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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My company has 17 branches, a HQ, and a data center at a colocation.  Our UCCX and Call Manager are located at our data center.  We have a PRI located at each location.  The main number for each branch is translated to a designated trigger on the UCCX on our voice gateway.  When our branches lose data connectivity back to the Call Manager and UCCX, SRST does not function properly.  I know that the phones go into fallback mode and outbound calls can be made, but incoming calls get a fast busy.  I believe there is a problem with the configurations for SRST on the gateway.

Question: Is there a way for the gateway's voice translation to change to an internal extension located at the branch rather than to the UCCX trigger if data connectivity is lost at the branch office?
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Good evening, how i can block transfer call for outside of company, the allow for inside only. For more information, we have two ISP, where one is a normal ISP and other is for SIP Trunk, now our plan is to block all transfer call for outside of company, and permit to transfer call only for inside (LAN) of company.

In future, we want to create a rule that permits a list of numbers to transfer to outside of company and deny the rest. Please, help me to solve this problem because we are receiving a big count to pay to ISPs and people (our BOSS) doesnt understand what real happend, then use to complain everiday.
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I've installed an HP 2530 48 port switch, and I'm having problems getting Cisco 8851 phones with a key management side car / add on.  The KEM won't power up on the phones.  On the rare occasion, I got one phone to power up the KEM but I haven't been able to duplicate the process.  If I were to restart that phone then I'm not sure if the KEM would start or not start.  I cannot tell if it's a config issue on the switch or something with the phone config.  Unfortunately, I don't have access to the phone config.

With the default switch configuration, the phones startup and indicate that the accessory (the KEM) cannot be started. When I look at the phone log, I find a line item indicating that a request for power is made but it is denied.  This is with the POE+ option TLV enabled.

If I turn off POE+ and just use LLDP-MED, the accessory still doesn't power up but there is no message about the accessory not being started.  When I look at the phone log, I find a line item about a timeout occuring when the request for additional power is made.

I've uploaded the results of "show tech all" for the switch.

I don't manage the phones.  I'm trying to get the HP switch in place to avoid having my client purchase the Cisco switches the phone vendor wants to install.  There's almost a 3x price difference !!!

Thanks for your help.
putty-session.log
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I am replacing my 3845 router with PVDM2-64 with a 4331 with PVDM4-64.

In the 3845 I had configured 8 sessions of transcoding, 4 sessions of conferencing. And for MTP " maximum sessions software 128".

With the new 4331 system with PVDM4-64 - what are the maximum number of transcoding and conferencing sessions I could configure? I am thinking to keep the maximum number of participants to ad-hoc conferences to eight.

Thank you.



!

dspfarm profile 2 transcode  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 maximum sessions 8

 associate application SCCP

!

dspfarm profile 1 conference  

 codec g711ulaw

 codec g711alaw

 codec g729ar8

 codec g729abr8

 codec g729r8

 codec g729br8

 maximum sessions 4

 associate application SCCP
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I''ve recently been given the responsibility of managing a FreePBX cluster deployed by an outside vendor.  I've been asked to add some extensions to a "hunt group" so when a particular extension in the IVR is pressed, the call will ring on any open extension in the "hunt group"?  The vendor already created several of these extensions which ring on a "hunt group".

In researching about FreePBX, I thought I was supposed to create a "ring group" but when I click on ring groups in the FreePBX UI, it doesn't look like there are any ring groups configured so I'm not sure how the vendor is currently accomplishing this functionality?  How can I accomplish this functionality?  There are queues configured.  Am I supposed to add the extensions to a queue?  If so, how do I do it?
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Attached is the sh tech from the router, also attached is the sh run and sh policy-map for the interface in question, basically we want to confirm the QoS policies we have currently configured, we had an incident yesterday where the traffic for the MPLS circuit had a brief burst which consumed the full allocated bandwidth which is 100mbps, when this occurred the voice RTP traffic was affected along with the call signaling for the Cisco phones. Just like to have a second pair of eyes for the QoS configuration on this router.
sh_policy-amp.txt
sh_tech.txt
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Hi,
I'm getting the below error message when booting my Cisco Call Manager:
*** An error occured during the file system check.
*** Dropping you to a shell; the system will reboot
*** when you leave the shell.
Give root password for maintenance
(or type Control-D to continue):

This link explain how to fixe it but it doesn't fixe the one I'm facing.

Can someone help me fixe this problem?? Give me the root password as I fail to log in??

Regards.
0
Hi,

We run an Avaya Ip Office 500, with 10 Avaya agents these people answer 5 hunt groups,   these hunt  groups run under the time profile
from the Avaya

Time Profile Mon - Fri 08:00 to 18:00

We have now started to work on Saturday, New Saturday profile Sat 10:00 to 16:00. The question how do I get our 5 hunt groups to look at the Saturday profile ?

regards,
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Hello,
we have a Cisco Unified CM ver 8.6.2 managing our VoIP calls. As test I was asked to configure one (and only one) DN line to use the code g722. Checking around the CUCM configuration I see that the g722 is indeed enabled (see attachment) but how can I configure only a specific DN to use this code without effecting the rest of the lines in production? And is there any configuration change also for the Cisco IP Phone  to use this codec?

thank you
codec.jpg
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I'm getting ready to deploy Skype for Business and replace our existing Lync 2013 Enterprise systems.  We've had an enterprise deployment with 2 front end servers since we migrated to 2013 from 2010.  We've used DNS load balancing as we don't have any hardware/virtual appliance load balancing systems.  Our plan is to use S4B Standard and just create a second server running as a failover pool in case of an issue.  

We have 280 Lync enabled employees, 150 of them with Enterprise Voice.  The Skype for Business server will be going on a server with 64GB of RAM, dual Intel Xeon E5-2650L v3 processors (total 2 Sockets, 24 cores, 48 logical procs, at up to 2.09GHz).

Will I have any issues moving the CMS back to a Standard Edition server?  Any other issues I should expect?
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I am running CUCM 10.5. I need a program that will allow users to click to call from SharePoint.  We are currently not running any IM&P servers and dont plan on it anytime soon. I looked up Cisco click to call but it seems that is no longer available. Does anyone have any suggestions?
0
Using RTMT I know that I can view the number of failed registration attempts. But how can I see the details of what mac addresses are attempting to register but failing for whatever reason? Thank you.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/8_5_1/rtmt/RTMT/rtconfcm.html#wp1296686
0
I have the Speedport Entry 2i router and have set up both a port forward and a DMZ zone which points to an ubuntu server that has apache2 and asterisk running. Pings and special software confirms that the ports are open (for testing purposes all ports are open) and I can successfully from any point in the world see my "It works" apache2 page. The issue comes when I try to register sip client onto the Asterisk server. The same client and other client register without any issues from LAN, so should not be a config issue. I initially though that the router blocks something, but tcptrack detects connections when I try to register, so not sure where the issue is.
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So to preface, Myself and my underling are not the guys who set this system up, nor are we by trade telecom guys.

Out pbx's digital ports are full, so we're testing adding voip phones to free up digital ports.  So we had extension 336 using port 01a1005, we changed it to a 4612 which changed the port so S00001.  However checking free ports there are still 0 returned.  What are we doing wrong?
0
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!
0
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?
0
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.
No-MOS-Example.png
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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: 10.5.1.10000-7
VMware Installation: 2 vCPU Intel(R) Xeon(R) CPU E5-2650 0 @ 2.00GHz, disk 1: 80Gbytes, 4096Mbytes RAM, Partitions aligned
0
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
xols_onhook_ringing
*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
ENT_TIMER]fxols_wait_ring_min_timer
*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
0
Hi
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
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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 (10.179.8.2) so it can then be port forwarded (PAT) back to the LAN 1 IP on the IPO.

192.168.168.30 = IP Office LAN 1 (LAN 2 not in use)
10.179.8.2 = firewall outside interface
10.128.2.40 = 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 (10.179.8.2).  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 …
0
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 (10.0.2.100)                                            
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:            
0
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.
0
Hi,
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
0
Is there a way to delete all user recorded greetings and voicemail from Unity?

thanks in advance


harbor235
0

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.