October 4, 2007 Just upgraded to 2114 and it fixed my email issues from 2108... I have a question about codecs and my sip.cfg... Here is the section in my current sip.cfg file... <codecs> <preferences voice.codecPref.G711Mu="1" voice.codecPref.G711A="2" voice.codecPref.G729AB="3" voice.codecPref.IP_300.G711Mu="1" voice.codecPref.IP_300.G711A="2" voice.codecPref.IP_300.G729AB="3" voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2" voice.codecPref.IP_4000.G729AB=""/> <profiles voice.audioProfile.G711Mu.payloadSize="20" voice.audioProfile.G711Mu.jitterBufferMin="40" voice.audioProfile.G711Mu.jitterBufferShrink="500" voice.audioProfile.G711Mu.jitterBufferMax="160" voice.audioProfile.G711A.payloadSize="20" voice.audioProfile.G711A.jitterBufferMin="40" voice.audioProfile.G711A.jitterBufferShrink="500" voice.audioProfile.G711A.jitterBufferMax="160" voice.audioProfile.G729AB.payloadSize="20" voice.audioProfile.G729AB.jitterBufferMin="40" voice.audioProfile.G729AB.jitterBufferShrink="500" voice.audioProfile.G729AB.jitterBufferMax="160" voice.audioProfile.Lin16.payloadSize="10" voice.audioProfile.Lin16.jitterBufferMin="20" voice.audioProfile.Lin16.jitterBufferShrink="500" voice.audioProfile.Lin16.jitterBufferMax="100" voice.audioProfile.Lin16.frequency="16000" voice.audioProfile.Lin16.payloadType="117"/> </codecs> I have a Polycom IP430... I want to use the best codec possible for sound, I have no preferences set on the Trunk.. Are the preferences in the above sip.cfg supposed to match with pbxnsip? If so what should they be set? What is a good default for trunk codec preferences? Thanks.. david Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 4, 2007 Author Report Share Posted October 4, 2007 The PBX takes the freedom to resort the list of proposed codecs according to its internal preferences. As long as the user agent promotes all available codecs it is fine. The 2114 build also has a global codec preference in the system admin/settings/ports section. Did you see that? Quote Link to comment Share on other sites More sharing options...
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