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About andrewgroup

  • Birthday 02/09/1957

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    IT Infrastucture Support with added VoIP services

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  1. The PBX snmp values are the same across the board. Linux has an snmp module, and if you want them on the same port "161" you may need to add a virual NIC net-snmp I believe is the linux module... a common command would read the linux values from the net-snmp linux module... snmpwalk -v 2c -c private . Please Note - The Snom SNMP stack in both the PBX and any Snom Phone will not respond or support the SNMP WALK... you can only get values with a properly formed SNMP GET command....
  2. Sent a PM with my contact information, send me a note with details..
  3. We definitely had issues of crosstalk within active calls, and aligning the ports fixed the issues, or it was EXTREMELY coincidental. So we've posted various ITSP Port recommendations in the past. Coredial out of Philidelphia recommends RTP Ports: Port Range Start: 10000 Port Range End: 30000, We've had no issues with them... Window Stream - Nuvox when using their SONOS servers gave us these recommendations long ago. I assume they can use multiple UDP ranges for Media RTP Media - udp range 16384 32767 udp range 49152 53247 Signal Protocol - udp 5060 tcp 5060 udp range 1718 1720 tcp range 1718 1720 udp 2427 tcp 2427 udp 2000 tcp 2000 We know have a large enterprise using the BroadSoft Platform on the Paetec Division of Windstream and working through the re-occurance of call issues. I personally called the client and appeared to have been connected to an in progress call..
  4. Thanks for the Snappy reply, but in the past, we've have situations where callers appeared to be cross connected, and the issues were resolved by matching the PORT range with the port range information provided by the carrier. IE Windstream / Nuvox Communications on their SONOS sip servers recommend the following ports. These SIP trunks are not behind NAT devices and making these adjustments fixed the issues at the time. RTP Media - udp range 16384 32767 udp range 49152 53247 We now make sure and ask the tech support groups at any carriers and see if they make recommendations. This new trunk, just added a week ago, in on a BroadSoft Platform, and they recommended the MEDIA ports to be 35000-40000 I'm sure the carriers have to create some sort of acceptable port values in their equipment, so it seemed logical to match that. This new PBX and SERVER is running Windows Server 2008, and their is a known issue with DNS grabbing a range of PORTS for DNS services, but I don't think that had any effect or issue within only the DNS client.. We'll dig some more and get a trace.
  5. When you have two trunks configured from two seperate providers and dial plans allowing the use of either trunk. How do we deal with the issue of each ITSP requesting the RTP ranges to be in two different Port Ranges. Example 35000 - 40000 for ITSP provider #1 and 49152-53247 for ITSP provider # Do we simply put 35000-53247 and let the SDP packets negotiate the final set of ports that will be used? Do we select the "FOLLOW RTP" option on the ports settings page? We recently added another SIP provider, and we are experiencing dropped calls... We think this may be the issue. (I just placed an inbound call, and was connected to what appeared to have been an active call hearing one side of an active call.
  6. On advise, I am making a second post about call accounting software options. We have a call center prospect and they have an interesting request. (outbound calling only) they would like a shift report that shows the IDLE time that this extension was making no calls, regardless of being on hold, talking or otherwise. they are a 30+ year old firm with ZERO technology, and a the owners believe this is the single most important report... From that report they can drill down to more information if needed. they operate on shifts so a reporting that shows EXTENSION IDLE or NO CALL TIME from 8:00 - 12:00 arriving at 12:15 will immediately highlight potential trouble spots. The second shift report from 13:00-17:00 arriving at 17:15 addresses the afternoon shift.. It seems the available Call Reporting Software only reports on activity (not the lack of calls) Sure a little math gives you a number, but having a canned report is not to much to ask. Your thoughts,,,
  7. We've recently completed a reasonable advanced and agressive setup to help a client.. We have a 10MB Fiber Trunk from Windstream Communications and the VoIP trunk originates on the PAETEC Broadsoft platform. On the trunk they provide 5 static non routable IP addresses of our choosing, 10.x.x.x, 172.x.x.x or 192.168.x.x These five IP addresses are the interface to our PBX... We have a PBX on 2 IP addresses and they have provisioned their SIP trunk to do out of band sip inquiries to the trunk. In the event the primary PBX fails to respond "200" the next inbound call we be delivered to the second PBX on another trunk. They continue to poll the first PBX and when it reponds, the call resume on the primay PBX. We also have an enterprise option, allowing the PBX's to deliver the originating Call ID to a number the call may be transferred too.. They have 24 x 7 emergency inbound calling and it's important the on-call folks see the original caller ID and the area codes they come from. A note of interest the second PBX can be configured to place calls against the primary and secondary trunk simultaneously. this environment creates a fully redundant system at the carrier environment as opposed to trying to make multiple servers failover and use the same trunk. If you have clients with this level of requirements, please PM us and we can get you in touch with the consultant and companies that can make this happen. (We have no vested interest, but wanted to share the positive experience.) Cheers - Andy
  8. actually I believe the testing of the SIP trunk is more advanced than that. We simply outpulse the appropriate AREA CODE ANI where we have the carrier provisioned numbers, and they deliver the call through the PSTN carriers to the local numbers. All of this occures on a single SIP trunk. We simply need to make sure we only present the correct numbers in the ANI stream. So we thought we would make extensions with the correct ANI's and register multiple phones against that number... Assume my explanations of the SIP trunk will work as described, we are wondering if their is a dynamic method to allow any extension to call into any area code. By way of an agent group, dial plan, or other interesting method. Thanks...
  9. We have an "Old School" manual outbound call center that focuses all of it's dialing into less than 2 dozen area codeds among a dozen calling agents. Groups of 2,3 or 4 agents dial into these areas during a campaign. Here is the plan to help improve there success. We have a national provider than can provision our SIP trunk to deliver these area codes on our SIP trunk and will outpulse and originate the outbound call to the area codes so the recepient sees a friendly area code. We hope to eliminate the need to dial the area code in one of several ways. 1. Agent Groups are for inbund calls only. No Help 2. We make Extensions with AREA code ANI's and register 1,2,3, or 4 phones on that extension. (This seems to be the best choice) This method is static and offers little flexibility 3. Create dial plans that insert the correct area code example 10555-5555 the diail plan see beginning two digitis [10] and inserts a three digit dial plan. [11] etc.... [12]..... This would allow any of the two dozen agents to participate in a campaign, but eliminates only 1 digit in dialing. What other methods might one suggest? Secondly the last post related to using Metropolis call accounting software was in 2011. An important report the agency wishes to have is Shift Idle time. IE Extension IDLE time from 8:00 - 12:00 delivered at 12:15 Seems most call accounting systems are good at calculating call activity, and does anyone have experience with a call accounting system that calculates the IDLE times by subtracting the activity from a time specific time period? All comments are welcome and greatly appreciated.. Andrew
  10. Yes, these problems arose last November after windstream - Nuvox did a software upgrade on one of their Sonus Softswitches. We quickly identified the issue after pressing them to tell us the differences between the .173 Switch and another that was on another client with no problems. We learned they had done a software upgrade, and a SIP trace they identified the problem that we were sending the call quality stuff to them that was causing the Sonus to reject the calls... We've had no problems since and make these our default settings..
  11. FYI. We have numerous clients using windstream (Formerly Nuvox) SIP trunks provided off of a T1 using a Cisco 2431 IAD. Windstream provides a Client Internet IP address on Wic1 and a seperate single IP in WIC2 for the SIP trunk... A problem began within the past day where calls to or from some cell phones would drop in 15 to 30 seconds. The problems was traced to a setting in the SIP trunk settings in the Sonus Server related to what they call a service Profile. The preferred service profile they call "Nuvox Sip-SipT-Noss" and this accommodates more codecs and keep alive settings for the cell carriers. For some reason this clients service profile was set to "Nuvox Transcode" The following are the settings we are using across the board for all SnomOne installations where Windstream - Nuvox using the Sonus switches are the carriers RTP Media - udp range 16384 32767 udp range 49152 53247 Signal Protocol - udp 5060 tcp 5060 udp range 1718 1720 tcp range 1718 1720 udp 2427 tcp 2427 udp 2000 tcp 2000 Trunk Settings: Type: SIP Gateway Dest: Generic SIP server Domain: Public IP of PBX Proxy Address: Codec: G.711U G729A Settings To Yes or Enabled: Lock Codec During Conversation, Interpret SIP URI as Phone Number Remote Party: Remote-Party-ID For DIDs use Send Call to Extension: !(.*)!\1!t! Wind Steam uses SONUS soft switches and they have a known issue. (SEE BELOW) http://wiki.snomone.com/index.php?title=RTCP-XR In order to disable these settings you must log onto the SIP PBX using the IP address and then move to the ADMIN settings page and paste, (1 by 1) the following URLs into the web. After each paste the page will switch to a status page, and you should go back to the settings page and do each of the remaining in the same way. These changes do not require a reboot, and take immediate effect. http://pbx-ip:8080/reg_status.htm?save=save&rtcp_loss_rle=false http://pbx-ip:8080/reg_status.htm?save=save&rtcp_dup_rle=false http://pbx-ip:8080/reg_status.htm?save=save&rtcp_rcpt_times=false http://pbx-ip:8080/reg_status.htm?save=save&rtcp_rcvr_rtt=false http://pbx-ip:8080/reg_status.htm?save=save&rtcp_voip_metrics=false
  12. Seems the previous install has installed a newer version of C++ libraries
  13. Any further development? Seems this fails instantly on w764/pro with this error Problem Event Name: CLR20r3 Problem Signature 01: attendantconsole.exe Problem Signature 02: Problem Signature 03: 4f91e78c Problem Signature 04: AttendantConsole Problem Signature 05: Problem Signature 06: 4f91e78c Problem Signature 07: c Problem Signature 08: a Problem Signature 09: System.Windows.Markup.XamlParse OS Version: 6.1.7601. Locale ID: 1033 Additional Information 1: 0a9e Additional Information 2: 0a9e372d3b4ad19135b953a78882e789 Additional Information 3: 0a9e Additional Information 4: 0a9e372d3b4ad19135b953a78882e789
  14. We've taken the following approach. Burn an invisible autoattendent and create the following service flag 601 Normal Closed (Sheduled close hours) NOT OPEN 602 Holiday Closed Hours 603 Manual Closed (gone home for weather of early close) Then place the service flags in the AA service flag field in least used order 603 602 601 Then brach to to the appropriate AA,HG,AG, or EXT.... in the matching order... Make the timeout settings so if none of these are set, then send the call to a CLOSED AA.... Burning this AA allows you to create other Manual or scheduled service flags.. (IE Extended Hours 17:00-19:00 M-F, this would be placed just after Holiday) This at least gets the following Logic in place.... IF CLOSED FOR BAD WEATHER go to the WEATHER ALERT AA, or if on Holiday go to the Holiday AA, or if OPEN goto the OPEN AA, ELSE we must be closed so play the Closed Greeting on this AA.. The logic is the stack whatever service flags you need in LEAST LIKELY to OCCUR Order.... We have begun to standardize our Service Flags and Matching Actions beginning at 601 through 610 for the3 service flags and 701 through 710 for the associated AA's, HG, and AG,,
  15. Thanks all, this may not be the perfect solution, but added the public IP alias to the Soho, removed the 192.xxxx LAN gateway, IP replacements were added, along with routes and appears to be OK. Seems to require the remote phone ports forwarded to the PBX. We will take another look to further simplify the install, but this seems reasonable in light of the goal... Once we regain some space on our upload portal, we'll provide better docs....
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