leviticus

Well, it's the call recording that poses the real issue. I'm not actually sending the CDR anywhere, but just observed that the call recording was entering the 'global' domain, instead of the 'final one. That's my core problem - I need to grant a domain admin access to THEIR recordings through the web interface. Any thoughts on how to get the system to detect that? -Levi

- We have a PBX that has multiple domains (lets say 20). - All calls come through a trunk that is in a 'central' domain - the trunk is 'global', and routes to DID numbers in other domains (routing works fine) However, when the calls are logged to CDR, they register under that 'central' domain, NOT under the eventual destination domain. We have an older instance of SnomONE running on another server with the exact same setup, and the calls get logged under the eventual destination domain. This is important for the purposes of having the call recordings readily available in the call log of that domain. Any thoughts? Levi Shores Digital Technology Partners

For the record, I'm also seeing this issue on two PBX installs, both 4.0.7 and 5.0.8, with multiple Snom 720 stations running 8.7.3.19. If I use the *78/*79 dial codes I see the same behavior: A hung call in the PBX that appears as from the phone I dialed to the PBX dial code, as well as a lit BLF light on the phone for the extension that placed the call. Would love to see a solution soon. Thanks! -Levi Shores Digital Technology Partners

I'm looking for a good way to log or alert via e-mail if a call goes unanswered in a hunt group. I attempted to route the call through the hunt group, and then for 'final stage' send to an extension with a setting 'send an e-mail when extension misses a call', however that doesn't send an e-mail if the extension is unregistered (I don't want to send the call to a registered extension). I'm also open to a reasonable web report, but all the call logs that I've seen lack a clear delineation between what calls got answered, and what calls were sent to a VM as 'final stage'. Any thoughts would be appreciated. -Leviticus Digital Technology Partners Atlanta, GA

Thanks! I guess we'll contact sales. Is there anything I could verify to see if we are licensed for it? (Or would this feature be so obvious that I wouldn't be asking about it if it were there? I suppose I'm just concerned that I just don't know how to do it if it is there, but I'm guessing you'd select the model in the provisioning section and it would provide you a provisioning URL? (for option 66 in Polycom's case) Thanks for your time/input.

We just converted a client over to Snom who has a conference phone (polycom soundstation ip 6000). I need to get this registered to our hosted SnomONE server. Is there a high level overview of provisioning Polycoms to SnomONE? I'm also concerned about NAT issues with this device working with the remote server. Any input would be greatly appreciated.

I'm curious if there's a way to view all customized xml/etc files throughout all domains in a SnomOne server. I'd even be comfortable doing a file search at the OS level. We customized several files (firmware push), and neglected to note exactly which we made (whether by phone or domain). Any thoughts/assistance is appreciated.

Well, of course there's a 'don't break me' check box! Thanks for the quick reply. Based on what I'm seeing this morning, it looks like the service must be restarted in order for a change to this field to take effect?

We've been steadily rolling out our SnomONE hosted solution (been working great). A few days ago, we crossed the mark of 100 remote extensions being registered. We got to 106, and seem to have hit a brick wall. No subsequent registrations can be made. We can un-register an extension, and the previously non-register-able extension right next to it at the remote site will suddenly register. Is there a possible hidden limit somewhere that is not readily visible in the system status tab? We rebooted the PBX and upgraded it to the latest build this evening with the same results afterwards. Also will to consider that it might be a limitation of our Firewall (Sonicwall NSA 2400 MX), OS (CentOS 64), etc. Just seems like way to clean of a number (see below: active SIP sessions=100). Version: 4.5.0.1090 Epsilon Geminids (CentOS64) Created on: Jun 19 2012 08:24:50 License Status: snom ONE Hosted <...> License Duration: Permanent Additional license information: Extensions: 168/500 Upgrade: 12 19 2013 Working Directory: /usr/local/snomONE MAC Addresses: <...> DNS Servers: 8.8.8.8 208.67.222.222 8.8.4.4 CDRs: Duration(7d): trunk = 4946, extension = 1785, ivr = 5884 Calls: Total 19/4, Active 0/0 Calls SIP packet statistics: Tx: 73823 Rx: 74685 Emails: Successful sent: 3 Unsuccessful attempts: 0 Available file system space: 38% Uptime: 2012/9/21 00:08:52 (uptime: 0 days 03:29:05) (96567 103215-0) WAV cache: 3 Number of HTTP sessions: Sessions: PAC=0, HTTP=1; Threads: SIP=100, HTTP=1 Domain Statistics: Total Domains: 21, Total Accounts: 329 Thanks for any assistance.

We have a client who has a Sangoma A200 with 4 FXO ports (3 in use). About 12 extensions, and simple daytime call routing - calls come in, go to ring group, if unanswered for 20 seconds, they go to the voicemail of ext 200. A few days ago, the voicemail seemingly stopped recording messages - the use would be prompted to record, hang up, and the voicemail would never appear. I cleaned up the extension, and replaced the voicemail greeting. Upon testing it now, the VM seemingly records and the handset is alerted, but upon listening to the voicemail, the recording is of 'If you would like to make a call, please hang up...', proceeded by off-hook sounds. I'm curious if this may be being caused by a configuration issue in the Sangoma. I currently have 'Connect Inbound Side Condition' set to 'ON-OUTBOUNDSIDE-CONNECTED' but... We are seeing an error on the Gateway of 'Could not start early media sine inboundcallhandling connectinboundsidecondition is not set to ON-OUTBOUND-SIDE-MEDIA-AVAILABLE. Please change this parameter to that value in the pstn configuration file if you want to let the early media flowing between the pstn and sip sides' I recall during the setup of this unit that these parameters had to be changed in order to get two way audio from the beginning of the call. No settings were changed on the gateway, so I'm curious if this is a gateway issue or something inside the PBX. Anyone have any thoughts?

I figured I'd come back here and post what has been working for us for several weeks now: Request-URI: Let system decide From: Custom "<sip:{ext-ani}@{domain};user=phone>" To: Let system decide All other options in Privacy indication are set to 'do not use header'. Everything seems to be functioning properly with NV now.

After Further testing the above setup does allow dialing of 800 numbers, however it creates nexvortex disconnects after 23 or 24 seconds consistently. Changing the setup to: Request-URI: Let the System Decide From: Based on incoming call To: Let the system decide p-asserted-identity: Extension ANI p-preferred-identity: Don't use Remote-Party ID: Don't use p-charging-vector: Use ICID Value Privacy Indication: Don't use Allows all calls except 800 numbers to be successful. Does anyone successfully use nexvortex in a mult-tennant setup, pulling outbound caller Id from the extension? What are you using for custom headers in your trunk setup? Thanks!

We're using a multitenant installation with a single global trunk configured for NexVortex. (so far... secondary SIP provider probably on the horizon). Discovered that 800 calling was not functioning with our setup and turned to the headers as a possible fix. This was my end result configuration, which successfully dials standard and 800 numbers, as well as pulls the outbound caller ID from the extensions (desirable for our setup). Figured I'd post here for commentary and if anyone had suggestions/thoughts on it. Request-URI: Let the System Decide From: Extension ANI To: Let the system decide p-asserted-identity: Don't use p-preferred-identity: Don't use Remote-Party ID: Based on incoming call p-charging-vector: Don't use Privacy Indication: Don't use Similarly related: Does the ANI 'rollover/up' if not defined? The behavior I've seen seems to indicate this. Example: If I've set the headers to deliver the extension ANI, but there is no extension ANI defined, but there IS a domain ANI, it will present the domain ANI? - This is very helpful, as we can set the domain ANI per tenant, and if ever need be, a particular extension is easily assigned their own ANI. If a user dials an 'emergency number', which ANI (Trunk, Domain, or Extension) ANI is presented?

I have no problem reproducing it. I just have to go into a tenant and try to re-number an existing extension to an extension number that exists in any other tenant. Is having 'localhost' as a domain alias a faux pas? Because we still do - I suppose I thought it was just a necessary evil? Here's the level 9 log-file (FQDNs redacted - <FQDN1>=original/localhost alias). I cleared the log just before attempting the re-number, and this was all that was posted in the log: [8] 2012/06/04 23:35:48: Last message repeated 3 times [9] 2012/06/04 23:35:48: 1(3): The alias name <FQDN1>, domain name <FQDN1> [9] 2012/06/04 23:35:48: 2(4): The alias name localhost, domain name <FQDN1> [9] 2012/06/04 23:35:48: 2(6): The alias name <FQDN2>, domain name <FQDN2> [9] 2012/06/04 23:35:48: 3(9): The alias name <FQDN3>, domain name <FQDN3> [8] 2012/06/04 23:36:00: Packet authenticated by transport layer [0] 2012/06/04 23:36:02: Create User: administrator from <my remote IP> is trying to create extensions [5] 2012/06/04 23:36:02: Create User: The name 43 is already in use [8] 2012/06/04 23:36:03: Packet authenticated by transport layer Current build info: Version: 4.5.0.1075 Delta Aurigids (CentOS64) Thanks for your help!

So, we're using a multitenant setup and in the course of testing/exploring I noticed an issue that begs two questions: 1. Is there an issue creating extensions with the same number (and even name) between tenants? 2. Is there a bug in the web interface when renumbering extensions? Basically, what I'm seeing is that I can CREATE the same extension number between multiple tenants, but if I renumber an extension the same as an extension in another client, I receive the error: "Error: At least one of the names exists. The names have not been changed" NOTE: I am ONLY changing the 'Account Number(s):' field, but is conceivable that in our client pool we will have extensions that share the same name and number ('200:Front Desk' between two tenants for instance.) An example: 1. <tenant1> has extension 200, 201, 202 2. I create new tenant <tenant2> and add extension 200 – Snom One doesn’t complain, creates extension. 3. I go back over to <tenant1>, renumber 200 to some other arbitrary extension (let's say one of the defaults: 41) 4. I then try to renumber that same extension BACK to 200 – no go; it complains that the name is in use (above error) 5. I delete the extension (200) in <tenant2>, go back to <tenant1> and I am then able to rename the extension 41 back to 200. It's not like renumbering extensions is going to be terribly common for us, but not being able to on the off-hand occasion is going to be annoying. More importantly, I'm curious if there are any issues with having extensions numbered the same between two tenants? Thoughts?

So, my core objective is to have 10 digit dialing without a prefix for a specific trunk. It seems the default dial plan requires a 1+10 digits. Anything else falls into the 2-digit extension dial plan. (I've been using 'North America 2 digit extension'.) I'm confused because I don't see how the dial timeouts play into this. Assuming all dial plans have 4 second timeouts (as defined by 'snom_3xx_dialplan_usa2.xml'), shouldn't I have 4 seconds to put in ANY combination before it's 'judged'? I've also put in a "<template match=".........." timeout="4" user="phone"/>" option in that same file. So, after a day of fiddling with this, I have to ask: -What is the relationship between dialplans.xml vs snom_3xx_dialplan_usa2.xml (and the other similar files)? -There also seem to be different syntaxes used between these files, how does that play in? -One piece of documentation refers to creating a 'custom' entry in dialplan.xml, but there doesn't seem to be an option in that file for timeouts. -Is that how timeouts are expected to work? Current behavior is that as soon as anything matching ([2-7][0-9]) is typed into the phone, it tries to submit it, resulting in an error. The rest of a 10 digit phone number can't be entered. Sorry for the rambling opening post. Any help in understanding how the system uses the dialplan definitions is appreciated.