E72 - GSM commands dialed over SIP

Normally GSM commands (e.g. *100#) are sent with the green phone key. As soon as a sip service is started, they are dialed over sip, instead of sent through the GSM network. I assume this is a bug and not a feature.
Nokia E72

IPv6 seems to be working fine on my 920.
there was some update that just came through and now my phone has a new setting offer
settings/access point/
and I can choose between T-Mobile or T-Mobile ipv6
you can also test to see if it is working by checking your ipv6 address.
open the web browswer and type:
whatismyipv6.com
if it shows up, you will be given your address and it will show ipv6.
if not, it will show you ipv4

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How to implement VoIP over SIP in J2ME

Hi All,
This is kapil from India and i want to implement VoIP over SIP in j2me.Any help will be appreciated.
I know about SIP API in J2ME. I want to use VoIP over it.Please help.I will be grateful to you.
regards
kapil

Why don't you give us the money your Boss pays you.. then we'll see what can be done ;)
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I have a customer that needs to have their phones automatically dial a particular number if no digits are pressed after 10-15 seconds. We are accomplishing this right now with our SCCP phones by having a blank translation pattern that points to an ambiguous hunt pilot.
For example, we create a blank translation pattern that translates to a HP with number 300. We also create a HP with number 3000. The translation pattern is triggered, but because 300 is ambiguous (the call could possibly be destined for 300 or 3000) CUCM waits for the interdigit timeout before calling 300.
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Thanks for your comments, but I'm afraid you guys are missing the issue.
Chris, you are correct that you can get a SIP phone to auto-dial a number via SIP dial rules. The problem is that it triggers immediately. I need the SIP phones to wait.
Here's a scenario - SCCP phones are in use all throughout a nursing home. Residents are told that they can simply knock their phone offhook if they need emergency help. If they don't dial anything within 15 seconds, their phone will automatically call into the local emergency oncall nurse. Sorta an "onstar" emergency help line for an old folks home. We definitely can't have that rule triggering immediately though, as residents would never be able to place a normal call, and there would be no way of distinguishing the emergency calls from the guy who is just trying to call for pizza. :)
This function works perfectly with SCCP phones - while they trigger the blank translation pattern immediately, CUCM doesn't actually begin dialing because there are multiple potential matches for what the translation pattern is dialing. SIP phones do not do this, however. If you have the SIP dial rule applied, it always triggers immediately, and it does not wait to see if callers give any input.
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Microphone streaming over SIP protocol.

Hello,
can I stream microphone voice over SIP protocol?
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thanks

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Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
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Thanks in advance for any help.
//Per

Thanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
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E72 restarts while browsing over WiFi / wlan

Hi
I've e72 for few weeks. In general,I'm satisfied with what I get. A major exception is a strange
resets I've almost allways during browsing over WiFi. Browser isn't an issue,it happens with both Nokia browser and Opera. It doesn't happen when I connect with mobile provider GSM.
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Hey Guys,
Probably your phone memory is full and your access points are overflowing. Do not store external data (like music, photos, videos etc. ) in your phone memory. And check whether your access points are full and delete unwanted access points. For this go to Settings >> Connectivity >> Access Points.
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Regards

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any adivce in Troubleshooting would be greatly appreciated.

Are you able to perform a CUCM SDI trace during a problem call?
Does that Line:Phone have the same Calling Search Space as the other phones on the remote site?
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CM Register over SIP Trunk

Hi guys,
would it be possible to allow sip users to register over the sip trunk on the Call Manager? or is this method not allowed?
Thanks.
Best regards

Hi Manish,
between sip client and webrtc gw -> ws and between webrtc gw and CM -> sip.
here are the sip messages.
both phones are registered, 9000 is a 7912 and 8080 is sip.
192.168.15.2 - CM
192.168.15.202 - webrtc
SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:[email protected]>;tag=400660433
To: <sip:[email protected]>
Contact: <sip:[email protected]:10060;ws-src-ip=192.168.251.105;ws-src-port=50731;ws-src-proto=ws;transport=udp>
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Content-Type: application/sdp
Content-Length: 978
Max-Forwards: 70
Authorization: Digest username="8080",realm="ccmsipline",nonce="gHqGqDWK4zTzv6Ijl6ixW58AK/Gm4yC6",uri="sip:[email protected]",response="352cb2e17e36b32ee4e0d52443d0a106",algorithm=MD5
User-Agent: webrtc2sip Media Server 2.6.0
v=0
o=doubango 1983 678901 IN IP4 192.168.15.202
s=-
c=IN IP4 192.168.15.202
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 58690 RTP/AVP 8 0 101
c=IN IP4 192.168.15.202
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:1YfBfgbhIdMB6YVtyZgJqc77QPHwm9o42aEPbkHD
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:fujGVOi70hQnKkeUimcFUw2bH3ajZ2iW0xKy5Nrw
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:4034073057 cname:c08c56217e96dbc1e8234373eb5d2fcc
a=ssrc:4034073057 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:4034073057 label:doubango@audio
a=ice-ufrag:uaektHZ6KFVn1fw
a=ice-pwd:HAj21nuOrDmIKl3ANXTc3K
a=candidate:tWR5PLw1x 1 udp 2130706431 192.168.15.202 58690 typ host
a=candidate:tWR5PLw1x 2 udp 2130706430 192.168.15.202 58691 typ host
RECV:SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:[email protected]>;tag=400660433
To: <sip:[email protected]>
Date: Wed, 19 Mar 2014 13:26:05 GMT
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Allow-Events: presence
Content-Length: 0
RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:[email protected]>;tag=400660433
To: <sip:[email protected]>;tag=856401750
Date: Wed, 19 Mar 2014 13:26:05 GMT
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Allow-Events: presence
WWW-Authenticate: Digest realm="ccmsipline", nonce="gHqGqDWK4zTzv6Ijl6ixW58AK/Gm4yC6", algorithm=MD5
Content-Length: 0
SEND: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:[email protected]>;tag=400660433
To: <sip:[email protected]>;tag=856401750
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 ACK
Content-Length: 0
Max-Forwards: 70
Receiving SIP o/ WebSocket message: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKZEk81zTwfVde8oImts6ZHiTzchfBWh1N;rport
From: "8080"<sip:[email protected]>;tag=XFKqC4zu0S9QfzzMzQ4u
To: <sip:[email protected]>;tag=1464334432
Call-ID: ecc84fa2-3de3-d953-527f-5e7515cabca3
CSeq: 29519 ACK
Content-Length: 0
Route: <sip:192.168.15.2:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Thanks.

Unity Connection not passing CallerID to CUCM over SIP Trunk

I'm trying to get CallerID working for Unity Connection Device Notification (and it seems everything else), however, when I run UC Remote Port Status Monitor and the Call-Out goes to CUCM for the Device Notification, no caller ID is presented to the CUCM SIP trunk.
06:06:02, New Call, CalledId=, RedirectingId=, Origin=16, Reason=1024, CallGuid=,
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[Origin=Unknown],[Reason=Unknown]
06:06:02,
Dialing '99254753'
06:06:32, Idle
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10:59:01.005: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x5B03
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                Standard = CCITT
                Transfer Capability = Speech
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I've looked through my SIP trunk on the CUCM side and for Inbound Calls, Connected Line ID and Presentation Name are set to "allowed" or "default" doesn't make a difference. RTMT Port Status also shows no "caller", so I'm thinking there is some way to set or allow the calling number on the Unity Connection (8.5) side.
Oddly enough, I also noticed that in Unity Connection> Telephony Integrations > Port Group, if I change the Contact Line Name from nothing to "Unity" (or whatever), the Q931 debug outbound doesn't show ANY "Calling Party Numer - = XXXXX" and the carrier throws out the BTN as the ANI.
10:46:00.837: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x5AFF
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                Transfer Capability = Speech
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                Exclusive, Channel 23
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THANKS!!
Mike.

I did not- my work around has been to put in a name for Contact Line Name under Port Group Basics Switch configuration in Unity Connection- this for some reason keeps CUCM from sending ANI TYPE/PLAN information in the Q931 message, and my carrier then sends a default ANI of the circuit's BTN. When I have time, I'll open up a TAC ticket.
Mike.

E72 unable to create new sip profile

I am trying to create new sip profile from options>settings>connection>sip settings
nohing happens when I select new sip profile using default profile
I hav already reset the phone using *#7780# but problem is still there
please help ?
Solved!
Go to Solution.

*#7780# does not reset the phone. It just restores factory default settings.
In order to reset the phone you need to use *#7370# instead. Back up everything except your settings beforehand because this does erase the phone's C: drive entirely.
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you!

How to make inline format command carry over to subsequent pages

Using Designer 5.7. Using inline formatting in the data file (\b. \b0.) to make part of the text bold (Example: bold text bold text regular text). The problem occurs when the end of page is reached. The text flows to the next page and if the text is split across pages after the bold command was turned on but before it is turned off, the text at the top of the new page is not formatted.
Example of Bottom of Page 1: This is an example of the text at the bottom of page one. This text is supposed to be
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Using 1 line subform and an identical overflow subform formatted to reserve space and a Switchpage override in in the Preamble.
It's not feasible to use a field formatted for bold and also be able to determine when it should be used during document generation since the text may be variable in length. Any information is appreciated.

For the time being, I can only respond to Pages for iOS. OS X I'll need to check when I get home (or perhaps someone else will beat me to it). You may wish to post in the Pages for Mac forum for better exposure.
In Pages for iOS you are limited to a single header and footer. There is no Section structure so no way to do different headers/Footers for different pages (as you can do in MS word for example). To modify the Header and insert the text you want: Tap the Tools icon (wrench in upper right), then Document Set up. Tap in the header field, then tap and hold to use the menu to insert pages numbers, then type the text you need.
Quite possibly. Check the app store. Or you may be able to set this up in Pages for OS X and import it into Pages on the iPad. I'm not certain the sections will import however.
Also consider using a different app, such as word for iOS or any of the numerous other word processing apps available.

Can LabVIEW overlay a dial over an image?

I would like to overlay a dial on top of an image in LabVIEW. It would be great if the dial could rotate about the center axis. I included an attachment of what It would possibly look like. Please let me know if any of you know how to do this.
Thank you!
Solved!
Go to Solution.
Attachments:
DIAL.jpg ‏55 KB

You need to set the other items on a normal dial to transparent and place it on top the image.
Omar
Attachments:
Gauge.vi ‏45 KB

Flash sockets over SIP-protocol: HTTP/1.1 501 Not Implemented

Hi there.
I'm developing a chat app based on IRC server and flash client (flash sockets). Everything is OK but for some ISP flash client doesn't connect to IRC server: There are some logs from chat while connecting to IRC server:
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There is only thing I'm aware - this issue occurs for SIP mobile network provider only. For other non-mobile providers issue doesn't occur and chat works good.
I would really appreciate for any thoughts on this matter.

Hi there.
I'm developing a chat app based on IRC server and flash client (flash sockets). Everything is OK but for some ISP flash client doesn't connect to IRC server: There are some logs from chat while connecting to IRC server:
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    Receive: Content-type: text/html
    Receive: Content-Length: 219
There is only thing I'm aware - this issue occurs for SIP mobile network provider only. For other non-mobile providers issue doesn't occur and chat works good.
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Register TANDBERG MXP 6000 over SIP

Hi, i have MXP6000 with 9.1 software. Cant make it register with SIP. No single packet comes from MXP to server.
Has anyone been able to make it register with SIP server?
Config is quite simple:
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*c xConfiguration SIP MNS Mode: Off
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Does it matter? NO REGISTER packets arrived to server, i was sniffing traffic.
Problem solved just after i entered valid DNS server address in IP parameters. Why would it need DNS if i'm using direct IP addresses...
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NICE recorder - one way or garbled audio over SIP trunk.

Customer with CUCM 10.5.2 trying to integrate with a NICE recording solution. Everything configured per NICE documentation and rechecked that several times with the NICE vendor. However, if calling to/from the PSTN or legacy Nortel PBX to a Cisco 78xx or IP Communicator soft phone - we get one-way audio pushed out the SIP trunk to the NICE system. If it's a Cisco to Cisco phone call - the audio is garbled.
Has anyone experienced this issue with this type of integration - or the same issue with a SIP trunk to the CUCM to another system at all? We're at a loss here.
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This document should help you:
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008009484b.shtml

E72 - GSM commands dialed over SIP

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