clarity

Still would very much like to get ahold of the version for Debian. How do I get it? There's no reference to where to download it here in this thread or in the general software download areas. The reference in this thread is only for Centos or CS410 or windows. Thanks, Steve

Is debian5 (Lenny) i386 (32bit) officially supported or semi officially supported? In other words should I proceed to use Debian5 since it is newer and considered stable at least from a Debian community standpoint or do I need to make special effort to go back and use Debian 4 with any pbxnsip including 4beta and forward? Thanks. Steve

Where can I download this debian version? Only Centos/WIndows and CS410 are mentioned above. THANKS! Steve

Thanks!

Can you please tell me.. For both the beta and pbxnsip 3.x versions. 1. is it supported in Debian? can we get a debian supported download of the beta? 2. is ithe beta supported in Centos version 5.3 both i386 and 64bit versions? 3. Is pbxnsip supported in any 64bit linux? b. Is there a benefit to using 64bit OS beyond huge ram? c. Is there a benefit to AVOIDING a 64 bit OS? 4. is pbxnsip supported in Debian 5 stable? 5. is it supported in Debian 64bit Linux? 6. is it supported in 64bit windows? THANKS!

What I think you are telling me is to simply have them do just the hot desking (4xx) extension numbers in your example. Make sure the 4xx phones are in the right agent groups but do not log onto the quoe so skip that step. They are already setup exactly like that.. just that they have been doing the log onto the que step as well as hot desking. So I should simply have them *not log onto the queue* and the just use the hot desking at 4xx (6xx) in our case. And this should clear up the problem. (make sure 4xx) (6xx) in our case extension numbers are in the right que(s) hen simply logg off hot desking if not wanting to be in the queue so to speak. I will have them try this. Thanks.. Steve

I had shared this directly with pradeep earlier and he had suggested there may need to be a software update to fix this: I am also cross posting this issue to the public forum per Pradeep's request. 3rdly I have opened a support case and copied this information there as well. THANKS! --- I’m not sure exactly how to go about trouble shooting this problem. I imagine it’s simple. We have a customer of whom I’ve set up hot desking where they log onto their extension number (601) Hot desk. Then they log onto the agent queue by pressing *64 (agent login) The person (601) is entered into a number of agent groups as defined in the agent groups themselves.. You’ll see “601” is part of maybe 5-10 agent groups. The customer is complaining that they are still getting inbound calls to their phone (from other agent groups) of which they are part of when they are already on the phone.. And that the other agents who are logged in are not getting the calls sometimes… In other words it does not always go to an available agent who is not on the phone but is ringing him (601) yet he’s already on the phone having answered a phone call earlier.. If you need to look on our server. The specific extension (hot desking acct) is 601, he’s logging onto phone (501) and then logging into the agent queue by pressing *64. Server hostname: ******.***.*** Username: **** Password: **************** His expectation is that he should not receive any agent queue type calls when he’s already on the phone but someone else who is not on the phone but in the agent group should be getting the second inbound call rather than him. Here are the call details that he was able to get me on the last time it happened… I’ve asked him to try and record the events when it happens and to see if it can be recreated. Do you see anything obviously wrong or have any suggestions on where I can begin to test? Thanks! [Pradeep's response:] Hi Steve, Here is the response from Christian. Looks like we need to add some code to make the behavior that you are looking for. Pradeep Subject: Re: FW: Pbxnsip Hot Desking Ongoing complaint from customer. Hmm. I believe it is a "feature". The agent is something "logical", while the phone is something physical. The PBX knows that the agent is busy talking, but is not sure about the specific device. I believe the workaround must be that the phones that are used as hot desking devices should not be part of a ACD.

This series of questions is geared toward using pbxnsip in a hosted pbx environment (providing services to many many clients). I have to prepare for and expect to be able to support 50,000 phones & up as our company grows & expands. This started out as an email question and per Kevin's suggestion I am posting it on the forum as well so Christian can have a look and answer. --------- I have a few questions you might be able to help me with regarding load balancing large numbers of pbxnsip servers and interrop with session border controllers. Do you have any experience with placing pbxnsip behind any of the well known commercially available and/or open source session border controllers? And can you share your experiences with us? how did it go? what works what does not work? I'm somewhat familiar with the process of using OpenSER as an SBC and placing asterisk & other servers behind it and doing all of the nat traversal/call translations etc in the SBC section of the network. I'm curious of your thoughts toward eventually employing this type of approach with PBXnsip versus having 50+ pbxnsip servers directly on public IP addresses where pbxnsip is handling virtually all of the NAT and call routing/translations. Do you think this approach makes sense? and do you think it would be relatively 'plug -n- play'? Or do you think we'd have a HUGE interrop/development task on our hands before it would work. Have you ever been through this interrop process with any SBC product? My thought today would be to put 10-50 pbxnsip servers behind OpenSER and have OpenSER handle all of the NAT traversal/internal network 'topology hinding' features and place the pnxnsip servers on private non routable IP space behind OpenSER.. yet still maintain and have ALL of the pbxnsip features be able to continue to work behind OpenSER. features like BLF intercom IM really all pbxnsip features that we have today... Do you think they will work behind an SBC like OpenSER? or do you think they would all break due to interrop issues with an SBC, where pbxnsip no longer has total control of the public Internet Interface? Thank you very much for your time and your thoughts!! I am currently working with OpenSER and will be steadily becoming much more familiar with all the ins & outs of this software as we move on. I've also been working on a daily basis with the Stratus ENTICE Softswitch/SBC for about 3 years now. Take Care! Steve

Don't rely on just those customer reports of audio dropping off to determine you are nearing your max CPU for PBXNSIP. Get an IP phone plugged directly into the lan or WAN of the server.. try both. During the busiest time of the day and during those types of loads test with a locally connected telephone and make calls into/out of voicemail. If it's not choppy and sounds perfect in both directions (recorded messages) I'd bet you have plenty of CPU/RAM/IO to spare. Also try PSTN calls if you have locally connected PSTN. (not distant voip). Just my immediate thoughts.. Maybe you have already done these tests. -Steve

Tested with Polycom Phones (IP-650) (IP-670) and IP330 If I change the top (preferred codec) to be G722 in PBXNSIP.. Either globally (settings) -or at the individual account.. G722 calls work great between supported phones and to PBXnsip IVR/Voicemail prompts. However all calls break to other phones that do not support G722 such as Polycom IP-330 or to the PSTN gateway which is G.711 This is not expected behavior. I'd expect if one leg of a call.. be it another phone that does not support G722 or the PSTN Gateway (G711 only) should cause both legs of the call to fall back to the common accepted codec (g711). This is not happening... the IP 650's and 670's are still showiing "HD" in the display and get NO audio with the other end of the call which is G711. G722 phones sem to 'have no clue' that the far end is not supporting G722. I can provide packet traces if needed. Our test box is on Debian4/Intel Xeon. 3.1.0.3043 (Linux) License Status: Demo License (3 Minutes) License Duration: Permanent Additional license information: Thanks! Steve

Looks like the problem(s) are that the examples given on that site do not work. I'm guessing the web interface tries to do it the same way which does not work. Further looking around shows that you have to issue the command like this: The -c and -p must come FIRST then the arguments. *this works* pbx:/srv/pbxnsip# taskset -c -p 0 3330 pid 3330's current affinity list: 2 pid 3330's new affinity list: 0 pbx:/srv/pbxnsip# Doing them the way that website suggests: # taskset -c 1 -p 13545 Absolutely does not work in my Linux installation as well as some others I had read about. I'd still like to be able to set it from the web interface.. Is that hard coded? or does it fire off an external script that I can 'fix'? ;-)

Logfile shows "Set processor affinity to 1 failed" on each service startup. How do I fix this? I really would like to lock it down to one CPU core and avoid RTP the associated jitter problems of 'core hopping'. This is a quad core Xeon server and I'm running PBXNSIP 3.0.1.3023 (Linux) Deban 4.0 up to date. SMP kernel. Any suggestions? pbx:/proc# cat version Linux version 2.6.18-4-686 (Debian 2.6.18.dfsg.1-12etch2) (dannf@debian.org) (gcc version 4.1.2 20061115 (prerelease) (Debian 4.1.1-21)) #1 SMP Wed May 9 23:03:12 UTC 2007 pbx:/proc# cat cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel® Xeon CPU 3.20GHz stepping : 1 cpu MHz : 3192.275 cache size : 1024 KB physical id : 0 siblings : 2 core id : 0 cpu cores : 1 fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips : 6388.27 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel® Xeon CPU 3.20GHz stepping : 1 cpu MHz : 3192.275 cache size : 1024 KB physical id : 0 siblings : 2 core id : 0 cpu cores : 1 fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips : 6384.06 processor : 2 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel® Xeon CPU 3.20GHz stepping : 1 cpu MHz : 3192.275 cache size : 1024 KB physical id : 3 siblings : 2 core id : 0 cpu cores : 1 fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips : 6384.05 processor : 3 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel® Xeon CPU 3.20GHz stepping : 1 cpu MHz : 3192.275 cache size : 1024 KB physical id : 3 siblings : 2 core id : 0 cpu cores : 1 fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni monitor ds_cpl cid cx16 xtpr bogomips : 6384.12

Not very easily in this case... the switch is expecting that I will send it the ttl expire value that I want in the registration requests. Every other sip endpoint I have used allows me to change this. And I've not had a problem setting it one any other equipment yet. Is there some way I can change this in pbxnsip a config file maybe? Thanks! Steve

Don't know if this helps or applies... But it works for me on the older Cisco phones and V7 or 6.5 Steve nat_enable: "1" voip_control_port: "5060" start_media_port: "11200" end_media_port: "11300

I think what we are really after here is a means to set the sip expiry value to 120 instead of 3600.... In most endpoints/phones that I work with when I set the re-registration period this is also set simultaneously on that device... I need to get pbxnsip to do this... I need to be able to set the expiry to less than 3600 (120 or 60 in this case). Updating ticket with actual sip header info: Steve