N95-2 Incoming VOIP Calls
Hi
Has anyone else had this problem and/or found a fix:
I have N95-2 updated to the latest firmware.
The Internet telephony works fine for making and receiving calls (I use Sipgate).
However if a caller rings in and withholds CallerID, the phone does not ring and the caller receives a message saying "The other person has hung up". If callerID is transmitted, the phone rings OK.
I have checked the routers and the call is being sent to the handset OK.
Any suggestions?
Thanks.
Is this device behind NAT ? You might be needing to forward some SIP ports on the firewall side to let the call come in. Additionally, there are available parameters on the PAP2T that will help you with the NAT issue , NAT mapping and NAT keep alive, use STUN or Outbound proxy if available.
Similar Messages
-
Problems setting up VOIP calls on Nokia E71
can anyone help on how to set up VOIP calls correctly on E71? what should I put in for Public user name? and then username and password??
please helpHi David,
Why not apply a access list to filter incoming traffic into the SG300 switch such as, via command line or GUI.
Here is an example below, by no means complete, just an example
Just remember, we are using reverse masking in the ACE;
config
ip access-list extended restrictGuest
deny ip 192.168.30.0 0.0.0.255 192.168.20.0 0.0.0.255
deny tcp 192.168.30.0 0.0.0.255 any 192.168.30.1 0.0.0.0 www
deny tcp 192.168.30.0 0.0.0.255 any 192.168.30.1 0.0.0.0 telnet
deny ip 192.168.30.0 0.0.0.255 192.168.10.0 0.0.0.255
permit ip any any
exit
interface gigabitethernet1
service-acl input restrictGuest
exit
Don't forget to save the configuration with the following command and respond to the prompt.
write
or do it via the GUI method
Step 1. Create a ACL name
step 2, Add the port based ACE which is the filter list,.
step 3. Apply or bind the list to a port so that the port can look at and filter pattern matches for traffic ingressing into the switch. I have given you an example of a ACE list above, you can be more creative in what you deny.
step 4. Now add or copy the entry to other switch ports.
Remember to save your configuration change.
Hope this helps.
regards Dave -
Block incoming anonymous calls
Hi,
I'm trying to block incoming blocked/unknown/anonymous callers over a sip trunk? I've creaed a translation rule and applied it:
voice translation-rule 5000
rule 1 reject /^$/
voice translation-profile CallBlock5000
translate calling 5000
dial-peer voice xxxx voip
call-block translation-profile incoming CallBlock5000
call-block disconnect-cause incoming invalid-number
To try it out, I'm dialing (from a normal/off network cell and landline) *67 and then the number. This does not work; only if I match the exact number I'm calling from, then it does get blocked.
When I show sip calls during the *67 call I see the calling number is blank.
Calling Number :
When I show sip calls during the regular call, I see the proper Calling Number.
As I understand it, with Call Manager and phones running SCCP, I cannot enable/use anonymous call blocking; so I do have to enforce the call blocking policy at this gateway device (UC520).
I'm very new to Cisco voice, so sorry I'f I'm missing something obvious. Thanks in advance!Thanks, I tried the rule--still private gets through.
*Feb 14 21:52:37.457: //394/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 207.2.123.180:5060;branch=z9hG4bK71b5401f596084541bb0894ae16bbbc8
From: ;tag=3538245046-512161
To: [email protected](CUBE external IP)>
Date: Tue, 14 Feb 2012 21:52:37 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Feb 14 21:52:37.457: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected](UCMBE IP):5060 SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy(UC500/CUBE IP)>;:5060;branch=z9hG4bK2B4F4E
Remote-Party-ID: [email protected](UC500/CUBE IP)>;party=calling;screen=no;privacy=full
From: "anonymous" ;tag=4C78DEC-1E4B
To: [email protected]>
Date: Tue, 14 Feb 2012 21:52:37 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0186175275-1452085729-2206375728-3952970709
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1329256357
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 246
v=0
o=CiscoSystemsSIP-GW-UserAgent 2063 472 IN IP4 yy.yy.yy.yy(UC500/CUBE IP)
s=SIP Call
c=IN IP4 yy.yy.yy.yy(UC500/CUBE IP)
t=0 0
m=audio 19566 RTP/AVP 0 101
c=IN IP4 192.168.20.5
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Feb 14 21:52:37.465: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Date: Tue, 14 Feb 2012 21:50:47 GMT
From: "anonymous" ;tag=4C78DEC-1E4B
Allow-Events: presence
Content-Length: 0
To: [email protected](UCMBE IP)>
Call-ID: [email protected]
Via: SIP/2.0/UDP yy.yy.yy.yy(UC500/CUBE IP):5060;branch=z9hG4bK2B4F4E
CSeq: 101 INVITE
*Feb 14 21:52:37.469: //395/0B18CF2B8382/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Date: Tue, 14 Feb 2012 21:50:47 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "anonymous" ;tag=4C78DEC-1E4B -
New FIOS customer with dropped VOIP calls and Internet connection
I am a new FIOS customer. Got my 50/25 connection a week ago, switching from a TWC 6/1 connection. Ever since the new connection, I've had numerous issues.
My VOIP (Ooma) connection constantly drops and re-connects during conversations
I've had random Internet connection losses, which picks up again after a few minutes
My home alarm starts chirping every once in a while
I've contacted Verizon several times due to these problems and have received varying answers with no resolution of the problem.
The first time I spoke with support, the tech logged into my router and changed the WiFi channel saying that would fix the problem. It didn’t.
The second time I contacted them, the tech ran a bunch of diagnostics and said everything looked fine so it must be an IP address conflict with my devices, because I had a couple devices using static IP addresses. He said everythinf should be DHCP and the last two digits could not be higher than 99 (192.168.1.99). He said FIOS does not support 3-digit numbers at the end.
So I changed all my devices to DHCP and ran some online VOIP tests. It showed a packet loss of 2-5% and MOS score of 1 (which is bad). I was still getting dropped connections, so I disconnected all devices and connected just one computer to the router and tested again. I was still getting packet loss.
Then I called support a third time, this time the tech said there were no 2-digit IP restrictions and that he was detecting there was no UPS baterry backup for the ONT which was probably causing the problem, so he dispatched a field tech to my house.
Today the field tech came (same guy as before), he took one look at the box and said it was too close to my Electric meter and the RF from the meter was causing interference to the FIOS connection and resulting in dropped connection.
He moved the ONT to another location and said that should fix it.
Well, I'm still seeing packet loss and low MOS score when I run the VOIP test.
I don't know how much of what the techs are saying is true and how much is made up stuff.
Has anyone had similar issues and have thoughts on solutions or likely causes for dropped VOIP calls and connections? Could RF be causing this?
I thought going from a 6/1 Cable connection to a 50/25 FIOS connection would be awesome, but this has turned out to be a nightmare, and I may have to switch back to cable if the problem is not resolved.
I would appreciate any help.
Thanks!Don't know where the packet loss is happening. I ran the VOIP test on myspeed.visualware.com and it shows a packet loss of 2-5% at different times and a MOS score of 1.
The report says MOS should be around 4 for good VOIP calls.
The Verizon tech who came to the house just blamed the electric meter box for RF interference and move the ONT farther away.
My concern is that I'm getting different answers from different techs at Verizon.
Regarding IP addresses. The Router shows a DHCP range from 192.162.1.2 to 192.168.1.254 as available for devices on the network. So, if I need to assign a static IP to a device should I use a number below 99 or above 151?
Thanks! -
Transfer VOIP Calls Between Cisco Desk Phone and Cisco Jabber For IPhone 9.5
Does anyone know how to transfer an active voip call from a Cisco IP Desk Phone to Cisco Jabber for IPhone? I can transfer a call from Cisco Jabber for IPhone to my Cisco IP Desk Phone no problem. I put the call on hold and then click "Resume" on my Cisco IP Desk Phone. However I cannot do the same but the other way around. If I put the call on hold on my Cisco IP Desk Phone, I see "no active call" on my Jabber client. The only information I could find slighlty relevant was using the Mobility Key/Remote Destination Profile feature however this defeats the object as this will forward to an external number, e.g. mobile and I just want to transfer the call within the VOIP environment between the two devices that are using the same directory number.
I am using Cisco Call Manager 9.1(2), Cisco Presence 9.1 and Cisco Jabber for IPhone 9.5.
Any help would be greatly appreciated.
Kind Regards,
Paul Parker.Did you ever find an answer to this ?
I am seeing the same behavior and trying so see if I can put calls on hold and pick them up both ways also.
The only answer I seem to have found is to use park instead
That would/should work but I would just prefer to hold/unhold
Just not sure why we would not be able to hold/unhold on what is essentially a "shared" line
Does anyone have this working for them ? -
A friend of mine has an iPhone with a cellular data plan, a Macbook Pro, and a mini-iPad. She's been using Facetime, but says that all of the devices used to ring at the time of an incoming Facetime call. Now only her iPhone rings. How do you set whichever device you want to ring?
James: I agree. Please keep them shorter.
The answers to both questions are no.
The iPad is NOT a computer replacement. It is a mobile device.
The iPad can not stream tro the iPhone nor an iPod.
Hope this helps.... -
Incoming Phone Call DND (Do Not Disturb) or ability to send call to voicema
Does anyone know a good way to put a DND (do not disturb) on incoming phone calls? The idea is to send calls directly to voice mail while still leaving the phone on to receive email & text messages. While at work or in meetings, it would be nice to not be bothered by calls especially from specific people. Ideally it would be really nice to one or all of the following:
-Function to send all calls to voice mail without ringing
-An option when an incoming call is coming in to send the person to voicemail with the choice of a prerecorded message such as "In a meeting" or "At work" etc.
-Send specific people to voice mail without ringing, possibly with a prechosen message.The only option for the iPhone not ringing when a call is received and the same when a text message is received while in a meeting is by turning the silent switch on located above the volume buttons.
-
SPA3102 - Line 1 - Outgoing VOIP calls fail
I recently reset my SPA3102.
After configuring Line 1; I am able to receive VOIP & PSTN calls on the analog phone attached but unable to make outgoing VOIP calls, I can only make outgoing calls via PSTN.
My dial plan is as follows:
(xx.|<#9,xx.<:@gw0>)
I can call the outside world by dialing #9 and then the phone number but I cannot call any VOIP extensions, when I try, I do I get the busy tone after a few seconds.
Please help, I’ve spend hours trying to figure this out to no avail.
Solved!
Go to Solution.Thanks for your response,
I finally resolved the problem after spending many more hours on it.
It turns out the problem had nothing to do with my dial plan, it had to do with a service provider (SPA3102) I had configured on my asterisk box; adjusting the configuration on that solved the problem. -
No voice during incoming outgoing calls
Hi, if anyone faced this issue before....there is no voice heard during incoming outgoing calls. However, we are able to talk on Skype or listen to music.
To me this sounds like the RTP stream problem - it probably does not pass the firewall / nat on the way between SIP provider and you.
What may have happenned is, that either on your Belkin router/NAT, or on internet ISP's router/NAT the SIP ALG (Application Layer Gateway) was enabled.
If this is the case, you need to disable SIP ALG on your router and/or ask your internet ISP to disable it on the firewall router.
Another issue may have been if you have (accidentaly) changed the port forwarding on your Belkin router, and/or the PAP2T local LAN IP address changed and the port forwarding no longer works.
So my proposals to check would be :
- check and DISABLE SIP ALG on your Belkin (if router has this feature)
- check the port forwarding on your Belkin router ... IF you setup port forwarding for 5060/61 you MUST set the same for RTP ports (16384-16483)
- if you don't have port forwarding on your Belkin, try to setup forward of 5060-5061 AND 16384-16483*.
*(If this is too many ports, set the the RTP portrange on PAP2T to 16384-16394 for example and then forward only that range)
- if neither of above helps, change your Line1 SIP port from 5060 to let's say 6070, and RTP ports to the range let's say 17300 - 17310
- last you may do on your side is to remove all port forwarding and put the PAP2T to the DMZ settings of your Belkin router. -
Only want VOIP calls, can it be made to do that?
Don't want a phone carrier. Rather purchase directly and use for VOIP calls. Possible? If so, where to buy?
Hi,
No question is ever a dumb one. Currently there is a video chat in the Nightly version of Firefox for desktop and mobile I believe. However if you are looking to purchase a Firefox OS device with no carrier you would need an internet connection.
Its expected to see this functionality in the later versions 2.0 or 2.1 Firefox OS, but is available in the Web API in the gecko.
Keep track of the Hello project: [https://wiki.mozilla.org/Loop] -
Making a VoIP call with the Cisco 837 ADSL router
I would greatly appreciate if could please provide some technical assistance to my questions below:
Is it possible to make a VoIP call between two 837 ADSL Cisco routers over a 1Mbps ADSL broadband connection?
If so, can I configure this VoIP connection using either a PPPoE or ATM WAN link?
Is it possible to make a VoIP call using a Cisco 837 Router while simultaneously surfing the Internet? In other words do I need two public IP addresses i.e. one for accessing the internet and one for making the VoIP call or is one static IP address obtained from my ISP sufficent.
It is possible to configure QoS parameters (e.g. RSVP, Voice precedence, Voice codec selection) on this 837 router using PPoE or can it only be done using an ATM WAN interface?
Does the Cisco 837 router support both the H.323 and SIP communication protocols? Do I need to purchase a certain IOS operating system version for VoIP calling?
Does the VoIP dial peers need to be configured with both a POTS and VoIP phone numbers or is only one number required?
Do I need to obtain a special VoIP number from my VoIP service provider? or can I use existing POTS numbers or made up numbers within the dial peers as this situation involves making a private VoIP call between two branch offices using 837 ADSL routers and not via a VoIP service provider.
Finally, can I use POTS ordinary telephones with the Cisco 837 for making VoIP calls or do I strictly need to purchase VoIP phones?
My apologies for the number of questions asked here but I currently need to know the technical ability of the Cisco ADSL 837 as I am thinking of employing these routers in my company organisation.
I await your feedback in due course.
Thanks,
Martin HealyHi,
I give you a sample config of my router.
class-map voice
match access-group 101
policy-map mypolicy
class voice
priority 128
class class-default
fair-queue 16
ip subnet-zero
gateway
interface Ethernet0
ip address 20.20.20.20 255.255.255.0
no ip directed-broadcast (default)
ip route-cache policy
ip policy route-map data
interface ATM0
ip address 10.10.10.20 255.255.255.0
no ip directed-broadcast (default)
no atm ilmi-keepalive (default)
pvc 1/40
service-policy output mypolicy
protocol ip 10.10.10.36 broadcast
vbr-nrt 640 600 4
! 640 is the maximum upstream rate of ADSL
encapsulation aal5snap
bundle-enable
h323-gateway voip interface
h323-gateway voip id gk-twister ipaddr 172.17.1.1 1719
h323-gateway voip h323-id gw-820
h323-gateway voip tech-prefix 1#
router eigrp 100
network 10.0.0.0
network 20.0.0.0
ip classless (default)
no ip http server
access-list 101 permit ip any any precedence critical
route-map data permit 10
set ip precedence routine
line con 0
exec-timeout 0 0
transport input none
stopbits 1
line vty 0 4
login
voice-port 1
local-alerting
timeouts call-disconnect 0
voice-port 2
local-alerting
timeouts call-disconnect 0
voice-port 3
local-alerting
timeouts call-disconnect 0
voice-port 4
local-alerting
timeouts call-disconnect 0
dial-peer voice 10 voip
destination-pattern ........
ip precedence 5
session target ras
dial-peer voice 1 pots
destination-pattern 5258111
port 1
dial-peer voice 2 pots
destination-pattern 5258222
port 2
dial-peer voice 3 pots
destination-pattern 5258333
port 3
dial-peer voice 4 pots
destination-pattern 5258444
port 4
end -
Blocking incoming collect calls in the voice gateway
Hello
I am using a C3825 router and I want to block incoming collect calls. I tried the command "double-answer" under cas-custom but it is not working. Does anyone have an alternative? I am using an E1 R2 digital.
Thank you
MarcosJonathad,
OP is not in the US, and does not have ISDN, has E1 R2 instead
E1 R2 has a method to block collect calls called double-answer. This method is supported and documented by Cisco.
But for some reason it doesn't work for OP.
In these case, it is necessary to involve an experienced consultant, if not TAC escalation directly. -
Spa3102 would not forward a voip call to pstn line
Good morning.
I've done the implementation provided here http://community.linksys.com/t5/VoIP-Adapters/SPA-3102-and-softphone-to-
make-calls-via-pstn-line/td-p/326390 .
It is a way to use for outgoing calls a given pstn line from anywhere I have internet (voip to pstn).
The spa3102 is connected to a router (with an active DHCP server and ip 192.168.1.1) from where it takes the internal
ip (192.168.1.3).On the same network is also a computer , connected to the router ( with ip 192.168.1.2). The spa3102
is set to bridge mode and thus inactivates the function of the router (on SPA3102), and it functions as a simple
network device . I have done port forwarding (from the router) to 192.168.1.3 (SPA3102) for the port 5061 (PSTN
LINE) ( but for 5060 for the LINE 1 also). I want to make calls from a voip softphone (x-lite 4) to the SPA 3102 and
this to forward the voip calls to PSTN line to which it is connected. In x-lite the SPA3102 is set as a proxy so that
i can type the phone number I want to call without being followed by the SPA3102's ip each time ( eg on x-lite I
give call number 2101111111 instead of 2101111111 @ wanip: 5061 where wanip is the external ip of the router).
When x-lite is running on the computer that is on the same network with the SPA3102 everything works as expected. A
voip call is made from x-lite ( using as a proxy the wanip everytime, or even for test purposes the dyndns domain
that i set up for this reason), this call is passese PSTN line and the phone of the called party rings . At x-lite
COMES indication "call established ".
The problem occurs when I do the same procedure from x-lite installed on a computer belonging to another network (
e.g. in another building with its own internet connection , own router, own computer , etc. ) . Always using the
wanip the x-lite makes the voip call to the SPA3102, writes "call established" ( meaning it connected to SPA3102) but
never routed the call to the called party ( the SPA3102 did not forward voip calls it receives to the PSTN line ) .
Trying to find what 's wrong I've tried to disable all firewalls (soft and hard from all involved machines ) . The
behavior is the same either the computer that makes the successful calls is connected to the network directly to the
router or through the port "ethernet" on the SPA3102.
What is the difference in these two voip calls to the SPA3102 and the one " triggers " it to forward the call to
PSTN line and the other does not ?
Thanks now for any ideas you give .The audio sound problem is more than likely also associated with the overall addressing problem initially encountered. As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets. In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets. The sound is exchanged by two separate streams of packets, one stream in each direction. The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.
In your previous posting you mentioned that you "set the minimum EXTernal rtp port at the sip tab". Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does. In my tests it didn't appear to affect the rtp port number used in a predictable manner.
The common changes to make for audio problems typically would be to setup a STUN server. A STUN server is an external server that echos back to the initial sender the external ip address and port number that the STUN server received with the message received by the server. This allows the sender (SPA3102 or softphone) to determine its external ip address and external port numbers for both the sip signalling and rtp packets.
A STUN server is commonly recommended to be setup with the following settings in the SPA3102:
PSTN Line Tab:
NAT Mapping Enable: Yes
Sip Tab:
Handle VIA received: yes
Handle VIA rport: yes
Insert VIA received: yes
Insert VIA rport: yes
Substitute VIA Addr: yes
Send Resp To Src Port: yes
STUN Enable: yes
STUN Server:
The following web page has a list of "Public STUN Servers"
http://www.voip-info.org/wiki/view/STUN
You are using CounterPath's XLite softphone. stun.counterpath.net is a STUN server on the list.
I see XLite also has a setting to use a STUN server on the "Topology" tab. -
Roaming charges for incoming international calls?
That's pretty much my question. I read somewhere on here that I would NOT be charged for long distance or roaming if my IPhone 5 receives incoming calls from Germany. Before I tell my family that they can call me, I want to be absolutely sure of that.
Thanks for your help!Just like a landline, an incoming international call will not have any add'l charges. But it will use minutes from your allotment, unless it is after 9pm (Or 7pm if you have an Alltel plan)
UPDATE: From VZW Website:
If someone calls me internationally from outside the U.S. will I be charged extra?
No, you won't be charged long distance or roaming fees if you're in the U.S and someone calls you from outside the U.S. -
Incoming & Outgoing calls number must view during making call & receive call
Under one contact name I have 4 to 5 Numbers (Local & International).
During the incoming & outgoing calls I would like to know that which no I receive and vice versa under that contact name.Called ATT...they verified the problem and said it was an intermittent signal problem....
only issue is I have to keep the phone off to make sure I received voicemails....
Definitely NOT an iPhone problem, but an ATT problem.
Maybe you are looking for
-
how do i send a pdf file i receieved in a email or can i go to my email and print it off
-
Hi I am on an iphone 5 IOS7 and when I look at the 'Silence' setting at the bottom of the Don Not Disturb' screen - then even though all of the settings on that screen are off i.e. not green - the 'Silence' -'Only when iPhone is locked' setting is ti
-
I have three Tables(Emptimesheet,comexpensetable,perexpensestable).I want to show otput something like below, EmpID,EmpName,StartDate,EndDate,Total,Chotel,CAirfare,Cfuel,WeeklCompanyExpenses,Percompanyexpenses rightnow in the below query I don't have
-
Event in maintainance view not working in View cluster
Hello All, I have created a view for a student table, which contains three fields. The third field should be automatically incremented, based on the entries in the first two fields. The functionality is working fine in the view created, a
-
Event loop based programming help in c#
Hello everyone . I was reading about node.js and how it does all the work using event loops using a single thread to do all the non-blocking socket I/O . I am trying to implement the same concept in my application which will be doing a lot of socket