Notifying PBX if Voip calls can't be completed from WAN

I have 10 2621XMs each connected directly to G3 PBXs. They supply VOIP call lines to the PBX via a direct T1 (NM-HDV-1T1-24). However, when the direct T1 is up between the PBX and the CISCO router, how do I tell the PBX that VOIP calls can't be completed? This, in it's simplest form, would be due to a Frame outage on the WAN side, or perhaps internal call issues in the cloud. Using basic configs, the PBX will only know it when the direct connect T1 is down not the WAN T1 on the other side of the router. Is there any proven ways to track other interfaces or routing and convey that to the voice-ports by seizing the unused ports or taking down the direct connect T1?

Hello there,
YES you can. The feature is known as "Busyout Monitor".
Please check at:
http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps1830/products_feature_guide09186a0080087b50.html
You can find gatekeeper enhancements for that feature at:
http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps1839/products_feature_guide09186a0080110ba7.html
Also for latest features available into that direction (monitoring packet loss, delay) please check the URL below:
http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps5012/products_feature_guide09186a00800879da.html
Those will surely help you to understand the feature.
REGARDS
Dragos

Similar Messages

ASA 5510 - how many concurrent VOIP calls can pass through?

Hi all,
I wonder how many concurrent VOIP calls can handle Cisco ASA 5510, any idea?
Gegham

hi Gegham,
Basically what the values of 50,000 and 130000 connections indicate are lab values tested with 80% TCP and 20% udp traffic. (according to table a-2 in the doc below)
http://www.cisco.com/en/US/docs/security/asa/asa72/configuration/guide/specs.html#wp1170941
RTP is udp traffic but in case of an asa and considering a customer scenario what happens is...
1 voip call = 1 control connection (h323,sip,sccp) + 2 or 4 rtp connections
-so a call will in total easily consume 5 or more connections depending on control connections you have set up .
-also this number differs depending on if the call is voice only or video.
So to simply answer your questions...
1>the number of connections that a call consumes depends on the above factors.
2>Also there is no hard number on the number of calls an asa can handle because this depends on the controls you use ...including nat and inspections.
Thanks,
Karthik

AS5300: VOIP call, can't listen to answering party.

I have a AS5300 running VOIP call without Gatekeeper, and my cisco is behind NAT.
Now the problem is, i can't listen to a particular country but answering party can hear me well.
More, this don't happen to the rest of the countries.
And, if tested with different AS5300 on else where IP (NAT) it's ok to this particular country.
How can i conclude it's not my error ?
P/S: i did a debug and found that all my voice is sent successfully to the answering party. But no return packet from them after connection.

I was getting the same problem , I did nat on router but it did work , so i used PIX instead of router. what PIX do is that it has the command fixup protocls which make it works.
let me explain you :
well H.323 put its source n destination packet inside the payload so when the distination router recieve the packet it looks to the inside one( payload adds) not the outer one that changed through natting. so the destination router doesnt know about the public add , he knows only abt the private address. so in this way the packets are not coming back.
Now I need to put router instead of router , can u explain me how is it working for other countries , it should works for any one.

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]

VoIP (UDP) and Video (mpeg4) Benefit from WAN Optimization?

From what I am reading VoIP and (Video)MPEG4 are "integrated" into a vendor's WAN acceleration/optimization app/device but I don't read VoIP or mpeg4 will benefit? Is this a true statement?

Sorry I have taken so long to reply but work got in the way.
This User Defined Firewall.
Can you tell us more about that.
Access Control = NO Valid Filtering Rule!!!
MAC Filtering Table = Disabled
URL Blocking = No entries
Schedule rule = No Valid Schedule Rule!!!
Intrusion Detection Feature
SPI firewall protection = ticked
Anti-DoS firewall protection = ticked
RIP defect = blank
Discard Ping To WAN Interface = blank
Stateful Packet Inspection
Packet Fragmentation = ticked
TCP Connection = ticked
UDP Session = ticked
FTP Service = ticked
H.323 Service = ticked
TFTP Service = ticked
When hackers attempt to enter your network, we can alert you by Email
Your Email Address :empty
SMTP Server Address :empty
Connection Policy
Fragmentation half-open wait: 10 secs
TCP SYN wait: 30 sec.
TCP FIN wait: 5 sec.
TCP connection idle timeout: 3600 sec.
UDP session idle timeout: 30 sec.
H.323 data channel idle timeout: 180 sec.
DoS Detect Criteria:
Total incomplete TCP/UDP sessions HIGH: 120 session
Total incomplete TCP/UDP sessions LOW: 115 session
Incomplete TCP/UDP sessions (per min) HIGH: 115 session
Incomplete TCP/UDP sessions (per min) LOW: 110session
Maximum incomplete TCP/UDP sessions number from same host:30
Incomplete TCP/UDP sessions detect sensitive time period: 900 msec.
Maximum half-open fragmentation packet number from same host:20
Half-open fragmentation detect sensitive time period: 10000 msec.
Flooding cracker block time: 300 sec.
DMZ = disable
There are NO ports set.
The NAT Mapping table is 16 pages long.
SIP/VoIP only mentions 5060 with no ranges
Well I hope that is the info you wanted
Cheers
Message was edited by: greatfinewine
Message was edited by: greatfinewine

N97 FW 21.0.0.45 voip calling issues

Yesterday,I have upgraded my firmware to v21.0.0.45 and since then I am facing difficulties in making an Internet call(Voip calls).
Problem is first voip calls will go fine but from 2nd time call will not at all connect.I have to restart the phone again.After restart again first call will go fine but 2nd one will again not connect.
My phone is N-97 type RM-505 having custome version 21.0.045.c01.01. Please help me out to solve this issue.With the previous versions voip calls were going fine... Nokia please release a patch for this urgently.This is not at all acceptable... Solve one bug insert another...We need urgent patch for this.
All,Please feel free to give your views

Hello, i'm in the same case, since upgrade to V21.0.102, I can't use voip for send calling, but i can only receive call.
The password, login, and paramètre are ok
If i go in "lastest error Sip" i say me error 403 méthode: invité
Somebody have a solution ?
I need it a lot,
Think a lot, for your help,
Didier.

Can't make voip calls 5060 blocked?

I can't make voip calls
rkmac-wl:~ rkaminen$ telnet jfk-primary.voicepulse.com 5060
Trying 64.61.93.190...
^C
rkmac-wl:~ rkaminen$ telnet jfk-backup.voicepulse.com 5060
Trying 67.108.9.165...
^C
rkmac-wl:~ rkaminen$ ping jfk-primary.voicepulse.com
PING nycproxy02.voicepulse.net (64.61.93.190): 56 data bytes
64 bytes from 64.61.93.190: icmp_seq=0 ttl=47 time=16.363 ms
64 bytes from 64.61.93.190: icmp_seq=1 ttl=47 time=16.008 ms
Request timeout for icmp_seq 2
64 bytes from 64.61.93.190: icmp_seq=3 ttl=47 time=17.725 ms
^C
I can ping my voip provider but can't reach port 5060
Is verizon deliberately doing this?

verizon doesnt block that port....
if it was down for 30 mins, and internet was working at the time.. it is most likely that the VOIP was at fault
Bob Robertson - Lighter Klepto - I steal lighters, not bandwith

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 ?

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.

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 Healy

Hi,
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

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.

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 help

Hi 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

VOIP call Failed (Error Code 15) in Prime Event log.

I'm seeing the following events in Prime Infrastructure related to Cisco 7925 phones.
VOIP Call Failed for "(Error Code 15) occurred on client "xx:xx:xx:xx:xx:xx' which was associated to AP <APName> on interface 802.11b/g.
I can't seem to find where this error code 15 is defined. The closest I was able to find was error code 15 under gatekeeper codes in the following document which may or may not be relavent.
http://www.cisco.com/en/US/docs/ios/voice/monitor/configuration/guide/vt_voip_err_cds_ps6350_TSD_Products_Configuration_Guide_Chapter.html
Any insight into this would be appreciated.
Jake
P.S. No voip call jokes on 802.11b/g please

It had 18Mbps set as the min phy rate. I also think it has to do with CAC issues at 18Mbps that are pretty well noted and differeing mandatory rates across the 802.11b/g/n and 802.11a/n networks on the controller. When the phone would roam onto a radio with 18Mbps was the min mandatory rate, it would drop and give this error.

Redirecting Voip call to another termination router

Hello All
The structure is very simple: There is one originating router (as5300 let say A) and
one terminating router (as5300 let say B) with 1 pri line. I want to route some calls whose have specific destination pattern from B to another router (let say C) to be terminated.
I have used "dial-peer voice xx voip" with "session target ipv4:x.x.x.x. on B.
But it failed. When I debug it on B ,it gives " can not gateway with C".
Do you think the idea is wrong for this structure . or how can I do this ( PS :I can administrate only termination routers )
thanks in advance..

to tapatel:
First of all thank you for your assistance
Router B is not making call origination, it is only termination router.
Router A (I do not manage router A), is sending voip calls to B and I want to route some calls which has specific destination pattern from B to C. The configuration for B is like this:
! the dial-peer 101 and 99 below is working properly
dial-peer voice 101 pots
destination-pattern 3054T
port 0:1
prefix ,054
dial-peer voice 99 voip
destination pattern 3054T
session target ipv4:10.2.2.2 (the ip address of router A)
! I want to route it to router C with the lines below
dial-peer voice 102 voip
destination-pattern 30549238.....
session target ipv4:10.10.1.124 (this is the ip address of router C)
May be I have to use gatekeeper for this project but i am not sure if it is necessary.

Set up VoIP call termination

Hi, my client is send to SIP G711, G729 Voip calls through the internet, how can I use cisco router 2811 for terminate the calls? I do not have a PSTN. Thanks

You just need to take one minute to search CCO for "SIP trunk configuration" to get the docs
Cisco CallManager Express (CME) SIP Trunking Configuration Example
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk

Notifying PBX if Voip calls can't be completed from WAN

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