djanjic
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Posts posted by djanjic
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Where can I check email logs if such thing exists? In the past few days I haven't received my regular call log reports and I haven't received few voice mail messages. I want to see where does this get stuck?
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I think the black-list would be appropriate? That would work for calls into an extension. For calls to other accounts, you could think about using a IVR node and do from-based routing.
I can't find black-list. That is exactlty what I want...
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I have searched the forum for this topic and couldn't find anything. Is it possible to block-reject an incoming call from a specific caller ID. The scenario is this:
An extension has a DID # and tel:alias associated with it. Is it possible to block an incoming call if the call dials DID number directly? I couldn't find it...
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has anyone used this product in a multitenant application (ITSP) I would like to know how many users IE concurrent calls on a dual quad core server 16 MB of memory either running Red Hat Linux or Windows server 2003. The most important question is reliability and the lack of idiosyncrasies in a hosted multitenant application.
Thank you very much
John Warren
585-419-8264
Hi John,
We are a relatively small ISP offering hosted SIP PBX services to our small business customers, we stay away from residential. For the most part our customers do not require too many features, so adding them to our offer is a nice bonus.
We’ve been running PBXNSIP since version 1.5. We are running it on win32 platform There were some problems wit earlier versions but right now it is super stable. Our last reboot was about 2 months ago, we had to reboot after one of Windows updates. Current version is 2.1.10.2474. During the business hours we get up to 20 concurrent calls. Our machine doesn’t have any trouble with this volume. I am assuming you meant 16 GB RAM instead 16 MB. I think that it would be a long time before you put that machine to a real test.
Few pointers.
1. Make sure you have good DNS solution in house.
2. Make sure you get your license key on a USB dongle, it will save you from lot of headaches if for some reason you need to move your config to another box.
3. Make sure to know how are you going to comply with 911.
4. Make sure you know how you are going to provide outbound caller ID. That turned out to be the single most important thing to our customers.
5. If you can always sell them the solution with SIP phones. We have excellent success with Snom’s. We have a customer with an extension in Chile using Snom phone to connect to their local office. We do not hear from them at all. I guess they have really good DSL in Chile.
Hope this helps?
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I do not see where can I add another tel:alias to account? What is the trick?
I have found answer to my question which works for us. In case someone has similar problem here is the solution:
in the alias field type:
tel: your 10 digit number space tel:seven digit number. Hope this helps...
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Remember that you can also use two (or more) tel:alias in one account. The first one will be the one that is used as ANI, but the second one is also used for inbound matching. Maybe that helps to solve the problem.
I do not see where can I add another tel:alias to account? What is the trick?
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In 2.1.10 you can do that by using the tel:alias or a per-account basis, this has higher priority than the DID number setting in the trunk. This solution is okay unless you have two extensions that should both show the same DID which is not the same as the DID for the trunk (that will be fixed in 3.0).
That is precisly the problem. We have to have tel:alias in order for incomming calls to reach proper domain, remember we are running in hosted mode with many domains. On top of that our povider requires that on outbound calls we send 10 digits to them. Only then they pass it along, if it is not 10 digit number they send the number asociated with PRI (our office main number). If we use 10 digit tel:alias our gateway doesn't know how to send incoming calls correctly since it is a local 7 digit call. So now everyone's caller ID appears as our main number, so we get a lot of calls from people that are "returnig" their calls since it appears that it came from us
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In the 3.0 release we (will) have a ANI settings that should solve many of these problems.
Here is our problem with Caller ID issue. We offer hosted service. We port many existing customers original numbers to our PRI (we have a provider who will do that) In many cases we have to assign either one of our DID numbers or ported number to an extension so that incomming call can find proper domain. In previous version we were able to assign DID on the trunk and present that as an outgoing caller ID. In our current version (2.1.10.2474) we can no longer do that. Is there a way to force PBX to send the caller ID from DID on the trunk?
How soon to the next version?
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There is something on the Wiki http://wiki.pbxnsip.com/index.php/SonicWall, a little bit outdated maybe. Maybe a time to update it and check what the latest versions are.
Yes I looked at that on the wiki and anything else I can find on the forum related to sonic wall. The problem seems to be either with sonicwall latest release or this particular appliance. Everyone including sonic wall support is bafled by it since all settings as they can see it are correct???
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We have a customer who has just installed sonic wall 190 appliance. They are using our PBX hosting service. No problems untill the sonicwall was installed. Sonicwall has the latest software 3.09. Snom 320 phones have 6.1.17. After the install the phones cannot register at all. Sonic wall appears to be open and even has our outside server added as a trusted IP for all traffic, still nothing. The customer has talked to sonic wall their tech support is confused as we are. Has anyone experianced this problem and if they have what was the solution?
Thanks.
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Probably it is easier to look into the concrete system that talking abstract.
I am little confused with your answers. Who do I contact to look in to the system?
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What is strange here is that 300 seems to be an account, but the PBX does not find it and uses the dial plan to dial an external number. Do we have a problem of mixing up domains here?
I don’t know if this is a question for me or a rhetorical one, but I believe that this is a bug. I have done some more testing and some domains work fine and some do not. Even in the same domain I have numbers that are routed properly and some that are not. Do I need to open a ticket with tech support on this?
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Do the trunks have an outbound proxy set? The "outbound" proxy is also the "inbound" proxy for identifying where the call belongs to. Check if you see anything in the log saying "Identify trunk XXX" (log level 5 in the trunk category) and if it matches your expectations.
Yes all trunks have outbound proxy set, and they are all the same. Here is what the log shows on the call that doesn't go properly:
[5] 2007/11/23 10:23:25: Identify trunk 73
[5] 2007/11/23 10:23:26: Attendant: Redirect to 300
[5] 2007/11/23 10:23:26: Dialplan: Match 300@rgv.vatelephone.com to <sip:300@lexotho.vatelephone.com;user=phone> on trunk Prescient
[5] 2007/11/23 10:23:28: INVITE Response: Terminate 41ed1980@pbx
[5] 2007/11/23 10:23:28: Dialplan: Match 300@rgv.vatelephone.com to <sip:300@callcentric.com;user=phone> on trunk callcentric
[5] 2007/11/23 10:23:31: INVITE Response: Terminate 7f2fa7e3@pbx
More onthe subject. I have some more details. I beleive this is a bug. This behavior occurs in some domains if the tel:extension is forwarding calls to either autoatendant or hunt group. What is weird about it is that in some cases this doesn't happen? Any suggestions?
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This issue is driving me absolutely crazy! We are running a hosted PBX for a small number of customers. We have a Cisco Gateway with several PRI-s attached to it. Each customer is designated their own domain. Each domain has a separate trunk (to provide caller ID for outgoing calls) and each customer has a very simple dial plan for outgoing calls. Basically anything that is local call is sent to Cisco trunk and then on to PSTN to terminate the call.
The issue is this. When calling from a PBX and dialing a number which belongs to a particular customer from a different domain some calls properly get sent to the outbound trunk (cisco) and then routed to the PBX but some don?t. They somehow get sent to our LD provider.
If I dial a number which belongs to domain Z from domain X I get to the right place.
If I dial a number which belongs to domain Z from domain Y I get half way around the world with the message that the number is invalid
I have checked settings for trunks, dial plans and I don?t see anything wrong on any of the settings. Here is the log from incorrect routing:
5] 2007/11/15 17:03:20: Attendant: Redirect to 500
[5] 2007/11/15 17:03:20: Dialplan: Match 500@lexortho.vatelephone.com to <sip:500@prescient;user=phone> on trunk SipVonWorld
[5] 2007/11/15 17:03:23: INVITE Response: Terminate 17747495@pbx
[5] 2007/11/15 17:03:23: Dialplan: Match 500@lexortho.vatelephone.com to <sip:500@callcentric.com;user=phone> on trunk callcentric
[5] 2007/11/15 17:03:25: INVITE Response: Terminate 650d6fcc@pbx
[5] 2007/11/15 17:06:37: BYE Response: Terminate 307d0dbf295ba49c336d6a984ed47598
Can someone help?
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Are you using 6.5.12 (http://wiki.snom.com/Snom320/Firmware/Release_Notes#6.5.12_release)? Seems there was a fix for lost subscriptions.
Yes, we are using that version on both snom 320 and 360. The results are the same...
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We are having issues running combination of Snom 360s and 320s with latest stable 6 firmware and Pbxnsip 2.0.3.1705. We are also assigning extension keys on buttons with an idea to monitor who is on a call or to pickup the call when the person is not around. The feature works for about 10 incoming calls and then the feature dies. The lamps no longer lit up and only the phone reboot will make it work for another 10 calls. The phones continue to work, only the feature is affected. Most of our customers like the feature because they want to monitor who is on the phone when but with this problem we are not implementing it. Can anyone help
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911
in Trunk Setup
This is a question for the forum and critically important to everyone who wants to provide hosted PBX services to residential and business customers, since FCC requires that 911 be provided to any VoIP customer. We provide VoIP service to local business and residental customers and we had signed up with a national 911 provider. We had set them up as a Trunk using option SIP gateway on the pull down menu. During the test here is what they received:
"CONTACT_URI=<sip:66.207.74.227:5060>") IP address of our PBX
"FROM_URI="RGV" <sip:208.71.179.10>;tag=61950") IP address of their gateway.
We made some more tests this time using SIP registration as an option on the trunk
This time the result was even more confusing, this is what they got in their logs:
Origin contact; dnis ; newfrom; new contact; ndnis ; pai ; sessionid
66.207.74.227 911 6@6 6@6 7052228221 6302395128 447ed917@pbx
New from and new contact fields are really strange 6@6. Has any one done this and what should proper fields be?
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I am having a trouble setting up a return caller ID call. Currently when an incoming call come in on SIP gateway the incoming caller ID is presented to the PBX as an area code plus seven digit number, for example 540XXX-XXXX. That is how the Snom phones receives it as well. If the user wants to return the call based on the caller ID the PBX doesn?t know how to send the call out. I will either need to add a 1 in front the whole number or strip area code from the number. The first option is more desirable. Can someone help me with this?
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I am not 100% sure of what is taking place. SIP Trunks act differently than POTS and I know POTS better than SIP.
Anyway.......
It sounds like the "Number has been disconnected" announcement is coming from the ITSP and not PBXnSIP.
You are "downstream' in the call flow sequence at this point while the ITSP has control over what happens when all 3 lines/channels are busy.
From that position, I don't think there is much you can do. The fourth caller, most likely, never makes it to your location.
What happens if you de-register the service and call the main number?
Does each line Register as its own trunk or is there one registration for all three?
Who is the ITSP?
Bill H
I beleive you are right and that the problem will clear itself out once the whole legal process of porting the numbers is complete. That usually takes 48 hours, in this case I think it is business hours in question since we have done number portability on Friday. thanks for your help.
Dusan
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We run 2.0.3.1705 in a hosted mode. We just turned up a new customer and have a strange issue. Their incoming calls come through our SIP gateway. Right now they have 3 incoming lines which have been ported from their old POTS provider. When all of their lines are in use and they get another incoming call instead of getting a busy signal the announcement comes on and says that the number has been disconnected. Is there a way to make PBX give a busy signal?
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The 2.0.3.1715 is the latest release. Later builds are builds, but no releases. 1715 seems to be pretty stable and unless you are not missing features there is no need to move to another version.
We deo not need any new features, we are trying to implement some SOAP applications and we were told that 2.0.2 had some bugs. That is the only reason for upgrade...
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What is the latest supported stable version of the software? Can someone keep the group informed about this? The releases seem to be flying on a daily basis. It is hard to have a confidence in an upgrade unless we know full details of the release. The latest post in the section on versions and updates is dated April 10th and talks about 2.0.3.1707. I am seeing posts which make references to 2.0.4 and 2.0.9. Hard to know what to use???
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I have a strangest problem. I am trying to implement Sipura SPA devices (1001, 2000, 2002 and 3000) behind NAT. I can make outward calls all I want, no problem with audio quality etc. My problem is incoming calls. The phone rings and then silence when we pick up the phone. I am assuming it is RTP issue and specific to Sipura (Snom phones do not have this issue) I have opened ports on the router, looked at every setting on SPA interface and just cannot find the problem. I tried to capture traffic using wireshark, simply cannot find it. It works like a charm with public IP address. Does any one have correct settings for Sipura ATA adapters behind NAT, or can someone point me to a direction where I can find it?
Ok folks, thanks to "StopCallingMePhyllis" we have fixed the problem. In case someone else is having similar problem here is what cleared it.
The problem is related to version 2.0.1.1624, upgrading to 2.0.2.1675 by replacing pbxctrl.exe had done the trick. Hope this helps someone if they have similar issue.
Dusan
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I have a strangest problem. I am trying to implement Sipura SPA devices (1001, 2000, 2002 and 3000) behind NAT. I can make outward calls all I want, no problem with audio quality etc. My problem is incoming calls. The phone rings and then silence when we pick up the phone. I am assuming it is RTP issue and specific to Sipura (Snom phones do not have this issue) I have opened ports on the router, looked at every setting on SPA interface and just cannot find the problem. I tried to capture traffic using wireshark, simply cannot find it. It works like a charm with public IP address. Does any one have correct settings for Sipura ATA adapters behind NAT, or can someone point me to a direction where I can find it?
Typo
in Mailbox
Posted
Recently I noticed that I am not receiving VM in to my email box. After some checking in lto ogs I realize that there is a typo for our mail server:
[3] 2009/01/29 22:00:10: SMTP: Cannot resolve mail.rockbridge.ne6t (should be mail.rockbridge.net)
I looke everywhere and cannot find it. Every screen appears to be correct. Is there a file on the server which holds this information so I acan manualy change it?
Thanks