andrewgroup

Is their a 50 address limit displayed to the user login? We can search for the 200+ speed dials, but cannot browse to list any more than the first 50. Are we overlooking a NEXT button or something?

To speed firmware upgrades we duplicated the Snom provisioning WEB site on in an internal server with a DNS record pointing to it so a group of Snoms would get the upgrade far fast on the LOCAL lan. Having rescued some phone with TFTP, the thought that TFTP can restore the rescue image pretty darn fast with no reboots or hassles. That said, moving from 5,6 or for that matter moving to the latest version seems to be done the fastest using the rescue image and TFTP.. Will check with Snom to determine if any hardware revisions on snom phones would prevent going from lower revs to the latest with the rescue image. Will post the reply here.

Might you know if you can use the Snom 7.3 emergency recovery image to upgrade a 6.x snom to the latest? hitting 321 and entering an IP address and TFTP server seems to restore a Snom 360 to the latest firmware about 100 times faster than any on-line upgrade.... Keeping an TFTP server up is a mighty fast way of upgrading phones.

http://forum.pbxnsip.com/index.php?showtopic=599 System has been operational since April 06, AMD 3100 512MB XP PRO, 3WARE IDE RAID card.....

Is the internal DNS server a Windows Server? Is that DNS authoritative on the main domain name? Did you create a Stub Zone? We were figuring SRV would allow us to make a DHCP VLAN for phones that would pull the public DNS info and have an internal IP as high priority on srv and a secondary lower priority public IP... We would set our router to allow a triangle route so internal phones could register against the public IP if it went to the lower priority entry..

Logging PNP events might help diagnose troubles... I also think different log files should be an option for different logging events..

When PNP works its great, when it breaks, it's a royal pain in the $#% and the diagnostic process to get it working again is a prayer. Does a clear explanation of how it works exist? This would include a basic flow of events and what happens along the way like the creation of the generated files and more... A clear understanding of the flow might help others (and me) diagnose problems.

Had this up for nearly 3 years.. 192.168.1.250 on an Internal LAN..... 10.10.1.2 on a second NIC with a SIP trunk to a PRI gateway on 10.10.1.1 and public IP x.y.w.z on 3rd NIC... Local LAN phones register against the 192.168.1.250 IP Remote offices hit the Public IP and a point to point SIP trunk with the PRI Gateway We've posted notes in best practices for an XP installation...Of Course no services are running on the PC except the bare minimum... Good Luck..

How many installation has the PBXnSIP community deployed where every user can unplug their office VoIP phone, carry it home and barring NAT firewall problems plug it in and Voila' ? Our Goal is to figure our how every installation past, present and future will work this way. To accomplish this the phones must be PNP, The Phones should register using SRV records On first glance, a common installation will likely have a Windows Server supporting DHCP... option 66 for configs is the obvious... but, the PBX might be on an internal lan, say 192.168.1.99, The pbx registration should pull the PBX registration and do a DNS SRV lookup, but it seems the Windows DNS server internally will need a SRV record to 192.168.1.99 sip_udp so that local phones stay on the local LAN. Carry the phone home, I assume the settings would retain the DNS look up info... Of course the real world DNS's would have a public IP srv records.. (The real world is the internal windows dns servers are not likely authoritative on the domain) I assume a firewall would port forward / ALG the sip stuff to the PBX, or the PBX would have a public IP also... I think this brings up a lot of misc. issues, and I would like to know if we've made this overly complicated, or have others wrestled with these issues?

With so many recommendations to change the default ports 80 and 443, what is the logic in not making the default ports something other than 80 and 443? Wouldn't this prevent a number of posts and potential conflicts? I think the CS410 is has defaulted to 8080 for a while.

Delete the Idendity seemed to do the Trick... Thanks Self, School of Hard Knocks So what must one do to restore PNP to a single extension?

The following is the log from an extension that was previously working fine. After attempting to get an M3 to parallel this ext, and trying to confirm the SIP password, the result is 403 Forbidden.. Immediately below is another extension.. So what must one do to restore PNP to a single extension? The MAC is correct, and all looks well.... Must one remove generated files, etc, and what was the actual cause? If you can determine from this post? Thanks REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.166:2141;branch=z9hG4bK-9b6ocpg84gks;rport From: "Kenny ACC MNG" <sip:302@localhost>;tag=a5pkwsegun To: "Kenny ACC MNG" <sip:302@localhost> Call-ID: 3c26701cbcf2-1rlt6ss1r4kn CSeq: 6 REGISTER Max-Forwards: 70 Contact: <sip:302@192.168.1.166:2141;transport=tls;line=yoxynt0d>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:4216b9d9-0985-478f-960e-2bb7b5bb915f>" User-Agent: snom360/7.3.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.166 WWW-Contact: <http://192.168.1.166:80> WWW-Contact: <https://192.168.1.166:443> Proxy-Require: buttons Expires: 3600 Content-Length: 0 [9] 2010/01/07 18:56:21: SIP Tx tls:192.168.1.166:2141: SIP/2.0 403 Forbidden Via: SIP/2.0/TLS 192.168.1.166:2141;branch=z9hG4bK-9b6ocpg84gks;rport=2141 From: "Kenny ACC MNG" <sip:302@localhost>;tag=a5pkwsegun To: "Kenny ACC MNG" <sip:302@localhost>;tag=48c390b7f1 Call-ID: 3c26701cbcf2-1rlt6ss1r4kn CSeq: 6 REGISTER User-Agent: pbxnsip-PBX/3.4.0.3201 Content-Length: 0 [8] 2010/01/07 18:56:21: Packet authenticated by transport layer [9] 2010/01/07 18:56:21: SIP Rx tls:192.168.1.145:2059: REGISTER sip:localhost SIP/2.0 Via: SIP/2.0/TLS 192.168.1.145:2059;branch=z9hG4bK-ktjxeqrfaf35;rport From: "Dan SALES 2" <sip:311@localhost>;tag=tmaab84wpy To: "Dan SALES 2" <sip:311@localhost> Call-ID: 3c26701cf29f-1rlt6ss1r4kn CSeq: 9712 REGISTER Max-Forwards: 70 Contact: <sip:311@192.168.1.145:2059;transport=tls;line=yoxynt0d>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:4216b9d9-0985-478f-960e-2bb7b5bb915f>" User-Agent: snom360/7.3.30 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.1.145 WWW-Contact: <http://192.168.1.145:80> WWW-Contact: <https://192.168.1.145:443> Proxy-Require: buttons Expires: 3600 Content-Length: 0 [8] 2010/01/07 18:56:21: Packet authenticated by transport layer [9] 2010/01/07 18:56:21: SIP Tx tls:192.168.1.145:2059: SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.145:2059;branch=z9hG4bK-ktjxeqrfaf35;rport=2059 From: "Dan SALES 2" <sip:311@localhost>;tag=tmaab84wpy To: "Dan SALES 2" <sip:311@localhost>;tag=18e05bf06e Call-ID: 3c26701cf29f-1rlt6ss1r4kn CSeq: 9712 REGISTER Contact: <sip:311@192.168.1.145:2059;transport=tls;line=yoxynt0d>;expires=181 Content-Length: 0

After fully backing up a 2.1.5.2357 to the latest 3.4.xx on the web, the installation asked that an upgrade occur. Selected yes. the installer ran to completion. Expecting to see 3.4x run, No Love, 2.x was still running, What happened? Cheers

RESOLVED.... Summary... Manually upgrading the 2.x controller to 3.4 controller the system immediately began sending old messages.... So I have safely assumed the problem is the 2.x controller. Restoring the 2.x controller and changing some logging options, I was able to get an error log showing an SMTP error and an associated IP address that was not the IP address of the mail server as configured in the appropriate spot. Several things have contributed to the fix, but the exact cause/affect is not known, Deleting a large number of messaged from the spool folder.. 20K.... The system failed and began saving messages at 1, so the blame can't be the number of messages in the spool folder, but V3.4 might handle that larger number better and it began sending them...That was the clue. Disabled a Public Facing NIC... so the route table would have 0.0.0.0 on the lan with the Mail server. Not sure it was wrong though.. Restored the 2.x controller and the email system began working as expected. (I'm guessing that the files in the spooler might actually contain some information about the mail server on how to send the message. (This might have been the indication I captured with a bad mail server IP address. This may have once changed, causing the backlog... V3 may do it differently.. Just a SWAG. Cheers

We did an in place upgrade from 2.1.5.2357 (Win32) to pbxctrl-3.4.0.3201 and the SMTP services immediately began working. A roll Back returned SMTP to non-functioning. We began getting a could_not_read_file error so we put it in reverse. What if any file formats may have changed? Cheers

If this was a member of a local domain, and yes a ton of packets are flying about when using Wireshark when trying to capture the ARP/SMTP queries. I assume the ARP table would be shared across all applications. An onsite visit is in order to find the cause../solution. Will post the fix here if it's not a LAN/MAC table problem in a switch or router. (wishful thinking)

for some unknown reason, the local windows applications like outlook express, telnet, mailtester, all seem to be able to successfully bounce email off of the smtp mail server at 192.168.1.5 but the PBX executable does not and creates the "CANNOT RESOLVE" entry in the Log file. The configuration jhas been set to the IP address, a DNS entry, a DNS entry with a matching entry in the local HOSTS file. We've replaced the original executable from the ZIP file, but no Love. This may prove to be an elusive LAN issue related to routers etc, Could a license expire and cause a problem like this?

Attached is a diagram showing a 3 NIC XP PBXnSIP... Before you jump on the DNS server is the problem hear me out. we can configure any number of email clients (outlook express) to use the 192.168.1.6 IP address as the mail server and it will successfully bound mail of of the SBS 2003 Mail Server. (Yes the IP address of the PBX is in the allow relay settings that allows outlook express to work correctly. Only the 192.168.1.250 has a gateway .1.... (Note .1 is a linux gateway with an SMTP mail server and Outlook Express can send via it too.. The 100.x network is a private network going to the Epygi T1 gateway only... The Public IP ISP provider has no Mail Server to use as a test.. This setup had worked for some time, I've restored a original copy of the pbxnsip executable to no avail... Thinking outside the BOX, does the PBXnSIP service somehow bind itself to a specific Interface and somehow that has happened that will not allow the DNS queries or the SMTP transactions to occur on the correct interface? mail settings have been set to the local FQFN of the mail server with matching entries in the hosts files DNS has been removed from all NICS and used the Mail Server IP's only DNS only on the LAN IP. Doing most of this testing remote, means we can't remove the gateway on the public IP...but could the LAN IP as a test. Just occurred and will do noew and post tommorrow. If fails, kind of thinking a rebuild.... or a LAN issue exists with a switch or router. Cheers PBX.pdf

Do you mean in terms of recording every inbound or outbound call simultaneously? Assuming that is what you mean, then this is a very technical question that is greatly affected by the platform running PBXnSIP. We designed an installation with plans to record up to 96 calls simultaneously, but the project fell through. We were very confident in our system design that we could record these and more with no big difficulties. Will be deploying the same design in December with 24 concurrent call path recordings. 17,000 minutes per month. Will update later.

could you provide details on the configuration of the environment? audio issues are commonly related to NAT routers and routers no properly configured to dynamically manage SIP calls. On a PBXnSIP server with a Public IP exposed 1-way audio is less troublesome, but can be affected by adjacent devices on public IP's not playing fair..Internally on the LAN 1-way audio is less troublesome but LAN switches not properly configured for 802.1X features can result in the RTP streams being affected. If these calls are just with an external caller, and you are using a SIP provider, then look to any router that you may have in front of PBXnSIP? RTP ports may need forwarding on less smart routers.

Will Do... Seems a likely candidate for and SNMP OID for FXO ports status.

CS410 OUT PULSING Control codes to Analog Ports. When using the CS410 it will common for you to use new or existing ANALOG lines from any given Telecom Carrier. Historically Telecom carriers offered features codes on their analog lines such a CALL FOREWARD, CALLER ID Blocking and more. It was common to use the “*” Star DTMF character at the beginning of these codes. PBXnSIP internally processed these star codes and will not pass these to the ANALOG Ports. Here is our standard method to pass these codes to Analog Ports using the CS410 and other Analog Gateways. Create a Dial plan that looks like this. THE SEARCH string is 1167\*([0-9]*)@.* The REPLACEMENT String is sip:\*\1@\r;user=phone We’ve assigned the 1167 to be our key to allowing passing of star codes. To send the *67 Block Caller ID you simply enter 1167*67 followed by the number.

In posts http://forum.pbxnsip.com/index.php?showtopic=2891 and http://forum.pbxnsip.com/index.php?showtopic=2946 A problem exists where an FXO port hangs and does not release.. An update 1. Download the new version from http://pbxnsip.com/cs410/sipfxo-8-6 was suggested but the results for not a success... We have an installation 150 miles from our offices with the continuing problem. We are going to swap FXO 1 with FXO 2 and this should isolate the problem to the LINE or to the PORT. In the event port 1 continues to freeze requiring a restart, what are the latests available updates for both PBXnSIP and the Gateway drivers? FYI a restart was required today.. Version: 3.4.0.3194 (Linux) License Status: Appliance Key License Duration: Permanent Additional license information: Extensions: 3/32 Accounts: 12/40 Working Directory: /pbx Calls: 0/0 (CDR: 7) 0/0 Calls SIP packet statistics: Tx: 1491 Rx: 1491 Emails: Successful sent: 0 Unsuccessful attempts: 0 Available file system space: 53% Uptime: 2009/10/22 17:07:42 (uptime: 0 days 06:18:00) (1975 2170304-0) WAV cache: 0 The sipfxo file is below comcerto:~# cat /etc/sipfxo-release sipfxo 0.86, MSP m828_v2_04, VAPI Library Release 2.4.2, API Version 4.0 0 rsion 4.0

Not familiar with that handset, but can you simply rename the ring tones within the phone to what you want? I like simple, can you tell?

We set "Trunk requires out of band-DTMF tones:" to NO in the trunk settings for all US installations.... also the "Inband DTMF detection:" settings in Admin general SIP settings is set the ON