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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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Just want to get folks opinions.  Anyone using them?  Any feedback?
Trying to implement sparkboards in every new office and eliminate things like conference phones, polycoms, and all that legacy stuff.

Thanks.
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Hello, I work with asterisk servers and mainly soft SIP clients. Customers would like to have the possibility to hang up an incoming call. That's the first step...no big deal. But what if the call is a "call group" call i.e. a call to multiple devices at the same time and you want the command from the client not only to hang up his device while the others keep ringing, but to hang up all legs of the call - caller and all callees. Is that possible?
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There is a fellow it guy that keeps forwarding his line to my extension when he is "busy"coding or something else, he always forgets to disable it and I've had enough of it so I want to disable his ability to forward calls, or disable my number from accepting forwards from him only if possible or altogether from anyone, I don't want do disable the function globally just in his ext number forwarding or mine receiving from him/any. ( I have console admin access)
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Question -
company's IP phones/voice managed by 3rd party.  No PBX on-site.  
Is SIP Trunking still required for the calls or its just calling over the Internet?
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Hi,

 I am considering "converting cable operator provided phone service to VoIP phone service".
 Can you recommend a vendor and explain why you like them?

 I am aware that there are multiple players - RingCentral, Vonage, 8x8 ... etc.

Thank you for your input in advance.
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I don't much about the Avaya IP-Office 500.  I've attached some pics that may help.  The first pic is the blue cable that connects to the phone but is no longer providing service.  
During our rack equipment move, we may have re plugged in one cable on the IP Office in the wrong port.  I'm not sure if that matters as I'm not proficient in PBX.  I had one of the staff members mark cables based on what phone goes out if we unplug it.  I'm not sure if that's done yet.  I left for another meeting.  
Anyway, are the patch cables placement very fickle for the IP-Office?  We may have accidentally plugged in one cable to the wrong port when moving things and that caused the phone to no longer light up.   Sorry, I'm walking blind here.  I would greatly appreciate some feedback.
blue_cable_from_face_plate_to_phone.jpg
a_and_b_drops.jpg
IP-Office_Ports.jpg
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For the Current PBX system, the Telco delivered an Adtran, off the Adtran there is an amphenol cable that is punched down to a 66 block.  There is also a CAT5 off the adtran that is also punched down to a 66 block.  

I am installing a Cisco VoIP solution,  but I am not sure how to connect my Cisco voice gateway router with a PRI/T1 card to the Adtran.  I'm not sure if the telco will have to hand the circuit off to me another way, or if I can just connect to the adtran using the ethernet port.  I'm having a lot of trouble getting info from telco.    Could anyone lend me some insight on this and their experiences?   I do have some VOIP experience, but no experience with traditional pbx, 66 blocks, etc.  Thanks.
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we create a sip trunk, cisco phones can call to avaya, when a try to call a cisco show "INCOMPATIBLE" im not a expert on avaya, ideas  ?

log avaya -> cisco

274138126mS PRN: ++++ END OF TCP MONITOR CLIENT DUMP ++++
 274160750mS Sip: SIP Line (17): License, Valid 1, Available 15, Consumed 0
 274160750mS Sip: SIP Line (17): sip_trunk_config_items 0002c10c, voip.flags 00040949
 274160750mS Sip: SIPDialog f172d9f4 created, dialogs 1
 274160756mS Sip: 0a3c1e8c0003346c 17.210028.0 33506 SIPTrunk Endpoint(f172f28c) received CMSetup
 274160757mS Sip: 0a3c1e8c0003346c 17.210028.0 33506 SIPTrunk Endpoint(f172d9f4) SetLocalRTPAddress to 10.60.30.140:46754
 274160759mS SIP Call Tx: 17
                    INVITE sip:50528337@10.120.200.20 SIP/2.0
                    Via: SIP/2.0/TCP 10.60.30.140:5060;rport;branch=z9hG4bKfdfadfc4705f6d30aeaf1ab49c1310a3
                    From: "Karina Bolado" <sip:SIPDefault@10.120.200.20>;tag=b25a03e400a8ebb0
                    To: <sip:50528337@10.120.200.20>
                    Call-ID: 554091e18ddc18667746c51e49f9a926
                    CSeq: 740975927 INVITE
                    Contact: "Karina Bolado" <sip:SIPDefault@10.60.30.140:5060;transport=tcp>
                    Max-Forwards: 70
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
                    Supported: timer,100rel
                    P-Early-Media: supported
                    Min-SE: 90
                    Session-Expires: …
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Hi

I've been wrapping my head around this for quite some time - still accomplishing nothing but headache. Here's the setup

Lync 2013 mediation server
Lync PSTN Trunk points to Avaya Aura RED
Avaya Aura RED routes PSTN traffic to Avaya phone central

We have landline numbers in Lync, and cell phone number with a PSTN provider

Lync <-> avaya RED gateway <-> Avaya phone system

Also - we've setup so that calls to cell phone in PSTN, will also simultaneous call Lync number.

so;
1. user call my cell phone (tel: 99 99 99) from his IP-phone/cell phone
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I answer the call with either cell phone or lync - the ringing will stop on the other device

so far so good.

But if this process is repeated, but this time the call is initiated from a Lync client at the internal office

1. user call my cell phone (tel: 99 99 99) from enterprise voice enabled lync client
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I …
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Hello,
Need a question answered being asked of me and our Cisco CUCM 11.1 phone system. I need to know if Cisco CUCM 11.1 will integrate with NetSuite CRM via TAPI and/or CTI. If so, any third party apps needed or can it be done within CUCM?
Thanks in advance!
Steve
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Hi All

I'm in the process of planning out implementing site codes dial codes within my company's CUCM environment. We currently use 4 digit dialing across all sites, but this is no longer scalable with the amount of expansion we're experiencing.

I've set this up in a lab environment, and it works as expected. However, I'm stumbling with the modifying the Calling Party ID when dialing between sites. So when someone in Site A dials an extension in Site B, the Site A caller ID is prepended to include the Site A dial code.

I'm think I have to create multiple translation patterns, intersite partitions, and CSS's to do this; but this also seems messy, so I'm wondering if there's a better way to accomplish this?
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I want to change time of stages of phones that ring on incoming calls.
Documentation on Incoming call system options. The main two I'm primarily interested in there behavior are 22-01-04 and 22-01-09.

04 - Normal DIL Incoming Call No Answer Time

09 - DID to Trunk to Trunk No Answer Time
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Hello Everyone

I'm currently in the process of migrating our current PBX system away from asterisk to Freeswitch. I am using FusionPBX on Debian 8. I am using the freeswitch webapi to originate calls. I am at the stage where when I execute the command, it rings the call centre agents phone and the customer automatically without the agent manually dialling the number. I would like the ability to manually specify a caller id number for the outbound leg of the call. At the moment it is not sending any caller ID. I can manually specify a caller ID number in the extensions page, and it works statically, however we have a need for the caller ID to be dynamic.

http://X.X.X.X:8080/webapi/originate?{click_to_call=true,origination_caller_id_name='Click to Call',origination_caller_id_number=1000,instant_ringback=true,ringback=\'%(400,200,400,450);%(400,2200,400,450)\',presence_id=630@X.X.X.X,call_direction=outbound,sip_auto_answer=true,domain_uuid=52b92yy9-7fb7-52ae-9e9e0595058bcdaa,domain_name=X.X.X.X}user/630@X.X.X.X &transfer('SOME EXTERNAL NUMBER XML X.X.X.X')

What do i need to add to this web address to get it to send a custom caller ID number to the customer outbound?

Many thanks in advance.
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I'm running CUCM 9 and Unity connection 9.  All screen's on my Cisco IP phones go dim (black) at 5pm.  I know this has to be a global setting in CUCM, as all phones do this, but I can't figure out where to go to change this.  We have recently extended the hours the office is open, so I need to change this.  Does anyone know where this setting is?  Thanks!
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SIPp is a free SIP traffic generating tool for Linux.
http://sipp.sourceforge.net/

SIPp user manual says you can install SIPp under CYGWIN on windows. However I am not experienced  with compiling applications to run under Linux and need help getting SIPp up and running under CYGWIN on a windows10 machine.

I have successfully installed CGYWIN and included the following packages (all successfully)
gcc-core
gcc-g++
gcc
libncurses
make

After the CGYWIN install, I put C:/cygwin64/bin in the win10 systems’ environment variable PATH – so far all ok and CGYWIN seems to be working fine.

In addition, the SIPp install instructions state:

Warning
SIPp compiles under CYGWIN on Windows, provided that you installed IPv6 extension for CYGWIN (http://win6.jp/Cygwin/), as well as libncurses and (optionally OpenSSL and WinPcap). SCTP is not currently supported.


QUESTION 1 -  Do you know what this is???    IPv6 extension for CYGWIN http://win6.jp/Cygwin/ 
is it a CYGWIN package, and entire install version??
What/how do I need to do to check/install?

QUESTION 2 – Nothing happens when I try to run “autoreconf -ivf” ...but this might have to do with Question 1 not being addressed yet.

 /cygdrive/c/Backup/tools/SIPp/3.3
$ autoreconf -ivf
-bash: autoreconf: command not found


+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
FROM DOC


Installing SIPp
•      On Linux, SIPp is provided in the form of source code. You will need to…
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We are on Avaya IP Office 8.x (?) ....   yes we are planning on an upgrade to 10.? later this summer.  Everything had been fine until about 3 weeks ago when all of a sudden our voicemails stopped going to email.  We are not having any luck with our vendor figuring out the issue.  Have stopped and restarted voicemail services.  I don't understand a lot about Avaya IP Office as I was more familiar with Cisco Call Manager.    We have many other services that send emails through our SMTP server so I don't think that is the issue.  Can anyone give me some insight or point me in the right direction?

Thanks
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Hi

We have a 3CX voip server that is hosted in the cloud and we have a T49G phone we need to configure to work with the 3CX server.

Now, 3CX server doesn't officially support T49G phone but they say it should work as a normal SIP phone without any provisioning.

I have tried simple SIP config by putting in the extension number and its password but the SIP registration keeps failing.

The packet capture on the 3CX server is showing 407 proxy authentication error.

Can someone help me configure Yealink T49G phone on a 3CX Voip server?

Happy to provide packet capture or any other logs you may need.
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When you use Microsoft PSTN calling - can you use your old Cisco 7945 phones provided you load them with the SIP firmware instead of the SCCP?
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the system was setup with an auto attendant and 3 call groups for voice mail and after the auto attendant the caller would choose 1 , 2 or 3  and that would send you to the proper mail box and then you would hear that message. i can not figure out how to rerecord the auto attendant message or change the message for the mail boxes that auto attendant send the caller to. i can do intercom 500 from every extension and that takes me to voice mail management, but I can not figure out how to manage another mail box other than the extension I am on or how to change the recording on auto attendant. I can access both systems with my laptop.
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Hi,

We currently have a Cisco CCM system which runs Vlan 10 as the Voice lan and Vlan 2 as the Data, the hosted platform we are going to can have vlan which is set via portal etc, it gets a normal IP and then boots into the correct lan ..... phone will connect ok but will no allow data pass through .... the Cisco phones are working fine but not sure why the hosted wont work ...... any ideas ?

Thanks
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Got a ip pbx and i want to send the voice mail via e-mail in the office we got a Miicrosoft 2011sbs standar with exchange 2010
altho i have create the account voicemail@xxxxx.org and configure the ip pbx the pbx is not able to send the email with the voicemail. i have testet the email created and work.  The pbx is nec sl1100
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Hey all, i have an issue that external calls has noise, delayed audio from the external side. Internal calls are fine.
We recently changed from the Telstra supplied modem to a Draytek Modem. All ports have been opened up the same as they were on the Telstra one, all lines and SIP registered straight away, however i have not been able to resolve the noise.
As Draytek have alot more advanced firewall settings, QOS, I'm not sure what feature / settings i need to change to test.
We are running freepbx 2.11.

Thanks
Matt
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Hello Experts,

I am in need of options to connect an analog phone to an old, Nortel PBX that is in another building with no physical, wired connection between the two buildings.  There IS network connectivity between the two buildings, and I've been told by our phone support \ vendor that there are devices available that would connect to the analog port on the PBX on one side, and to the phone itself on the other side, and then both devices would connect to the LAN to provide connectivity between the PBX and the phone.  Our vendor, however, has very little info on these.  I am writing to see if anyone here has any familiarity with this concept, and if so, can provide any recommendations or guidance on makes \ models they have used successfully in the past?

Thanks in advance for any help that can be offered,

Russ
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When ever i use skype for basic chat the output on the screen shows first in non-italic characters and then repeats itself in italic characters.  Is there any way to shut this off and have it all appear as just one non-italic output?
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We have ISDN PRI line 30 channel terminated into Digium TE133 card. Sometimes when we make outbound call on any mobile number we got error "All circuits are busy now". Once we redial the same number the call connects.

 How to capture  debugs  ?
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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.