andrewgroup

Assuming manual cionfig PBX is 10.10.10.250 phones are in the same /24 subnet

Can someone post a sample speed dial entry for a Snom 3xx phone speed dial settings. Examples of using the Action URLS would be useful too.. Cheers

It would be very nice and convienent of a special account could be created. MOH. extensions with access could easily access and record new MOH files whenever necessary. Perhaps allowing the creation of multiple MOH accounts Say "Christmas, Haloween, etc) and set them to a service flag by date.

This may be perfectly normal and may be associated with the NAND memory and the linus kernel operations. Only the OEM manufacturer of the hardware platform can truly confirm if this is a problem.

How and where do you bind the button to the indentity? We have 1 of 7 Snom 360 phones that has only 1 identity registered, and when the incoming call is answered on ext40, it cannot manually part a call *85101 or use the part 101 button but all others can answer and park and retrieve. Just disabled all other indentities, and will report back.

We've ran paid versions of AVG in the past, but long ago went naked in this regard. I wrote a best practice on Windows installs and our standard install has the common Windows XP or Server tweaks, followed by all services set to manual allowing only the minimal services. This kills networking, Dcom, winstall, and more.. Aside from the pbxctrl app, we have 4 or 5 windows services and the Raid controller. More than a few machines on have had PUBLIC IPV4 addresses on the web for well over 2+ years.... (no known problems) "cross my fingers haha) Cheers

Our most reliable CS410 is below. 90 days of uptime - only 5 extensions and no email sending of email attachments. This system is attached to 4 Analog Lines provide from a CISCO IAD 2400FX16 from a Nuvox FlexIP circuit. The FXOSIP update mentioned in this post is a replacement for the previously provide FXO update installed via the WEB UPDATE? Correct. Version: 3.2.0.3143 (Linux) License Status: Appliance Key License Duration: Permanent Additional license information: Extensions: 10/32 Accounts: 20/40 Working Directory: /pbx MAC Addresses: 001915683FD0 001915683FD1 Calls: 1610/58 (CDR: 1156) 0/0 Calls SIP packet statistics: Tx: 786047 Rx: 786325 Emails: Successful sent: 274 Unsuccessful attempts: 0 Available file system space: 67% Uptime: 90 20:34:41 (7561 8481024-0) WAV cache: 1

Do specifics exist for these options in the Wiki or the support FAQ's Unfortunately the client didn't know this was happening and either email or snmp alerts should address these potential troubles.

Ignore the incorrect time, as this is a common problem discussed in other posts... A restart fixed this issue...on this rev. Version: 3.4.0.3194 (Linux) License Status: Appliance Key [5] 2008/01/04 18:39:13: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:39:19: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:39:19: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:44:19: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:44:24: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:44:24: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:46:51: Last message repeated 2 times [5] 2008/01/04 18:46:51: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:52:00: Last message repeated 4 times [5] 2008/01/04 18:52:00: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:52:30: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:52:45: Last message repeated 2 times [5] 2008/01/04 18:52:45: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:52:45: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:52:51: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:52:51: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:52:57: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:52:57: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:53:03: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:53:03: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:53:09: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:53:09: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:53:15: PSTN: Continue Dial Tone detected on 2 (version: 2.4.2) [5] 2008/01/04 18:53:15: PSTN: Continue Dial Tone detected on 1 (version: 2.4.2) [5] 2008/01/04 18:53:18: PSTN: Busy Tone detected on 1 (version: 2.4.2)

We too have deployed 3.4.0.3201 (Win32) Both unicast and mutlicast are working fine...

REALLY - JULY 9 has come and gone - Any Love yet?

Would it be possible to associate the indexes to the release versions? Also, with respect to registrations ( .2 ) is this the number of VoIP phone registrations and/or sip trunks? 14 registrations on a system with 6 phones, what do we make of this? Could an index be created specifically to the number of registered VoIP handsets?

We only install the version below. Of the 872 calls during the last 34 days, we have seen 5 to 10 of the disconnected calls...resetting a box each night isn't something we would do. Version: 3.4.0.3194 (Linux) License Status: pbxnsip CS410 License Duration: Permanent Additional license information: Extensions: 7/10 Accounts: 17/20 Working Directory: /pbx MAC Addresses: 0019156BF914 00191XXXXXX Calls: 872/112 (CDR: 316) 0/0 Calls SIP packet statistics: Tx: 578617 Rx: 579546 Emails: Successful sent: 160 Unsuccessful attempts: 10 Available file system space: 69% Uptime: 2009/8/15 16:12:51 (uptime: 34 days 05:58:06) (22622 23714944-0) WAV cache:

I've missed some OID's in my own records, could you post the entire present sent.. (I lost count after .10.. Thanks

We commonly see these disconnects, the client isn't complaining, so we kind of assume the call hung up but not detected and the call was disconnected after a time out.. Could this type of error be turned into an SNMP OID? reset daily or weekly? We ocassionally see these regardless of the telco line vendor.... The call between sip:81XXXX3166@localhost;user=phone and sip:312@localhost;user=phone has been disconnected because no media session was establised (source=0.0.0.0:0)

Nice Post - We too have strived to make every IT project over the last 25 years be as perfect as it can be. Unfortunately PBXnSIP doesn't control the Kernel on the CS410 Mindspeed and it's partners manage that, and in a previous post, a new kernel was mentioned.... so lets wait and see. The Sheeva looks very promising? Done that yet? Cheers

While VLANs are useful for segmenting traffic types, the VLAN protocol has no provisions for traffic control. To properly manage traffic flow you use 802.1P and Q, and most managed switches support a few choices regarding queues and priorities. Given the choice, we strongly suggest mastering the P's and Q's regarding IP control. While you might consider anything less than VLAN as half baked, please visit what might arguably be the #1 commercial seller of VoIP PBX's (Shoretel) and you'll find the vast majority of their deployments are done with dump switches. Cheers

What if you used the LAN port on the VLAN tagged Port Based VLAN (set security of IP registrations to the IP addresses on the that SEGMENT, Then attached the WAN port on a new IP address and created a static Route on the LAN gateway to talk to this IP address, and using this for the WEB interface, and security wise all UDP/RTP traffic would be in on the VLAN segment, while the WEB console is affectively on another. With the upper limit of the CS410 being about 25 phones, we've yet to see any reason to deploy VLANs in smaller installations.

DLink L2 switch support automatic VLAN routing with Static IP Routes, with a much lower cost than L3 devices.

All Things are possible. - but this one is narrowed down to PBXnSIP The Local Raid controller can successfully bounce mail off of the Exchange Server using the same mail settings set in the domain settings. Outlook Express Will Successfully Send email using the same settings as below. The PBX executable is MD5 the same as the original executable in a local folder We easily capture and display SMTP, ARP traffic when either of the above programs send email but the PBX application makes no attempt to communicate on the LAN segment... The wks dns cache correctly identifies the mail server by ip address A full restart fails to resolve The minimum # of windows services are running... Viewing the TCP connections the PBXctl process never attempts to connect on port 25 really Strange and after 5 hours of analysis, the log still reports SMTP: Cannot resolve. we are convinced it's now time to reinstall // update.. Successfully been running 2.1.5.2357 for some time, and this problem surfaced several weeks ago.... I hate not knowing the exact cause of a trouble but in this case, I know the shortest distance to the end goal is likely now to be the reinstall.... Yuck , but we'll improve on the overall system in the process... Cheers.

The Mail Server entry is in fact x.x.x.6 as the mail server entry. From windows or command prompt an SMTP test tool and command line works fine, proving the Exchange server x.x.x.6 accepts and relays the mail. We made a hosts file entry for mail.domain.com and changed the mail server entry and the same results. Pinging by name or IP operates fine. What did you mean by including an "@" ? Cheers,

Any suggestions on what may be happening are helpful.. PBXnSIP windows Server with IP x.x.x.250 no longers relays mail via x.x.x.6 Exchange email server. From the PBX a full telnet email session can be sent via Telnet and the Exchange server successfully sends these emails. the PBX log has the following entry a na few moments later the error Sending CDR email to <test email address> SMTP: Cannot Resolve Wireshark traces do not reveal any SMTP packets, meanwhile the 40+ phones and all calls are being handled fine... The PBX service has been restarted and no change to the symptoms. We'd like to reveal the cause if possible. A total system restart is planned, but not very hopeful as SMTP work fine with telnet to the exhange server... Any Deep Thoughts? - Cheers.

A quick read of the TZ180 manual tells me this should do all that needed. I think you should be able to set these up for SIP/5060 to pass across your VPN, while letting TLS UDP RTP packets to run freely and by using the Bandwidth management features youy can enforce a rule so that that RTP packets will take extreme precedent over all other packets (Both ingress/egress) This might prevent the running out of bandwidth on the current setup, but bring all other traffic to a halt.... Figure 90K per ULAW call path up/down 32K for 729. I have no personal experience with the TZ180 but the manual looks as if its quite capable.... Cheers...

Your results are as expected. IPSEC VPN does add on average 15 to 20% packet overhead for the encryption/encapsulation. Also problems you may have will result from IPSEC adding it's 20 to 30 bytes to packets and commonly caused packets to fragment on MTU 1500 interfaces as packet sizes increase. Pushing RTP traffic within an encrypted TCP/IP stream removes the natural operations of RTP as suddenly all packets are delivered, checked, reassembled.. Learning a lesson from Verizon Business, they require IPSEC/VPN for the SIP 5060 traffic and hand off RTP to the proxy/gateway assuming you use TLS then all traffic would be protected still... I would also consider the following options; 1. G729 to reduce BW requirements 2. Move to L2TP on a quality Router that supports BW management with source or destination IP

Update on this Topic.... MP118 Audio Code.. We had a client accept our proposal, and the original proposal had assumed the the current 8 line configuration would become an 8 line incoming hunt group. We now suddenly have to deal with duplicating the old config.... Lines 1 thru 4 go to a hunt Group.. Lines 5 through 8 each are stand-alone lines that are to ring specific extensions.... to complicate matters, we now must forward calls to a third party voice mail and if possible us the CENTREX single line transfer feature using the FLASH Option.... did the registration of different lines on the MP11X work as expected? Cheers - A.T.