Trunk lines

Hi All ,
Just a quick one, i need to connect 32 Analog trunk lines to a PABX and 4 PRI's to the PSTN. any Suggestions on which Cards i could use on the ISR to achieve this
Thanks in Advance

Hi,
from the PBX point of view, trunk lines are FXO, that is, analog lines that normally go to the pstn. This means the voice gateway will need FXS ports.
If this is your requirement, be aware that the integration may suffer of problems, for example you will not be able to give DID to the PBX.
Anyway, considering the density, the minimum you can use is a 2821 with EVM-HD-8FXS/DID + 2 x EM-HDA-8FXS, 2 x VIC-4FXS, 2 x VWIC2-2T1E1, 1 x PVDM2-32.
Hope this helps, please rate post if it does!

Similar Messages

Connecting the UC560 to a Two Loop Start Trunk Line

I just installed a new UC560 and am ready to convert from our old Panasonic DBS 84 system. The Panasonic is fed by 4 trunk lines that go into two RJ14 trunk connectors. so that each is a Two loop start trunk line plugged into the Panasonic. My question is do I need to modify the two lines in order to connect them to the UC560 and if so, how. If not where and how do I plug them into the UC560 and configue the incoming lines so that calls come in and go out similar to what we have now. Obviously I am new at this, so please forgive the blatant ignorance. Thanks I uplaoded the Panasonic manual if it helps.

You would terminate each line in an individual RJ11 connector, and fit them in the ports labelled FXO on the UC500.
Regarding configuration, if you are the end user, and you have no previous experience or certification, it would be much, much better that you engange a reputable consultant, or certified Cisco partner, because the UC500 is meant to administered by such a professional, and is not very user friendly.

4 FXO lines on a 504G

So I have setup the UC320 with 4 FXO trunk lines and deployed 12 504G phones. I assumed since they are 4 line phones that they would be able to see all 4 trunk lines at the same time.
Upon getting everything configured I have discovered that the first programmable button seems to be permanently assigned to the users personal extension. Which leaves only 3 remaining buttons for shared FXO lines. So I was forced to make the 4th trunk line a hunt group with all the phones.
It's such a simple setup I'm surprised the 4 line IP phones can only actually monitor 3 of the shared trunks leaving the 4th working but hidden from view. There doesn't appear to be anyway to change or remove the first button from being the personal extension button, which I find to be redundant.
All I would like is to have the phone monitor/display the status of all 4 trunk lines on the phone. Any help for this situation? Maybe I'm overlooking something obvious here?

Adam,
You are correct, even in Key System Mode the UC320W assigns an extension on the first line button. This allows for internal dialing between users without utilizing a Shared FXO line.
The other 3 line buttons can programmed via the UC320W.
The SPA504G is a 4 line phone and can be configured with multiple SIP based call control systems, when used with UC320W then the phone allows for 3 user programmable line buttons.
Regards,
Randy

Why Native VLAN exists on a Trunk?

Basically, A Native VLAN carries untagged traffic on a trunk line.
A trunk line allows mutiple VLAN traffic ( tagged traffic). So Why Native VLAN exists on a trunk.
Why Native VLAN was created?
I'm pretty confused up with VLANs.

Hi,
The second question - why PC Network adapters support VLAN tags - is actually easier to answer :)
First of all, with regards to VLANs and frame tagging, there is actually nothing special to support on a network adapter! The VLAN tag itself is in fact stored in the payload of a tagged frame. Even to the most dumb network adapter, a tagged frame looks like any other - Destination MAC, Source MAC, EtherType (set to 0x8100), Payload (holding the rest of the VLAN tag, the original EtherType and the original Payload), and the FCS. Supporting VLANs and frame tags is possible on a purely software level. The fact that network adapters often do have hardware support for VLANs is related to performance reasons: With hardware VLAN support, the tagging, de-tagging, filtering and/or sorting frames based on the VLAN tag value is faster and it allows offloading these operations from the computer's CPU to the network card. However, even if the network adapter did not have any kind of VLAN support, the VLANs could still be implemented purely in the card's software driver.
Ordinarily, you would not see a common PC send and receive tagged frames. However, there are situations in which even a PC would send or receive a tagged frame. One of reasons is the presence of the Class-of-Service (CoS) bits in a VLAN tag. You surely know that basic Ethernet frame format does not include any kind of priority marking. There is no field in an Ethernet header that would allow you to indicate that this or that frame requires a preferential treatment. VLAN tags, on the other hand, have a 3-bit CoS field that allows you to indicate the priority of the tagged frame. Should a PC need to send a frame that needs to be explicitly marked as more important than others, it can be done by inserting a VLAN tag into this frame and setting the CoS field to a non-zero value (with 3 bits, the maximum CoS value is 7).
Another reason for a computer to send and receive tagged frames would be when the computer itself would be intentionally placed into multiple VLANs. For example, the router-on-a-stick performing inter-VLAN routing is not a concept just for dedicated hardware routers. For example, any computer running Linux can be used in place of a Cisco router to perform inter-VLAN routing. Just like on a Cisco router, you would configure the Linux with subinterfaces for each VLAN it should be able to route from and to, assign IP addresses, and voila - you have a cheap and powerful inter-VLAN router. Yet another reason for receiving and sending tagged frames on a computer would be virtualization: You could have a server that runs multiple virtual operating systems in VirtualBox, VMWare, Xen or some other virtualization solution, and you want each of these virtual PCs to have a "separate" network card so that they can not talk to each other when they communicate with the outside world. You would do this again by creating subinterfaces on the physical machine, and bridging the individual virtual PCs with unique subinterfaces so that each virtual PC gets its own subinterface onto which it is bridged. As a result, the communication of individual virtual PCs would be tagged on the physical link depending on what virtual machine was speaking.
So, while not exactly a typical situation for an ordinary office PC, it is nonetheless quite normal to see a computer being connected to a trunk port. This, however, is always done with the prior knowledge that the computer will indeed need to talk to several VLANs simultaneously and directly. Otherwise there's no need for that.
Regarding the native VLAN on trunks - well, this is a neverending debate. I wish the native VLAN was never invented but well, it's here so we have to fight with it. Often, it is explained as "the VLAN that will save you if you happen to connect a normal PC to a trunk", and you have asked quite correctly - why on Earth would I want to connect a normal PC to a trunk, if not for reasons stated above? And you would be perfectly right - you wouldn't. The reason for native VLANs is different. If you try to study the IEEE 802.1Q standard you will learn that it does not recognize the terms access port and trunk port. It has no distinction for these port types. Instead, 802.1Q considers each port to be possibly associated with multiple VLANs at once. One of these VLANs is called the Primary VLAN, its number (ID) is called the Primary VLAN ID (PVID), and this VLAN is considered to be the one that is always associated with the port and thus does not need to use tags. Any other VLAN that is in addition associated with the port obviously has to use tags to be distinguishable. From this viewpoint, a port that is associated just with its PVID would be what Cisco calls an access port, and a port that is associated with VLAN IDs other than just its PVID would be what Cisco calls a trunk port, with the PVID being the trunk's native VLAN ID.
So in the way IEEE defines VLANs and their usage, PVID (= native VLAN ID) is a property of each port, so any implementation that claims compatibility with 802.1Q has to implement the PVID. Cisco decided to have a twist on the understanding of VLANs, and came up with access ports (i.e. ports associated just with their PVID and no other VLAN ID) and trunk ports (i.e. ports associated with many VLAN IDs including PVID), and so each trunk port must have its PVID - and that is what we call native VLAN and why we need to at least support it - although we do not need to make use of the native VLAN on trunks.
Quite convoluted.
Best regards,
Peter

Please confirm.. Whether Cisco CCM support SIP trunking !

Hi,
We r in the process of VoIP implementation nd during that phase came to know that Cisco doesn't support SIP trunking at gateway level.. Can anyone suggest me whether it's true or not??? Cisco says its support SIP then what is the meaning of that?? I m really confused with the twisting of words in different docs...

Thanks to all of you for your reply...When you say SIP trunking is supported then whether I can use SIP at gateway level towards making outward call.. Sorry if I am getting confused myself but when one says SIP trunking is supported what's the mean of that?? If I have a trunk line(PSTN) and a T1 line acting as a gateway, is it possible to design a network only depending upon SIP protocol using CISCO UNIFIED Components and call manager 5.0 ??
Sorry once again if my question sound something ..... but it just to clear my confusion too as I am listening so many stuff from different people when it comes to SIP overall implementation for a Enterprise network using CISCO Unified communication equipments...
With best regards,
Mani

Site to Site calling issue - Cisco 2911 Dial Peer Configuration

My customer dials from remote site to main site to their main site number, the call by-passes their auto attendant and goes directly to any random available party.
At first fingers were pointing to the their PBX, however we noticed one of their sites that wasn't managed by our company did not have the issue. We cut that site over to our service and the issue started right up. I believe it is possibly due to the way the dial peers are configured and how the calls route into the PBX. Unfortunately I do not understand much about them and curious to know if anyone has any history on a issue similiar to this or any input whatsoever?
Cisco equipment/Dialpeer config below ........
co IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2) - Cisco CISCO2911/K9
dial-peer voice 100 voip
description --- VoIP Dial-Peer ---
translation-profile outgoing 7digit
huntstop
preference 1
service session
destination-pattern .T
progress_ind setup enable 3
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 99
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
fax nsf 000000
ip qos dscp af41 signaling
no vad
dial-peer voice 150 voip
permission none
description 900 block
huntstop
destination-pattern 1900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 151 voip
permission none
description 900 block
huntstop
destination-pattern 900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 101 pots
description --- INCOMING Calls from PBX ---
incoming called-number .T
direct-inward-dial
dial-peer voice 1001 pots
description --- Calls to the PBX ---
preference 3
destination-pattern .T
port 0/0/1:23
forward-digits 4
Here is some ISDN debug information
BAD CALL
Protocol Profile = Networking Extensions
0xA11C0201420201008014484152545F20484F54454C535F434C4159544F4E
Component = Invoke component
Invoke Id = 66
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, ''6551''
Plan:Unknown, Type:Unknown
Aug 19 16:10:47.242 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: RX <- CONNECT pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x2B15
GOOD CALL
Protocol Profile = Networking Extensions
0xA116020144020100800E475245454E204D4F554E5441494E
Component = Invoke component
Invoke Id = 68
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, 'XXXX''
Plan:Unknown, Type:Unknown
Aug 19 16:15:07.999 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB17
Channel ID i = 0xA98381
Exclusive, Channel 1

I done the configration via CCA and the running conf i can see two voip dial peer. this is the site where all trunk line roured. Customer from other site2 needs to call outside by taking line from site1.
dial-peer voice 2100 voip
corlist incoming call-internal
description **CCA*INTERSITE inbound call to SITE 1
translation-profile incoming multisiteInbound
incoming called-number 82...
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
dial-peer voice 2101 voip
corlist incoming call-internal
description **CCA*INTERSITE outbound calls to SITE2
translation-profile outgoing multisiteOutbound
destination-pattern 81...
session target ipv4:192.168.50.1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
no dial-peer outbound status-check pots

Using a wildcard in the contact number for a messenging service.

I am a volunteer firefighter. Our fire company uses a messenging service to send MMS notifications when there is a call. All of the messages come through as 1(410)000-nnn where nnn = the sequential message number. E.g. let's say I get a text at 06:00, the sender is 1(410)000-236 Then, I get another call at 11:00 and the sender is 1(410)000-237, etc.
I would like to add the number to my contacts using a wildcard so that any text from, say, 1(410)000-### is listed as "Fire Company". Is there any way to accomplish this?

It's the same story with phone numbers. E.g. let's say I am a Geico customer and because they are so big, they have their own trunk line. Thus, any number from 123-456-nnnn is going to be Geico. I don't really care to program 10,000± numbers into my phone. I just want to know that it's Geico calling me. Programming 123-456-#### or something for "Geico Insurance" would be nice.

I CAN'T GET RID OF QUICK TIME 6.5 ON A WINDOWS XP/HOME COMPUTER

I have a Mac at work with a T-4 trunk line so I sometimes use QT. But, since my genealogy program only works with Windows, I have a Win XP/Home at home using dial-up. As soon as I bought this computer, I downloaded QT 6.5 not realizing that I couldn't use it. I understand there are security flaws and I want to remove it.
Under Add or Remove Programs, there is no mention of Quick Time or Apple. I downloaded Windows Install Clean Up and neither QT nor Apple is listed. When I go to Start>All Programs>Quick Time, there are 4 options: 1). QT Player, 2). QT Read Me, 3). QT Updater and 4). Picture Viewer. The only place I can find Quick Time is in my Control Panel between Program Updates and Regional Language Options. There is the QT icon.
I can't believe that Apple would not include an un-installer. Rather than spend 4+ hours downloading a newer update (and not sure if it would un-install the older version) and not being able to use it anyway, I just want to get QT 6.5 off my Windows computer.
Thank you
Sony Windows XP P-4 HT 3GHz

Hello thehordepvp, type %appdata% in Run dialog box and delete the Mix Dj folder if exist.
then try the [https://addons.mozilla.org/en-us/firefox/addon/searchreset/ SearchReset add-on], the add-on backs up and then resets your search preferences and home page to their default values, and then uninstalls itself.
thank you

If anyone can shed some light on this I would appreciate it.

For starters my laptop is getting almost 3 years old and the hard drive was filling up. I decided to reformat and move my itunes and subsequent music folders to an external hard drive. I left it on my external and then I left for a 2.5 month deployment with the US Navy. Upon coming back my external no longer had anything saved on it and i cannot locate my music folder from itunes anywhere on my computer. Is there anyway i can retrive the music that i have been legaly purchasing off itunes over the past 3 years. There are still purchase receipts on my itunes account. Anyone have ideas... or am I s.o.l.
thank you in advance for any information.

A CTI Port is a virtual port. It is analogous to a trunk line in a traditional ACD or PBX setting. i.e The CTI Port allows access to the Post Routing capabilities of the IP-IVR.
A CTI Route Point is a virtual device that can receive multiple simultaneous calls for the purpose of application-controlled redirection. Once a CTI Route Point has been created, lines (directory numbers) can be added and configured. Applications that use CTI route points include the Cisco IP Interactive Voice Response (IP-IVR) system and Cisco JTAPI with ICM. Calls are directed to CTI Route Points, which subsequently route the call to an available CTI port, the front end of the application. A CTI Route Point is the number a caller dials to access the application. A CTI Route Point must be configured for each application.

How to connect keysystem/pbx

hi guys i m new to voip and just started my cvoice, i have a confusion plz help me out
1) if i want to connect a pbx/keysystem with a CO switch over analog lines then wat type of interfaces will be used at both end ?? i m asking this coz in cvoice book its written that if u want to connect keysystem to CO switch thru analog trunk lines the signalling used will be winkstart and all does that mean that the cards will also be E&m trunk cards ??? am i right in this ??
2) wat are the possible ways of connecting pbx/keysystem with a CO switch ( i mean physical interfaces and the signalling )
plz tell me in a bit detail coz i m very much confused
thanks in advance

hi sir i m needing this for cvoice, see i have completed my first chapter but still not able to understand the analog connections b/w pbx and co switch, i m narrowing my question, can u plz tell me that if i want to connect a pbx with a co switch what are the possible ways of doing it ?? like wat interfaces will be used at both end ? wat signalling ? somewhere its written that groundstart signalling is used and somehere its written that i have to use E&M interfaces so wat shall i conclude, i really dont have any other book besides cvoice and voip fundamentals, if u could plz refer me a site or answer me anyhow i will really be grateful
thanks in advance

DSL is Brutally Slow

My Verizon DSL connection has become very slow in the last several months. FIOS is not available here so DSL for Verizon is my only option except for switching to my cable provider for internet service which I am about to do if my DSL speed cannot be consistantly improved.
I have spent hours on the phone with customer service (they are always very helpful and patient) as well as speaking with Tier 2 technicians. I have recievled a new modem and my trunk line connection has been changed several times at wherever that takes place (somewhre in Maryland, I think). Mys DSL is still slow and inconstistant. Speedtest.net sometimes shows good speeds but most of the time my DL speed is 1.0 or less (a lot less msot of the time) and my UL is okay at 0.4 or 0.5.
I have a sales job and work from home 90% of the time and my wife works from home about 20% of the time. We need good speeds or our company VPN's become unusable.
I do not want to change ISP's but I'm at the end of my rope with my DSLservice. Please help if you can.
Thank you.

Hi Stephen306,
Please go to your profile page for the forum by clicking on your name, and look at the top of the middle column where you will find an area titled "My Private Support Cases". There you will find a link to the private board where you and the agent may exchange information. This should be checked on a frequent basis as the agent may be waiting for information from you before they can proceed with any actions. Please keep all correspondence regarding your issue in the private support portal.

Phone out of order!

help, both my droid and my land phone out of order at same time.

Who is the landline through?
I do wonder if someone cut the trunk line or fiber optic line feeding both the tower and the branch your landline is fed through.

To configure VOIP for this 2FXO card

We are having one Cisco 3640 Router with One Serial Port and 2FXS(2nos) Voice card. We have configured the 2FXO card to the best of our knowledge. These 2 ports are configured in EPABX. Now we are able to get the dialtone but we are unable to make calls to US . Kindly let me know the commands to configure VOIP for this 2FXO card.
Thanks in advance
Regards
Sapna shah

We are having one Cisco 3640 Router with One Serial Port and 2FXS(2nos) Voice card.We have configured voice card FXO but now due to some unknown reason we are not able to call the US office. Kindly help us .Here is the configuration:
Can u let us know what errors we have made in configuruing steps .
Here are HYD-RTR#sh voice port
Foreign Exchange Station 1/0/0
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is Direct line for IOC
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Foreign Exchange Station 1/0/1
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is Connected to Alcatel PABX thru trunk line
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 1 s
Region Tone is set for US
Analog Info Follows:
Currently processing unknown
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Foreign Exchange Station 1/1/0
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is Second Direct Line For IOC
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Foreign Exchange Station 1/1/1
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is Second Line Connected to Alcatel PABX thru trunk line
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing unknown
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Foreign Exchange Office 2/0/0
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is plar
Connection Number is 100
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is groundStart
Number Of Rings is set to 1
Supervisory Disconnect active
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Type is pulse
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Foreign Exchange Office 2/0/1
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is groundStart
Number Of Rings is set to 1
Supervisory Disconnect active
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Please enlighten us why we are not able to make calls .IS there any fault in the configurations made by us !
Thanks in advance

SPA3102

I have some questions as below listed:
1. Is SPA3102 can handled more than one call session at moment?
2. How can i share the VoIP account in the LAN for other users?
3. How many IP phones can connect to one SPA3102 for concurrent call sessions? (we have only one VoIP carrier account from lowratevoip.com)
4. please let me have a solution for multiple users and concurrent calls in the LAN with one VoIP carrier account?

1. Is SPA3102 can handled more than one call session at moment?
the SPA3102 can handle 2 lines for line 1 , 1 for main call and 1 for call waiting.
from the analog phone, you can also setup a dial plan for dialing out to 4 other gateways. the PSTN line has a 1 gateway call.
2. How can i share the VoIP account in the LAN for other users?
if you have for that one ATA, you can set those others to make call without regsiter to yes and answer call to yes, and regsiter to no.
3. How many IP phones can connect to one SPA3102 for concurrent call sessions? (we have only one VoIP carrier account from lowratevoip.com)
this one is not possible. the SPA3102 cannot be used as a voip gateway or PBX
4. please let me have a solution for multiple users and concurrent calls in the LAN
with one VoIP carrier account?
consider the SPA9000 PBX system that can can have mutiple sessions depending on the trunk line provided by the VOIP provider for you.

SF 300 Serires switch not participating in spanning tree?

I just purchased an SF300-24 managed switch and I am running it in layer3 mode. I am testing it out right now and have it connected to two 2950 switches. The SF300 is connected to each 2950 with a four port etherchannel running LACP. When looking at spanning tree all three switches are configured the same when it comes to hello, forward, max age and all three are in RSTP mode. I adjusted the priorities so that the SF300 would be the root but that is not happening.
I only have one VLAN as of right now set up and connectivity between the three switches is fine. The only problem seems to be that the two 2950 switches are the only two switches involved in the determination of the root bridge. Additionally it was the same way before I configured the etherchannel and had the switches connected over single trunk lines.
I would appreciate if someone can expain to me why this is?
Thanks in advance.

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Thanks for your help but know I still cannot get the three devices to talk MST either,it is getting frustrating. If i add a redundant link and directly connect the two 2950's they immediately talk and configure MST. But when I remove that link no info is passed and both 2950's think they are the root even though the SF 300 priority is 0 on all three MST instances. On the SF300 I have the following settings:
Spanning tree: enabled
STP Operation Mode: Multiple STP
BPDU Handling: Flooding
Path Cost: Long
Region name: test
Revision: 1
Max Hops: 20
Max-age: 20
Hello Time: 2
Forward Delay: 15
MST instance 1 Vlan 100
Bridge Priority 0
Designated Root Bridge: Self
Root port: 0
Root path cost: 0
MST instance 2 Vlan 2-5
Bridge Priority 0
Designated Root Bridge: Self
Root port: 0
Root path cost: 0
MST instance 0 all vlans not in instance 1 and 2
Bridge Priority 0
Designated Root Bridge: Self
Root port: 0
Root path cost: 0
For MST interface Settings (both LAGs/instances are thesame)
Int Priority: 128
Path Cost: 20000
Port State: Boundary
Mode: RSTP
Type: Boundary
Designated port ID: 128
Designated Cost: 0
Remain Hops: 20
Forward Transitions: 1
The 2950 switches: (The only difference on the other switch is that the priority is 8192, and the MACs of course)
MST00 is executing the mstp compatible Spanning Treeprotocol
Bridge Identifierhas priority 4096, sysid 0, address 000b.460e.e040
Configured hello time 2, max age 20, forward delay 15
Current root haspriority 0, address 6c50.4dcb.334b
Root port is 65 (Port-channel1), cost of root path is 50000
Topology change flag not set, detected flag not set
Number of topology changes 7 last change occurred 00:18:54 ago
          from Port-channel1
Times: hold 1, topology change 35, notification 2
          hello 2, max age 20, forward delay 15
Timers: hello 0, topology change 0, notification 0
Port 65 (Port-channel1) of MST00 is root forwarding
   Port path cost 50000, Port priority 128, Port Identifier 128.65.
   Designated roothas priority 0, address 6c50.4dcb.334b
   Designatedbridge has priority 0, address 6c50.4dcb.334b
   Designated port id is 128.1000, designated path cost 0
   Timers: message age 4, forward delay 0, hold 0
   Number of transitions to forwarding state: 1
   Link type ispoint-to-point by default, Boundary RSTP
   BPDU: sent 571,received 568
MST01 is executingthe mstp compatible Spanning Tree protocol
Bridge Identifierhas priority 4096, sysid 1, address 000b.460e.e040
Configured hello time 2, max age 20, forward delay 15
We are the root of the spanning tree
Topology change flag not set, detected flag not set
Number of topology changes 9 last change occurred 00:18:55 ago
          from Port-channel1
Times: hold 1, topology change 35, notification 2
          hello 2, max age 20, forward delay 15
Timers: hello 0, topology change 0, notification 0
Port 65 (Port-channel1) of MST01 is boundary forwarding
   Port path cost 50000, Port priority 128, Port Identifier 128.65.
   Designated root has priority 4097, address 000b.460e.e040
   Designated bridge has priority 4097, address 000b.460e.e040
   Designated port id is 128.65, designated path cost 0
   Timers: message age 0, forward delay 0, hold 0
   Number of transitions to forwarding state: 1
   Link type ispoint-to-point by default, Boundary RSTP
   BPDU: sent 598,received 0
MST02 is executingthe mstp compatible Spanning Tree protocol
Bridge Identifierhas priority 4096, sysid 2, address 000b.460e.e040
Configured hello time 2, max age 20, forward delay 15
We are the root of the spanning tree
Topology change flag not set, detected flag not set
Number of topology changes 9 last change occurred 00:19:50 ago
          from Port-channel1
Times: hold 1, topology change 35, notification 2
          hello 2, max age 20, forward delay 15
Timers: hello 0, topology change 0, notification 0
Port 65 (Port-channel1) of MST02 is boundary forwarding
   Port path cost 50000, Port priority 128, Port Identifier 128.65.
   Designated root has priority 4098, address 000b.460e.e040
   Designated bridge has priority 4098, address 000b.460e.e040
   Designated port id is 128.65, designated path cost 0
   Timers: message age 0, forward delay 0, hold 0
   Number of transitions to forwarding state: 1
   Link type ispoint-to-point by default, Boundary RSTP
   BPDU: sent 611,received 0
I notice that on MST01 and 02 they are not receiving BPDU’s,but I am not sure why or if that is the problem. It appears that the SF 300 is not sending BPDU packets for MST01 and 02, but is sending them for MST00. I also attached a capture. I captured the VLAN info for VLAN 100 which is in MST1. on the SF300, it appears that the SF 300 is recieving STP traffic but not generating any.

Trunk lines

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