Configure goandcall SIP calls on E61i

Hi All,
I have configured goandcall SIP service and it got registerd on my nokia E61i. The problem is I cannot make outgoing calls. On the other hand when I configure the same on Fring with username, password, proxy : sip.goandcall.com
I am able to make both mobile and landline calls.
Please advice if there is any advance settings I have to make on my mobile. I have installed the nokia SIP_VoIP_Settings_v1_0_en software and set priority of codec G729 but no help.
Regards,
Abhi.

I found the solution for this.
On Nokia E series devices, you need to install a software from Nokia SIP_VoIP_Settings_v1_0_en at the URL : http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...
When you install it, you need to go to Menu -> Installation -> SIP VOIP Setting -> NAT firewall Settings.
Here you need to configure the Domain parameters for sip.goandcall.com and put the value of STUN server as provided on your website.
Please note, we do not need to configure the outbound proxy in E61i ( referred as Proxy Server).
The reason we need to install the above software is because I could not find advance settings in Nokia to configure STUN server.
Hope this info helps.
-Abhi.

Similar Messages

  • AAA and MD5 Configuration on SIP Calls

    Olease can anyone help in AAA and MD5 configuration on Cisco 3640 running SIP. My carrier told me that the only way that my calls can be Authenticated is thru AAAor MD5, eg -
    Host:
    Authentication ID:
    Secret:
    Please I need your help thank you in advance.
    Knmezi

    MD5 authentication works similarly to plain text authentication, except that the key is never sent over the wire. Instead, the router uses the MD5 algorithm to produce a "message digest" of the key (also called a "hash"). The message digest is then sent instead of the key itself. This ensures that nobody can eavesdrop on the line and learn keys during transmission.
    These protocols use MD5 authentication:
    OSPF
    RIP version 2
    BGP
    IP Enhanced IGRP
    For AAA configuration refer to following url;
    http://www.cisco.com/en/US/products/sw/secursw/ps2138/products_configuration_example09186a008017ee15.shtml

  • How can i configure SIP call using uc320 in india

    Hi,
    I have one uc 320w box. now wannt to call my itally office in cost effective way.
    how can use it?
    i heaard about SIP calling. is it avalaible in india? or suggest me the possiblw ways.
    Thanks
    Sujish Sudhakar

    Hi Sudhakaran,
    I think this article about SIP Configuration would help you as a step-by-step process.
    Generic SIP Configuration on UC320W

  • Is it possible that Exchange UM could be configure with two call managers over the same sip?

    Hi,
    I have Cisco call manager 8.2 integrated with Microsoft Exchange Server 2010 Unified Messaging.
    Call manager has primary and secondary server. I created a sip trunk and linked primary CUCM with Exchange. Users can leave and get voice mails.
    Problem: In case that primary server is down (WAN is down) the users registered on secondary server but they cannot contact to Exchange Unified Messaging.
    I added  new UM Dial Plan with the same pilot and associated it to the secondary CUCM server. UM answered but do not recognize the extension number "is not a valid mailbox extension".
    Is it possible that Exchange UM could be configure with two call managers over the same sip, the same pilot number, different associated UM servers and get access to the same voice mail boxes?
    If not:
    Does exist a way to configure Exchange UM that will work if one CUCM server is down?
    Thank you,
    Peter

    Hi,
    I have Cisco call manager 8.2 integrated with Microsoft Exchange Server 2010 Unified Messaging.
    Call manager has primary and secondary server. I created a sip trunk and linked primary CUCM with Exchange. Users can leave and get voice mails.
    Problem: In case that primary server is down (WAN is down) the users registered on secondary server but they cannot contact to Exchange Unified Messaging.
    I added  new UM Dial Plan with the same pilot and associated it to the secondary CUCM server. UM answered but do not recognize the extension number "is not a valid mailbox extension".
    Is it possible that Exchange UM could be configure with two call managers over the same sip, the same pilot number, different associated UM servers and get access to the same voice mail boxes?
    If not:
    Does exist a way to configure Exchange UM that will work if one CUCM server is down?
    Thank you,
    Peter

  • CVP Opsconsole: Patterns for RNA timeout on outbound SIP calls - Dialed Number (DN) text box does not take any input

    Hi there,
    I'm having problems modifying the 'Dialed Number (DN)' text box under 'Advanced Configuration->Patterns for RNA timeout on outbound SIP calls' of the SIP tab in the Cisco Unified Customer Voice Portal 8.5(1) opsconsole. In a nut shell, I need to change the RNA timeout but some reason when typing into the Dialed Number text box, the input is not taken. The reason I want to change this settings is because my ICM Rona is not working with CVP:
    https://supportforums.cisco.com/thread/2031366
    Thanks in advance for any help.
    Carlos A Trivino
    [email protected]

    Hello Dale,
    CVP doesn't allow you to exceed the RNA more than 60  Seconds. If you want to configure the timer for DN Patterns you should  do it via OPS console, It would update the sip.properties files in  correct way, the above way is incorrect.
    Regards,
    Senthil

  • SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?

    Hi,
    I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
    Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
    I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
    London CME (local SCCP phones 22xx):
    interface FastEthernet0/0.100
    encapsulation dot1Q 100 native
    ip address 10.0.10.250 255.255.255.0
    voice class codec 101
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    codec preference 3 g711alaw
    dial-peer voice 25 voip
    description *** SIP Peer to India ***
    answer-address 400.
    destination-pattern 400.
    voice-class codec 101
    session protocol sipv2
    session target ipv4:192.168.15.10
    incoming called-number 400.
    no vad
    India CME (Local SSCP phones 400x):
    interface FastEthernet0/0
    ip address 192.168.15.10 255.255.255.0
    voice class codec 100
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    codec preference 3 g711alaw
    dial-peer voice 10 voip
    description *** SIP Peer to London UK ***
    answer-address 22..
    destination-pattern 22..
    voice-class codec 100
    session protocol sipv2
    session target ipv4:10.0.10.250
    incoming called-number 22..
    no vad
    The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
    A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows “From: <sip:[email protected]” when I need this to be the internal IP 192.168.15.10.
    Can anyone offer any assistance with this?
    Regards,
    Chris

    Hi,
    thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
    The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
    Chris

  • E72 and SIP - unable to make sip calls

    I have my E72 configured with SIP using the nokia sip client. Receiving calls work fine, however being called on my sip number causes the calling phone to go ' busy' when I pick up on the E72
    Anyone a clue?
    No software updates available for my E72
    Thanks

    Try another sim even if yours works on ither phones. Failing that, you could try resetting your phone. remenber to backup your spdata beforehand. If the reset does not fix the fault then I suggest having it checked out at your local Nokia care point. You can locate your nearest care point using the link below.
    www.nokia.com/support

  • Configuring Level3 SIP trunk with Lync 2013

    Hi, I ran into some issues trying to configure SIP trunk from Level 3 and I was hoping someone here can help. We have our mediation server collocated with FE and SIP traffic goes from public IP, port 5060 via NAT, to local IP on FE, port 5060.
    Level 3 provided us with one signaling IP and two RTP IPs.
    I tried multiple trunk configuration settings and I can see that when I'm placing a call from Lync to an outside number I'm getting INVITE from Level 3 signaling IP, the session is established, phone rings, but there is no audio on either side. There's also
    a METHOD NOT ALLOWED message coming from them, which doesn't tell me much about what's happening.
    If I call to a Level 3 DID (assigned to my Lync user account) there's also INVITE from their side, but later I receive a CANCEL from them due to idle session. The phone never rings.
    Questions:
    1) Does anyone have Level 3 SIP trunks configured and can share their Get-TrunkConfiguration settings? What settings should I have for encryption, refer, sessionTimer / RTCP, and others? Level 3 refuses to provide any additional information besides IPs.
    2) Do I understand this correctly that when configuring PSTN gateways in topology, one of the RTP IPs should be entered in the  "alternate media IP" field? We have SIP trunks from another provider (which work fine), and they only use one IP
    for everything, so I don't have any experience configuring separate SIP and media IPs with Lync.
    Thanks, and let me know if I should provide additional info.

    Hi,                                                              
    On Lync topology PSTN gateways interface, please check if you enter gateway listening port 5060 and enable TCP option.
    Please also check if you enable refer support on Lync Server Control Panel, if you enable it please uncheck it.
    You can compare the trunk configuration for Level 3 in the part “Sample Trunk Configuration for Level 3” in the link below with yours’, it is for Lync server 2010 but similar for Lync server 2013:
    http://blogs.technet.com/b/nexthop/archive/2013/04/10/configuring-lync-2010-server-to-work-with-level-3-sip-trunking-services.aspx
    Best Regards,
    Eason Huang
    Eason Huang
    TechNet Community Support

  • Configure the SIP account ... (VoIP)

    How to configure the SIP account on my E61i to use it in my company(VoIP)?

    This could be better answered by your company as they have all the setting info, don't you think?
    Show the KUDOS button some love.... Hit that bad boy.... It don't hurt....
    Apple iPhone 5,
    Retina MacBook Pro, iPad Mini, Nikon D4

  • Cisco ISR G2 SIP Calls Capacity

    Dear all,
    We're planning for Cisco Voice Gateway configuration with SIP trunk, till now no E1s are used.
    I would like to know how can we calculate the number of simulataneous calls that a cisco ISR G2 router (1921. 2921.3945,etc...) can support ?
    How much sip simultaneous calls each ISR G2 model can support ?
    Is it better to use SIP or we must get into E1 PRI ?
    Regards,

    The Q and A below has the call capacity you are looking for
    Table 1. Number of IP-to-IP Calls per Platform
    Platform
    Maximum Number of Simultaneous Calls (Flow-Through)
    Cisco 3945E
    2500
    Cisco 3925E
    2100
    Cisco 3945
    950
    Cisco 3925
    800
    Cisco 2951
    500
    Cisco 2921
    400
    Cisco 2911
    200
    Cisco 2901
    100
    Cisco ASR 1004; and Cisco ASR 1006 Router Processor 2 (RP2)
    5000; 16000*
    Cisco ASR 1002, ASR 1004, and ASR 1006 RP1
    1750
    Cisco AS5350XM and AS5400XM
    600
    Cisco 3845
    500
    Cisco 3825
    400
    Cisco 2851
    225
    Cisco 2821
    200
    Cisco 2811
    110
    Cisco 2801
    55
    http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/prod_qas09186a00801da69b.html
    Please rate all useful posts
    "opportunity is a haughty goddess who waste no time with those who are unprepared"

  • Cisco CUCM to Alcatel PBX SIP calling issues

    Hi All
    I have configured a SIP trunk between my cucms and an Alcatel old pbx on a remote site, they are all identical configs.
    However one of them, the remote site Alcatel can call my cucm and voice is ok
    But when we try to dial from the CUCM to the Alcatel we are getting the fast busy tone!
    Codecs are set etc! as it works one way fine!
    any ideas what thsi could be ?
    cheers

    Hi here is a snip of the trace for the call
    the calling phone was ext 448 the called number over the sip trunk is 88044615
    cheers
    16
    2015/01/23 08:15:31.897|CC|REJECT|26821723|26821724|476|8804615|8804615|1
    2015/01/23 08:15:43.577|CC|RELEASE|26821726|26821727|16
    2015/01/23 08:15:54.907|CC|SETUP|26821728|26821729|476|88044615|88044615
    2015/01/23 08:15:54.909|CC|OFFERED|26821728|26821729|476|88044615|88044615|SEPC4641301122E|DELHI-SIP-TRUNK
    2015/01/23 08:15:55.273|SIPT|26821729|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,13,53766610.2^*^*|13409968|[email protected]|INVITE
    2015/01/23 08:15:55.640|SIPT|26821729|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651678^172.20.65.5^*|13409969|[email protected]|100 Trying
    2015/01/23 08:15:56.012|SIPT|26821729|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651679^172.20.65.5^*|13409970|[email protected]|403 Forbidden
    2015/01/23 08:15:56.012|SIPT|26821729|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651679^172.20.65.5^*|13409971|[email protected]|ACK
    2015/01/23 08:15:58.668|CC|RELEASE|26821728|26821729|67108885
    2015/01/23 08:16:02.133|CC|SETUP|26821730|26821731|4133320804|467|467
    2015/01/23 08:16:02.136|CC|OFFERED|26821730|26821731|4133320804|467|467|172.24.32.38|SEPC464130114C0
    2015/01/23 08:16:18.712|CC|SETUP|26821733|26821734|568|487|487
    2015/01/23 08:16:18.714|CC|OFFERED|26821733|26821734|568|487|487|SEPC464130117E9|SEPC4641301147E
    2015/01/23 08:16:20.151|CC|SETUP|26821730|26821737|4133320804|467|1999
    2015/01/23 08:16:20.157|CC|OFFERED|26821730|26821737|4133320804|467|1999|172.24.32.38|CiscoUM1-VI54
    2015/01/23 08:16:20.159|CC|RELEASE|26821731|0|0
    2015/01/23 08:16:28.151|CC|RELEASE|26821730|26821737|16
    2015/01/23 08:16:31.997|CC|SETUP|26821738|26821739|476|88044615|88044615
    2015/01/23 08:16:31.998|CC|OFFERED|26821738|26821739|476|88044615|88044615|SEPC4641301122E|DELHI-SIP-TRUNK
    2015/01/23 08:16:31.999|SIPT|26821739|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,13,51956482.95772^172.24.48.180^SEPC4641301122E|13409978|[email protected]|INVITE
    2015/01/23 08:16:32.366|SIPT|26821739|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651682^172.20.65.5^*|13409979|[email protected]|100 Trying
    2015/01/23 08:16:32.764|SIPT|26821739|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651683^172.20.65.5^*|13409980|[email protected]|403 Forbidden
    2015/01/23 08:16:32.764|SIPT|26821739|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651683^172.20.65.5^*|13409981|[email protected]|ACK
    2015/01/23 08:16:39.148|CC|RELEASE|26821738|26821739|67108885
    2015/01/23 08:16:44.261|CC|SETUP|26821740|26821741|4133320804|465|465
    2015/01/23 08:16:44.263|CC|OFFERED|26821740|26821741|4133320804|465|465|172.24.32.38|SEPC46413011466
    2015/01/23 08:17:26.622|CC|RELEASE|26821733|26821734|16
    2015/01/23 08:17:33.320|CC|SETUP|26821743|26821744|568|536|536
    2015/01/23 08:17:33.322|CC|OFFERED|26821743|26821744|568|536|536|SEPC464130117E9|SEPC46413011477
    2015/01/23 08:17:44.673|CC|RELEASE|26821706|26821707|16
    2015/01/23 08:18:38.248|CC|RELEASE|26821713|26821714|16
    2015/01/23 08:18:51.306|CC|RELEASE|26821740|26821741|16
    2015/01/23 08:18:53.509|CC|SETUP|26821746|26821747|447|033255961|033255961
    2015/01/23 08:18:53.513|CC|OFFERED|26821746|26821747|447|033255961|033255961|SEPC46413011482|172.24.32.38
    2015/01/23 08:18:53.739|SIPL|0|TCP|IN|172.24.32.34|50

  • H323 to SIP calls

    Can someone explain how h323 to SIP calls work & vice versa.

    The following messages are mapped:
    SIP <---> H323
    INVITE - SETUP
    100 Trying - Call Proc
    180 Ringing - Alerting
    183 Session Progress - Progress
    200 OK (for INVITE) - Connect
    BYE - Release Complete
    With H323 to SIP CUBE, if fast start occurs on one leg, early offer needs to happen on the other (and vice versa).  Most SIP devices these days to early offer (SDP in invite) so you typically need fast start enabled on both directions of the H323 leg for this design.
    Check out this link for more information:
    http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-h323sip_ps5640_TSD_Products_Configuration_Guide_Chapter.html

  • Need help regarding configuring the WebService Call from RTD to Siebel

    Hi All,
    Can someone help me with the information on how do i configure a Webservice Call from RTD to Siebel?
    Any high-level or granular details on this would be very helpful as I am new working on this product. How can a jax-ws be utilized to achieve the same?
    Thanks in advance.
    Best Regards,
    Hariharan

    If you actually need a portal service though, this will not work. However, you could have the portal service return a Document object, which is basically the text of the HTML file you want to display. Then, when calling the portal service, you can simply output the text to the IPortalComponentResponse object
    I hope this helps
    Darrell

  • Cannot change library path or name when configuring a library call function

    I cannot change the library path or name when configuring a library call.  I browse to the new location, click OK.  The new path appears  in the box.  I click OK to dismiss the configuration.  Then I re-open the configuration and the old path has magically appeared.  I wouldn;t care but I get a fatal error whenever I execute the call.  (Another VI with the same call but to a different library works fine.  So I am trying to get the failing call to use the one that works.)

    I am using version 7.1
    It turns out that everything works fine in LV 2010 but the customer doesn't have that rev.
    I am remotely debugging this using GoToMeeting.  Sounds weird, but I fell into that.  It works fine, I have a screen which looks exactly like the screen in the customers site.  I can do everything, albeit with a little time drag.  The thing is, I e-mailed the VI from his site to mine so that I could send it to you.  But it works fine here.  So the question is what could be different there that would cause this?
    The main problem that I am trying to solve is that when I run the VI, I get a message
    "Labview:  An exception occurred within the external code called by the Call Library Node.  This might have corrupted Labview's memory. Save any work to a new location and restart Labview"
    If I set the Library Node on my machine to the correct function, it runs without error.  It seems, however, to start with a different function and then I get the same error.  I would think that if there is a parameter mismatch that the library would return an error status, but this appears not to be the case.  (I did not write the library, HP (Agilent) did)  The function is correct at the customer's machine.

  • Incoming sip calls are not working but outgoing is working with cme

    I have CME setup with voip.ms on my 2800 router, my outgoing calls are working  but my incoming calls are not.  Below is my config, please let me know if it is something with my config:
    voice translation-rule 3
     rule 1 /^9142281\(...\)$/ /\1/
    voice translation-profile INCOMING_CALL_1
     translate called 3
    dial-peer voice 1 voip
     translation-profile incoming INCOMING_CALL_1
     session protocol sipv2
     session target sip-server
     incoming called-number .%
     voice-class codec 1
     dtmf-relay rtp-nte
     no vad

    I made the change, but I am getting no output from debug voip ccapi inout.  What does concern me from debug ccsip messages is:
    Aug 31 12:42:04.195: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 400 Bad Request - 'Invalid Host'
    Via: SIP/2.0/UDP 107.6.67.238:5060;branch=z9hG4bK000d3c36;rport
    From: "+19144410197" <sip:[email protected]>;tag=as7439b9c1
    To: <sip:[email protected]:1061>;tag=829C8-2532
    Date: Sun, 31 Aug 2014 12:42:04 GMT
    Call-ID: [email protected]:5060
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Reason: Q.850;cause=100
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    I also am getting this:
    voicertr2#debug ccsip error
    SIP Call error tracing is enabled
    voicertr2#
    Aug 31 12:45:07.359: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr
    Aug 31 12:45:07.359: //-1/78AE76E98009/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE

Maybe you are looking for

  • Error while creating vendor contact person using vmd_ei_api

    Hi, while craeting vendor contact person using maintain_bapi of vmd_ei_api class iam getting error like 'Specify address number or address handle'. code : CALL FUNCTION 'BAPI_PARTNEREMPLOYEE_GETINTNUM' EXPORTING quantity = 1 IMPORTING * RETURN = cont

  • Some photo's in an events folder will not sync to my iPhone5

    Some photos I have in my "Home Improvement" event folder will not sync to my iPhone5. Out of 140 phots, 50 or so will not sync. I think it has something to do with the "descriptions". Is there a way to check if the photo is .jpeg or some other format

  • WRT54GL Can't set WPA or WEP and connect

    Okay, brand new WRT54GL, setup on the network, can connect, no problem.  If I goto setup the WPA or WEP and enter the corresponding info on the Windows machine, I can't connect to the router.  It leases an IP, and I can see in ipconfig (dhcp client)

  • What's the difference??? and WHY

    The below select query returns a row .... SELECT vdd.disp_desc vcInsStatusDesc, 'W' vcCustomerType, NULL vcInsOwnerFullname, cpd.cpd_no vcValidationPostCode, cpd.cpd_options_selection vcOptionsSelection, cpd.cpd_complexity vcComplexity, cpd.cpd_fl_ma

  • Material Document Number Range

    Hi, What is the meaning of setting number range as Year    From Number    To Number     Current number 9999    4916000000       4916999999   4916568199 My understanding is we do not need to set sepeate number for every year once we set 9999. Is that