DECT phone outgoing call issue
Has anyone had issues with DECT phone outgoing calls - My new BT Studio 3500 can receive calls ok, but dialling out returns "this number is not recognised" - using my old analogue phone and it suffers no such issues. Is there something up with my phone line - I have ADSL connected, this works fine with microfilter and analogue- even when I disconnect the ADSL line the DECT phone still has an outgoing call issue. I even replaced the DECT phone with a another model previous, but the issue still remains.
Im not contracted to BT but go via TalkTalk, and just want to find out if I am setting the DECT phone up incorrectly!
So rather than try and troubleshoot your problem, or take the phone in to have it checked by Apple for a hardware fault, you're just going to wait for something to magically change... OK, then.
Similar Messages
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SIP trunk incoming and outgoing calls issue
Hi Everyone,
We recently installad a SIP trunk and terminated on CUBE and CUCM but we have issues on incoming and outgoing calls, When someone dial in from outside he keeps listening the dailing ring even after we pick up the phone and at the end the callers time exipres and call gets disconnected.
For Dailing out, the dialed number rings and caller hear the dailing ring as well but if someone pick the phone it apprears that call is connected but no audio in it, dead air.
Our call flow is as
IP Phones => CUCM --->SIPTRUNK--->CUBE=>SIPTRUNK=>SP
I have attached the config for CUBE and debug ccsip messages output for both incoming and outgoing calls.
Please if some help in sorting out this issue, Thanks in Advance
TasneemInbound call>>>>
The reason you are experiencing this is that your CUBE is requesting PRACK and your provider is not responding to it..
Here we have your cube sending 180 ringing with "Require 100rel"..This was sent several times and your ITSP didnt respond probably because they do not support 100rel...(It is Huwaei after all, they do what they like)
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Date: Thu, 16 Jan 2014 13:31:34 GMT
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Require: 100rel
RSeq: 2507
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "TEST STC" ;party=called;screen=yes;privacy=off
Contact:
Record-Route:
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Length: 0
AFter the CUBE didnt get any response, it then replied with Gateway Timeout...
Jan 16 13:31:54.550: //31880/5A4406E48184/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 504 Gateway Timeout
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Reason: Q.850;cause=102
Content-Length: 0
I suggest you disable this parameter..and test again
voice service voip
sip
rel1xx disable
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay" -
Sip outgoing calls issue, getting 403 forbidden sometimes.
Hi!
This is my connection to provider
ISP---Sip---CUBE---Sip---CUCM---Sip---7821
I've got a problem with otgoing calls.
On sucessfull calls my CUBE sent to ISP Invite with SDP, then gets 100 trying, 183 progress sdp, 180 ringing and rtp after it. The call is ok. Hold, transfer, redirect works fine.
But sometimes (maybe 30-40% of outgoing calls) I sent Invite with SDP, then get 100 trying and after it my cube receives 403 forvidden from provider, cause code 57. Session breaks. What's the matter with it? Incoming calls are ok.
Also I disabled providers MOH on this cube by using no ip cef on it and using software mtp on sip trunk fom CUCM.If the provider is the one sending that to you then it would be best to get with them. The initial INVITEs and TRYINGs look good from what you can tell? Is there any dialplan differences between the phones? Is it the same phones causing this problem or is it random? These are things you need to look into.
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CME SIP phone outside call issue
Dear all,
i have cme version 9.1 on router 2921 with 7962 sccp phones and 3905 sip phone.
when i place outside call ( to pstn) using the below dial peer, call is processed.
when the call is answered by the autoattendent of the called company ( assume i called x company) , i cant press any other numbers using the sip phones.
i mean if i want to press zero for help or internal extension of the x company, these pressed numbered are not recognized by the analog panasonic PBX of the x company.
Sccp phones works well.
Any help please and below is the dial-peer.
dial-peer voice 1003 pots
trunkgroup 1
corlist outgoing CITIES
description CALLING CITIES
destination-pattern 90[1-9]......
forward-digits 8
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/2.10
bind media source-interface GigabitEthernet0/2.10
registrar server expires max 36000 min 600
voice class codec 5
codec preference 1 g729r8
codec preference 2 g711ulaw
voice register global
mode cme
source-address 10.100.4.20 port 5060
max-dn 200
max-pool 100
load 3905 CP3905.9-2-1-0.loads
authenticate register
timezone 31
date-format D/M/Y
voicemail 177
tftp-path flash:
create profile sync 000473524028932A
conference hardware
voice register dn 1
number 109
allow watch
pickup-call any-group
pickup-group 170
shared-line max-calls 3
voice register pool 1
id mac 6C99.8984.9678
type 3905
number 1 dn 1
template 1
dtmf-relay sip-notify
voice-class codec 5
username SFD1 password SFD1
thanksHi Yahsiel,
firstly thanks for help, secondly if you don't mind i want to ask you the below if possible:
1- in my cme, is there a way when i call an internal extension (e.g 110) from an internal phone it rings normally but when i call from outside-->autoattendent answers-->when i press 110 it get transferred to another phone (e.g 111)....????
2- when i call from outside(pstn) to the cme -->when the plar command is directly to the internal extension the caller id appears but when the autoattendent answers and then transfer to the operator (by pressing zero) the caller id appears as unknown number ??????
3- is the 3905 sip phone support 1Gbps when connected to the PC, as after connecting the phones to the PCs the speed decreased up to 100Mbps?? or it is another matter?
(poe switches is cisco SG200)
regards, -
SIP trunk INCOMING/OUTGOING calling issue
Dears,
i am facing issue i.e inbound / outbound calls are not wokring after configuring SIP trunk . the call flow is
SERVICEPROVIDER--------sip trunk------VG --------------sip trunk ----------CUCM
For Outbound calls ---error-----tot tot tot
For Inbound Calls -----error--- Silence
Below are the configuration and debugs:Appreciate if some provide me Sip configuration and troubleshooting guide...Thankyou
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HOW DO U FIND UR OUTGOING CALL LOG ON VERIZON.COM, IM TRYING TO LOOK AT MY HOME PHONE OUTGOING CALL LOG
I've been nagging Verizon on this one alot. If you have one of the international long distance plans, you get 300 or 500 minutes of international long distance, then any time over this allotment is billed by the minute. The problem is that you have NO WAY of knowing how many minutes you have used, until the bill comes at the end of the month.
I've had customer support say:
- You need to track the time yourself - we suggest a stop watch.
- I get several calls a month about this , but we don't have an answer.
- We are trying to resolve this issue.
- You don't need to know how many minutes you used. (They indirectly imply this)
- We can see how many minutes you used, but can only tell you to resolve a fraud alert issue regarding unusual long distance activity. (This bugs me as much as suggesting I use a stop watch)
- If you switch to the more expensive 500 minute plan, you won't have to worry as much.
Call assistant would be the logical place to put outgoing calls, with international plan minute totals, but what do I, a customer, know.
CM
Message Edited by ComposMentis on 05-18-2009 07:26 AM
Message Edited by ComposMentis on 05-18-2009 07:27 AM
Message Edited by ComposMentis on 05-18-2009 07:42 AM -
My phone made an outgoing call on its own anyone else with this issue?
My iphone made an outgoing call somehow on its own which is impossible since i have password protected and my phone was on me at the time. There was no way i pocket dialed or someone else called because it was in my hand. Doesn't make sense! Has this happend to anyone else?
Odd problem, but there are all kinds of problems with the wall chargers. If you want a new one, make an appointment with your local Genius Bar at the apple store, bring in your misfunctioning one, and they will swap it out for a new one. They do this routinely for charging bricks, sync cords, and headphones.
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When I make an outgoing call the recipient cannot hear me very well as though the volume was turned down very low
However when I activate the speakerphone the recipient can hear me loud and clear
Is there a fix for this?
Thanks
TeresaHey Tlabounty,
Thanks for the question. I understand you are experiencing issues with the microphone on your iPhone 5. To troubleshoot this issue, you may find the information in this article helpful:
iPhone: Microphone issues
http://support.apple.com/kb/ts5183
Note that speakerphone and regular calls utilize seperate microphones on the device. It sounds like you have isolated the issue. Furthermore, if you determine that the issue is hardware related, the following resource may help you get started with the service process:
iPhone Repair and Service - Apple Support
http://www.apple.com/support/iphone/repair/
Thanks,
Matt M. -
My 4s has no sound on most of the outgoing calls i make. It will ring the other phone I am calling but I cannot hear it nor can they hear me when they answer. Its just dead air but my phone timer is in use. How can this be corrected?
It's very possible that you have stuff running in the background that you don't even realize you have running. For example the default is for all notifications to be on and you usually need few to none of them.
That said, while an iPad can run warm, too hot to handle isn't right. ALso you should get about 10 hours....depending on what you're doing of course. If you're doing processor intensive stuff, like gaming and videos it can drain faster and generate more heat.
I would say if you have an apple store near by, make an appointment and have them check it out. If the issue is due to what they determine to be manufacturer's defects then it could fall under the warranty. But that's up to the store to determine. -
Incoming calls issue in Third Party SIP Phone
Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
ThanksDear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
To:
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info:
Contact:
Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^* -
CUCM: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls
Hi Team,
we are running CUCM 9.1(2a),
we have integrated Third Party SIP Phone(Avaya 1230 SIP Phone) with CUCM,
Issue: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls, we are able to see only the dailed Number,
When "A" calls to "B", "A" can see only the dailed number of "B" but not the "Caller ID"
Regards
AnanthakumarAre A and B both Avaya phones?
So it looks like you're not seeing the alerting name/connected name getting updated then? Do you have alerting names configured on the directory numbers? Might need to take a look at the SIP messaging to see if the alerting name/connected name is being sent to the Avaya phones and maybe they just aren't displaying it. Might just be something that needs to be tweaked in the 46xxsettings.txt file. -
Cisco 3905 failed but outgoing call still ringing on mobile phone
Dear all,
We setup Cisco Call Manager 10 with Cisco Voice gateway 2911 ( E1 port) and 3905. All outgoing call & incoming call are ok. But we meet a small issue as below:
-Using 3905 do outgoing call to mobile number. Mobile number ringing.
-3905 failed without power but the call still ringing on mobile phone.
Anyway to fix that ?
Thank you & regards,
ThànhNTHello,
So, your config looks good.
Pelase, can you explain what you mean when you say:
"-3905 failed without power but the call still ringing on mobile phone."
You can try a "debug isdn q931" on your gateway and see if the disconnect is ok. Use the documentation bellow to understand the debug:
http://www.cisco.com/c/en/us/td/docs/ios/12_2/debug/command/reference/122debug/dbfipx.html#wp1018126
If the disconnection is good, you will need to understand your call flow.
Phone -> CUCM -> Gateway -> PSTN -> Called Party
The final disconnection point is provided by your telco, and it can send the disconnect late.
Please, let me know the result of the debug.
best regards. -
Cisco IP phones with only incoming calls not outgoing calls
The problem in one of our client place where we implemented the Cisco IPT using Cisco Callmanager 4.1 running on MCS 7800 server with the IP phones being used over there are 7940G, 7912G. is that sometimes the phones are unable to call outside but they can receive calls from outside. When on the Cisco CallManager Administration web console reset the gateway (by going to Devices> Gateway, then check the box next to the gateway and click reset) the outgoing calls can be performed again.
Where will the problem be either in callmanager or the voice gateway router 1760?
Please let us know the solutions to this issues asap.
Thanks
swamyI would check the IOS version you are running, first off, and use Bug Tool or downloads to see if there are VOIP issues with the Feature Set or Train you are using. I know from the past, I personally had MANY problems with the 1760 router platform and had to try several IOS Ts to find one that worked consistently. The other thing you ought to look at is whether it is an MGCP or H.323 gateway. I would suggest redeploying as an H.323, as it can be a bit more manageable. The difference is:
MGCP => CallManager controls dial plan with Part / CSSs
H.323 => the router control dial plan with dial peers in IOS
HTH, -
When making an outgoing call from my i phone, an icon of a microphone with a slash across appears. I can hear the recipient speakin but he cannot hear me. What has been turnedd off, and how do I turn it on?
I think mute is on, tap the microphone icon with the slash running through it
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i have downloaded ios7 but now the recent calls (all and missed) in phone is not showing any of the received or outgoing calls. How do I get this to show?
First try a Reset ( No Data will be Lost )
Press and hold the Sleep/Wake button and the Home button at the same time for at least Ten seconds... (But can take Longer)... until the Apple logo appears. Release the Buttons
If no joy... Try a Restore...
1: Connect the device to Your computer and open iTunes.
2: If the device appears in iTunes, select and click Restore on the Summary pane.
Restoring > http://support.apple.com/kb/HT1414
Make sure you have the Current Version of iTunes Installed on your computer
iTunes free download from www.itunes.com/download
3: If the device doesn't appear in iTunes, try using the Steps in this article to force the device into Recovery Mode.
Note on Recovery Mode.
You may need to try this More than Once... Be sure to Follow ALL the Steps...
After you have Recovered your Device...
Re-Sync your Content or Restore from the most Recent Backup...
Restore from Backup > http://support.apple.com/kb/ht1766
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