TimB
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Posts posted by TimB
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That would be very difficult - a SIP phone does not send a notify to the PBX when it goes offhook.
so paging through the phones is useless.
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Version: 1.5.1.1.a
All Snom 300's Version: snom300-SIP 6.5.6
If you are in conversation and some does a Page to all of the phones it puts the caller on hold and the page comes through the speaker. Is there any fix for this that would not send a page to those phones that are offhook and only send the page to the idle phones?
Thanks
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That sounds like SRTP trouble. What versions are you using?
version is 1.5.1.1a
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The system I have at Big Brothers is having a problem with the 2 remote extensions. They are connected over a VPN thats working very good but inter office calls from office A to office B here a static sound about 6-7 minutes into the call then it drops. Same if a remote extension at office B grabs an outside line from the office A and is talking with a client 6-7 minutes into the call it drops. Any thoughts????
Many Thanks
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35 views and 0 suggestion. I love this product!
I have a client who's voicemail stops working all the time. You restart the pbxnsip service and it starts working again. Any idea what causing this and is there a solution, people are getting pissed off at me a regular basis these days.Thanks
((BUMP)))
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In mid conversation you get a Static Interuption like TV static sound for about 10 seconds then the audio comes back to normal. Any ideas whats causing this?
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I have a client who's voicemail stops working all the time. You restart the pbxnsip service and it starts working again. Any idea what causing this and is there a solution, people are getting pissed off at me a regular basis these days.
Thanks
((BUMP)))
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Have a page group 700 setup and when dial it does an all call page to all the phones. If your on the phone it puts the caller on hold and you hear the page. Is there a way to disable that so if your on the phone you wont get the page and can continue talking. All Snom 300 phones.
Thanks
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you are good to go on the VPN .. the default gateway will route between the 2 sites and provide internet .,
otherwise you would need to do some route statements ..
should be easy ... remember, if you can ping it .. you can ring it !!
yori
Yes, it worked great as soon as the VPN was up and running just had to change the the IP scheme from to 192.169.2.X so they were not both 192.168.1.X
One thing went right today, I'm very happy!
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There are days when it doesn't pay to get out of bed. It is my job to support PBXnSIP installations, and the people using them. That isn't so bad until I have to do things like write documentation on the star codes in a user-comprehensible manner, documentation I feel should already exist. Repeated requests for PBXnSIP documentation in this forum from many people, and notes from VAR's that they've had to create their own should be a wakeup call. PBXnSIP does _not_ have the installation base to compete with Asterisk, and Digum if it's going to be a "just as good", or "not quite as good" solution. PBXnSIP is not open source, and since I'm paying for it there is no impetus to give back the work I'm doing which should have already been done for a commercial product. PBXnSIP is not going to be viable if I'm paying thousands of dollars for a platform which has less documentation than Asterisk.
PBXnSIP is a convergance of computing and telephones which means it needs twice the documentation. It needs users guides which are comprehensible for someone who has never used a PBX. It needs instructions for the web interface for the users. It needs the nice cheat cards that come with Avaya, Nortel, Panasonic, Cisco, and Alcatel. All these companies provide extensive administrative and customer documentation. All these companies are also making a ton of money. Post hoc, ergo propter hoc, of course applies here. If I am going to need it to sell PBXnSIP to the customer, PBXnSIP needs to provide it. Pre-install questionaires explaining PBXnSIP's features. Word templates for documenting the phone system layout and features which are customer specific, with fields for things like extensions list, and hunt group/agent group lists without ever having to make that technical distinction to the user.
Please figure out what your competition is doing right if you want their customers, or don't, and they'll take yours away from you.
Boy you nailed this one, documentation for this product is weak and relying on this forum for real support is a joke. It's nice when people reply but there really is not much from the pbxnsip admin. The big advantage with this product over Asterisk was SLA when selling a key system replacement. But now Asterisk 1.4.5 supports SLA and is way more flexible than pbxnsip. Also when you pay for a 25 user license it should be 25 extensions, counting trunks and AA as licenses is wrong. Just my opinion!
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This should be plug and play. This is for varying values of should that equate to: it depends on the type of of VPN, the OSI model layer the VPN operates on, the VPN routers support for dealing with lots of UDP traffic, the conditions of the internet between the two sites. I'd get this in place before it is needed and do some testing but it SHOULD work. Opening router ports definately isn't necessary unless the firewalls are really anal.
Thanks for the feedback, both routers will be Linksys/Cisco RV042. What do you mean by the type of VPN?
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We are setting a VPN between two offices, office B will have 2 extensions of of the pbxnsip server at office A
Once the VPN is up and configured will it be plug and pray (play) for these 2 remote extensions or will I still to open router ports?
Many Thanks.
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Something like that usually happens if you have deleted that trunk already. Maybe there is something screwed up in your database. Hit the save button on the trunks and the dial plans to make sure that the references are okay. Maybe also a restart will fix the references.
What's happening is when I set it to Gateway Mode the trunk does not register with the ISTP, only registers in Sip Registration Mode.
Makes no sense to me becouse when the trunk is registered under Sip registration Mode under the outbound dial plan I can create 6* and from and phone dial 6 and the number to the remote Snom phone and it rings through.
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Thats what I did I set up a seperate trunk as 8280003 and named it co6. I'm the ISTP so I know it's registering one time not multiple times. I have set ext 61 to only dial out on co6. I will have to check on the gateway mode settings.
I created an dial plan called Kennebunk. in the trunk drop down menu I select VoIP2 and hit save but it keeps defaulting back to AudioCodes. Any reason it's doing this?
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If you want to dedicate a "line" you must set up a seperate trunk for this purpose. In gateway mode, that should be no problem.
But in registration mode, there will be a problem as the PBX will try to register the same account twice from the same IP address. Your ITSP might not be able to deal with this...
Thats what I did I set up a seperate trunk as 8280003 and named it co6. I'm the ISTP so I know it's registering one time not multiple times. I have set ext 61 to only dial out on co6. I will have to check on the gateway mode settings.
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I want to forward an extension to an outside number on a permanent basis. So I use (redirect) (call forward all) but how can I force it to dial out on co6 only?
Many Thanks.
This is what I have and trying to accomplish. 1 SIP trunk (8280003) into the system and registered and working correctly. Snom 300 at a remote site registered with a SIP trunk (8280004) from the same ISTP.
I created an extension in PBSNSIP ext 61 and created an Outbound dial plan for 8280003.
I did a redirect on ext 61 to forward all calls to 8280004 the remote Snom phone.
I have told ext 61 to use the 8280003 outbound dial plan.
With this setup shouldn't I be able to pickup and extension at the main location dial ext 61 and be forwarded to that remote Snom phone?
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I want to forward an extension to an outside number on a permanent basis. So I use (redirect) (call forward all) but how can I force it to dial out on co6 only?
Many Thanks.
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I ran the WireShark on the PBX and tried to dial in from the remote extension. I didn't see anything obvious come up in the capture. Is there a specific entry I should be looking for?
Just to recap the client has a sonicwall TZ170, with the latest Windows Build of PBXnSIP running at the corp office. I've installed a Linksys SPA942 at the home office. The phone does register with the PBX and I do get a dial tone. I can even make call but person can't hear me nor can I hear them. All phones at the corp office (which are also Linksys SPA942) are working fine.
Thanks again for the help.
I'm having a similar problem with a client. The remote phone has a snom300 and it can dial back to the main office extensions w/out any issues and can dial out to the pstn with no problems. But when you call the remote phone from the main office you get one way audio. It's driving me nuts, we are looking to install a vpn next week and hopefully that will make it work.
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When the remote office initiates the call it works great. Its when the main office initiates the call to the remote office theres a problem.
tried with the stun server and I loose audio completely.
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stun.xten.com I think should work ok.
I think its because the Snom 300 initiates the call and opens up the necessary pin-holes in the router/firewall - so symmetric RTP is working. When the remote office initiates the call, the sessions dont exist so are dropped. It's very hard to say though - it varies with different models of router and UA.
When the remote office initiates the call it works great. Its when the main office initiates the call to the remote office theres a problem.
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Without a SIP trace/ethereal dump it's very hard to see what's going on. Can you maybe run Ethereal on the PBX and on the softphone PC, configure them both to capture and initiate the (failing) call. The one way audio sounds like the softphone is replying to the INVITE saying "send you audio here", the PBX then sends the audio to a) The wrong IP
The incorrect port or c) The router has restrictive NAT policies. STUN should discover the type of NAT for you.I will run the SIP trace, going back onsite today. Can you recommend a STUN server? Its not a softphone but a Snom 300 at the remote site. It just baffles me that the remote site can call the main office and it works fine but when the main office calls the remote site there's only one way audio.
Would a VPN between locations solve my problems?
Thanks
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WOW. Respect!
Well I thought I had this figured out, NOT! The remote extension can dial another extension back at the main office and it works fine, they can also grab dial tone from the main office and use the trunks for out bound calls. But when an extension at the office calls a remote extension I can hear them but they cannot hear me. I have played around with firewall ports on the remote end routers with no results.
Any suggestions?
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"Also, from our experience dynamic IP addresses will cause a lot of random instability and problem searching."
SIP is not HTTP...
FYI-We have it working fine now. Its nice to have a static but we have figured out you can make this work with a dynamic IP with pbxnsip NATED by port forwarding in your router. With good bandwith 3 remote phones should be fine QOS wise.
time warner dynamic IP cable modem> linksys router>pbxnsip on the lan.
remote phone registers with the dynamic wan IP address for the time warner cable modem and registers correctly.
If the wan IP changes you just change it in the phones GUI.
You could also use a free dns service like dyndns.com and give your server a name voip.whatever.com and register your remote phones that way too.
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does anyone know of any user groups for pbxnsip that have their own forums?
The incorrect port or c) The router has restrictive NAT policies. STUN should discover the type of NAT for you.
Page Interuptions while offhook
in General Setup
Posted
I didn't say when another call comes on.......When someone sends a PAGE through PBXNSIP it puts the caller on hold and the PAGE comes through the speakers of all the SNOM phones. This is bad if you are on the phone with a client and suddenly without warning they get put on hold.