MGCP Dialling Problem

I have a CCM 4.0 and an MGCP gateway cisco 1760 with VIC-2FXO. I can receive the calls from both IP network and PSTN but i am unable to call out to the PSTN. What could be the problem?

There could be several things causing this, here are the troubleshooting guides for callmanager:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guides_list.html

Similar Messages

Voice dialling problem in N97

Among myriads of problems with my N97 the freshest one is really annoying. As you know, N97 is supposed to automatically tag all the contacts with a voice tag allowing them to be dialled by voice. The contact which I use the most has lost its voice tag. It can't be dialled in this way and when I choose Options>voice tag details it says that "There are no voice tag details for this contact". I've tried everything: erasing this contact and entering it from scratch, restoring all contacts from the copy and with no results.
What's more, I've noticed that any new contact I enter also lacks its voice tag.
I wonder if any of you has ever met such a problem and perhaps someone knows a way to solve it.

Same issue here. Nokia N97 mini with latest firmware. 959 contacts.

MGCP flow problem - ASA

The following is the setup and the problem we have:
* MGCP call agent -- ASA 5510 -- vpn tunnel -- linksys -- MGCP gateway
* VPN tunnel setup is in aggresive mode (initiated as needed from linksys).
* When MGCP call agent is up but the vpn tunnel is still down, MGCP traffic (udp/2427) tried to reach gateway and failed --> this is as expected.
* Next, linksys brings up the vpn tunnel
* Any other IP traffic can go through the tunnel, but not MGCP traffic from call agent to gateway.
* Checked from packet tracer: it passed phase 1 & 2 (checking the flow), it found existing flow and used that flow. Then it stuck there, looks like it didn't know where to go next (no ACL or route checking).
We noticed that the annoying following workaround will make the MGCP goes through.
* shutdown MGCP call agent
* disconnect vpn tunnel from linksys
* system reload the ASA5510
* after ASA5510 is up, linksys initiate the vpn tunnel
* after vpn is up, bring up MGCP call agent
* MGCP traffic goes through the tunnel
It seems like ASA MGCP flow table is not updated if there is status change in VPN tunnel. Any idea how to fix this ?

Thanks for the reply.
However that wasn't mgcp ispection issue, it was more on dynamic vpn configuration issue. The setup is dynamic vpn initiated from remote sites (aggresive mode).
Worked with Cisco TAC and got the workaround. Following is the summary, hope it's useful.
1. Previous dynamic vpn config was:
crypto dynamic-map outside_dyn_map 20 set transform-set VPNHigh
When the tunnel was down and the call agent (central) was sending traffic to the call gateway (remote), the ASA created a connection to send the traffic out in the clear as it didn't match any crypto access-list due to the tunnel being dynamic.
2. The call agent (cetral) continued to attempt to send traffic to the call gateway (remote), which kept the conn up, even when the tunnel came up.
3. When the tunnel came up, the ASA continued to use the existing conn and sent the traffic out in the clear.
4. This is very similar to bug: CSCse36327 - GRE Packets are no longer encrypted and are sent out in the clear
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCse36327
Even though that mentions GRE, it's the same issue.
5. After the tunnel was up and "clear local-host " was issued, the conn was torn down and a new conn was built to go over the tunnel.
6. To avoid this situation, a crypto acl for this dynamic tunnel was created:
access-list dyn-mgcp line 1 extended permit ip
crypto dynamic-map outside_dyn_map 20 match address dyn-mgcp
crypto dynamic-map outside_dyn_map 20 set transform-set VPNHigh
crypto dynamic-map outside_dyn_map 40 set transform-set VPNHigh
Now, when the call agent (central) sends traffic to the call gateway (remote) when the tunnel is down, the ASA drops the packets and doesn't build a conn until the tunnel is brought up.
7. This was tested by bringing down the tunnel and watching incoming packets from the call agent (central) to the call gateway (remote) and verified they weren't being sent out in the clear. The tunnel up was brought up and traffic started being passed between the call agent and the call gateway.

External drive refuses to work and the other is very slow

I was not sure if I should put this in Lion or macbook pro section since thats what I am trying on (mid 2010) version.
Hi,
I purchased maybe 4-6 months ago a Toshiba E05A100SBU3EW_C STOR.E ART 4 1TB USB 3.0 2.5 inch Hard Drive - Black from amazon and one day it just suddenly stopped working and I was wondering if any one knew how to get it working again on my mac or windows machine, please as it has a lot of important data
it is this hard drive: http://www.amazon.co.uk/gp/product/B004NE76KQ/ref=oh_o01_s00_i00_details
it makes no sounds or noises on my mac but when I plug it into my windows machine it does make a noise but then there is nothing in my computer and it will not recognise the device at all
Toshiba E05A100SBU3EW_C STOR.E ART 4 1TB USB 3.0 2.5 inch Hard Drive - Black
warranty is valid but I really hope I can save the files
I am getting a message from my macbook pro saying that its disconnecting it due to too much power or not enough, why would it say this and how can I re enable the drive?
Also I have a second DW or is it WD I forget but it works but its just runing 10 times slower then normal and I cant figure out why as I never touched them they just one day started acting this way after I had my macs scren replaced one of the ports is 3.0..
do you think the guy could of swapped out the usb t a 2.0 one? but then why would my mac keep telling me that its ejecting the drive

Have you tried updating your software? The issue with the keys will either be a software issue or an issue with your hardware, if you're on the newest software try re-installing it. It's very simple to do using the desktop software, if this does not work it seems to be a hardware issue, meaning repair or replacement is needed.
Also, updating/re-installing your software should also resolve any slow/freezing issues and hopefully stop your dialling problem.
Hope this helps
Adam
EE Blackberry Specialist

I just instaled firefox 4 on my Acer netbook with Windows Vista Basic. Now the typing is very slow, ie. cannot keep up with my typing speed, which is not fast. Any ideas?

I have an Acer netbook, with 2GB ram and Vista Basic. I installed firefox 4 and now the machine cannot keep up with my typing. The letters are delayed appearing on the screen in all apps, but especially in email, and words just get lost if I type too fast, and I am not a fast typist. Should I remove firefox 4?

Have you tried updating your software? The issue with the keys will either be a software issue or an issue with your hardware, if you're on the newest software try re-installing it. It's very simple to do using the desktop software, if this does not work it seems to be a hardware issue, meaning repair or replacement is needed.
Also, updating/re-installing your software should also resolve any slow/freezing issues and hopefully stop your dialling problem.
Hope this helps
Adam
EE Blackberry Specialist

Confused by basic SIP Trunk configuration.

I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing. I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
If someone can make some sense of this, I'd truly appreciate it!

Hello Aditya and thanks for the consideration!
I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production.
I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
==============================================
Branch_SIP#show run
Building configuration...
Current configuration : 3529 bytes
! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Branch_SIP
boot-start-marker
boot-end-marker
! card type command needed for slot/vwic-slot 0/2
enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 10.0.10.1 10.0.10.10
ip dhcp excluded-address 10.0.30.1 10.0.30.10
ip dhcp pool Data
network 10.0.10.0 255.255.255.0
default-router 10.0.10.254
option 150 ip 192.168.5.250
dns-server 192.168.5.200
ip dhcp pool Voice
network 10.0.30.0 255.255.255.0
default-router 10.0.30.254
dns-server 192.168.5.200
option 150 ip 172.16.0.1
ip dhcp pool data
option 150 ip 172.16.0.2
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
bind media source-interface Loopback1
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2851 sn FTX1031A2FM
redundancy
interface Loopback1
ip address 2.2.2.2 255.255.255.255
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 10.0.10.254 255.255.255.0
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address 10.0.30.254 255.255.255.0
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/3/0
no ip address
shutdown
clock rate 2000000
interface Serial0/3/1
ip address 172.16.0.1 255.255.255.0
clock rate 250000
interface Internal-Service-Module0/0
no ip address
shutdown
!Application: CUE Running on AIM2
hold-queue 512 out
router eigrp 1
network 0.0.0.0
network 2.2.2.2 0.0.0.0
network 10.0.0.0
network 10.0.10.0 0.0.0.255
network 10.0.30.0 0.0.0.255
network 172.16.0.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 172.16.0.2
tftp-server flash:term45.default.loads
tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
tftp-server flash:cnu45.8-5-3TH1-6.sbn
tftp-server flash:apps45.8-5-3TH1-6.sbn
tftp-server flash:dsp45.8-5-3TH1-6.sbn
tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
control-plane
voice-port 0/0/0
voice-port 0/0/1
mgcp profile default
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
codec g711ulaw
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
sip-server ipv4:192.168.5.250
telephony-service
codec g711ulaw
max-ephones 24
max-dn 48
ip source-address 172.16.0.1 port 2000
system message SIP Branch Site
cnf-file location flash:
load 7960-7940 P00308010200.bin
max-conferences 8 gain -6
transfer-system full-consult
ephone-dn 1
number 4008
ephone-dn 2
number 4005
ephone 1
device-security-mode none
mac-address 001D.A21A.2065
button 1:1
line con 0
exec-timeout 0 0
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
speed 115200
line vty 0 4
password cisco
login
transport input all
line vty 5 15
password cisco
login
transport input all
scheduler allocate 20000 1000
end
Branch_SIP#show debug
TFTP:
TFTP Event debugging is on
CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)
Branch_SIP#
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4B6C5C28
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 4008
Called Number            : 5005
Source IP Address (Sig ): 172.16.0.1
Destn SIP Req Addr:Port : 192.168.5.250:5060
Destn SIP Resp Addr:Port : 192.168.5.250:5060
Destination Name         : 192.168.5.250
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 2.2.2.2
Source IP Port    (Media): 19472
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503
Branch_SIP#

Letter "E" will not work and the phone is very slow.

HI Everyone,
I would greatly appreciate the help of anyone who can help me solve my issse with my phone. i have a blackberry 8520 which was working great until yesterday.
Now i cant do the following:
1. Press the lettee E as it does not work
2. The ALT key when pressed does not work which means i cannot use a comma, full stop etc..
3. The phone seems to have a life of its own where it automatically speed dials the no 2 (letter E key) wheneever i just scroll my finger on top of it.
4. It also is quite slow and keep coming up with the jar telling me to wait. This is not soo much an issue, but just means i have to wait for the processor (in the phone if any) to get its act together.
i would really appreciate any insights you may have available.
thanks, Niraj.

Have you tried updating your software? The issue with the keys will either be a software issue or an issue with your hardware, if you're on the newest software try re-installing it. It's very simple to do using the desktop software, if this does not work it seems to be a hardware issue, meaning repair or replacement is needed.
Also, updating/re-installing your software should also resolve any slow/freezing issues and hopefully stop your dialling problem.
Hope this helps
Adam
EE Blackberry Specialist

My new updated N73(ME)has problem with Voice Diall...

Hellow to every one,
I ve just take my N73 from NOKIA service center, after 2 months im waiting for. The 1st problem was that after the latest firmware update my phone doesnt start at all. The 2nd problem was and still is that the voice dialling never works. My contacts have a nick each and are stored at phones memory. At service center says that every problem was fixed and for that they send my phone from Greece (where i live) to Sweeden (i think). What else can i do to make works voice dialling? Is that a bug of my phone so i must live with it?
PLEASE HELP!!!!
Regards
V 3.0705.1.0.31
26-01-2007
RM-133
Nokia N73 (07)

There is no volume adjustment that can be made.
Do you have a cover on you iPad that may be covering the mic hole?
If you frequently use FaceBook, you might consider getting an external microphone that plugs into the headphone connector.
 Cheers, Tom

MGCP problem with Marconi Softswitch

I have a problem configuring MGCP on Ciscos Residential Gateway(IAD 2430)wit Marconi's Softswitch.I have two main issues with 2430. Ineed to know :
1- How can i set MGCP ver 0.1 with NCS 1.0 in IAD2430 as this is what is expected from Softswitch side.
2- How can i set MGCP endpoint Name on 2430 as per my own convention ?
I want to configure things on the residential gateway side and want to avoid any changes on my softswitch side.Can anyone shed some light on my problems ?
Thanx
Tanveer

This url shoul help you, It speaks about Cross-Platform Release Notes for Cisco IOS Release 12.2 T, Part 3: New Features and Important Notes:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/prod_release_note09186a0080087e11.html
Understanding prov-sync, prov-dply, and pom.dataSync and troubleshooting PGW standby OOS issues :
http://www.cisco.com/warp/public/788/products/prov_sync.html

Vg224 mgcp problem

Good morning, I've a callmanager 8.6.2.
All works well excepet a vg224 mcgp .
The vg is correctly registered via mgcp on callmanager, and I've configured one fxs port to connect to an analogic phone.
The problem is: if I start a call from analogic phone to ipphone the call wors well, but i I start a call from an ipphone to the analogic phone the call fails.
Eachone can help me?
in the attached file the vg 224 config
interface FastEthernet0/0
ip address 10.116.1.27 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip default-gateway 10.116.1.1
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.116.1.1
no ip http server
control-plane
voice-port 2/0
voice-port 2/1
voice-port 2/2
voice-port 2/3
voice-port 2/4
voice-port 2/5
voice-port 2/6
voice-port 2/7
voice-port 2/8
voice-port 2/9
voice-port 2/10
voice-port 2/11
voice-port 2/12
voice-port 2/13
voice-port 2/14
voice-port 2/15
voice-port 2/16
voice-port 2/17
voice-port 2/18
voice-port 2/19
voice-port 2/20
voice-port 2/21
voice-port 2/22
voice-port 2/23
ccm-manager redundant-host 10.116.1.13
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.116.1.22 10.116.1.13
ccm-manager config
mgcp
mgcp call-agent 10.116.1.22 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0
mgcp profile default
dial-peer voice 99920 pots
service mgcpapp
port 2/0
dial-peer voice 99921 pots
service mgcpapp
port 2/1
dial-peer voice 99922 pots
service mgcpapp
port 2/2
dial-peer voice 99923 pots
service mgcpapp
port 2/3
dial-peer voice 99924 pots
service mgcpapp
port 2/4
dial-peer voice 99925 pots
service mgcpapp
port 2/5
dial-peer voice 99926 pots
service mgcpapp
port 2/6
dial-peer voice 99927 pots
service mgcpapp
port 2/7
dial-peer voice 99928 pots
service mgcpapp
port 2/8
dial-peer voice 99929 pots
service mgcpapp
port 2/9
dial-peer voice 999210 pots
service mgcpapp
port 2/10
dial-peer voice 999211 pots
service mgcpapp
port 2/11
dial-peer voice 999212 pots
service mgcpapp
port 2/12
dial-peer voice 999213 pots
service mgcpapp
port 2/13
dial-peer voice 999214 pots
service mgcpapp
port 2/14
dial-peer voice 999215 pots
service mgcpapp
port 2/15
dial-peer voice 999216 pots
service mgcpapp
port 2/16
dial-peer voice 999217 pots
service mgcpapp
port 2/17
dial-peer voice 999218 pots
service mgcpapp
port 2/18
dial-peer voice 999219 pots
service mgcpapp
port 2/19
dial-peer voice 999220 pots
service mgcpapp
port 2/20
dial-peer voice 999221 pots
service mgcpapp
port 2/21
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999223 pots
service mgcpapp
port 2/23

Duplicate
https://supportforums.cisco.com/discussion/12245471/problem-vg224-mgcp

Problem dialling certain digits with Lumia 800

In order to find out how much pre-paid phone credit I have with my carrier, I can dial #123#, which works with my SIM in my iPhone (unlocked) and I get a response giving my available credit. However, when I try to dial this combination of digits with the same SIM in the Lumia 800 (unlocked), I always get the message 'failed'. Can anyone suggest how I might dial #123# on the Lumia? Thanks

Hi GeoffN,
Usually service codes are in the format *[number]#
For instance by dialling *100# on my Lumia 800 I will get my calling credit with my carrier.
Hope this helps,
Kosh
Press the 'Accept As Solution' icon if I have solved your problem, click on the Star Icon below if my advice has helped you!

Problems with MGCP

Im having some issues getting MGCP to register on our CCM (Version 8.6.2.21900-5) with an 800 series router (Specifically a C888SRST-W) if i bind the BRI interface to the CCM, Layer 2 will not activate. If I remove the binding command then the link activates as expected however I need this bound to the CUCM manager.
On the 888 router, the BRI port is known as BRI4 rather than the expected 0/0/0 format. On the CUCM the only options are 0/0/0 etc. Could this be the source of my problems? We have many 1800/2800/3800 series routers with BRI modules where this config works fine.
Thanks
Craig

If you add a pots dial peer with service mgcpapp and the port number it will work.

Problem when dialling with Xperia Mini x10

Hi,
I am having issues with my phone when dialling numbers in that I don't hear a dialling tone or the person I am calling, so unsure when I am connected to my contact other than by looking at the timer on screen. I can hear the dialling tone through headphones and when the load speaker is on, but that's not any good when you're having a private call.
I have tried resetting factory settings, but this has not changed a thing. Any suggestions on how to resolve or does this mean my phone is broken?
Thanks

Hi,
I noticed your post hasn't had any responses. Are you still experiencing these problems?

MGCP with FXO Problems

I have configured MGCP to access an FXO port used for our paging systems.
When I dial the access code, I get the Intercom Tone briefly then the audio goes away and I am showing connected. The voice port status even shows the interface as being up. When I hang up it shows idle.
Any Ideas?
We are running 12.2(15)T on the router. 2651

After alot of debugging, I found the problem, but it doesn't make sense. I had to change the route pattern to remove the digits predot.
Example *5.
It was sending the *5 to the Intercom.
Can anyone enlighten me to the below the command?
mgcp dtmf-relay voip codec all mode out-of-band
I thought this command on the router removed the digits sent to an MGCP Gateway?
If I am wrong, please explain. Any explaination would help me.
Jason

Hi ... Suddenly I have this problem with my iPhone4 . Whenever I am dialling or receiving calls, I am not audible at the other end , but if I am putting phone on speaker then I am audible to other person .Not sure if any setting is changed or device issue

Hi ... Suddenly I have this problem with my iPhone4 . Whenever I am dialling or receiving calls, I am not audible at the other end , but if I am putting phone on speaker then I am audible to other person .Not sure if any setting is changed or device issue. Please suggest or help.

There's an Apple help page on the DNS problem - at http://support.apple.com/kb/TS3408
The first thing to try is changing your DNS settings - and for that you need to open System Preferences, then the
Network section.
In Network - click on Advanced - then the DNS tab
under DNS servers - use the + button to enter two extra lines
either these, from google dns
8.8.8.8
8.8.4.4
or from OpenDNS
208.67.222.222
208.67.220.220
then click OK & then 'Apply'
You would need to do that seperately for the Cable (Ethernet) settings and Wireless (Airport)

MGCP Dialling Problem

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