Connecting to a PBX

Hi,
Is there a way to connect a Mac to a PBX system.
We'd like to use a mac for public announcements via the paging function of our office phone system.
Thanks,
b.

As per me, you need fxo card as well at cme router to achieve both side calling. Actually, FXS generate the dial tone so connectivity should like below..
CME (FXS)------PBX (FXO) (Call from PBX to CME)
CME (FXO)------PBX (FXS) (Call from CME to PBX)
May someone else confirm it but I have done this kind of integration between Vega Gateways.
Suresh

Similar Messages

SPA 8000 Connection with IP PBX

I have SPA 8000 8 Port Analog Telephone Adapter , i want connect this with IP PBX and use Analog telephone in my home (Assign extension number for each room)
Can you please explain to me how to configure and get extension number from IP PBX to ATA adaptor
Thanks in advance
Shamsheer

Hello,
as far as I know, FAX and VoIP aren't the best friends - those technologies are not designed to work together smoothly ...
You may try following tuning :
Set your fax machine to 9600 (not 14400),
Make sure you only use G711a (Europe) or G711u (US) codec (fax will not work with others).
Activate T38 FAX support if your device FW offers such possibility.

Connecting Asterisk SIP PBX to Skype

Hi,
I'm a new Skype Business customer, I have a few user registered and I wish to connect my company Asterisk PBX to Skype. In details, I bought a Skype-In number, I would like that calls to such number will arrive to one of my PBX extension. Is there some way to do this?
Thank you in advance.

Could someone help me?
I'm having problems with my Elastix (Asterisk) / Skype Connect Configuration. I always get "All circuits are busy now..." message. My configuration is:
Trunk Name: skype_in
type=friend
username=xxxxxxx
fromdomain=sip.skype.com
fromuser=xxxxxxx
realm=sip.skype.com
host=sip.skype.com
dtmfmode=rfc2833
secret=password
nat=no
insecure=invite
qualify=yes
disallow=all
allow=alaw&ulaw
amaflags=default
trustrpid=no
sendrpid=yes
context=skype_in
Register String: [email protected]/xxxxxx From asterisk I get:
elastix*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
skype_in/99051000147265 204.9.161.164 5060 OK (177 ms)
and
sip show registry
sip.skype.com:5060 990510001472 105 Registered Thu, 06 Oct 2011 20:46:30
Every looks good but when I try to make a call... I get the busy message:
SIP Debugging Enabled for IP: 10.168.16.115:5060
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-
Max-Forwards: 70
Contact: 6001>
To: 20551141256555>
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 255
v=0
o=Zoiper_user 0 0 IN IP4 10.168.16.115
s=Zoiper_session
c=IN IP4 10.168.16.115
t=0 0
m=audio 8000 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 10.168.16.115 : 5060 (no NAT)
Using INVITE request as basis request - MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
<--- Reliably Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2f3cba10
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60db2ce3"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.' in 32000 ms (Method: INVITE)
Found user '6001'
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-
Max-Forwards: 70
To: 20551141256555>;tag=as2f3cba10
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-
Max-Forwards: 70
Contact: 6001>
To: 20551141256555>
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest username="6001",realm="asterisk",nonce="60db2ce3",uri="sip:[email protected];transport=UDP",response="ed79aee161a6cc8a7e520cd011afe0bb",algorithm=MD5
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 255
v=0
o=Zoiper_user 0 0 IN IP4 10.168.16.115
s=Zoiper_session
c=IN IP4 10.168.16.115
t=0 0
m=audio 8000 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 10.168.16.115 : 5060 (NAT)
Using INVITE request as basis request - MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
Found user '6001'
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.168.16.115:8000
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.168.16.115:8000
Looking for 20551141256555 in from-internal (domain 10.168.16.3)
list_route: hop: 6001>
<--- Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: 20551141256555>
Content-Length: 0
<------------>
Audio is at 10.168.16.3 port 17408
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2650cb3c
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: 20551141256555>
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 14425 14425 IN IP4 10.168.16.3
s=session
c=IN IP4 10.168.16.3
t=0 0
m=audio 17408 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
<------------->
Scheduling destruction of SIP dialog 'MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 484 Address incomplete
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2650cb3c
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-
Max-Forwards: 70
To: 20551141256555>;tag=as2650cb3c
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
elastix*CLI> sip set debug off
SIP Debugging Disabled
Could someone help me?
Best regards
Eduardo6001>20551141256555>20551141256555>6001>20551141256555>20551141256555>6001>20551141256555>20551141256555>6001>6001>6001>20551141256555>6001>6001>20551141256555>20551141256555>6001>6001>20551141256555>6001>

How to use more than VWIC- 1MFT E1 to connect to a PBX

I have a 2691 Router with a VWIC-1MFT module being used to connent to my Siemens HiCom PBX, (running QSIG). Right now this can support a maximum of 30 channels between my router and the PBX. To add to this i am using VOIP-(extra info).
However if i require to add another 30 channels between my Router and the PBX, can i use another VWIC-1MFT E1 module.
And how can i configure the routing of calls to using both E1 controllers?
Thank You in advance.
Mohannad

According to below link. It do support two MFT-E1. However, you have to install additional AIM-VOICE module to support the additional voice channel.
http://www.cisco.com/en/US/products/hw/modules/ps3115/products_data_sheet09186a0080194e20.html
http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800b6e0f.shtml
Hope this helps.

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

FAX through PBX connected to Router

I do implementation in which PBXs are connected to Cisco routers using E&M ports at several sites. A FAX machine is connected to an extension on the PBX needs to communicate with another FAX machine connected to the PBX at another site. The settings of the E&M ports are here:
voice-port 0/0/0
trunk-group e&m
type 5
dial-type pulse
timing wait-wink 800
description E&M-PBX
A voip dial-peers are configured for the destination router.
dial-peer voice 95 voip
destination-pattern 95..
session target ipv4:192.168.125.2
I tried some chnages but it did not work. I will try to configure the codec to G711 and no VAD and no EC on all ports and test it. But it will cost bandwidth.
Do anyone have another solution?

Thank you Paolo
I configured fax protocol pass-through under the service level and it worked. I wait the feedback from the customer if any problems arise.
voice service voip
fax protocol pass-through g711ulaw

Route E1 connect PBX problem

Dear All:
       I have got a big problem of 2921 use E1 connect NEC 8100 PBX for VOIP.
      Topology:
      C2921-----E1--------NEC------Analog Phone
                                  |
                                  |
                                  -----------Digit Phone
1. When a call form Analog Phone to other site using E1 cross NEC and 2921, other site phone can ringing, but the Analog phone can't hearing ring back.
2. When other site user off hook phone, he can hearing the Analog Phone speaking, but local site can't hearing other site speaking.
3. When Digit Phone call other site, every thing is OK.
4. When Analog Phone call local Digit Phone, cross PBX and not using E1, it is OK(Can hearing ring back and Digit Phone speaking.)
5. When Analog Phone call local Digit Phone, cross PBX and E1, it is same as call other site, can't hearing ring back and Digit Phone speaking.
6. When Other site Phone call Analog Phone, every thing is OK.
======
this is E1 configuraiton:
controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-17
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
voice-port 0/0/0:15
echo-cancel coverage 32
cptone CN
dial-peer voice 9001 pots
destination-pattern 111....
port 0/0/0:15
do you have got any idea about this?!
Thanks~!

The problem is resloved. I add "isdn overlap receiving" on interface s0/0/0:15
The PBX looks like didn't support bloc send number.

Remote Connection to PBX

Theoretically, you could connect the 'trunk' of a 3CX install to an extension of the Panasonic, but it's not very elegant.
The simplest way would be to create a VPN between the sites, and follow Panasonic's networking guide.

We are setting up a remote office that is in a separate country to our main office. In the main office we have a Panasonic KX-TDE100 for our PBX. What we are looking for is either some software that can run on a server and connect to the PBX, or some hardware that could do the same thing?
The one solution that I have sort of found is the soft PBX from 3CX. I read in a post of someone using it to remotely extend their Panasonic PBX, but I haven't been able to find any documentation on it so I don't know if it is possible...
All suggestions welcome! Thanks :)
This topic first appeared in the Spiceworks Community

NOT RECEIVING THE COMPLETE DIALED STRING FROM AN ISDN TRUNK PBX - AS5850

Hi
The problem i'm facing is the next:
I have a Cisco AS5850 with an Euro ISDN trunk connected to a PBX. The pbx send out the calls in order the A5850 terminate these calls in a remote gateway. I receive only a maximum of 12 digits of the complete string dialed in a phone hooked up in the PBX side.
Is it normal to receive only a max of 12 digits or there is something to do to receive as much digits as those come from the PBX??
My dial-peer looks like this:
dial-peer voice 13 pots
incoming called-number 0T
destination-pattern 0T
direct-inward-dial
port 2/7:D
dial-peer voice 14 voip
destination-pattern 0T
session target ipv4:209.210.174.17
tech-prefix 1051
fax rate 9600
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
and the output from the debug dialpeer is this:
1w4d: destination pattn: 1234567891 expanded string: 1234567891
1w4d: MatchNextPeer: Peer 7 matched
1w4d: MatchNextPeer: Peer 8 matched
1w4d: dpMatchPeersMoreArg: Result=0 after MATCH_ORIGINATE
Thanks in advance for your help!
Oscar

Do a debug isdn q931 and see what comes in from the PBX. Post it here so we can see. They could be sending the digits using overlap sending , so the intial setup may not have all the digits that you want or need.
Also, your VOIP and POTS dial peers have the same destination pattern. There is nothing stopping an inwards ISDN call from being hairpinned back out the PRI if it has a leading 0. You might want to make your dial plan a bit more specific.

Is it possible to dial an extension on a remote PBX system?

One thing I've never been able to figure out how to do in Lync is to dial an address book contact that has an extension configured-e.g. where I need to dial a main number, and then enter a 4 digit extension once connected. Most PBX systems let you
dial something like 19055551234,,,1234 (where , is a 2s pause), or even 19055551234^1234 (where ^ is wait for connect). In Lync though, there doesn't appear to be any way to interpret any of these characters. I've tried having a dial plan normalize
these to e.164 (tel:+19055551234;ext=1234), but this gets ignored by the gateway (in this case an Audiocodes M1000). Should that be supported? Is there a way to get Lync (or the GW) to dial DTMF after a connection?

In Lync, not so easy. With AudioCodes, you might have an answer! If you want to send digits that are present in the SIP To header such as an extension (and you can ensure they are with Lync normalization) after the phone goes off
hook, you can enable "Enable Digit Delivery to Tel". This is under "Configuration -> VoIP -> SIP Definitions -> Advanced Parameters."
From there it's about manipulations, you can insert pauses by using the letter "p". Check out the below from the AudioCodes 6.6 manual, or search for Digit Delivery in whatever version of the manual that matches your firmware. Or, just open a
ticket with AudioCodes support and tell them you need help with Digit Delivery. :)
Enables the Digit Delivery
feature, which sends DTMF digits of the called number to the device's port
(analog)/B-channel (digital) (phone line) after the call is answered (i.e.,
line is off-hooked for FXS, or seized for FXO) for IP-to-Tel calls. <o:p></o:p>
§ [0]
Disable
(default). <o:p></o:p>
§ [1]
Enable =
Enable Digit Delivery feature for the FXO/FXS device. <o:p></o:p>
<o:p></o:p>
For digital interfaces: If the
called number in IP-to-Tel call includes the characters 'w' or 'p', the device
places a call with the first part of the called number (before 'w' or 'p') and
plays DTMF digits after the call is answered. If the character 'w' is used, the
device waits for detection of a dial tone before it starts playing DTMF digits.
For example, if the called number is '1007766p100', the device places a call
with 1007766 as the destination number, then after the call is answered it
waits 1.5 seconds ('p') and plays the rest of the number (100) as DTMF digits.
<o:p></o:p>
Additional examples:
1664wpp102, 66644ppp503, and 7774w100pp200. <o:p></o:p>
Notes: <o:p></o:p>
§ For this parameter to take effect, a device reset is
required. <o:p></o:p>
§ For analog interfaces: The called number can include
characters 'p' (1.5 seconds pause) and 'd' (detection of dial tone). If
character 'd' is used, it must be the first 'digit' in the called number. The
character 'p' can be used several times. For example (for FXS/FXO interfaces),
the called number can be as follows: d1005, dpp699, p9p300. To add the 'd' and
'p' digits, use the usual number <o:p></o:p>
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications

ISDN(BRI) to PBX

Hi, I have a small problem with an NM-8B-S/T connecting to a PBX. (Ericson with Basic Rate Card in it).
All ISDN Layers are active, when I dial into the router, I get the following in the debug. (see attached txt file.)
Anyone have any ideas on this one? What does this imply?
*Mar 6 1993 10:09:38.535 UTC: ISDN BR2/0 **ERROR**: accept_incoming_csm_call: modem problem Requested circuit/channel not available(0x2C): b channel 0, call id 0x175
I did an ISDN test call from the BRI2/0 interface and the call went out successfully, Dial in does as per the below and never succeeds.

I suggest you go through the ISDN troubleshoot flowchart. It can helps.
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a0080094bb8.shtml
Otherwise, did you confirm there is no configuration issue at PBX side ?
Referring to the "Dial commands reference", marked bad may be the firmware problem of the modem, try "no modem bad" and upgrade the modem firmware.
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_reference_chapter09186a00801a7f1e.html#wp1115576

Interconnect 2 NON IP PBX's over MPLS

I have 2 Mitel PBX's that currently communicate over channels 19-24 of a PtPTt1. The Routers are configured to take 19-24 and point them out another T1 interface on the router where that is directly connected to the PBX T1 interface. these 2 MItel PBX's are not IP.
So we are now going to MPLS and once this is done, I lose the physical PtP circuit and thus lose that channellization. So the question is How can I configure VoIP router to router and interconnect my 2 PBX's?
Thanks!
Bryan

If the two PBX's use "proprietary" signaling you could set two gateways linked via TCCS. Here is the doc, it's old but still valid
http://www.cisco.com/en/US/docs/ios/12_1t/12_1t3/feature/guide/dt_tccs.html

How can I connect the phone network to LYNC Server 2013?

Greetings,
We have a working implementation of Lync Server 2013 (pc to pc), our objective is to connect lync to our phone central, receive and send calls from lync clients (pc, phone, etc), basically enterprise voice services.
I would like to know the following:
Hardware that I need for this beside the server.
Extra configurations that must be done.
If you could recommend me the hardware needed it would be helpful.

You'll need a gateway typically. You can find a list of qualified IP gateways here: http://technet.microsoft.com/en-us/office/dn788945.aspx
I typically lean towards the AudioCodes Mediant line, though many here love Sonus as well. You'll need to pick a method to connect to your PBX before you begin, such as via a T1/E1 trunk, FXO ports, or a SIP trunk depending on what your PBX will
support.
You'll need to configure the PBX to route calls through the T1 towards Lync and vice versa. You'll also need to configure the gateway itself, so I'd go with a consultancy with experience here or purchase remote implementation support from the hardware
manufacturer.
On the Lync side, you'll need to configure dial plans, voice policies, usages, and routes:
http://technet.microsoft.com/en-us/library/gg398272.aspx
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

PBX interoperatibility with router

Dear ALl
I have a confusion regarding the PBX integrations with Router.
Now if i am integrating my 2801 router with any PBX trunk ports (by trunk ports i mean the ports used to connect with PSTN with normal RJ-11 connector) , do i really have to worry about the compatibility between router and PBX.
The reason is that if the PBX can terminate the lines from the PSTN (RJ-11) than it should work with my router too .It is with e1 when we should worry about compatibility .Am i correct.
So what ports do i need on the router side?FXS.i thnok so as when we connect pstn with PBX than PSTN provides the tone ,so now my router should provide the tone to PBX trunks.
My scenario is that eight lines from PSTN are terminated in the norstar PBX and as i am proposing the IPT system the lines will now be terminated into the router and than the router should forward the call to the PBX if it is for any analog phone .
So my real question is do i really need to worry about the PBX interoperability when i am dealing with analog trunks.
Backbone

Hi,
the short answer is if even FXO is barely acceptable to connect to the PSTN (you should really get at least ISDN BRI), FXS/FXO connnection to the PBX will be certainly too limiting. Many problems of double dialing, potential stuck ports, no passing of information, etc. etc. The very miniumum is E&M to the PBX but real good would be again, ISDN BRI or PRI.
If the PBX is missing the necessary cards and is not worth to buy them, the you should at some point look at replacing the PBX completely.
The answer to real question is yes, you should really worry about analog trunks if you want the integration be of a professional level.
Hope this helps, please rate post if it does!

Skype connect & VoIP Gateway

Hi,
Could you please any give me more information about VoIP gateway, and how to connect it with PBX and how to buy it and what is the price?
Thanks,
Mohamed ElDieb

Hello ToshibaJoe,
If your PBX doesn’t support SIP check with your manufacturer to see whether a SIP gateway, module or upgrade is available for your existing PBX, that would enable you to use Skype Connect/SIP.
Also, it is highly recommended that a Skype Connect certified device is used, to gaurantee all features and functions work. Following is a link to that lists all Skype Connect certified products:
http://www.tekvizionlabs.com/3rdpartyprograms/skype/skype_verified_products.php
Please let us know if you have further questions.
Thank you,
MariaA
Skype Enterprise Support

Connecting to a PBX

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