Vodia PBX
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Yes, it is on the short list. But it will be in the version 3.
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It would probably better to split this topic off to a new topic, as it is not directly related to the 2.1.7 release.
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how cool would this be to have a binary version for mac os x?
this should just be a recompile with xcode no?
There are a couple of differences. MAC OS is a little bit like BSD. Not sure if MAC is a significant market, we were also thinking about Solaris. Sun has a very good reputation in the server market, and that is what you want to hear when you are talking about a PBX platform.
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It would be nice if it would also work with pidging/adium, they are just chat clients and thus only support SIP/SIMPLE and not a full SIP stack.
Do you have a LOG with the SIP messages? No problem if they don't support other messages, they only need to support REGISTER, MESSAGE and properly reject incoming INVITE messages (e.g. with 500 Not Implemented).
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The SIMPLE support is very simple. The PBX practically works like a SIP proxy in this case, it just takes a MESSAGE request and sends it to all registered user agents. There is no store/forward. I would say, register the device and give it a try.
So far I have seen it working with Counterpath (did not try anything else).
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When I hear "static" the first thing that comes to my mind is SRTP trouble. Is there a crypro heder in the SDP? If you can still hear the real audio it is not SRTP.
The PBX itself is "pure digital" and does not introduce noise into the RTP stream. Does it make a difference what you are calling? Maybe there is a audio loop if you are talking to something in the room or in the office.
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RTP-RxStat: Dur=25,Pkt=259,Oct=8288,Underun=0
RTP-TxStat: Dur=5,Pkt=259,Oct=8288
That indicates that the PBX received the same amount as it send (probably the 20 seconds time difference come from ringing). From that persoective, it does not "sound" like one-way audio problem.
The packet size for the 200 okay is in the 1200 bytes range. There are only 300 bytes left until you will experience UDP fragmentation. That might be a problem when you offer more codecs or add another Record-Route into the packet (or just dial a very long name).
Also check http://wiki.pbxnsip.com/index.php/One-way_Audio.
It would probably better to split this topic off to a new topic, as it is not directly related to the 2.1.7 release.
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You are correct, it only offered camp on when calling from a cell phone associated with an internal account. Now that I realize that, I can accept it, however I cannot accept that it offers when there is no answer.
Why not for "no answer"? I think it is nice if someone is out for lunch, you want to talk to him and get a call back after he is back from lunch and finishes the first call.
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That is a "famous" problem... Some operators accept foreign caller-ID, others (most) don't. I think in SS#7 is it quite okay to have one "display name" caller-ID which is presented to the end user, and at the same time the network-asserted caller-ID, which is primary used for billing purposes and maybe legal stuff, and which is (usually) not presented to the end user.
Tricky topic, as there are companies that spoof caller-ID this way and make people call back on expensive numbers. That is why operators are so scared about presenting any caller-ID, they might be afraid of being liable for these fraud cases.
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Ehh... actually I have some experience with that from a previous project. Problem is, we gotta support that! I agree, it is extremly flexible, but it also means that you have to deal with endless loops, stack overflows, in other words with the programming language comes the debugger. The next thing is VoiceXML. Out intention was to create a PBX that can be used by people who don't want to do programming, it should just cover 95 % of the cases which are out there. If you want to customize to a higher lever, you can still use an external VoiceXML processor and run the call there...
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With a second caller on a call, they are disconnected when you call another number and press tones to access a menu. The second call continues.
Is that a phone problem or a PBX problem? 2nd call coming in is usually a problem for the phone. Or is it the one-way media timeout because the call on hold is not being refreshed? I would say DTMF interference should be extremly unprobable.
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You should receive a black one. They use a new SoC chip which was a major step regarding stability. If you receive a white one tell us and we'll take care about this. All new devices sold should be black.
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Is there a way for each individual to set their zero out from their ext? For ezample one of the VP's want the zero out once in his vm to go to his admin asst.
There is a "mailbox escape account" (see http://wiki.pbxnsip.com/index.php/Extension#Redirection) that should do that job.
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I am running 2.1.8 and the PBXnSIP is now offering call camp when the extension does not answer, and rolls to voicemail. It is also offering call camp to external callers. Is there a way to make it only offer when the call comes from the inside, and the extension is in use?
You mean from any external number? Or just numbers that are listed as cell phone in an extension?
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How can I disable the "press 1 to receive a callback" when an extension is busy, I just want the user to hear the MoH and hear the "the extension is busy at the moment" after some seconds, and then just wait till the extension becomes available.
How can I configure this? if not possible, please include it, our clients are asking this many times:)
There is a settings called "Offer Camp On" in the admin settings. This will turn this feature off.
You mean the clients prefer to hear music on hold while the other side is talking talking talking talking? What if it takes 5 minutes, 10 minutes, one hour?
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Can you define what "not ready for VoIP" means? We have not seen this message in previous versions of pbxnsip.
Well if the router creates a NAT binding, it is supposed to keep that binding when there is traffic on that binding. When it drops the binding, and during that time the PBX sends a request to the old binding port, the request will not be forwarded to the phone. This creates a blind spot.
We have seen routers that have exactly 32 entries for NAT. If you open the binding 33, then it just drops binding 1. This especially happens when a PC behind that router does some kind of activity, practically kicking out SIP during that time. If you see the log message in the PBX, you should check if the router is responsible for that. If you have several routers in operation, it should be easy to check the model and maybe easy to isolate the router that is causing this kind of problem.
The ugly thing is that is works fine most of the time and it is extremly difficult to find out why devices are offline from time to time. That is why we put that log in.
BTW The same applies to changing IP addresses. Service providers in Asia now start to change/recycle IP addresses every 4 (four!) hours because if the address shortages there. I think everybody understands that this is not increasing the registration stability.
Bye bye, NAT. Here you go, IPv6!
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This is a serious message because the NAT router is not really ready for VoIP. There are some routers on the market that change the port during re-registration, leaving a blind spot for the reachability for the registered device. In the above case, the registration changes every 90 seconds - don't be surprised if calling that phone will result in random call drops.
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So did you read http://wiki.pbxnsip.com/index.php/Installing_in_Linux? Be careful copying the script, copy & paste is sometimes tricky...
Windows XP is fine, you can also use it.
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I was just wondering where it could be hiding in the software..I would like to not see that error..
Check the spool directory, there is probably an old email lingering. Just delete the file...
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Exactly, DST like in CET. This feature can be done in next releases?
The next release (3.0) will have it.
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The -30000 is according to the RFC (see http://www.ietf.org/rfc/rfc2822.txt, paragraph 3.3). We had another case where they changed the email client...
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Well "host" is the command to find out what DNS would respond to the provided argument (in Linux). What I was trying to say - there is no DNS record...
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Whow. SNMP should have almost no impact on CPU performance, therefore I believe that the packet I/O would cause this. Ethereal would show it.
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We use our own logo for PBXnSIP. Let's say I wanted to right justify or center the logo, is there a way to change the position of the logo?
Everything can be customized, question is if it is worth to spend a significant amount of time on it. Version 3 will change the logo again, keep that in mind.
Change SMTP adress
in Email Topics
Posted
Well, there are two things.
First, there are two places for settings the SMTP server. One is on admin level, and there is another one on domain level.
The other thing is that once that emails are being put into the spool directory, the email server is fixed. That means, you might have to manually delete the emails from the spool directory to get them out of the system.