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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We have a Cisco UCS560 Phone System that has been setup with a auto attendant to welcome all incoming callers a menu:

Option 1 - Sales and Ordering
Option 2 - All other callers

Today the phone system has decided not to route any calls when the option 2 has been selected by the caller. Need immediate advice. TIA!
I have a Polycom VX300 that was from vonage and now I have a different provider.

I can't reset the phone I tried everything. is there a way to force a flash to the phone to bypass the lock on the phone?
I am getting complaints of poor voice quality from my users at a new site. What's new about the
site is that they are using 7821 Cisco phones with SIP instead of 7945 phones with SCCP. I am
stuck at the outset trying to troubleshoot as the product I use to measure call quality - Solar
Winds VOIP and Network Quality - sees no MOS score information. Could that be a problem
of using SIP instead of SCCP? Or is it because of the model of phone? Other possible issue?

Update: I don't think SIP is the problem for no MOS scores as there's a conference phone
there - an 8831 - running sip8831.10-3-1SR2-2 which does report MOS scores ok.
I bought new phones and when I went to add them to call manager the option was not there for the new model.
My call manager has the 8861, but not the 8865.

here is the info about my call manager
Cisco Unified CM Administration
System version:
VMware Installation: 2 vCPU Intel(R) Xeon(R) CPU E5-2650 0 @ 2.00GHz, disk 1: 80Gbytes, 4096Mbytes RAM, Partitions aligned
Dear Sir,
i installed primacell to my VG router it is working MGCP with the call manger 10.5 and the caller id of the inbound calls show as O the below the configuration of the port that are connected to the primacell:
voice-port 0/1/3
 cptone EG
 timing hookflash-out 50
 timing guard-out 1000
 impedance complex2
 description line#8
 caller-id enable
 caller-id mode DTMF start A end D
 caller-id alerting ring 2

and ceck the below debugs:

*Jun  1 08:36:29.816: htsp_dsp_message: SEND_SIG_STATUS: state=0x0 timestamp=212
19 systime=463530589
*Jun  1 08:36:29.816: htsp_process_event: [0/1/3, FXOLS_ONHOOK, E_DSP_SIG_0000]f
*Jun  1 08:36:29.816: htsp_timer - 125 msec
*Jun  1 08:36:29.944: htsp_process_event: [0/1/3, FXOLS_WAIT_RING_MIN, E_HTSP_EV
*Jun  1 08:36:29.944: htsp_timer - 10000 msec
*Jun  1 08:36:29.944: htsp_timer3 - 5600 msec
*Jun  1 08:36:29.944: [0/1/3] htsp_start_caller_id_rx: Warning!! DTMF callerid r
equires line reversal alerting.
*Jun  1 08:36:29.944: [0/1/3] htsp_start_caller_id_rx:Mode DTMF. Alerting 0x4
*Jun  1 08:36:29.944: htsp_start_caller_id_rx create dsp_stream_manager
*Jun  1 08:36:29.944: [0/1/3] htsp_dsm_create_success  returns 1
*Jun  1 08:36:30.716: htsp_dsp_message: SEND_SIG_STATUS: state=0x6 timestamp=221
15 systime=463530679
*Jun  1 08:36:30.716: htsp_process_event: [0/1/3, FXOLS_RINGING, E_DSP_SIG_0110]
*Jun  1 08:36:30.900: htsp_dsp_message: SEND_SIG_STATUS: state=0x4 timestamp=223
I'm trying to add a virtual extension to trixbox ce (v2.6.1) but cannot find how to do this
Basically I want an extension number which will play a recording

E.g Someone rings me , I answer and then forward him to the virtual extension where he will hear a recording

Can anybody help please
would like to get your take on a resolution to my SIP trunking dilemma.  i am configuring a SIP trunk  to a provider with the network architecture as:

PBX LAN 1 -> firewall -> provider network

we are successfully registered and inbound and outbound calling is working.  however, audio inbound to the IP office is not working.  so in other words, when you call into the IPO you can hear the person but they cannot hear you.  after much packet analysis on the provider side and on the IP office side it seems that we have narrowed the issue down to SDP.  the provider is seeing SDP information to send audio back to the IP address assigned to LAN 1 (192,168,168.30) of the IPO which is behind the firewall.  i need the IPO to send SDP information to let the provider know to send audio back to the IP address of the firewall outside interface ( so it can then be port forwarded (PAT) back to the LAN 1 IP on the IPO. = IP Office LAN 1 (LAN 2 not in use) = firewall outside interface = provider switch (ITSP domain name)

below is a packet capture on an outbound call that completed showing the SDP info on the IP office side and in red are the IP address parameters that need to be changed to the outside interface of the firewall (  is this possible?  i have never had an issue with this before and have confirmed other SIP trunks i have setup are not affected by the private IP of the IPO LAN 1 showing in the SDP info and …
We have one Avaya IPOffice (Digital) system in a building which is across the street from another Avaya IPOffice (VOIP) system.  The network is connected via a FluidMesh bridge.  The two systems have been configured to talk to each other and we use pre-programmed buttons to reach colleaugues  or transfer calls to either side of the street.  This worked initially, and will often work after rebooting a phone, but is not reliable and more times than not you will hear silence before your call is dropped when attempting to reach someone on the other side of the streeet/bridge.

I logged a recent attempt to reach a colleague from my ext. 233 to his at 110 as follows:  (any and all help would be appreciated in figuring this out).

Extension Status, AdminBldg (                                            
5/25/2016 10:45:18 AM                                                              

Extension Number: 233            
Slot: 2                          
Port: 6                          
Active Location: None            
Telephone Type: 9508              
Current User Extension Number: 233
Current User Name: John  
Forwarding: Off                  
Twinning: Off                    
Do Not Disturb: Off              
Message Waiting: Off              
Number of New Messages: 0        
Phone Manager Type: None          
Packet Loss Fraction:            
Hello all, we have cisco call manager 7, and i was wondering how can i see the log of calls made by a user on a certain period of time.
I would like to lock the Cisco phone.So that whnever a user tries to dial a mobile number or an international number it should ask for a pin..I know we can achieve this from CAC, but not sure if it can be locked only for mobile and international calls.

Please let me knw the steps
Is there a way to delete all user recorded greetings and voicemail from Unity?

thanks in advance

I have a UC560.  Wehave checked all of the configurations and they appear to be correct.  We checked with our SIP provider and they say that everything is correct and that calls are coming to the UC560 but all incoming and outgoing calls are receiving a busy signal.  Any help will be appreciated.
Hi,  we have a mitel 5000 system in one of our facilities.  I am trying to come up with a configurable way that we can make an extension ring though the paging system because it needs to let people on the plant floor know someone is calling.  I know I could open up a phone and solder on some wires to the speaker and attach it through the paging system... but before I plug in my soldering iron, I thought I would ask you find folks if you know of another way.

Thanks in advance!
Despite my pleas to upgrade, I still have a client that has a Cisco UC500.  the are changing the answering service, and I can't for the life of me figure out how to change the number we forward to in CAA.  (night service bell object).  Assistance is aprpreciated.

We have a system that we changed the IP details on. When we entered the details we entered the "IP address" the "subnet mask" and (what the engineer though was the gateway) the "Primary Trans IP Address". This last part was a boo boo as all IP traffic including the configuration program now gets traffic directed to the gateway and not the IP Office unit.

I can see lots of methods to change IP and passwords etc through the DTE Port but unable to work out how to remove the "Primary Trans IP Address"

Anyone got any tips?
when we call from our cisco phones to outside numbers or mobile phones , the caller display on the mobile is of our main Pri number and not the extension number of the user..Also when we call back the main pri number it dosent ring..

How can I solve this problem..Please assist in what is the configuration required on my voice gateway or call manager

We are using Cisco router as voice gateway and call manager ver 10.5

We recently purchased a new Avaya IP Office 500 V2 phone system for our new office.  It's running version build 995.

The handsets we purchased for the new phone system have a voicemail button for Visual Voicemail.  The problem is that this is currently only working sporadically but it does however work on occasion.   We've reached out to Avaya support but they just suggest that we upgrade all our other phone systems.

We currently have other Avaya phone systems in other branch locations but they're running older versions.  All of the phone systems tie back to our corporate location via MPLS connections.  Our Corporate hub is where the voice mail servers are located.

Has anyone else out there ever encountered this issue?
Trying to figure out why my switches are dropping ARP packets from one switch to another. The switches are connected thru a leased fiber line from a 3rd party. i will try to explain it as best as i can. it gets confusing. Location 1 calls location 2 and location 2 picks up the call, they can hear me but i cant hear them. we get a delay between 3 up to 15 seconds before i can hear them. If i call them back right after we have finished the conversation it works fine. What i have seen in wireshark is that both phones do an ARP to find each other. in wireshark those arp's are showing as lost between the 2 switches. not all the arp's but enough where i have to start there to try to figure out the issue. I have had numerous calls with the leased fiber operator and network and mitel engineers but so far no luck they are all blaming each other so i am just caught in the middle.  Any ideas as to where to look would be highly appreciated. I have QOS on the specific vlan enabled and i dont have this issue on switches that are running dedicated fibers.
Conference calling is not working with numbers outside my company..But within office conference calls work
we are using Call Manager 10.5.

Need help to resolve this please
Strange problem.. Have a network with 4 locations.  All the locations run Atcom PBX's, all go through the same phone provider, and three of the locations use Zywall USG 20 routers.  My problem is this..Every  week or so, the pbx's  will freeze up at the locations with the Zywall's though not all at the same time.  When I attempt to access the Web GUI, right before the system goes totally down, the log in splash page moves from the center of the page to the top left, and I wont be able to log in.  My only solution is to reboot the phones.  
I am not sure where to start on the troubleshooting as they go down at random times.  oh yes, and all sites are connected via VPN connection through the routers.  The only sire that doesn't seem to experience this issues is the site without the Zywall router, yet I run different tings on the other routers and they never have any issues.  I am at a loss to know which way to proceed
We have been experiencing an on-going issue with our network since the installation of a Mitel cloud-based VOIP phone system.

The phones all get DHCP addresses from the Active Directory DHCP server. We only have one domain controller which acts as the DNS and DHCP server too. This is a Windows 2012 Standard server.

All workstations are set –up to use Office 365 E3, so all email is stored in the Microsoft cloud.

The network consists of a Cisco ASA5505 firewall, a Barracuda Web filter immediately on the inside network, then 3 HP 2530 network switches (just installed in the last 3 weeks). We also have 2 Aruba Wireless Access Points. There is only one VLAN so no Trunking has been setup. The firewall is configured with NATing for certain servers, the security and camera based network, these being controlled by an external provider.

One Xerox copier/printer/scanner is setup on a static IP address to scan to email using Office 365.

During the day, when staff are in the office, random outages occur which involve workstations being blocked from all internet access, so no web browsing, email, etc. This includes at times the Xerox printer being able to scan to email. There is no fixed pattern to the block on internet services. One day it will be one workstation, another day may be 5 other workstations. The workstations remain connected to the switch (link light is on) and can access anything internally.

I have cleaned up DNS, which did have some multiple entries in …
I have a user that no longer shows their voicemails in Communicator. Everything else looks fine. The history, contacts, etc. The voicemail tab is completely blank (see screenshot). We tried restarting the application and changing the voicemail password. Voicemails are available for this user from the handset.

I have raw Asterisk 11 set up as a standalone voicemail.  I have been asked to direct the voicemail email MWI to an alternate server for translation to an SMS but it keeps bouncing.

My host IP is in sip.conf

I have my subscribers configured in voicemail.conf :
4474XXXXXXXX@ (IP of SMS server)

When I do a TCPdump it shows:

        Via: SIP/2.0/UDP;branch=z9hG4bK634d57b7
        Max-Forwards: 70
        From: "asterisk" <sip:asterisk@>;tag=as64d0fb15
        To: <sip:4474XXXXXXXX@>
        Contact: <sip:asterisk@>
        Call-ID: 3b51d5e1569070454e3c94b406af2388@172.3123.5:5060
        CSeq: 102 NOTIFY
        User-Agent: Asterisk PBX 11.17.1
        Event: message-summary
        Content-Type: application/simple-message-summary
        Content-Length: 92
        Messages-Waiting: yes
        Message-Account: sip:asterisk@X.X.X.X
        Voice-Message: 6/1 (0/0)

and checking the maillog shows:

Mar 15 09:44:13 lm-vms-de-2 postfix/pickup[12201]: 87A227FC6E: uid=0 from=<root>
Mar 15 09:44:13 lm-vms-de-2 postfix/cleanup[12270]: 87A227FC6E: message-id=<Asterisk-6-1281486269-4474XXXXXXXX-1338@lm-vms-de-2>
Mar 15 09:44:13 lm-vms-de-2 postfix/qmgr[12202]: 87A227FC6E: from=<root@domain>, size=647, nrcpt=1 (queue active)
Mar 15 09:44:13 lm-vms-de-2 postfix/error[12273]: 87A227FC6E: to=<4474XXXXXXXX@>, …

We are running IP Office 500 version I have a bit of a funny,  I have set up an Hunt group with no voicemail on it. I have 5 CCR agents enabled on this CCR hunt group. Then I have an external number pointing to this Hunt group, when the an agents comes free this should ring the hunt group on the first call we can hear the caller, then the next caller in the queue we can't hear the call? It like the call has not been accepted by the IP Office ...Help

Group CCR Agent
Queuing ON
Over flow Off
Fall Back None
No voicemail
No recoding
Announcement Off

Where can I view the status of a ad-hoc Active Directory Synchronization? I'm not seeing a user that was added an hour ago and I doubt the sync is still running but where would I verify this?

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.