Voice Call issues

Hello,
We have Exchange 2010 and Blackberry 5 in our environment. This query is for a single user(User A)only who is having exchange active sync and Blackberry.
When users call him via Blackberry address book, the call lands to user B instead of User A.
If the user is called directly from voip, landline the issue does not occur.
The issue is observed only if any user calls this user A to his blackberry. However, the issue is not observed on I phone. The phone has been wiped alreday.
Regards
Ajit

Can you share the SIP dialog between CUCM and the MCU? INVITE, 100, 180/183, 200 OK, ACK?
CUCM can do the packet capture directly from the CLI as long as you run it from whatever node the SIP trunk is being sourced from (primary of that trunk's CMG).

Similar Messages

IPhone 4 voice call issues

My iPhone 4 sometimes goes into Facetime in the middle of a voice call. I don't have a screen protector that might block the proximity sensor. I have done a full reboot and am on the latest iOS. Any ideas what is wrong?

I've been having a very similar issue that just started last night. I keep dropping calls and when I try to call out nothing happens at all. I cannot browse Safari through 3G either. It keeps coming and going. I was able to finally place a call out and then tried making another call later and it wouldn't work again. I tried the restore as well and nothing works!
I do not know if I should contact AT&T or Apple for the issue. I do have the extended warranty that I purchased with the phone but about a week after I got the iPhone 4 I had to have it replaced for issues I am not even sure that they would replace it again. I do not have a home phone I rely solely on my cell so I cannot wait to have it fixed or replaced but I am not going to keep a phone that doesn't work especially after paying so much for it!

Voice call issue

I have had my Pixi for just over a week. I've noticed that during voice calls on my pixi (talking to someone or even calling my voicemail, both w/ speakerphone on or off), there is some weird noise while the other person is speaking (not sure how to describe it, other than its kinda of a static sound & there isn't any distortion of voice). It's distracting & makes it hard to understand everything that is being said (not super bad though but still). The best way I can describe the noise it that it is like the volume is turned up to loud, but it does it no matter what volume it is set at, just less so if lower & more so when someone enunciates something more.
There is no issue when I watch YouTube videos or vids I took with the phone or Sprint navigation, or ringers. Only during calls. I do not use Bluetooth or anything like that, have not even used headphones with the phone. I should have good signal, as I never had issues before I got this phone & although sometimes it says my signal is low, it still works (the internet, apps, etc) just fine. But the signal varies (sometimes as much as a 3-4 bar difference) sometimes even when I am staying in the same spot.
I've looked around to see if anyone else has a similar issue w/ a solution & can't seem to find an answer. Any ideas? I would greatly appreciate it!

Hello raelynjo and welcome to the forums...
If the other applications are using the speaker fine. Does this issue happen when you use the speaker on a call? If so, it could be the device. Otherwise, if the issue only happens when your on a call, whether using the speakerphone, headset, or hold it to your ear, it could be the carrier.
I would first try a restart on the device through the Device Info> Reset Options> Restart.
There is also an interactive test for the audio on the device. If you go to Device Info> tap on the top left Preferences> Tests> Interactive Tests> Audio. This will test your device for all the audio outputs.
If you have a failure or any errors reported from the tests, you can send the report to Palm through the device. (instructions will appear if the test fails)
I would consider replacing the device if it is a device info. It sounds like a hardware issue or a carrier issue.
Thanks,
Colapop

HD voice call issue

Hi
Hoping someone can help as I've tried calling 150 and they suggested a new sim but that hasn't worked.
So the issue is that both myself and my partner are on EE, I have an iPhone 5s and he has an iPhone 6. Both of us can successfully hd voice call other people on EE but when we call each other it's just a standard call and won't go in HD. Both of us are in a strong 3G area when testing. Any ideas how this can be fixed?
Thanks

Basic troubleshooting steps outline in the manual are restart, reset, restore from backup, restore as new. Try each of these in order until the problem is resolved. If you've been through all these steps and the trouble persists, you'll need to bring your phone to Apple for evaluation. Be sure to make an appointment first at the Genius Bar so you won't have a long wait.
Best
GDG

IPhone 5s Voice Call Issues

I've had an iPhone 5s (upgraded to 8.1.3) since mid-November and have been experiencing difficulties on voice calls ever since (I switched carriers as well). When calling specific contacts (family, friends etc.), the calls sound as though as I am on speakerphone, and there is a great deal of static and distortion present as well. The beginning of the call sounds as though the network is about to drop it before recovering it in a much lower quality. To the best of my knowledge, I have tried absolutely everything at this point, including: swapping the handset (three times), sim card (three times), resetting the device and even changing my number (was informed that it could have something to do with the "porting" of a number profile from one carrier to another). None of these moves have worked and I am still trying to decipher why I am encountering this problem (I have been in constant contact with my carrier who appears to have exhausted all ideas).
Any suggestions or thoughts are greatly appreciated.

I was having the same problem and also loosing battery life - it suddenly happened. I went online and saw where someone said to get rid of all the unused apps still sitting there, deleted all the old posts and yea! suddenly the problem has been fixed. Hadn't deleted the unused apps in quite sometime, so obviously draining everything.

IPhone 5S: does anybody has this problem. During a successful voice call all of a sudden people can't hear me while I can hear them

iPhone 5S: does anybody has this problem. During a successful voice call all of a sudden people can't hear me while I can hear them. Signal is strong and call quality (me hearing the other) is still very good but it looks as if I’m going on mute while I’m not muting my iPhone. When dropping the call and re-dialing it works fine again. It’s a brand new iPhone 5S but it now happened already 4 or 5 times in 2 days.
It feels like a hardware/iOS problem and not a network issue. It happend with iOS7.03 but I just see 7.0.4 is available so will install that one now.
Thanks in advance for feedback
Best regards,
Marco

Hello Jeff,
Thank you for providing so much detail about the issue you are experiencing with the audio on your iPhone. I would be concerned too if the audio on my iPhone was not working unless using speaker phone. I found an article with some steps to take when you are experiencing issues like this:
iPhone: Can't hear through the receiver or speaker
http://support.apple.com/kb/TS1630
Thank you for using Apple Support Communities.
Best,
Sheila M.

IPhone 5 voice call quality fade in and out. Noise Cancellation interference?

I've been having this huge problem with the iPhone5. The evidence i've collected seems to indicate this is a problem with all iPhone5's.
On calls in certain places both indoors and outdoors, the callers voices over the iPhone5 fade in and out. Their voices occasionally completely fade completely away, and only come back momentarily. The low volume for 50% of the time forces me to raise the volume, but raising the volume does not help because when the voice does come back, the volume is too loud for my ears.
I went back to the Apple store and they are not able to help me. They suggested a clean restore, which I did. They offered to replace the phone, which I did, but it is the same problem.
- iPhone 5 vs. iPhone 4S, the fading does not occur. I make the call to the same voice mail system I have, and the sounds from the other side are consistently the same volume for the 4S, but fades in and out for the 5.
- I connected up a 8mm jack between my iPhone 5 and my mac in an attempt to record this problem. In that instance, the sounds from the voice mail DO NOT fade in and out. I did find out that the dynamic range on the iPhone 5 is MUCH greater than that of the iPhone 4S, for some strange reason.
- When I hold the phone up to my ear, the fading in and out does occur, but it is less noticeable. But it's still there.
- When I leave my work area the problem goes away. However, because the iPhone 4S on the same AT&T service does not have this problem, this leads me to believe it's NOT a lack of transmission power from the AT&T cell tower. Rather, it is a reception issue at the iPhone5 side.
- I turned off LTE, data and then made the calls. Same issue.
I use this phone to make 60 minute calls for conference calls frequently. I cannot use it. Please advise. I am seriously considering returning this phone. Except i've sold my old iPhone 4 already. I'm in a bad place. Please help.
Here's another post of a fellow with the same exact problem.
http://forums.macrumors.com/showthread.php?t=1456404
Apple, please look into this. I believe this is a problem with the noise cancellation software.

I've got an update on this problem with fading volume. I've isolated it to the set volume on the phone for voice calls when using headphones. When the headphone volume is set between 1 and 6 (out of 16 squares) the callers volume will consistently fade in and out. The fading which is normal to zero volume at 2-5 Hz is serious enough that normal conversation is impossible. When the volume is raised above 6/16 then the callers voice stabilize and I can hear all their words again. Unfortunately, at 6/16 volume their voices are too loud for the headphones.
Furthermore, when headphones are plugged in, and I initial a call after the iphone5 was idle for several minutes, the first dial tone has preceded by a very loud pop an the tone, as though there was too much pressure (charge) that was dissipated all at once.
Other details:
- I used 3 different earphones, iPhone 4 headphones, altec earphones, an the new iPhone 5 earbuds. Same.
- I turned on and off lte, data, Bluetooth, wifi. Problem persists unaffected.
- when switching to iPhones earpiece and mic, the fading is gone.
- I have 3 out of 5 bars where I teste this, in the city outside and on hwy 5 in central California. Same symptoms in all places I've tested.
- I have exchanged my phone at the apple store. Same issue with both iPhones an I suspect all ipjone5s.
- the ipjone4s does NOT have this problem when the volume is set lower. I can tell that the iPhone 5 must have a new audio filter to make the sounds have larger dynamic range. I liked the 4s better, because it is easier on the ears!

IPhone 4 voice call problems on 3G

I noticed at the weekend that my iPhone 4 was having trouble with voice calls. Incoming calls would sometimes hang up when I answered them, outgoing calls would take a very long time to connect (or freeze), and when I did get through to someone the call quality was extremely low (lots of stuttering and cutting out).
After trying just about everything recommended on various help pages (including reseating the sim card, refreshing the network settings, and restoring the phone), I still could not get it to work properly. I then tried turning off the 3G and, lo and behold, the phone worked absolutely fine! Voice calls were clearer than ever!
I contacted Tesco Mobile hoping they might be able to shed some light. They said there are no faults with the network and have arranged for me to return my phone. That could take up to 10 days though, which I can't really afford.
Does anyone know what the problem is here? Does the fact that it works fine with 3G disabled suggest that it is not a hardware problem?

I've been having a very similar issue that just started last night. I keep dropping calls and when I try to call out nothing happens at all. I cannot browse Safari through 3G either. It keeps coming and going. I was able to finally place a call out and then tried making another call later and it wouldn't work again. I tried the restore as well and nothing works!
I do not know if I should contact AT&T or Apple for the issue. I do have the extended warranty that I purchased with the phone but about a week after I got the iPhone 4 I had to have it replaced for issues I am not even sure that they would replace it again. I do not have a home phone I rely solely on my cell so I cannot wait to have it fixed or replaced but I am not going to keep a phone that doesn't work especially after paying so much for it!

Iphone 4S - Voice Call

Hi all,
When I call a people by my iphone 4S, there is a problem of sound.
My voice, after a few minutes, becomes metallic and i listen to a series of sound noise.
I tried to update software version to IOS 5.0.1, but there is the problem yet.
How I can resolve this problem ?
Thank you,
Best regards

OK, so i have tracked it down to a single AT&T Cell Tower location in my instance, but it also just so happens to be at my office, which means I am affected by it all day. What is happening is that at this particular location, the AT&T tower is misconfigured or experiencing some type of technical issue, resulting in the inability of the cell tower to handle simultaneous voice and data. Siri uses data to match the contacts, names etc and hands off to the Phone dialer, which immediately experiences a call failure as a result of the 3G Data still being in use by Siri. You can verify simultaneous voice and data connectivity by placing a voice call and then attempting to use 3G data (you must not be using WiFi to determine if this is a problem with the Cell network). While in a voice call, launch the Speedtest.net Mobile Speed Test app. You will clearly see that 3G data services have all but ceased if this is your problem. End the Voice Call and run the Speedtest.net Mobile Speed Test again and 3G data will work fine.
I have reported the problem via AT&T's Mark the Spot App numerous times and had my colleagues do the same and called AT&T Wireless customer service to no avail. They simply refuse to acknowledge the issue is on their end. I personally know at least 5 other iPhone 4S users in my office who are experiencing the same issue and the results can be consistently recreated on all of their devices. And again, once they leave the office in the afternoon, all of their services return to normal and they can again make and place Voice initiated phone calls without the dreaded Call Failed. Thanks AT&T.

IPhone 5s continue ringing after answering the voice call

Hi,
After I installed the update iOS 8.0.2 my iPhone 5s continues ringing 0.5-1 sec more after I answered the voice call.
What can I do to fix it? It is pretty uncomfortable, because I use to swipe and answer, but now phone continues ringing in my ear for a some time.
Thanks in advance!

This has just started happening to me as well! After two months of relatively non-faulty use, now this affect. Really rattles me, too.
Yesterday I got a warning triangle out of the blue that said the phone was shutting down due to overheating, but it was stone cold! I reset then and it has not re-appeared. Never saw that with two years of use on my iPhone 3G. Now this affect, they must be related, but does this mean that there is a software issue or a hardware issue?
I still have proximity sensor issues after the iOS 4.1 upgrade

IPhone 4 Microphone not not working on voice calls but voice memos, viber, skype & facetime work just fine!! Any suggestions?

Hi Everyone,
I have encountered the following problem which I think a lot of other fellow iPhone users are also encountering however mine seems to be fairly pecuilar.
Here's the issue;
When making or receiving voice calls via my network operator's service, no one can hear me, i.e my audio input. I can hear the person on the other end of the phone fine but they can't hear me whatsoever.
However when I use the likes of viber, skype and facetime, my audio input is perferctly fine and the person on the other end of can hear me.
Note both viber and skype calls work just fine and 2 way communication is audbile regardless if I have a 3G or Wifi connection. Plus I have full singal bar strength when near a mobile tower so that elimantes antenna issues.
Also have taken alternative turns to cover both the top mic and the bottom mic during a viber or skype call and my audio input can still be heard.
I've done a reset, a hard reset of the phone and a reset of the network settings, still to no avail.
Have updated iOS to the latest version 5.1.1
Any suggestions would be much appreciated?
Thanks

I have the exact same problem, I have already restored the phone and it didn't work. The microphone works with facetime, skype and the voice memos. What just doesn't make sense is that when I use my earphone's microphone when calling, it doesn't work either. If it was a hardware problem the earphone's microphone would have worked I suppose.
This is the second iPhone 4 I have. The first one shut off after 9 months so I got a replacement phone. The replacement phone lasted 10 months when the microphone stopped working. I went to the apple store and they just couldn't give me a solution. They understood my situation and told me that it was very weird but all they could offer me was that I had to pay for a replacement. This is absolutely unjust, why do we costumers have to pay for the companies mistakes.

Calling issue with Cisco 7937 conference station

Hi Friends,
I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
When making calls from once remote site to another using Cisco 6921 phones calls working fine
When making calls from once remote site to another using Cisco 7937 conference station to make call any phone at remote office, calls are getting disconneted, remote phone rings when calls, but its gets fast busy tone when other party picks up the phone and not able to talk.
I suspect the issue with Codec but we have configured transcoders in VG and registered with CUCM
Please help me if any one experience such issue earlier.
Regards
Siva

hi Basant,
1. Actually tow phones A and B are registerd with centralized CUCM, A and B are located in two different locations, RTP traffic between And B pass through service provider.
Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
Show Run
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...
Current configuration : 12139 bytes
! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname eucamvgw01
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
boot-end-marker
card type e1 0 0
logging buffered 51200 warnings
no logging console
no aaa new-model
no network-clock-participate wic 0
no ipv6 cef
ip source-route
ip traffic-export profile cuecapture mode capture
bidirectional
ip cef
ip multicast-routing
ip domain name drreddys.eu
ip name-server 10.197.20.1
ip name-server 10.197.20.2
multilink bundle-name authenticated
stcapp ccm-group 2
stcapp
stcapp feature access-code
stcapp feature speed-dial
stcapp supplementary-services
port 0/1/0
fallback-dn 5428025
port 0/1/1
fallback-dn 5428008
port 0/1/2
fallback-dn 5421462
port 0/1/3
fallback-dn 5421463
isdn switch-type primary-net5
crypto pki token default removal timeout 0
voice-card 0
dsp services dspfarm
voice call send-alert
voice call disc-pi-off
voice call convert-discpi-to-prog
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 10.198.0.0 255.255.255.0
ipv4 152.63.1.0 255.255.255.0
address-hiding
allow-connections sip to sip
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
fax-relay ans-disable
sip
rel1xx supported "track"
privacy pstn
no update-callerid
early-offer forced
call-route p-called-party-id
voice class uri 100 sip
host 41.206.187.71
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 ilbc
codec preference 4 g729r8
codec preference 5 g729br8
voice class codec 20
codec preference 1 g729br8
codec preference 2 g729r8
voice moh-group 1
moh flash:moh/Panjo.alaw.wav
description MOH G711 alaw
multicast moh 239.1.1.2 port 16384 route 10.198.2.9
voice translation-rule 1
rule 1 /^012237280$..$/ /54280\1/
rule 2 /^012236514$..$/ /54214\1/
rule 3 /^01223651081/ /5428010/
rule 4 /^01223506701/ /5428010/
voice translation-rule 2
rule 1 /^00$.+$/ /+\1/
rule 2 /^0$.+$/ /+44\1/
rule 3 /^$[0-9].+$/ /+\1/
voice translation-rule 3
rule 1 /^9$.+$/ /\1/
rule 2 /^\+44$.+$/ /0\1/
rule 3 /^\+$.+$/ /00\1/
voice translation-rule 4
rule 1 /^54280$..$/ /12237280\1/
rule 2 /^54214$..$/ /12236514\1/
rule 3 /^\+44$.+$/ /\1/
rule 4 /^.54280$..$/ /12237280\1/
rule 5 /^.54214$..$/ /12236514\1/
voice translation-rule 9
rule 1 /^$....$/ /542\1/
voice translation-rule 10
voice translation-rule 11
rule 1 /^\+44122372$....$/ /542\1/
rule 2 /^\+44122365$....$/ /542\1/
voice translation-rule 12
voice translation-rule 13
rule 1 /^$[18]...$/ /542\1/
voice translation-rule 14
voice translation-profile MPLS-incoming
translate calling 10
translate called 9
voice translation-profile MPLS-outgoing
translate calling 11
translate called 12
voice translation-profile PSTN-incoming
translate calling 2
translate called 1
voice translation-profile PSTN-outgoing
translate calling 4
translate called 3
voice translation-profile SRST-incoming
translate calling 14
translate called 13
license udi pid CISCO2921/K9 sn FGL145110RE
hw-module ism 0
hw-module pvdm 0/0
username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
redundancy
controller E1 0/0/0
ip tcp path-mtu-discovery
ip scp server enable
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description internal LAN
ip address 10.198.2.9 255.255.255.0
duplex auto
speed auto
interface ISM0/0
ip unnumbered GigabitEthernet0/0
service-module ip address 10.198.2.8 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 10.198.2.9
interface GigabitEthernet0/1
description to TATA NGN
ip address 115.114.225.122 255.255.255.252
duplex auto
speed auto
interface GigabitEthernet0/2
description SIP Trunks external
ip address 79.121.254.83 255.255.255.248
ip access-group SIP-InBound in
ip traffic-export apply cuecapture size 8000000
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
no ip address
shutdown
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.198.2.1
ip route 10.198.2.8 255.255.255.255 ISM0/0
ip route 41.206.187.0 255.255.255.0 115.114.225.121
ip route 77.37.25.46 255.255.255.255 79.121.254.81
ip route 83.245.6.81 255.255.255.255 79.121.254.81
ip route 83.245.6.82 255.255.255.255 79.121.254.81
ip route 95.223.1.107 255.255.255.255 79.121.254.81
ip route 192.54.47.0 255.255.255.0 79.121.254.81
ip access-list extended SIP-InBound
permit ip host 77.37.25.46 any
permit ip host 83.245.6.81 any
permit ip host 83.245.6.82 any
permit ip 192.54.47.0 0.0.0.255 any
permit icmp any any
permit ip host 95.223.1.107 any
deny ip any any log
control-plane
voice-port 0/1/0
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/1
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/2
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/3
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
no ccm-manager fax protocol cisco
ccm-manager music-on-hold bind GigabitEthernet0/0
ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
ccm-manager sccp local GigabitEthernet0/0
ccm-manager sccp
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
sccp ccm 152.63.1.19 identifier 4 version 7.0
sccp ccm 152.63.1.100 identifier 5 version 7.0
sccp ccm 172.27.210.5 identifier 6 version 7.0
sccp
sccp ccm group 2
bind interface GigabitEthernet0/0
associate ccm 4 priority 1
associate ccm 5 priority 2
associate ccm 6 priority 3
associate ccm 3 priority 4
associate profile 1002 register CFB_UK_CAM_02
associate profile 1001 register XCODE_UK_CAM_02
associate profile 1000 register MTP_UK_CAM_02
dspfarm profile 1001 transcode
codec ilbc
codec g722-64
codec g729br8
codec g729r8
codec gsmamr-nb
codec pass-through
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 18
associate application SCCP
dspfarm profile 1002 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1000 mtp
codec g711alaw
maximum sessions software 200
associate application SCCP
dial-peer cor custom
name SRSTMode
dial-peer cor list SRST
member SRSTMode
dial-peer voice 100 voip
description *** Inbound CUCM ***
translation-profile incoming PSTN-incoming
incoming called-number .
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 500 voip
description *** Inbound TATA MPLS ***
translation-profile incoming MPLS-incoming
session protocol sipv2
session target sip-server
incoming called-number ....
incoming uri from 100
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 510 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 54[013-9]....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 520 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 5[0-35-9].....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 200 voip
description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 0122365....
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 201 voip
description *** Inbound M12 *** 012237280XX
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 012237280..
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 202 voip
description *** Inbound M12 *** 01223506701
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 01223506701
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 210 voip
description *** Outbound M12 ***
translation-profile outgoing PSTN-outgoing
destination-pattern +...T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 211 voip
description *** Outbound ISDN for SRST and emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 9.T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 212 voip
description *** Outbound ISDN for emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 11[02]
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 2000 voip
description *** Outbound to CUCM Primary ***
preference 1
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.19
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2001 voip
description *** Outbound to CUCM Secondary ***
preference 2
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.100
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2002 voip
description *** Outbound to CUCM Teritiary ***
preference 3
destination-pattern 542....
session protocol sipv2
session target ipv4:172.27.210.5
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 999010 pots
service stcapp
port 0/1/0
dial-peer voice 999011 pots
service stcapp
port 0/1/1
dial-peer voice 999012 pots
service stcapp
port 0/1/2
dial-peer voice 999013 pots
service stcapp
port 0/1/3
sip-ua
no remote-party-id
gatekeeper
shutdown
call-manager-fallback
secondary-dialtone 9
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.198.2.9 port 2000
max-ephones 110
max-dn 400 dual-line no-reg
translation-profile incoming SRST-incoming
moh flash:/moh/Panjo.ulaw.wav
multicast moh 239.1.1.1 port 16384 route 10.198.2.9
time-zone 22
time-format 24
date-format dd-mm-yy
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 131
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
line vty 5 15
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
scheduler allocate 20000 1000
ntp server 10.1.30.1
end
eucamvgw01#
Sh SCCP
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.198.2.9
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.198.2.9, Port Number: 2000
Priority: 3, Version: 7.0, Identifier: 3
Call Manager: 152.63.1.19, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 4
Trustpoint: N/A
Call Manager: 152.63.1.100, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 5
Trustpoint: N/A
Call Manager: 172.27.210.5, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 6
Trustpoint: N/A
MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1000
Reported Max Streams: 400, Reported Max OOS Streams: 0
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1001
Reported Max Streams: 36, Reported Max OOS Streams: 0
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: g722r64, Maximum Packetization Period: 30
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1002
Reported Max Streams: 16, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070080
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
eucamvgw01#

Unity Connection Voice Mail Issue

Hi,
I have a weird Unity Connection Voice Mail issue and would appriciate any help i can get. Many thanks in advance.
I have Unity connection 7.0 and CUCM 7.0 integrated in a lab enviroment and here is what happens.
When I place a call internally say from 2001 to 2002, things work as expected, 2002 rings and it goes to voicemail where I can leave voicemail and listen to it from 2002.
However, if I place call from the PSTN to the same number 2002 (or any other number in other sites etc) the call again rings in 2002 and goes to voicemail, unity cnx plays the greetings for 2002 and says record your message as usual. Everything up to point is fine, then when the time on the PSTN phone is showing around 14 seconds into the call unity starts playing, "to send this message press one", if I press one nothing happens.
I have rebuilt unity and cucm, even just configured the bare minimum in the lab and still getting the same result. I tried calling from E1 connection and T1 connection but with the same results.
I have run out of ideas...
===============================
Here is a call trace from an internal call:
CallData, 1, CallerId=2003, CalledId=2002, RedirectingId=2002, Origin=16, Reason=4, CallGuid=8B7859FD6C16417A9A07F507418DD25B, CallerName=, LastRedirectingId=2002, LastRedirectingReason=4, PortDisplayName=PhoneSystem-1-001
Application, 1, 2003, AttemptForward
State, 1, 2003, State - AttemptForward.cde!Dummy
State, 1, 2003, Event is [NULL]
Application, 1, 2003, PHTransfer
State, 1, 2003, State - PHTransfer.cde!LoadInfo
State, 1, 2003, Event is [TrueEvent]
Application, 1, 2003, PHGreeting
State, 1, 2003, State - PHGreeting.cde!PlayGreeting
Display, 1, 2003, Call answered if needed
Display, 1, 2003, Playing greeting for Subscriber: hq2
Display, 1, 2003, No DTMF received
Display, 1, 2003, Playing greeting for Subscriber: hq2
State, 1, 2003, Event is [RecordMsgEvent]
State, 1, 2003, State - PHGreeting.cde!RecordMsg
State, 1, 2003, Event is [NULL]
State, 1, 2003, State - PHGreeting.cde!RunEditMsg
Application, 1, 2003, -->MessageEditing
State, 1, 2003, State - MessageEditing.cde!CheckMsgMenuOpt
State, 1, 2003, Event is [EditMessageMenuEvent]
State, 1, 2003, State - MessageEditing.cde!PlayEditMenu
State, 1, 2003, Event is [HangupEvent]
State, 1, 2003, State - MessageEditing.cde!CheckMsgLength
State, 1, 2003, Event is [ManyEvent]
State, 1, 2003, State - MessageEditing.cde!SendMsg
State, 1, 2003, Event is [TrueEvent]
State, 1, 2003, State - MessageEditing.cde!ConfirmSend
State, 1, 2003, Event is [HangupEvent]
Application, 1, 2003, <--MessageEditing
State, 1, 2003, Event is [HangupEvent]
Display, 1, 2003, Idle
Display, 1, , Dialing (MWI) '2002'
Display, 1, , Idle
and here is a trace from an external (PSTN) call
Trying 142.100.64.13, 5000 ... Open
CallData, 1, CallerId=911, CalledId=2002, RedirectingId=2002, Origin=16, Reason=4, CallGuid=CA3DFD90846C4FE7B0D68298A7698287, CallerName=PSTN Emergency, LastRedirectingId=2002, LastRedirectingReason=4, PortDisplayName=PhoneSystem-1-001
Application, 1, 911, AttemptForward
State, 1, 911, State - AttemptForward.cde!Dummy
State, 1, 911, Event is [NULL]
Application, 1, 911, PHTransfer
State, 1, 911, State - PHTransfer.cde!LoadInfo
State, 1, 911, Event is [TrueEvent]
Application, 1, 911, PHGreeting
State, 1, 911, State - PHGreeting.cde!PlayGreeting
Display, 1, 911, Call answered if needed
Display, 1, 911, Playing greeting for Subscriber: hq2
Display, 1, 911, No DTMF received
Display, 1, 911, Playing greeting for Subscriber: hq2
State, 1, 911, Event is [RecordMsgEvent]
State, 1, 911, State - PHGreeting.cde!RecordMsg
State, 1, 911, Event is [NULL]
State, 1, 911, State - PHGreeting.cde!RunEditMsg
Application, 1, 911, -->MessageEditing
State, 1, 911, State - MessageEditing.cde!CheckMsgMenuOpt
State, 1, 911, Event is [EditMessageMenuEvent]
State, 1, 911, State - MessageEditing.cde!PlayEditMenu
State, 1, 911, Event is [HangupEvent]
State, 1, 911, State - MessageEditing.cde!CheckMsgLength
State, 1, 911, Event is [NULL]
Application, 1, 911, <--MessageEditing
State, 1, 911, Event is [NULL]
State, 1, 911, State - PHGreeting.cde!AfterMsg
State, 1, 911, Event is [NULL]
Display, 1, 911, Idle

Sounds like one way audio from PSTN to your Unity Connection, couple of things to check:
1. ensure your protocols are bound properly on the GW, i.e. SIP/H323/MGCP
2. Make sure IP routing is OK between Unity and the voice gateway
HTH,
Chris

External and internal mikes are not automatically switching over either recording or on voice calls

My laptop model name is HP Pavilion dv4-1100ea which is shipped with Vista Home premium 32 bits and has got service pack 1.
Restored the laptop to factory setting since then I am having the following problems; I had the same problem when my laptop was brand new and whenever I reset the laptop to factory setting I get the following problems:
1. External and internal mikes are not automatically switching over either in middle of the recording using sound recorder or while the call in progress on voice calls (skype):
   Using the sound recorder if I start recording the sound with external mike and in-between recording if I switchover from external mike to inbuilt mike and later on when I play back I can only hear the sound up till where I used the external mike during recording, after the switchover to inbuilt mike I cant hear any sound.
   But if I start the recording with inbuilt mike and in between recording if I plug in the external mike and later on when I play back I can only hear the sound up till where I used the inbuilt mike during recording , after the switchover to external mike I cant hear any sound.
   So in brief both my external and internal mikes are working fine, only problem is that if I start recording (using sound recorder) or voice call with one specific mike, I have to continue with it till the end. I can’t switchover to another mike in between the conversation (voice call) or recording, if I do so, I have to select the mike manually in chat software but while recording I cant even select manually because in laptop, it takes the mike whichever is in current use as default mike in recording tab(sound window). The green tick automatically (in recording device tab) switches over according to the use of mike. Though the green tick in the recording tab is switching over automatically according to use of mikes, its not picking up the sound after switch over during recording.
2. And also when I click on recording device tab in sound window and plug in external mike, though the green tick automatically switchovers from internal to external mike, while I speak both internal and external mikes volume meter respond to the sound inputs by rising and falling but if I take out external mike, green tick goes to internal mike and when I speak only internal mike volume meter respond to sound rising up and down not the external mike.
To resolve the issue I have tried following steps with no luck:
1.I have checked the mikes(internal and external) properties, the both mike shows to be enabled in general tab, in level tab the volume is set to 100 and in advanced tab , all options are selected.
2. In device manager I have got only one audio driver named as “IDT High definition Audio CODEC”. I have uninstalled the audio driver and reinstalled it using scan for hardware option
3. Uninstalled the audio driver in device manager and reinstalled the audio driver using recovery manager > advanced option> hardware driver re-installation.
3. I have updated the BIOS(Insyde F.65, 12/02/2010).
4. I tried to update the audio driver using below link but things went more worse so I did system restore (not factory setting though).
http://h10025.www1.hp.com/ewfrf/wc/softwareDownloadIndex?softwareitem=ob-67051-1&lc=en&dlc=en&cc=us&...

Sounds like you need to upgrade to the iPhone 5s
The 5s has Touch ID
You can unlock your phone with your finger instead of typing in a key code
No swiping to unlock either, just touch the home button
You can enrol multiple fingers as well
Here is a video of it in action
http://www.apple.com/iphone-5s/videos/#video-touch
Or wait and see what iPhone 6 has to offer
That being said, as desiel vdub posted if the phone is up to your face, the proximity sensor should turn the screen off
And when you lower the phone turn it back on again
Not sure about the phone locking when your on a call doesn't sound right

Browsing while voice call is in process

Hi Masters,
I've recently updated my Curve 3G 9300 from OS5 to OS6 and have read a lot about the common issue that OS6 doesn't give us the flexibility to choose which browser to use. But it seems that my problem is uncommon since I haven't been able to see any solution to it in last 3 days of googeling.
My problem is that I'm unable to use internet browser while talking to someone. Everytime I'm on a call and I open my BB broswer it shows ''A voice call is in process. Browsing will not be possible". I was able to use browser while being on a call in OS5 but not in OS6. However, I'm able to use broswer while being on call if using it through WIFI but not with BIS.
Kindly Help...
Ankur

Plz help Masters..

Voice Call issues

Similar Messages

Maybe you are looking for