SIP configuration in Nokia N 97
I am unable to configure SIP in my N 97, whereas E 71 phone was ver easy to configure. Anyone can suggest any steps to help me. the configuration is as below:
SIP Server Or SIP Proxy Server: ata.nymgo.com
SIP User ID: nymgo username
Authenticate ID: nymgo username
SIP password: nymgo password
SIP Port: 5060
Codec Used: g729
STUN Server: stun.nymgo.com
STUN port: 80
Try search there are many many posts on this here.
I love my purple screen, 1/2 day battery, sketchy touch screen and $800 price tag - I call it my Lumia 900
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VoIP (SIP) configuration guide for Nokia 5800xm
hi.everybody i would like to ask how to configure this
VoIP (SIP) configuration for my n5800, because when i saw and follow this link http://www.elisanet.fi/craig/sipvoip/nokia_n97.html ,i only configure my SIP setting and not my net setting which i tried to find in my phone. can someone help me find this Net setting in my handset and can you put a link on how to call from phone to phone using this "VoIP" and is this free of charge. thank..more powerHi, don't have an answer to your question, but Skype for S60.5 was announced today, works fine, will give Skype to Skype calls free, and low cost calls to non Skype numbers, but both Skype or Voip will incur data charges if you don;t have a data plan !
Good Luck
Skype.com/m with your phones browser will take you to Skype
If I have helped at all, a click on the White Star is always appreciated :
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Hi, I want sip settings of actionvoip in my Nokia E63 i wanna make new server profile plz help me if every one know this problem... I shall b very thankful to you for that.
I recommend the following steps for setting up your Nokia E63 for VoIP over Wifi (and perhaps for other S60 based phones as well), for Voip.ms, the realm settings were the same as the others:
How to configure SIP profile in Nokia Eseries devices for VoIP calls?
This text is a cut-and-paste from Nokia's FAQ.
The Nokia Eseries mobile devices support SIP VoIP internet calls. This article describes a sample configuration when user authentication is enabled in the SIP proxy and registrar servers. The proxy and registrar servers are located in the same device in this sample.
Before configuring the mobile device, the following information should be obtained from the SIP service provider (some other parameter values may also need to be checked if not using the default values):
SIP proxy and registrar server address ("sip.voipservice.com" in this sample)
SIP proxy realm ("inphone.com" in this sample)
SIP registrar realm ("sip.voiceservice.com" in this sample)
User name ("1234567" in this sample for both the proxy and registrar)
User password ("1234")
Public user name ("[email protected]" in this sample)
In order to establish a VoIP call, a WLAN access point needs to be defined in the mobile device access point settings and the mobile device must be within the reach of the WLAN network.
Start the mobile device SIP configuration by going to menu Tools > Settings > Connection > Sip Settings and by creating a new SIP profile. The following parameters are defined in the profile:
Profile name: Siptest (can be freely chosen)
Service profile: IETF
Default access point: WLAN access point used for the VoIP connections
Public user name: [email protected] (note that the public user name must be given in this format, including the domain name)
Use compression: no
Registration: when needed (set "always on" if you want the registration to occur automatically)
Use security: no
Proxy Server:
Proxy Server Address: sip.voipservice.com
Realm: inphone.com
User name: 1234567
Password: 1234
Allow loose routing: yes
Transport type: UDP
Port: 5060
Registrar:
Registrar Server Address: sip.voipservice.com
Realm: sip.voipservice.com
User name: 1234567
Password: 1234
Allow loose routing: yes
Transport type: UDP
Port: 5060
Note that the realm name is often the same for both the proxy and registrar but the name may also be different, as in this sample. If the registrar realm name is not correctly defined, the registration will fail. If the realm name of the proxy is not correct, the registration may succeed but the internet calls will fail. -
Please help with SIP configuration on 2801 router
Hi All.
Please help me to setup a SIP account. I’m already struggling to do that for a few days, and can’t find out how to finish that. We have 2xISDN lines running, so I need to add a SIP trunk to existing config.
The information from our SIP provider:
We have issued the following DDI range: 018877000 – 99
There is no need to register the DDI’s as these will be offered to your PABX IP address provided to in the completed SIP trunking form.
Configuration details are as follows:
Our Primary Proxy:- 99.234.56.78
Codec supported:- G711Alaw, G729 (G711Alaw is the preferred codec)
Fax Support:- T38 and G711Alaw
DTMF:- RFC2833 and INFO
CLI Method:- Remote-Party-ID
Trunk doesn’t require registration; you just need to send Invite. In cisco this is done through Dial-peer session-target command. We are authenticating your IP address for outgoing calls and incoming calls we then forward to the IP mentioned in the sip form.
This is a SIP configuration on Cisco2801 router (I used outgoing calls only):
translation-rule 10
Rule 0 ^90 0
Rule 1 ^91 1
Rule 2 ^92 2
Rule 3 ^93 3
Rule 4 ^94 4
Rule 5 ^95 5
Rule 6 ^96 6
Rule 7 ^97 7
Rule 8 ^98 8
Rule 9 ^99 9
interface FastEthernet0/0.1
description ***DATA VLAN***
encapsulation dot1Q 1 native
ip address 10.1.1.101 255.255.255.0
interface FastEthernet0/0.2
description ***VOICE VLAN***
encapsulation dot1Q 2
ip address 192.168.22.1 255.255.255.0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
call start slow
sip
bind control source-interface FastEthernet0/0.2
bind media source-interface FastEthernet0/0.2
registrar server expires max 36000 min 600
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 1 pots
description ### External Dialling via BRI ###
preference 7
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/0
forward-digits all
dial-peer voice 2 pots
description ### External Dialling via BRI ###
preference 2
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/1
forward-digits all
dial-peer voice 9000 voip
description ** Outgoing calls to SIP **
preference 1
destination-pattern 9T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:99.234.56.78:5060
dtmf-relay rtp-nte
codec g711alaw
no vad
sip-ua
timers connect 100
sip-server ipv4:99.234.56.78
I used debugging commands to troubleshoot the calls.
2801(config-dial-peer)#
094509: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=211, Called Number=, Voice-Interface=0x65FA35B4,
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
094510: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20018
094511: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
094512: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9
094513: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094514: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094515: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90, Peer Info Type=DIALPEER_INFO_SPEECH
094516: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90
094517: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094518: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094519: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=908, Peer Info Type=DIALPEER_INFO_SPEECH
094520: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=908
094521: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094522: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094523: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9086, Peer Info Type=DIALPEER_INFO_SPEECH
094524: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9086
094525: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094526: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094527: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862, Peer Info Type=DIALPEER_INFO_SPEECH
094528: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862
094529: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094530: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094531: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=908621, Peer Info Type=DIALPEER_INFO_SPEECH
094532: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=908621
094533: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094534: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094535: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9086215, Peer Info Type=DIALPEER_INFO_SPEECH
094536: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9086215
094537: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094538: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094539: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157, Peer Info Type=DIALPEER_INFO_SPEECH
094540: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157
094541: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094542: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094543: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=908621577, Peer Info Type=DIALPEER_INFO_SPEECH
094544: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=908621577
094545: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094546: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094547: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9086215777, Peer Info Type=DIALPEER_INFO_SPEECH
094548: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9086215777
094549: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094550: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094551: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094552: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
094553: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
094554: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
094555: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
094556: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774T
094557: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
094558: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
094559: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094560: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
094561: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
094562: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
094563: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
094564: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
094565: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094566: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
094567: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
094568: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
094569: Jan 24 09:27:10.719: fb_get_reject_cause_code: ERROR cause_code NULL
094570: Jan 24 09:27:10.727: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397230
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094571: Jan 24 09:27:11.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397231
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094572: Jan 24 09:27:12.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:12 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397232
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094573: Jan 24 09:27:14.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam" <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:14 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397234
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I made some changes in the router configuration.
I removed FA0/0.2 Voice interface from Voice service voip configuration (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2). And now it’s using ip address 10.1.1.101 (data ip).
The debugging is changed now. I can send and receive a respond from SIP server. But It shows an error: SIP/2.0 404 Not Found
Then it moves to ISDN line, and use this line to make a call.
102988: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
102989: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774T
102990: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
102991: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
102992: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102993: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
102994: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
102995: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
102996: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
102997: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
102998: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102999: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
103000: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
103001: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
103002: Jan 24 14:45:47.298: fb_get_reject_cause_code: ERROR cause_code NULL
103003: Jan 24 14:45:47.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Seam" <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3989446920-1171263969-2466545983-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327416347
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 2438 9821 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 19412 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
103004: Jan 24 14:45:47.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
From: "Sam "<sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Call-ID: [email protected]
CSeq: 101 INVITE
Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK4875CB9
Content-Length: 0
103005: Jan 24 14:45:47.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
From: "Sam " <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
103006: Jan 24 14:45:47.374: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up
103007: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=211, Peer Info Type=DIALPEER_INFO_SPEECH
103008: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=211
103009: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
103010: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20018
103011: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=0862157774, Peer Info Type=DIALPEER_INFO_SPEECH
103012: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0862157774
103013: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
103014: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=NO_MATCH(-1)
103015: Jan 24 14:46:08.815: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down
2801(config-dial-peer)#
Then I removed SIP-UA as I was told there is no registration necessary, only Dial-peer configuration.
But it didn’t affect anything.
Then I add translate-outgoing called 10 command to dial-peer 9000, nothing happened.
Really stuck and don't know where to look at.
Any help will be highly appreciated.
Thanks.Hi Dan.
Yes, I saw that RTP debugging, but what can I change there? Maybe I need to open more ports on ASA for RTP like 19412?
I use Cisco ASDM for ASA to make changes.
There are static NAT rules for: Server source IPs(10.1.1.100) to Outside(translated IPs, 88.99.77.44) for a few ports.
Also I added Security policy access rules for LAN: Any to SIP, and Outside: SIP to any.
For NAT:
I can't add this: for LAN: STATIC ROUTER IP 10.1.1.101 (AS SOURCE) UDP 5060 TO OUTSIDE IP 88.99.77.44
(AS TRANSLATED) UDP 5060
Because there is already translation for the Server.
Debugging looks like that now. There is no Received: SIP/2.0, but I can make an outside call with no audio.
116013: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
116014: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
116015: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
116016: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=90862157774
116017: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
116018: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=9000
2: Dial-peer Tag=2
3: Dial-peer Tag=1
116019: Jan 25 15:28:25.588: fb_get_reject_cause_code: ERROR cause_code NULL
116020: Jan 25 15:28:25.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:25 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505305
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116021: Jan 25 15:28:26.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:26 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505306
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116022: Jan 25 15:28:27.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:27 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505307
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116026: Jan 25 15:28:57.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam" ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389preference 1771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505337
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I'll add Incoming dial-peer now.
Not sure what kind of NAT rule should I put into ASA to allow in and out sip traffic.
Appretiate your help.
Thanks a mill. -
Hi
Can someone tell me where to find the Sip configuration for VYKE system?Profile name: Arbitrary
Service profile: IETF
ACCESS POINT : [your wifi access point]
PUBLIC USER NAME: sip:[username]@sip.vyke.com
use compression: no
registration : “always on” or “when need”
SECURITY: NO
PROXY SERVER
PROXY SERVER: sip:sip.vyke.com
REALM: sip.vyke.com
USER NAME: :[sip_user]
password: [sip_pass]
allow loose routing: yes
transport type: UDP
PORT: 5060
REGISTRATION SERVER
REGISTRATION SERVER: sip:sip.vyke.com
realm: sip.vyke.com
user name: [sip_user]
password: [sip_pass]
transport type: UDP
PORT: 5060
N86 8mp: RM-485,0590552; Version 20.115 - 6/29/2009. - -- E71-2; RM-357; 0569371; Version 100.07.76 - 6/08/2008.
N8; N900 -
Hi,
On my gateways 5400, SIP configuration is :
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface Loopback6
bind media source-interface Loopback6
But on one of my 5400, line "bind control source-interface Loopback6" doesn't appear. When i implement it, i have no error message.
SIP calls doesn't complete (dead air). Dial-peer is well configured (calls completed on another gw).
Thanks for your help.
PhilCompare the IOSes running on these AS5400s.
Are they different? :)
If they are, choose one that you consider as being most stable and make the other running it :) -
How to configure SIP settings on Nokia E51
Hello to all.
I need help. I have
Username
Authentication Username
SIP address
SIP Password
Domain
Outbound Proxy
in one of big SIPs provider but i cant configure my phone work with it. Please help me. I allready installed SIP Voip settings from nokia store.Have you checked out this post upon forum http://discussions.nokia.com/t5/Connectivity/Nokia-E51-SIP-Settings/m-p/481535/highlight/true#M24073 or this resource?http://www.voip-connections.com/wp-content/uploads/2011/06/How-to-configure-Nokia-E51-and-E71-to-con...
Happy to have helped forum in a small way with a Support Ratio = 37.0 -
Nokia N91 SIP Configuration (problem)
Hi all,
As it ays in the subject, I own a nokia N91 handset and i'm having problems accessing SIP Settings.
Basically when I click on SIP Settings, and view SIP providers. I cannot delete, edit, or add any settings at all! When I do select (for example), options - delete - delete confirmation (yes) The SIP profile doesn't delete.
I have done a full factory reset on the phone (*#7370#), but still no luck19-Apr-2008 06:11 PM
os2lover wrote:
Hi,
No the *#7370# doesn't affect your harddisk, only the phone memory.
Best regards,
Olivier Baum
ohh thanks for this info... Im just curious, how do we format the hard drive? Thanks
CeS
"The best index to a person's character is how he treats people who can't do him any good, and how he treats people who can't fight back" -
SIP configuration on N95 problem
Hi,
I'm trying to set a SIP profile on my N95 using Nokia native SIP. My goal is to run a VoIP service named Spikko.
I followed the Skippo instruction sheet (to be enclosed below), but fail to establish a connection.
I checked and rechecked the SIP parameters and found a difference between my inputted paramters and the ones which my N95 uses. Whenever I define an address, in my case SIP.SPIKKO.COM, the parameter shown is SIPIP.SPIKKO.COM (SIP: added).
Can this explain my failure? If so, how do I solve this problem? If that's a harmless change, what can explain and solve my problem?
Here's the configuration instruction sheet:
Connecting to Spikko using Nokia SIP:
Nokia N95, E71, E61, and several other models include a Nokia “Native SIP” application. The configuration differs slightly between models. The following is the N95 configuration:
• Go to “Menu » Tools » Settings » Connection » SIP settings » Options » New SIP profile and use the following settings:
--Profile name: Spikko
--Service profile: IETF
--Default Access Point: Choose “WiFi” or “3G internet access point” - לא קיים
--Public username [email protected] (for instance: [email protected])
--Use compression: No
--Registration: When needed
--Use security: no
--proxy server:
--Proxy Server Address: sip.spikko.com
--Realm: spikko
--Username: your Spikko username
--Password: your Spikko password
--Allow loose routing: yes
--Transport type: Automatic
--Port: 5090
--Registration server: Same as proxy server
• Now, go to “Menu » Tools » Settings » Connection » Internet-Tel” and choose the Spikko profile. Your Nokia now is ready to make and receive calls.
• See additional information at http://www.spikko.com/Nokia/Movie.aspx.Hi,
I'm trying to set a SIP profile on my N95 using Nokia native SIP. My goal is to run a VoIP service named Spikko.
I followed the Skippo instruction sheet (to be enclosed below), but fail to establish a connection.
I checked and rechecked the SIP parameters and found a difference between my inputted paramters and the ones which my N95 uses. Whenever I define an address, in my case SIP.SPIKKO.COM, the parameter shown is SIPIP.SPIKKO.COM (SIP: added).
Can this explain my failure? If so, how do I solve this problem? If that's a harmless change, what can explain and solve my problem?
Here's the configuration instruction sheet:
Connecting to Spikko using Nokia SIP:
Nokia N95, E71, E61, and several other models include a Nokia “Native SIP” application. The configuration differs slightly between models. The following is the N95 configuration:
• Go to “Menu » Tools » Settings » Connection » SIP settings » Options » New SIP profile and use the following settings:
--Profile name: Spikko
--Service profile: IETF
--Default Access Point: Choose “WiFi” or “3G internet access point” - לא קיים
--Public username [email protected] (for instance: [email protected])
--Use compression: No
--Registration: When needed
--Use security: no
--proxy server:
--Proxy Server Address: sip.spikko.com
--Realm: spikko
--Username: your Spikko username
--Password: your Spikko password
--Allow loose routing: yes
--Transport type: Automatic
--Port: 5090
--Registration server: Same as proxy server
• Now, go to “Menu » Tools » Settings » Connection » Internet-Tel” and choose the Spikko profile. Your Nokia now is ready to make and receive calls.
• See additional information at http://www.spikko.com/Nokia/Movie.aspx. -
Hi All,
I have Nokia N 80 IE .i want to know how to configure VOIP Through SIP.
I have Callcentric account i try to configure but its showing Registraion Failur.
Pls Help me
BinoIt is more likley they're saying which phones are getting N-Gage support, rather then saying which ones are not. That leaves us wondering forever
A full list of future planned N-Gage support for all phones would be nice, rather then the "coming soon" ones. Just to make sure if it will ever happen to E71 (and other models ofc).
Message Edited by tmani on 16-Feb-2009 04:49 PM -
Mail Configuration in Nokia C5-00
I am facing problem in configuring Email in my new Nokia C5-00. I am following below steps, Menu->Application->Email->New mail box, Entering Mailid (with domain [email protected], [email protected]) & Password, Next screen is getting with "Validating e-mail account" and "Detecting email settings", After few seconds, its going back to email main page, without creating mail box. Its not showing any error/warning during or after configuration.
I want to do email through nokia c5 ,i have gprs of bsnl punjab .please send to me details of email from nokia c5
-
CME Extension Mobility, SIP configuration
Hi,
Need help with CME Extension Mobility with SIP Phones (7841). I'm using CME 10.5 and I configured the parameters below for extension mobility but the phones won't register right after I put the logout profile in the voice register pool.
They work normally when not in Extension Mobility though. Please help I need to deploy this to my customer soon.
hostname Router
boot-start-marker
boot system flash:c3900-universalk9-mz.SPA.154-3.M2.bin
boot-end-marker
no aaa new-model
no authentication logging verbose
ip dhcp excluded-address 192.168.1.1 192.168.1.20
ip dhcp excluded-address 192.168.1.254
ip dhcp pool Phones
network 192.168.1.0 255.255.255.0
default-router 192.168.1.254
option 150 ip 192.168.1.254
no ip domain lookup
ip cef
no ipv6 cef
multilink bundle-name authenticated
cts logging verbose
voice-card 0
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/1.10
bind media source-interface GigabitEthernet0/1.10
registrar server expires max 600 min 60
voice register global
mode cme
source-address 192.168.1.254 port 5060
max-dn 110
max-pool 110
load 7841 sip78xx.10-1-1SR1-4
time-format 24
date-format D/M/Y
service https
url authentication http://192.168.1.254/CCMCIP/authenticate.asp
tftp-path flash:
create profile sync 0002641841434163
voice register dn 1
number 6001
name Poh Huat - 6001
label Poh Huat - 6001
voice register dn 4
number 6005
name Coordinator - 6005
label Coordinator - 6005
voice register pool 1
logout-profile 100
busy-trigger-per-button 2
id mac 547C.69D6.1AB6
type 7841
voice register pool 4
logout-profile 100
busy-trigger-per-button 2
id mac 547C.69D6.1A2F
type 7841
voice logout-profile 100
pin 1234
user 6000 password 12345
number 6000 type normal
speed-dial 1 999 label "EMERGENCY"
voice user-profile 1
pin 12345
user richard password richard
number 6001 type normal
speed-dial 1 996506901 label "Richard"
voice user-profile 2
pin 12345
user 6005 password 12345
number 6005 type normal
license udi pid C3900-SPE100/K9 sn FOC16145MQA
license boot module c3900 technology-package uck9
username xtra privilege 15 secret 5 $1$STRs$Qsuesm8dF23Okof.vRyf5.
redundancy
ip ftp username xtra
ip ftp password xtra2006admin
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address dhcp
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
interface GigabitEthernet0/1.10
encapsulation dot1Q 10 native
ip address 192.168.1.254 255.255.255.0
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
no ip http secure-server
ip http path flash:
nls resp-timeout 1
cpd cr-id 1
tftp-server flash:PHONES/sip78xx.10-1-1SR1-4.loads alias sip78xx.10-1-1SR1-4.loads
control-plane
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
gatekeeper
shutdown
telephony-service
authentication credential 6000 12345
em keep-history
max-ephones 110
max-dn 110
service phone webAccess 0
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 05 2015 15:50:52
I have turned on debug ip http all and debug voice em-profile on and right after I entered the logout profile 100 under pool i get the following logs.
Router(config-register-pool)#
Mar 5 16:15:28.299: Thu, 05 Mar 2015 16:15:28 GMT 192.168.1.21 /CMEserverForPhone/serviceurl ok
Protocol = HTTP/1.1 Method = GET Query = locale=English_United_States&name=SEP547C69D61A2F
Mar 5 16:15:28.299:
Mar 5 16:15:28.299: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:28.299: SIP phone 4 found with contact IP address 192.168.1.21
Mar 5 16:15:33.363: Thu, 05 Mar 2015 16:15:33 GMT 192.168.1.21 /CMEserverForPhone/serviceurl ok
Protocol = HTTP/1.1 Method = GET Query = locale=English_United_States&name=SEP547C69D61A2F
Mar 5 16:15:33.363:
Mar 5 16:15:33.363: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:33.363: SIP phone 4 found with contact IP address 192.168.1.21
Mar 5 16:15:37.539: Thu, 05 Mar 2015 16:15:37 GMT 192.168.1.21 /CMEserverForPhone/extensionmobility ok
Protocol = HTTP/1.1 Method = GET
Mar 5 16:15:37.539:
Mar 5 16:15:37.539: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:37.539: SIP phone 4 found with contact IP address 192.168.1.21
After this the phones are still not registering, I'm suspecting it is the url authentication command, as i can't put the application-name and password after the command, any suggestions would be appreciated. THanks in advance.
-richardTry adding:
voice register global
url authentication http://192.168.1.254/CCMCIP/authenticate.asp secretname psswrd
if still doesn't work try to compare your config with the reference guide here:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmemobl.html#pgfId-1163414
-Terry
Please rate all helpful posts -
PROBLEMS IN J2ME IN RECEIVING SIP MESSAGES IN NOKIA E60
Hello guys!
I'm currently developing an instant messaging for sip messages in the NOKIA E60 platform.
Registration process to the proxy functions properly, sending of messages to target URI functions properly, however, when a softphone send a sip mesage to the proxy-routing to my application. My application doesn't receive anything. I used the ethereal to see incoming frames in the proxy and i found out that Nokia E60 send a "bad media type mesasge" as a reply to the message. Any feedback in this matter ? My code is written below:
import javax.microedition.io.*;
import javax.microedition.sip.*;
import java.io.*;
import javax.microedition.lcdui.*;
public class UserForm extends List implements CommandListener
//private variables
private Command m_exit=null, m_select=null, m_back=null;
public Databank m_data=null;
public IM_SIP m_sip=null;
public WriteForm m_writeForm=null;
public ReadForm m_readForm=null;
public PhoneBook m_book=null;
public RegisterThread m_register=null;
public void Stop_MainThread()
//m_game.stop_MainLoop();
public void cleanresource()
try
catch(Exception ex)
System.out.println("UserForm"+ ex.toString());
public Display getscreen()
return m_sip.m_disp;
public void commandAction(Command cmd, Displayable disp)
int index;
index=this.getSelectedIndex();
if(cmd==m_select)
//determine what the user wants
switch(index)
case 0://write message option
m_sip.m_disp.setCurrent(m_writeForm);
break;
case 1:
m_sip.m_disp.setCurrent(m_readForm);
break;
case 2:
m_sip.m_disp.setCurrent(m_book);
break;
case 3:
break;
else if(cmd==m_exit)
try{
m_sip.destroyApp(false);
m_sip.notifyDestroyed();
m_sip=null;
catch(Exception ex)
//System.out.println("UserForm Exit" + ex,toString());
System.out.println("UserForm_Exit:" + ex.toString());
else if(cmd==m_back)
this.setTitle("Instant Messaging SIP");
//if(m_continue==false || multi==true){
this.deleteAll();
this.removeCommand(m_back);
this.add_other_items();
this.addItems();
this.addSoftButtons();
//else
// this.addContinue();
private void addSoftButtons()
this.addCommand(m_exit);
this.addCommand(m_select);
private void removeSoftButtons()
this.removeCommand(m_exit);
this.removeCommand(m_select);
private void addItems()
//this.append(new String("Write Message"), null);
//this.append(new String("Inbox"), null);
this.append(new String("Phonebook"), null);
this.append(new String("Options"), null);
public void add_other_items(){
this.append(new String("Write Message"), null);
this.append(new String("Inbox"), null);
public UserForm(IM_SIP p_sip)
super("Instant Messaging SIP", List.IMPLICIT);
//initialize all the important variables
System.out.println("INITIALIZING....");
m_sip=p_sip;
//this.addItems();
m_exit=new Command("Exit", Command.EXIT, 1);
//m_instform=new Instructionform("Game Instruction", this, new String("- Objectives: Get the flag of the enemy at all cost\n- For every you beat an officer, your score increses\n- Controls: Press the directional keys, 2, 4, 6, 8, press the ok button to confirm an action."));
m_select=new Command("Select", Command.SCREEN, 2);
m_back=new Command("Back", Command.SCREEN, 1);
m_data=new Databank();
System.out.println("FINISHED DATABANK INITIALIZATIOn...");
System.out.println("FINISHED DATABANK2 INITIALIZATIOn...");
m_book=new PhoneBook(this);
System.out.println("FINISHED DATABANK3 INITIALIZATIOn...");
//m_game=new GameSurface(this, m_data);
//m_view=new ScoresView(m_data, this);
this.addCommand(m_exit);
this.addCommand(m_select);
this.setCommandListener(this);
System.out.println("FINISHED DATABANK3 INITIALIZATIOn...4");
//m_register=new RegisterThread(this, m_readForm);
m_register=new RegisterThread(this);
m_register.start();
//m_readForm=new ReadForm(this);
//m_writeForm=new WriteForm(this, m_readForm);
/** Register Message Thread */
class RegisterThread extends Thread implements SipClientConnectionListener {
private int recTimeout = 0; // do not wait, just poll
public String message="";
//public String proxyAddress="[email protected]";
public String msgSubject="";
private UserForm form=null;
//private ReadForm m_read;
private SipClientConnection scc = null;
RegisterThread(UserForm p_form){
super();
form=p_form;
// m_read=p_read;
public void notifyResponse(SipClientConnection sc) {
try {
scc.receive(recTimeout);
form.add_other_items();
form.addItems();
form.append("Response register received: " + sc.getStatusCode() + " "
+ sc.getReasonPhrase(), null);
scc.close();
form.m_readForm=new ReadForm(form);
form.m_writeForm=new WriteForm(form, m_readForm);
//form.append(new String("Write Message"), null);
//form.append(new String("Inbox"), null);
} catch(Exception ex) {
form.append("MIDlet: exception " + ex.getMessage(), null);
ex.printStackTrace();
public void run() {
try{
//try {
// if (waitFor!=null) {
// waitFor.join();
// } else {
// } catch (InterruptedException ie) {}
try{
scc = (SipClientConnection)Connector.open("sip:[email protected]:5060;transport=udp");//;transport=udp");
catch(Exception ex){
form.append("p1"+ex.toString(), null);
// form.append("P1", null);
try{
scc.setListener(this);
catch(Exception ex){
//form.append("p2"+ex.toString(), null);
// form.append("P2", null);
// scc.initRequest("REGISTER", m_read.m_read.scn);
try{
//scc.initRequest("REGISTER", m_read.m_read.scn);
scc.initRequest("REGISTER", null);
catch(Exception ex){
Alert alert = new Alert("Error");
alert.setType(AlertType.ERROR);
alert.setTimeout(3000); // display the alert for 3 secs
alert.setString(ex.toString());
//form.append("P3", null);
//form.append("SEND REGISTRATION1", null);
String adr = "<sip:[email protected]>";
try{
scc.setHeader("To", adr);
scc.setHeader("From", adr);
scc.setHeader("Contact", "sip:[email protected]:5060");
scc.setHeader("Content-Length", "0");
scc.setHeader("Max-Forwards", "6");
}catch(Exception ex){
Alert alert = new Alert("Error");
alert.setType(AlertType.ERROR);
alert.setTimeout(3000); // display the alert for 3 secs
alert.setString(ex.toString());
// uaStatus.setStatus(REGISTERING);
try{
scc.send();
}catch(Exception ex){
form.append("p4"+ex.toString(), null);
//form.append("SEND REGISTRATION2", null);
System.out.println("SYSTEM REGISTER SEND");
// uaStatus.waitUntilChanged();
// progressGaugeFinished = true;
}catch (Exception e) {
Alert alert = new Alert("Error");
alert.setType(AlertType.ERROR);
alert.setTimeout(3000); // display the alert for 3 secs
alert.setString(e.toString());
m_sip.m_disp.setCurrent(alert);
form.append(e.toString(), null);
// failMessage = e.getMessage();
// commandAction(failedCmd, currentDisplay);
// return;
}Hey!
1. Nokia hides some folders containing system stuff, so that you don't accidentally delete something and mess things up. The computer sees more of them and can count them in, so...
2. It happens because your wallpapers are stored on the Mass storage/ Memory Card drive... And if you connect your phone to the computer, it won't be able to read any files on E/F drives as the computer acceses them.
3. They are! You just have to scroll down the list that pops up when you click on the widget
4. Try this one: to snooze an alarm, turn the phone on the other side, so that the scrren faces the table- it will shut up!
5. Nope, they're usable straight away
6. Don't really see what you mean... You want the messaging app shortcut to open up with folders (outbox, inbox, stuff like that?) I'm afraid it's not possible You'll have to access them via the options menu -
Hello,
I just received a cisco telepresence SX10 device which uses SIP protocol only and I have no experience at all with this. Older devices that I have worked with used H323 protocol which is very easy to configure. Looking for some assistance on how to setup the SIP. Your help will be greatly appreciated
Thank youFor sip you need to configure the following under sip profile
1. sip uri...This is usually the extension of your sx10 ex..[email protected]
2.Display name: used for identifying this device when making a call
3.Authentication: login name and password if you have one set up on your vcs
4. Default Transport: TCP/UDP/TLS..Change this to whatever protocol you are using on your proxy (VCS)
5. Outbound..You can enable your sx10 to use multiple connection to your proxy servers by specifying the proxy addresses here. or You can turn this off and use the proxy option to configure which proxy servers you want your device to connect to
These are the basic that should get your device to register on your proxy server. Check more details here..
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/endpoint/sx-series/tc7/administration-guide/sx10-administrator-guide-tc71.pdf -
Gprs configuration in nokia 6300
m using nokia 6300.when i configure gprs setting manually in personal configuration settings , it doesn't seem to work, and any third party web appliacation like opera mini cant access internet.however the native browser is running well.
well ...i cheked other nokia phones .when u configure manually in personal configuration the access point does appear in preffered access points automatically.and in case of my nokia 6300 manual configurations are not appearing in preffred access points...
any help wd b highly appreciatedGet your Configuration settings here
Gadget
Remember to mark all correctly answered questions as Solved. A forum is only as great as the sum of its parts, together we will prevail.
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