UC540 random no CLID in incoming calls

We have UC540 with 4 FXO Software pack: 8.6.1. sometimes it shows caller ID for incoming calls and sometime it doesn't. we have also tried changing the required rings for CLID, but no success.
Need Help

Hello Mike,
Could be because of a lot of different things you may open a case for this issue with SBSC and resolve it faster.
Best regards,
Alex

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SPA 2102 after RV082, Drops sending on incoming calls at random

Hi, I have read how it should be possible to have a SPA 2102 working together with a RV082 on this page
http://www.cisco.com/en/US/products/ps9923/products_qanda_item09186a0080a365b1.shtml
Have followed how it should be configured. Opened ports on the RV082 router and configured the SPA2102 for static IP and enabled it to work with NAT.
The phone works however, kind of random it drops the incoming calls on my transmitting side. So if someone calls, randomly it seems, the other person can loose my transmitting side and not hearing me talk anymore, and I can hear "Hello? You still there?"If hang up and the person dials again, there is incoming signal, I lift the phone and it totally quiet, can't hear a thing.
Anyone has ideas what to do with this problem? They are both Linksys products, and as there are even instructions how to make them work together, it should be possible right?
By the way, I only use one of the Internet ports, don't use the second one. Have a direct Ethernet link for the Internet 100/10 so there is no DSL involved.
I could have them rigged up the other way around SPA2102 to RV082, but there are some issues then when it comes to port handling, and if I remember correct the SPA2102 is only 10/10.
-settings-
I have set the SPA2102 as DMZ on my RV082 router.
Have forwarded port 3478, 5060-5080,5004,10000-20000,16384-16482 (this should proboly be obsolete)
Have set port 8 as it is conneted to as High.
Have set Firewall/Accesss Rules for this IP to be Priority 1 and allow all traffic.
Have included jpg screenshots of settings for my SPA2102.
Any help would be appreciated.

If your forwarding all those ports to the device and it is still acting up. I would try a factory reset of the spa2102 and reconfigur it. Have you checked to see if you have the latest firmware on the spa2102. If not upgrade that and reset the spa2102 and reconfigure it. With it being in the dmz host section on the router, it should not have any issues. The dmz host does not have any firewall rules or any blocked ports pertaining to it so it should be fine with no blocks. Give that a shot and lets us know.

UC540 - Flip to a Camera Service URL from Incoming Call

I have a client who says he has seen this before on a UC520 w/ 7975s, and he wants it repliated on his UC540 w/ 7975s. i.e. an incoming phone call from an access gate on an FXO port would cause the phone to ring, and force a page flip to a services URL or some oher mechanism where the camera is visible (Axis camera server that is then converting the images to PNG via a realtime script on a Unix host).
I know that cameras can be viewed at the services URL using a conversion method to PNG as above, but to the best of my knowledge you can't force the phone to flip automatically, or replace the background image automatically based on an incoming call.... Can somebody please confirm, or if it is now possible, explain the steps.

If that was done with CTI or TAPI and phone SDK, etc as custom application, yes it's possible. You would have to contact the author to know about the details, or start from almost zero to replicate it yourself.

I was sitting here and my phone randomly said incoming call from my friend that is an iphone user. I answered it and nothing was there. He texted me the screen shot on his phone that said I called him, but it showed on my phone as an incoming call. Why?

I was sitting here and my phone randomly said incoming call from my friend that is an iphone user. I answered it and nothing was there. He texted me the screen shot on his phone that said I called him, but it showed on my phone as an incoming call. Why?

Heyc123 wrote:
I had bought an iphone from this guy . ... . I had try to call him but he wouldnt answer and went ghost on me . Is there another way to activate the phone without putting in his apple id
Nope.
http://support.apple.com/kb/TS4515

Iphone 5: Incoming Calls won't go through.

Hello everyone,
I purchased the Iphone 5 about 1-2 weeks ago and so far i love it. However as of recently i've been expeirencing a few issues where incoming calls won't go through.
I'll try my best to break it down:
Someone calls me.
Phone "rings" on their side.
My iPhone doesn't ring.
I get no notification of a missed call, and maybe a few minutes later i'll be alerted to a missed call or a voicemail.
I've tested various things that i assume would cause the issue, but so far none of them seem to be the case.
I even went to Verizon (a few minutes ago actually) and they did a soft-reset (pretty much what i did prior) and soon as i got home i called the phone and nothing went through. I waited about a minute, called again, and then i rang. However seconds later i tried calling again, and it didn't ring.
Also it happens are random times. First insident was a few nights ago around 10 PM. Then again a few days ago around 3 PM and finally this morning around 8-8:30 AM.
No my "Do not disturb" isn't on.
I have good (if not great) connection.
At home the phone is always on WiFi
Yes i've tried turning off LTE.
Yes i've done a network reset.
Yes, phone is up to date.
One question i have is could it be only having this issue at my house? I have no way of testing if it only happens at my home (cause no one really has that kind of time to waste with me lol) so i'm not sure.

I want to ensure those important calls get through Huxable! I understand why you are concerned!
My first recommendation would be to make sure you are power cycling the device(turning off and on) at least once a day. This allows the device to refresh with the network and update the tower information on 4G handsets.
Second, it will definitely be valuable to test the device in different locations to determine whether it is a location issue versus an equipment issue. If this happens in multiple locations for example, it is likely device related.
If you find the issues occur mainly inside your home, the network extender accessory is an excellent option to boost indoor signal strength.
http://bit.ly/AKkC
Sincerely,
JonathanK_VZW
VZW Support
Follow Us on Twitter@VZWSupport

IPhone 2.0 incoming calls

Since I have updated my iPhone to the 2.0 software incoming calls have been not ringing and or causing my phone to freeze. I was wondering if anyone else was having the same problems. Also SMS text and calendar and notes all have been opening very slowly. I have restored my phone and this does not seem to help. I was wondering if it is my phone or if more people are having these problems. Yes kind of but I have the original iPhone so it has nothing to do with 3G it will just randomly not receive a call, or freeze when it does causing it to vibrate out of control.
Message was edited by: iPhone236545

My iPhone is also not ringing most of the time. I have noticed that the ringer volume is often turned off in spite of the fact I keep tuning it up. This happens even if the phone is locked.

Curve 9360 not ringing with incoming calls

Good day. I am using a Blackberry Curve 9360 Smartphone.
Hope you can help me, I have inadvertently ticked the "All Alerts Off" option instead of the "Silent" option.
The phone rejects my attemps to change it back to All Alerts on, with a reply "Alerts Off cannot be editted or deleted"
There's a red tick mark next to the icon, and the icon is also highlighted in red.
The "Normal" option is ticked at the moment but the "All Alerts Off" option remains the same.
Incoming calls go straight to voicemail without a ringing alert, and a sms informs me of a missed call.
Please advise me how to resolve this problem, URGENTLY !!!!

Hi and Welcome to the Community!
That is strange indeed...typically, only one profile at a time can actually be active...from what you describe, both Normal and All Alerts Off are active at the same time...not normal. Lets try this...
Anytime random strange behavior or sluggishness creeps in, the first thing to do is a battery pop reboot. With power ON, remove the back cover and pull out the battery. Wait about a minute then replace the battery and cover. Power up and wait patiently through the long reboot -- ~5 minutes. See if things have returned to good operation. Like all computing devices, BB's suffer from memory leaks and such...with a hard reboot being the best cure.
Now try to reset your sound profile as you desire.
Best!
Occam's Razor nearly always applies when troubleshooting technology issues!
If anyone has been helpful to you, please show your appreciation by clicking the button inside of their post. Please click here and read, along with the threads to which it links, for helpful information to guide you as you proceed. I always recommend that you treat your BlackBerry like any other computing device, including using a regular backup schedule...click here for an article with instructions.
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Handling incoming caller ID from legacy PBX

Hello,
At my company we're currently in the process of running Lync 2010 EV through a POC. Everything is working great at the moment except for incoming caller ID from the legacy PBX. When a call comes into Lync from the legacy PBX (either from a desk phone or
from outside the company), the legacy PBX replaces the actual 'from' number with the company's main number. Apparently this is due to the legacy PBX being old and masking the outbound caller ID information at a global level. (So basically ANY call that goes
outbound from the legacy PBX has the caller ID masked with the main number, no way to make exclusions. This has been verified by the vendor that services the legacy PBX.)
The problem this introduces on the Lync client is that in the address book everyone has the company's main number as their phone number, except with an extension at the end. So when a call comes in from the legacy PBX the Lync client quite literally randomly
picks someone from the address book and says the call is coming from the randomly picked person.
So far in my attempts to fix this, I've tried creating a contact in AD with a phone number of just the main number. This did not change anything from the Lync clients perspective, however when I miss a call that was sent from the legacy PBX, Outlook now
indicates that the missed call is from the contact object with the main number instead of saying I missed a call from some random person from the address book.
I've also looked into seeing if a MSPL script could intercept incoming audio requests and change the From parameter to something else if the From value was the main number. So far I haven't seen any examples of this, nor do I have a test environment to test
MSPL scripts in yet.
Does anyone have any suggestions on what can be done for the POC we're trying to do? Obviously if (more like 'when' really...) we decide to move everything over to Lync EV and pull the plug on the legacy PBX this won't be an issue, but right now I can't
even move forward with doing a POC outside of my team until I figure out something to do with the inbound caller ID.
Thanks,
Cody

How are you connected to the legacy PBX? It might be easier to put a modification in a gateway if you're using one.
Also, why Lync 2010 instead of 2013?
What do you have set for the lineURIs of the users? Is it the main number with extensions or a DID?
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications

PAP2T: Incoming calls being blocked by Router.

Hi,
I am trying to use an unlocked PAP2T Adapter with Telus home network in Canada and am not able to receive calls. I can make outgoing calls to any number in the world, but whenever I get a call back, the phone doesn't ring & later I find it in Firewall log as being blocked under "Low Risk Attack". While trying to find a resolution with the VOIP provider I had at times picked up the phone randomly & found the caller on the line. So apparently the calls have no prblem reaching my PAP2T Adapater, it is just that (1) there is no ring when they do & (2) the incoming calls get blocked by Firewall & dropped.
I have tried port forwarding (UDP & TCP) as well DMZ mode for the Adapter, no luck. Later i tried disabling almost all of the internet provider's 2wire modem/router firewall features (UDP ports scan, packets , etc.) still didnt work. I have run out of ideas & hoping that someone can help me find a way out of this problem ? Appreciate your help & time. Thanks.

I changed to the following settings that did the trick for making the phone ring,
Click Regional on the top:
Under Ring and Call Waiting Tone Spec
Change Ring Waveform to Sinusoid
Change Ring Voltage: 90
Change Ring Frequency: 20
However, when I pick up the ringing phone, there is no voice going through the line. Also, as per my VOIP provider instructions, when I dial a local number followed by # sing & hang up in 5 seconds, I don't get any callback either. I have even updated the PAP2T to the latest firmware 5.1.6. Any thing else I can change to get the voice across & get the call back ? thanks for your help again.

Analog line (FXO) Incoming calls getting connected after 3 rings

     HI,
we are having 4 Analog line (FXO)...Every time when callers call the number they hear 3 rings & after that call frwds to AA or any extension.
In show voice port summary, we can see that voice port is getting connect at the first ring but after 3 rings only phone rings.
here is the o/p of voice port.
Foreign Exchange Office 0/0/0 Slot is 0, Sub-unit is 0, Port is 0
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is plar
Connection Number is 250
Initial Time Out is set to 15 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Power Denial Disconnect Time Out is set to 1000 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Companding Type is u-law
Region Tone is set for AE
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming): INCOMING_CallerID_PROFILE
Translation profile (Outgoing):
lpcor (Incoming):
lpcor (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is enabled
Number Of Rings is set to 1
Supervisory Disconnect is signal
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 65 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
Secondary dialtone is disabled

hostname VGUAE001
no aaa new-model
clock timezone UAE 4 0
ip cef
ip domain name yourdomain.com
no ipv6 cef
multilink bundle-name authenticated
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
voice-card 0
voice call send-alert
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
voice class cause-code 1
no-circuit
voice translation-rule 1112
rule 1 /^9/ //
voice translation-rule 3265
rule 1 // /9\1/
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 50
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
license udi pid CISCO2901/K9 sn FCZ173992Z8
hw-module pvdm 0/0
hw-module pvdm 0/1
username cisco privilege 15 secret 4 opjnnkXqCr4kCOa9DuALcNpBOMetBAc/usnpSWADsCI
username godiva privilege 15 secret 4 cH8b8z.ioYu/pMv/AKuEcBd/f6g9v/vm/s3aXeqUAd6
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.31.2 255.255.255.0
ip helper-address 192.168.31.11
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.31.2
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 192.168.31.1
control-plane
voice-port 0/0/0
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
groundstart auto-tip
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/1
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/2
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/3
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 250
caller-id enable
mgcp profile default
dial-peer voice 2000 voip
destination-pattern 2..
session target ipv4:192.168.31.11
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 10 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fire**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 997
forward-digits all
no sip-register
dial-peer voice 11 pots
trunkgroup ALL_FXO
description **CCA*UAE*International Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 900T
forward-digits all
no sip-register
dial-peer voice 12 pots
trunkgroup ALL_FXO
description **CCA*UAE*Eitisalat**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9101
forward-digits all
no sip-register
dial-peer voice 13 pots
trunkgroup ALL_FXO
description **CCA*UAE*Water or electrical emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 971
forward-digits all
no sip-register
dial-peer voice 14 pots
trunkgroup ALL_FXO
description **CCA*UAE*Police and emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 999
forward-digits all
no sip-register
dial-peer voice 15 pots
trunkgroup ALL_FXO
description **CCA*UAE*National area codes**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[1-579].......
forward-digits all
no sip-register
dial-peer voice 16 pots
trunkgroup ALL_FXO
description **CCA*UAE*Mobile Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 90[5-6][0-7].......
forward-digits all
no sip-register
dial-peer voice 17 pots
trunkgroup ALL_FXO
description **CCA*UAE*toll-free**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-9]00T
forward-digits all
no sip-register
dial-peer voice 18 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fixed Line Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-8]T
forward-digits all
no sip-register
dial-peer voice 19 pots
trunkgroup ALL_FXO
description **CCA*UAE*808**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9808T
forward-digits all
no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/0/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/0/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/0/2
no sip-register
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/0/3
no sip-register
Debug vpm signal:
Nov 23 19:31:31.556: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
Nov 23 19:31:31.556: htsp_timer - 125 msec
Nov 23 19:31:31.684: htsp_process_event: [0/0/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
Nov 23 19:31:31.684: htsp_timer - 10000 msec
Nov 23 19:31:31.684: htsp_timer3 - 5600 msec
Nov 23 19:31:31.684: [0/0/0] htsp_start_caller_id_rx:Mode BELLCORE. Alerting 0x1
Nov 23 19:31:31.684: htsp_start_caller_id_rx create dsp_stream_manager
Nov 23 19:31:31.684: [0/0/0] htsp_dsm_create_success returns 1
Nov 23 19:31:33.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:33.604: fxols_ringing_not
Nov 23 19:31:33.604: htsp_timer_stop
Nov 23 19:31:33.604: htsp_timer - 10000 msec
Nov 23 19:31:37.284: htsp_process_event: [0/0/0, FXOLS_RINGING, E_HTSP_EVENT_TIMER3]fxols_snoop_clid_stop
Nov 23 19:31:37.284: htsp_timer_stop3
Nov 23 19:31:37.516: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0000]
Nov 23 19:31:39.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:39.604: fxols_ringing_not
Nov 23 19:31:39.604: htsp_timer_stop
Nov 23 19:31:39.604: htsp_timer_stop3
Nov 23 19:31:39.604: [0/0/0] htsp_stop_caller_id_rx. message length 0htsp_setup_ind
Nov 23 19:31:39.604: [0/0/0] get_fxo_caller_id:Caller ID receive failed. parseCallerIDString:no data.
Nov 23 19:31:39.604: [0/0/0] get_local_station_id calling num= calling name= calling time=11/23 23:31 orig called=
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=250
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x3CE27724, Call Info(
   Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=50, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=FALSE,
   Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: cc_get_feature_vsa count is 1
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218944,feature_id:83
Nov 23 19:31:39.604: //83/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown))
Nov 23 19:31:39.608: [0/0/0] htsp_dsm_close_done
Nov 23 19:31:39.608: htsp_process_event: [0/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
Nov 23 19:31:39.608: fxols_wait_setup_ack:
Nov 23 19:31:39.608: [0/0/0] set signal state = 0xC timestamp = 0fxols_check_auto_call
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
   Event=0x22ACD828
Nov 23 19:31:39.608: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 250
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetContext:
   Context=0x230F9C10
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 83 with tag 50 to app "_ManagedAppProcess_Default"
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=FALSE, Mode=0,
   Outgoing Dial-peer=2000, Params=0x230FB0D0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Destination Pattern=2.., Called Number=250, Digit Strip=FALSE
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=TRUE,
   Guid=B583C95F-53AC-11E3-8093-C8EEBDE4256A, Outgoing Dial-peer=2000
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=250
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x22847B14, Interface Type=1, Destination=, Mode=0x0,
   Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-peer=2000, Call Count On=FALSE,
   Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, tg_label_flag=1, Application Call Id=)
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: cc_get_feature_vsa count is 2
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218720,feature_id:84
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccCallSetContext:
   Context=0x230FB080
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=2000
Nov 23 19:31:39.612: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc
Nov 23 19:31:39.612: htsp_timer - 120000 msec
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGetMediaClassTag:
   media class tag 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_call_proceeding:
   Interface=0x22847B14, Progress Indication=NULL(0)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_delay_xport:
   CallInfo(delay xport=TRUE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
   Interface=0x22847B14, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
   Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
   Call Entry(Responsed=TRUE, Alert Sent=TRUE)htsp_alert_notify
Nov 23 19:31:39.628: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alert
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_notify:
   Data Bitmask=0x5, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallNotify:
   Data Bitmask=0x5, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.672: //84/B583C95F8093/CCAPI/ccIsInfoRingback:
   Returning dpRingBack=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
   Interface=0x22847B14, Data Bitmask=0x1, Progress Indication=NULL(0),
   Connection Handle=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
   Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_notify:
   Data Bitmask=0x7, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Network, Params=0x0, Call Id=83
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   (confID=0xFFFFFFFF, callID1=0x53, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
   (confID=0xFFFFFFFF, callID2=0x54, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   Conference Id=0xFFFFFFFF, Call Id1=83, Call Id2=84, Tag=0x0
Nov 23 19:31:39.700: htsp_call_bridged invoked
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_bridge_done:
   Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
   Destination Call Id=84, Disposition=0x0, Tag=0xFFFFFFFF
Nov 23 19:31:39.700: //84/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
Nov 23 19:31:39.700: cc_api_get_xcode_stream : 4819
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x21, Destination Call Id=84)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x21, Destination Call Id=83)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
Nov 23 19:31:39.700: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
   Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
   Caps(Codec=0x1, Fax Rate=0x1, Fax Version:=0, Vad=0x1,
   Modem=0x2, Codec Bytes=20, Signal Type=3)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallNotify:
   Data Bitmask=0x7, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x21, Call Id1=83, Call Id2=84
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
   Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
   Caps(Codec=0x1, Fax Rate=0x2, Fax Version:=0, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ack:
   Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_caps_ack:
   Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
   Progress Indication=NULL(0), Data Bitmask=0x1
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
   Call Entry(Connected=TRUE, Responsed=TRUE)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
Nov 23 19:31:39.704: htsp_timer_stop
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
   Call Id=83
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
   Call Entry(Context=0x230F9C10)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
Nov 23 19:31:39.932: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x22847B14, Call Id=84
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
   Conference Id=0x21, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
Nov 23 19:31:48.860: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
   Destination Call Id=84, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.864: htsp_timer_stop3
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x22847B14, Tag=0x0, Call Id=84,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: :cc_free_feature_vsa freeing 3D1B9898
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: vsacount in free is 1
Nov 23 19:31:48.884: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release
Nov 23 19:31:48.884: htsp_timer_stop
Nov 23 19:31:48.884: htsp_timer_stop2
Nov 23 19:31:48.884: htsp_timer_stop3
Nov 23 19:31:48.884: [0/0/0] set signal state = 0x4 timestamp = 0
Nov 23 19:31:48.884: htsp_timer - 2000 msec
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3CE27724, Tag=0x0, Call Id=83,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: :cc_free_feature_vsa freeing 3D1B9978
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: vsacount in free is 0
Nov 23 19:31:49.156: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0100]

Incomming Caller ID

Dear Sir vasi here,
I am from Kingdom of Saudi Arabia. kindly find the attachment. i am using PSTN Trunks
Pls help me out with this issue. I am not getting caller ID for incoming calls it is showing Anonymous. Not having any issue with ISP I checked all individual phone line by connecting phone it is showing incoming number. But when I am connecting phone lines in UC540 it is not showing.
I configured only Hunt Grop. pls try to resolve this issue by GUI mode in priority or else CLI.

Sorry to say that I've removed my postings here due to a disrespectful attitude by the OP.
I recommend that he checks my reputation and status in this community and refrains from behaving like that again, to avoid being reported to administrators.

Droid 2 Global Issues - notif. bar and incoming calls

Ok, I've had my Droid 2 Global for about a year now, and for the most part it has been fine. However, on the latest couple of updates, some really annoying "features" have come up. On the current update, the notification bar disappears randomly. Not known cause. Also, if the notification bar disappears, when I receive and incoming call, it doesn't display the "answer call" screen, showing the contact. This makes it difficult to answer the call, especially when the screen is locked. I have to go unlock the screen, and go to a screen like I'm going to call someone, and THEN I can see who is calling and maybe answer. If it hasn't gone to VM already. This is really stupid - first and foremost, this thing is a FREAKING PHONE. I should be able to answer a call.
Frustrated. Any ideas/help?
Thanks,
-Dirv
Details
=============
System version: 4.5.629.A956.Verizon.en.US
Android version: 2.3.4
Kernel version: 2.6.32.9-g177d714 mcbk83@il93lnxdroid01 #1
Build number: 4.5.1_57_D2GA-59

Ok, I've had my Droid 2 Global for about a year now, and for the most part it has been fine. However, on the latest couple of updates, some really annoying "features" have come up. On the current update, the notification bar disappears randomly. Not known cause. Also, if the notification bar disappears, when I receive and incoming call, it doesn't display the "answer call" screen, showing the contact. This makes it difficult to answer the call, especially when the screen is locked. I have to go unlock the screen, and go to a screen like I'm going to call someone, and THEN I can see who is calling and maybe answer. If it hasn't gone to VM already. This is really stupid - first and foremost, this thing is a FREAKING PHONE. I should be able to answer a call.
Frustrated. Any ideas/help?
Thanks,
-Dirv
Details
=============
System version: 4.5.629.A956.Verizon.en.US
Android version: 2.3.4
Kernel version: 2.6.32.9-g177d714 mcbk83@il93lnxdroid01 #1
Build number: 4.5.1_57_D2GA-59

1.3.5.1 Updated, and Now Issue With Incoming Calls

Since updating to 1.3.5.1, any time I mute the ringer on an incoming call, it mutes the ringer until right before the missed call notification, but then plays the ringer again briefly. This defeats the purpose of being able to mute the ringer in the first place.
Post relates to: Pre p100eww (Sprint)

This occurrs any time I press the volume rocker key to mute the ring of an incoming call. After the initial time when it played an extremely brief snippet of my ringtone (factory installed "Older Phone" ringtone), it has randomly switched between the short ringtone and a electronic squeal noise.
Since I know the slightest software change can kill a seemingly unrelated feature, I'll try to be as detailed as possible. For my phone setup, it is completely stock (phone no.5 after misc. slider/screen/charger door issues, haven't had time to change anything yet) My ringtone volume is set to maximum with vibrate also on, system sounds are also set to maximum with the default tone. I have no non-app catalog programs or patches installed. I use a Touchstone back cover, although the phone was not on the charger when I noticed the issue.
The only downloaded apps on the phone are Pandora, Tip 'em!, Stop Watch, Fandango, Good Food, Mileage Monitor and gDial Pro (which has not been set up yet). With the exception of Pandora, none of these apps have been used since I received this replacement phone. They were all just automatically downloaded because my previous phone had them. I have e-mail accounts through Cox, Yahoo, and Gmail, none of which are synced to contacts, calendars, messaging or tasks. I do have my Palm profile and EAS through my work in a full sync capacity.
Hope this helps.
Message Edited by kx250ryder on 01-07-2010 07:29 AM
Message Edited by kx250ryder on 01-07-2010 07:30 AM

Blackberry 9300 Freeze During Incoming Call

Device: Blackberry Curve 3G 9300
OS: 6.0 Bundle 2475 (v6.0.0.546)
Application Storage Free Space: 43.3MB
Carrier: Maxis Communications (Malaysia)
My phone has been freezing up randomly during incoming calls for the last 3 weeks. Sometimes I am able to pick up the call but most times it just freezes up. During the freeze, all the buttons will not work and the phone will just keep ringing until the call has been canceled/dropped from the other end.
There are 3 scenarios:
1. Phone works as normal.
2. Incoming call comes in, phone freezes up, I pull the battery out, phone reboots and everything is fine until the next freeze. I can recieve SMS and all application works as normal.
3. Incoming call comes in, phone freezes up, I do nothing and let the call end. Screen will black out after a few seconds and none of the button works. Next incoming call comes in, the screen lights up to show the incoming call but none of the button works just like before. In the end, I have to pull out the battery.
Here are some of the things I have tried:
1. Deleting applications from my phone to free up memory (Facebook, Yahoo Messenger, Google Messenger)
2. Deleting pictures to free up memory
3. Updating the phone to the latest software version. Previously it was on version 5 but the freeze happened then too.
4. Reseting my phone to use the original theme.
I have never dropped my phone (not even once) and I have been using this same phone for the last 10 months. I still have 2 months left before my warranty expires. I really look forward to getting help from the experts here before I take the last option of bringing my phone to the service center.

Basics from the manual are restart, reset, restore from backup, restore as new.

Prefixing a 9 and 91 to incoming calls from SIP provider for callback

I am wondering what would be the best options for prefixing a 9 or 91 to incoming calls over a sip connection to allow callback from missed calls and recieved calls. The setup is
callmanager 7.1.5 >>>>sip trunk>>>>>>>>>>>>CUBE>>>>>>>>>>>sip to ITSP
I am thinking voice translation rules is the only option for this? any configuration examples for this would be greatly appreciated.
would this work?
voice-translation rule 1
rule 1 // /9/
voice-translation profile prefix_9
translate calling 1
dial-peer voice 101 voip
destination-pattern ???????...$
voice-class codec 1
session protocol sipv2
session target ipv4: to callmanager
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 1001 voip
translation profile incoming prefix_9
destination-pattern T
session protocol sipv2
session target ipv4: to sip provider
incoming called-number ???????...$
dtmf-relay rtp-nte

Your config should work fine, except your profile is only applied to one dial-peer, make sure you apply it to the one that is used to redirect the call to CUCM.
Also, you did not mention what country you are in, but if this is US you may want to prefix 91 to national calls as carriers don't provide 9 as part of the CLID delivery, also what about your international calls, you may would be more explicit in your first rule to match for national digit string and then have another rule for international.
HTH,
Chris

UC540 random no CLID in incoming calls

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