Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

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  • Have a deployment of 3 servers using dns loadbalancing.
  • we have a trunk setup via an sbc.
  • users have deskphones (CX600)


both inbound and outbound calls work as expected.


however, when a call is placed on hold after 30 seconds the call drops.

the same thing occurs when a call is parked..also after 30 seconds the call drops.



i should mention MOH IS SETUP in this deployment, however when an external call is parked there is just a beep sound.

when a call is parked between users MOH does work and the call does not get dropped.

refer and bypass are set to false on the trunk as well.  Trunk settings



in the snooper log i see references to "this call leg has been replaced"  in the same message as the BYE:
ms-diagnostics-public: 10026;reason="This call leg has been replaced";component="MediationServer"

the trace from the sbc shows that the mediation server is dropping the call so i haven't mentioned that here.

have the snooper trace if needed.


any suggestions appreciated.
0
How do I go about moving from a phone number hosted with Grasshopper.com to Google Voice?
0
Is there a portal for AT&T SIP trunk customers? We've asked our AT&T sales rep and Tech consultant over and over for this information but they can't seem to find it. I want to be able to forward DIDs which are part of our SIP trunks. So simple - yet so complex for the behemoth.
0
Hello,

Can anyone point me in the right direction? There are a few google hits for this problem, so it seems like a known issue.

///////Summary: On outbound calls, we are sending out caller ID, and it is showing as "restricted" on HD enabled mobile phones only. Older mobile phones and land lines display our caller ID correctly. The telco PSTN providers have all pointed to our call manager as not providing the correct 1TU-T E.164 standard. This is occurring on all 5 MGCP gateway PRI's across multiple local PSTN providers.


///////System Parameters
We have 2 call managers in our HA cluster, 1 publisher, 1 subscriber.
Cisco Unified CM Administration

System version: 10.0.1.12900-2

VMware Installation: 2 vCPU Intel(R) Xeon(R) CPU E5-2609 0 @ 2.40GHz, disk 1: 80Gbytes, 4096Mbytes RAM, Partitions aligned


///////Troubleshooting Steps
Here is the support forum posting of the same issue, I have performed the changes advised in this posting with no success:
https://supportforums.cisco.com/discussion/12746556/debug-caller-id

1) Under Service Parameters - Clusterwide Parameters - Calling Party Number Screening Indicator

Set this value to Callmanager Provides Calling Number (No success)


2) I have performed the changed to Call routing information - Outbound calls, tested with Calling Party IE type as both "national/ISDN", Calling Numbering plan "ISDN". No success.

Here are our test call DNs appearing in the debug isdnq931:
May  4 11:44:02: ISDN …
0
I stumbled on this 'free' service through a website that promotes deals on the web.

Figured I'd give it a try and it can't hurt to have a spare sim with live service but I have a bunch of questions and their tech support is the most useless I have ever experienced.

Wonder if others here have used this service and can play tech support for them since theirs functionally doesn't exist.
0
could i see the
do not call list


how can individuals know which numbers not to call; if they cant see the list


I am not sure which zone this question should be in so please add zones.
0
Sip phones do not register to CME when connection to UCM is lost. can any one assist me.

my configs below
cts logging verbose
!
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
!
voice service voip
 no ip address trusted authenticate
 allow-connections h323 to h323
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
  no h225 timeout keepalive
  call preserve
!
voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g711ulaw
!
voice class h323 1
  h225 timeout setup 3
  call preserve
!
voice class custom-cptone exit
 dualtone conference
  cadence 400
!
voice class custom-cptone join
 dualtone conference
  cadence 200
!
!
!
!
voice translation-rule 1
 rule 1 /^.*/ /81\0/
!
!
voice translation-profile 81code
 translate called 1
!
!
!
license udi pid CISCO2911/K9 sn FGL204110KH
hw-module pvdm 0/0
!
!
!
!
redundancy
!
no cdp run
!
ip tcp synwait-time 10
ip tftp source-interface GigabitEthernet0/0
ip ssh time-out 60
ip ssh authentication-retries 2
ip ssh version 1
!
!
!
!
interface Loopback0
 ip address 172.25.12.5 255.255.255.252
 no ip redirects
 no ip unreachables
 no ip proxy-arp
!
interface Embedded-Service-Engine0/0
 no ip address
 shutdown
!
interface GigabitEthernet0/0
 description Suam-suame Link to Radio
 no ip address
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip flow ingress
 ip virtual-reassembly in
 duplex auto
 speed auto
 no …
0
Does anyone know why the Avaya Phone System would give an error "DHCP ACK" error bootup? I noticed that it seemed to happen during the day when it was busy and towards the end of the day, it worked fine. When it did occur, I ran a ping test and noticed that it had timed out so it led me to believe that it was probably related to network congestion.  Also, not sure if the IP Office phone system's port is running only at 100MB (and don't think that system can handle more). We will be monitoring this some more through out the week.
0
On a network there are VoIP phones, the workstation are connected through the phones.

Is it possible to create a Vlan for the phones and a VLAN for the other devices?

I will enter the phones MAC address , and that list of MAC will have their own VLAN. all other devices will be on a different VLAN.

if yes how do I accomplish that?

( I want to use Sonicwall)
0
I call 1800 numbers and press 0 many times and then get disconnected

and then I call back and press 1 many times


is there an automated solution to call 1800 number press (0001, 0002, 0003 , 0004... ) and alert me a human has picked up phone
0
Hey Guys,

How do i backup the address book on a Cisco SPA525G phone?
0
I have CUCM and Unity Connection at my main office.  We just acquired a small office in another state, and want to put them on our VOIP system at our Main office and have a couple questions:

1.  Will the telco allow me to port numbers over from one state to another (different area codes)?
2.  If they do allow that, how would I ensure those numbers we ported have there own opening greeting?  So if someone called those numbers we ported over - they want it to ring a separate open greeting.
0
Hi I need a little help please I have a Cisco VOIP box running fine with a UK BRI Isdn voice card in it on a 3725 with an NM-1V interface. I am making outgoing calls just fine and receiving incoming calls everything is perfect.
However I need to utilise some of the additional UK Global phone numbers allocated to the ISDN line so calls from line 20 say get the outgoing caller id changed to 01234 567567 say instead of the default ISDN Number and when 01234 567567 is rang the call Just goes to Extension 20. I am guessing I need transform sets in the config can someone post me a little example please
I have at the moment:
isdn switch-type basic-net3
voice-card 2
 no dspfarm
!
interface BRI2/0
 no ip address
 isdn switch-type basic-net3
 isdn point-to-point-setup
 isdn voice-priority always
 isdn incoming-voice voice
 isdn static-tei 0
!
voice-port 2/0/0
 compand-type a-law
 cptone GB
 connection plar 20
!
Thanks
0
If a home user subscribes to FIOS, do they need to purchase new phone equipment for the apartment (i.e. answering device, headset, conference phone, etc.)
0
I am running Call Manager 9.1.2 and Unity Connection 9.1.2 in my main office.  We have a small branch office, we ported their phone numbers over to the main office about a year back, and their Cisco IP phones now register over a VPN to our main office.  This works great until there is a WAN outage, then they are without phones.  So I was reading about SRST.  So lets say I get 4 analog lines installed at this branch office and install an SRST router (2911).   This will allow them to call outbound during a WAN failure, but what about inbound calls?  Their main numbers are not going to work because their main numbers were ported over to our main office.  So how will they still get their inbound calls?  What do people do in this situation?  I feel like it has to be common, but i can't seem to figure it out.
0
I have a Cisco Linksys SPA 8000.  I was wondering if there was a way to export and import the current config.
0
Hey Guys,

We have a Cisco SPA525G2 phone here, it's just stuck on the CISCO screen even after reboots. Does anyone have a fix for this?
0
I have the following at my main office:

1.  CUCM 9.1.2
2.  Unity Connection 9.1.2
3.  Cisco 2921 Router (H.323) Voice gateway.

We have a small branch office, that is not on the VOIP system.  They just have 5 old school phones connected to analog lines.  They just have a receptionist that answers the phone.  They want to be put on our VOIP system, so they can have autoattendant, VM to email, etc.  I have a few questions about this:

1.  Could I just install a Cisco voice gateway router at this branch office with two VIC2-4FXO cards and connect to phone lines lines to it?  
2.  Would I then add this new gateway to CUCM?
3.  Could I configure separate call handlers that would pick up when someone calls their phones lines coming off their gateway router?
0
We have a cloud base Meraki APs at this site. We incorporated 7941 Cisco IP phone with the 7925G. we get frequent disconnects were user on the 7925g cell phone roams either won't get the call, or the call will drop, or the call will hang. We have pretty good wifi coverage in the office. Distance between the AP's is pretty close to each other. It can happen once a week or it will happen few times a week. From the logs of the cell phone I get this:

2017-04-05 11:29:10:0510 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 
2017-04-05 11:29:10:0520 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 
2017-04-05 11:29:12:0030 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 
2017-04-05 11:29:12:0040 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 
2017-04-05 11:29:13:0430 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 
2017-04-05 11:29:13:0440 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 
2017-04-05 11:29:14:0770 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 
2017-04-05 11:29:14:0780 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 
2017-04-05 11:29:47:0490 CP-7925G user.err secd: SSL_accept:error in SSLv3 read client certificate A 

Open in new window


Not sure what this means exactly. I also included an image from the cell phone trace logs for review. I also included SSID config which has very minimal overhead. Only this phone can access this SSID.

Could that SSL cert error be contributing to the issue? There is no traffic filtering in place. We have a lot of bandwidth available. Not sure what to look for at this point
2.JPG
1.JPG
3.JPG
0
I have 2 networks running call manager express. The 2 routers are connected VIA serial connection. I have voice vlans setup on each side with dhcp pools, and 1 phone on each network. The phones receive IPS, I can ping them from each router and the opposite side. I'm using EIGRP routing protocol for my networks. I have my dial-peer setup, destination-pattern, and session target with the opposite side's CME IP address. I get a dial tone once I dial the number I get a fast busy tone. Could you please help further troubleshoot this


ip dhcp pool HQ-15 VOICE
   network 10.10.15.0 255.255.255.248
   default-router 10.10.15.1
   option 150 ip 10.10.15.1

interface FastEthernet0/0.15
 description HQ-VOICE
 encapsulation dot1Q 15
 ip address 10.10.15.1 255.255.255.248
 ip helper-address 10.10.15.1
 ip virtual-reassembly in

interface Serial0/3/0
 bandwidth 64000
 ip address 192.168.100.245 255.255.255.252
 ip nat inside
 ip virtual-reassembly in
 no fair-queue
!
!
!
router eigrp 1
 network 10.10.6.0 0.0.0.255
 network 10.10.15.0 0.0.0.7
 network 10.10.18.112 0.0.0.15
 network 10.10.25.24 0.0.0.7
 network 10.10.100.244 0.0.0.3
 network 50.194.52.168 0.0.0.7
 network 192.168.100.244 0.0.0.3

dial-peer voice 1 voip
 destination-pattern 1...
 session target ipv4:192.168.15.1
!
!
!
!
telephony-service
 max-ephones 6
 max-dn 20
 ip source-address 10.10.15.1 port 2000
 system message Labs-HQ
 load 7960-7940 P00308000500
 time-zone 5
 keepalive…
0
Hi Experts,

We just implemented a new VoIP telephone system and we are experiencing so random cracking and static on some of the phone calls.


The implementation guys are saying this is a network issue but I do not think this is the case since we did a voice readiness test before we started the implementation phase and the report did not show any issues.


Any suggestions on what ma be causing his issue?

Any suggestions on a tool that I may use to monitor and track down any phone quality issues?

Thanks
0
We have a Cisco 3560G Switch - connected is 6 PC's, 6 ShoreTel IP phones, a Server and 2 Gateways; Sonicwall Router for Data and Frontier DSL modem/Router for Voice.  I want the phones and PC's to plug into the Cisco 3560G and be able to Route to each perspective Gateway.

I would like to create VLAN 10 for Data and VLAN 20 for phones.

I am weak on the Cisco programming and tried using the Cisco Network Assistant to achieve this setup without any luck.

What would be the simplest way to achieve this network setup on the Cisco 3560G Switch?
Does it have to programmed all using command line, if so where would I find this information in concise format, or can I use a GUI like Cisco Network Assistant App?

Thanks Experts
0
Hello

Currently we are using Lync 2013. Our voice backend is Cisco Call Manager 10x

We'd like to look into integrating the two, mainly so that we can :

1. Users can click to dial on Lync desktop to call contacts

2. On a personal mobile device, we can deploy the Skype for Business mobile client and leverage voice functionality to reduce call costs to the user

Does anyone know if the above are even possible with our set up?
0
Is their a way I can increase the microphone in my laptop so that even some one wisper, I can hear it from the other laptop on internet
0
Hello

Our users have two numbers for the clients essentially;

1. Their desk number
2. Their mobile number

I'd like to look into a solution where if their client wants to call them, they only need to ring one number which will ring on either desk or mobile depending on the owner's choice.

I believe this is called Single Number Reach in Cisco's terminology.

Is anyone aware of any actual implementations and what's needed to support this?
0

Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).