[HQ plus remote CS410a]

Keeping CS410's with stock ext 40 through 49 and having 4 remote CS410...

 

A simple dial plan allowing HQ to call remote CS410's and accessing the main AA #70 or to specify the extension would be better.

 

We created SIP gateways pointing domain to the FQDN of the remote CS410

 

Created a Dial PLAN do dial the first remote office

 

^50([0-9]{2}) this is in the pattern match designed to allow HQ users to dial a 2 digit remote office and the 2 digit extension

 

replacement is \1 (this should be the 2 digit that was dialed within the parens..

 

Assign this a dial plan value lower than our 7 digit or 10 digit trunk and this should go.. (did I miss something?)

[Deploying 3 new CS410's next week (maybe)]

I suspecting that myself, because last week we deployed a 410 on a Major Telecom carrier (Dynamic IP T1) circuit providing dynamically allocated analog lines for 4 POTS, and a FAX. They use the Cisco IAD gear too. I'll see if I can get the real configs of one for comparison.

[Deploying 3 new CS410's next week (maybe)]

This should be using the PSTN default trunk. resaving button will be performed and retested.

 

Upgraded to release 2448

 

Again the CO light relights and the previously used FXO port will not answer an incoming call until that light extenquishes on the phone. Our 4 POTS lines are being provided off of a CISCO 2400 series IAD from out ITSP. They have a large number of these same solutions provided and installed on many traditional PBX's with POTS interfaces. I've had them reconfirm the settings.

 

Previously posted the SIP log from the Phone and The CS410 covering the entire test dial period that should indicate why the Snom 320 received a message to light the light.

 

[5] 2008/02/17 11:54:23: PSTN: RTP destination=100007f

[5] 2008/02/17 11:54:23: PSTN: RTP destination=53674

[5] 2008/02/17 11:54:23: PSTN: RTP OOB codec=101

[5] 2008/02/17 11:54:23: PSTN: Start call on 0

[6] 2008/02/17 11:54:23: Last message repeated 2 times

[5] 2008/02/17 11:54:23: PSTN: Country Code set to 64

[5] 2008/02/17 11:54:23: PSTN: Tone Detection set to 64

[3] 2008/02/17 11:54:23: PSTN: Channel 0 going to DIALLING

[7] 2008/02/17 11:54:24: PSTN: DTMF: 2

[7] 2008/02/17 11:54:27: Last message repeated 7 times

[3] 2008/02/17 11:54:27: PSTN: Channel 0 going to TALKING

[5] 2008/02/17 11:55:03: PSTN: Received BYE message on channel 0

[3] 2008/02/17 11:55:03: PSTN: Channel 0: Hangup

[5] 2008/02/17 11:55:03: PSTN: Channel 0 goes onhook

[5] 2008/02/17 11:55:03: PSTN: enable_callerid 0

[3] 2008/02/17 11:55:03: PSTN: Channel 0 going to GO_ONHOOK

[3] 2008/02/17 11:55:04: PSTN: Channel 0 going to IDLE

[5] 2008/02/17 11:55:05: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[5] 2008/02/17 11:55:05: PSTN: Response code: 100

[5] 2008/02/17 11:55:05: PSTN: Response code: 200

[MOH info]

We were thinking if we should start another project that plays MP3 files and convers them to linear, then streams then via RTP to a local MoH port. But maybe it makes sense to search if such a project already exists and just properly document how to use it. Unfortunately, MP3 players seem to assume they run for a few hours maximum, so they don't have to take care about memory leaks.

 

MOH files have become extension of the total marketing efforts to tell a story. In traditional phone systems, these files are stored on Tape, and Now small audio devices and they continuously stream the MOH our via an Audio jack and people placed on hold will pick up on the marketing messages randomly along the 5 minute track that's made up of small 20 second messages intermixes with courtesy announcements like, "Some one will be with you momentarily" and another short advertisement.

 

Creating a random start point within the MOH file would be nice... MP3 and all that other stuff in my opinion would be a distraction to continous improvements on the core product...

 

Or perhaps allowing a string of MOH files seperated by Colons or commas, and being able select;

Linear play (start at the beginning)

Linear Random (play sequencially start on random file)

Random

 

Or maybe designate anothe account type as Message Box with these features, and it can be the designated MOH file for Different Autoattendents..

[2446, 2448 releases]

Should we be applying the 2446 or 2448 releases on three production systems in the next 2 weeks?

All Identically programmed with 4 snom 320 extensions 40 thru 43

All with 4 Analog Lines from Local PSTN

All with a Broadband connection for casual sip trunks to HQ.

 

Is there a release note on fixes or enhancements for these incremental releases?

[Deploying 3 new CS410's next week (maybe)]

Did you use one trunk for the four lines? If you change the trunk after the buttons, then you should press save on the buttons again. There are internal references that point to the account, and it can get screwed up if you change something afterwards.

 

This should be using the PSTN default trunk. resaving button will be performed and retested.

[Deploying 3 new CS410's next week (maybe)]

yes, buttons were created using button configs.

yes, PnP was used to deploy the phones,

Extensions were assigned with MAC to extension assignments

 

4 Analog lines are attached...

 

The left most top 4 buttons represent FX01 thru FXO4 The top right most Buttons Represent 4 extension 40 thru 43

 

Press a button and make a call, the other phones accurately show that line 1 is in use and the appropriate extension is in use.

 

Hang up from that call, within 2 seconds, the line line reappears on that phone....(the FX0 l.e.d.) that was just used by the PBX is extinquished, however, place a call immediately back into that line, and the FXO lamps rings and flashes but the PBX does not answer the line...(until the line light goes off on the phone)

 

The previously uploaded files were the complete PBX log for a single test call and the other upload was the Sip trace from that same phone for that same test call.

[Deploying 3 new CS410's next week (maybe)]

I'm hoping to learn this our problem...

 

almost fully default programming...4 buttons assigned to 4 snom 320's (v7.1.30)

 

or allow autoline select..

 

hang up via X button on go on-hook with handset... the just release line light on the phone lights, and the line that was used shows the incoming ring but the PBX will not answer that call. Wait 30 to 40 seconds. the sla lamp extenquishes and all is good..

 

2 files have been uploaded the pbx log and the snom phone log...

 

This is a 100% duplicatible problem and we are scheduled to install three identical systems next week. All over 100 miles away from our offices....(I'm not feeling so good, but I hope that this is a simple problem)

 

Cheers - I'll be drinking

[Single Extension (makes but can't take a call)]

Win V 1.5.1

 

1 extension using Snom 360 6.5.8 firmware suddenly can't recieve a call, but can make calls all day long...

Easy fixes were first reset phone to defaults, then a restart of PBX service...

 

Gee we gotta work now....

 

Maybe this is a Snom Issue, but perhaps not. Any Diag Tips helpful..

 

A snom Phone SIP trace contains these references to dropped SRTP for bad MAC...

 

8] 20080214164720: route_pending_packet -1368: entry=a udp 192.168.1.112 2057

[8] 20080214164720: route_pending_packet -1368: entry=udp 192.168.1.112 2057

[8] 20080214164720: Send Packet 200

[8] 20080214164720: route_pending_packet -1369: entry=url sip:123@192.168.1.121:2051;line=kr4lp4nz

[8] 20080214164720: route_pending_packet -1369: entry=udp 192.168.1.121 2051

[8] 20080214164720: Send Packet NOTIFY

[8] 20080214164720: route_pending_packet -1370: entry=url sip:117@192.168.1.117:2051;line=gv8x1x75

[8] 20080214164720: route_pending_packet -1370: entry=udp 192.168.1.117 2051

[8] 20080214164720: Send Packet NOTIFY

[8] 20080214164720: route_pending_packet -1371: entry=url sip:101@192.168.1.101:2066;line=gv8x1x75

[8] 20080214164720: route_pending_packet -1371: entry=udp 192.168.1.101 2066

[8] 20080214164720: Send Packet NOTIFY

[4] 20080214164720: Dropped 1 SRTP packets with wrong MAC

[4] 20080214164721: Dropped 10 SRTP packets with wrong MAC

[4] 20080214164722: Dropped 100 SRTP packets with wrong MAC

[8] 20080214164726: route_pending_packet -1372: entry=a udp 192.168.1.112 2057

[8] 20080214164726: route_pending_packet -1372: entry=udp 192.168.1.112 2057

[8] 20080214164726: Send Packet 200

[8] 20080214164726: SIP Rx udp:192.168.1.112:2057:

BYE sip:112@192.168.1.254:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.1.112:2057;branch=z9hG4bK-b95k4u02rhjv;rport

From: "Billy" <sip:112@localhost>;tag=vaf21dr489

To: <sip:1115550200@2AA.64.32.4>;tag=479

Call-ID: 1f013968@pbx

CSeq: 1 BYE

Max-Forwards: 70

Contact: <sip:112@192.168.1.112:2057;line=wwndntd5>;flow-id=1

User-Agent: snom360/6.5.8

RTP-RxStat: Total_Rx_Pkts=258,Rx_Pkts=258,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=255,Tx_Pkts=255,Remote_Tx_Pkts=0

Content-Length: 0

[MOH info]

I assume you're after some kind of disclaimer to be played before being connected to someone?

 

Not Quite, but close. Assume the MOH message is the best thing since slice bread and callers simply want to be put on hold to here the message by choosing option 7 on the autoattendent, but they would prefer the message randomly start within the 2 or more minute long file. The MOH message is full of 10 second spots and they would like any caller to begin listening at random spots...This file may grow to be 20 minutes in length. Or we'll need to break this up into many files or move this to a streaming input to get randomness.

[MOH info]

Assuming you have a 2 minute MOH wave file, does it always begin playing at the beginning when the first person is placed on hold?

 

Does the second person get it's own 2nd stream or simply hear the current location of the first caller on hold?

 

Assuming your MOH file was important, and you wanted to make it available as a choice on the AA can you place a caller on hold from a button choice?

[Using 1 2 and 3 for direct dial and using 3 digit extension]

Very Creative,

 

Quick way to burn accounts, getting the real problem fixed would be nice too.

[Using 1 2 and 3 for direct dial and using 3 digit extension]

!^0$!8240! !^9$!400! !^(1|2|3|4|5|6|7|8|9|0)(1|2|3|4|5|6|7|8|9|0)(1|2|3|4|5|6|7|8|9|0)$!\1\2\3! !E!401!

 

Not well versed in IVR nodes, short of getting a real fix to the fundemental problem, doing something like this may be to best route, but how does the IVR note know when to stop reading digits.

 

Press 1 for sales

Press 2 for Support

Press 3 for Accounting

 

and all of the extensions are 2xx through 2yy

[New Post Old Subject - Sorry]

How has the community been dealing with the need to change and modify scheduled service flags? I hope a previous thread about the need to dynamically change service flags leads to a solution allowing an extension to dial a service flag account and flip it.

 

Could I have this put to the top of the development list? Cheers.

[Diagnosing and improving hangup detection]

We had two problems that we can't identify. In the shop before deploying a new 410, when you hung up a phone, the SLA reappeared on the phone, and subsequent calls into the device on the previously used line, went unanswered. More testing on this today.

 

A 410 that was deployed last week has required a reboot already. Tech on site this morning to remove all logging, and to setup PNP on snom 360's. Does a error log exist in the file system that might identify what happened and how to prevent this?

[2.1.5]

If you add a 1 to the 10 digit phone number stored on the account settings - it will treat them normally.

 

Voila' I wonder if these means we can force inbound callers from a defined list of Cell Phones (50 employees to alway come into a BACK Door Auto Attendant?)

rather than bother the busy receptionist.

[2.1.5]

We have created 6 accounts for a client to directly send all calls to the user Cell Phones.

 

When those Cell phone users dial the main number, that are given DISA access to make outbound calls. Go to voice mail or go to the Autoattendant

 

The client would prefer that these users calling from the cell phone are treated as any other caller and no DISA outbound calling capability.

[Diagnosing and improving hangup detection]

What options are available in the new BLACK CS410's to control call hang-ups?

[Using 1 2 and 3 for direct dial and using 3 digit extension]

I would say unless there are white spaces in either field - or that the extensions don't exist - but I have to assume they do because you are obviously well-versed in pbxnsip... that it should work the way it's laid out.

 

Well, I deleted and retyped each entry hoping to find a whitespace, but to no avail the problem occurs...

 

The log file for media just when the user presses the 2 in an attempt to dial 201 is as follows.

 

9] 2008/02/11 13:59:05: DTMF: Power: 97 6 0 0 0 2 99 1 0

[9] 2008/02/11 13:59:05: DTMF: Power: 100 6 2 0 2 3 100 0 0

[6] 2008/02/11 13:59:05: Received DTMF 2

[8] 2008/02/11 13:59:05: Play audio_en/aa_not_existing.wav space20

[Using 1 2 and 3 for direct dial and using 3 digit extension]

I just verified it on latest 2.1.6 and it works for me - at least initially...

 

Call me crazy, but is the attached image as you would expect it to be?

 

Also using 2.1.6.2442 (Win32)

[Using 1 2 and 3 for direct dial and using 3 digit extension]

Supposedly using the # sign after the Direct Dial extension alleviates this - but I've had problems - especially when the direct dial extension goes to another AA.

 

The AA greeting is Press 1 for sales, 2 for Support

direct destination settings are

1 sends to a hunt group with three extensions

2 sends to the sales persons extension in this case 201

 

placing a #after the 201 entry on the direct destination options results in the caller getting, "I'm sorry you are not allowed to place this call" message

 

placing a # in the direct destination field after the 2 results in "This extension number does not exist" message

[Using 1 2 and 3 for direct dial and using 3 digit extension]

Customers like using Press 1 for this, Press 2 for that, and 3 this something else. They also like having extension like 101, 201, 301v etc...

 

what must be done in the auto attendent to allow this so that when the person calling presses the first digit in a valid extension it doesn't automatically send that caller to the quick access dial in the AA?

 

Some sort of delay is necessary to prevent the automatic redirect to an extension.

[Backup pbxnsip installs]

Did anyone check out http://www.linux-ha.org/GettingStartedWithHeartbeat? I think that does solve the problem for Linux-based installations.

 

In 30+ years from the IBM 360's, NetFrame Servers, the growth of RAID, clusters, etc.. I've never seen a system that is truely redundant that has has no single point of failure somewhere. Keeping it simple reduces the failure points to a mimimum allow you to have a easy to plug in disaster backup plan. The MTBF of devices is a function of rhe number of devices and the fewer the better. NVR ram in appliances as in CS410 and the future of bootable OS's on motherboards is going to shrink this stuff further.. Less is Better.

[DTMF In vs. Out]

If you have a FAX, check out the latest image http://www.pbxnsip.com/protect/pbxctrl-2.1.6.2443.exe. Set a direct destination of "F" to your FAX extension in the auto attendant and see if the PBX redirects the call to the FAX device.

 

I don't have a FAX here... Obviously I was able to survive without it!

 

We'll use both a Linksys supura ATA and a Patton ATA for testing. We'll get this tested Monday or Tuesday

 

How about a screen capture of what that Direct destination "F" is to look like...?

[DTMF In vs. Out]

BTW we are also looking at the FAX CNG tone (that is the tone in the middle of the 9 values), that should make it possible to detect FAX in the auto attendant.

 

Anything we can do to help, we'll be glad to.