SIP problem
I am angry, i have tryed hard to get my E72 SIP-settnings working steady, but not. It just doesent work, after a while it wont register, and than it lock "SIP setting" so i can not access it, I have to hardware reset and make the installation again (ALL installations, contacts, programs etc) Its very very frustating. After new installation I can create a new SIP and it works shortly again.
"SIP-settings" is allready in the phone
To access "Advanced VoIP" I downloaded an installed "SIP VoIP 3.x"
(Look here: http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin... (The other programs there is not comatible)
But as i said earier, it just doesent work. I have read a lot here on this forum, and around internet "Step by step" instructions but just doesent work.
Sometimes it works a couple of times, but never stable and after a while not at all.
What to do?? Somone have any ideas? Is there realy somone who have successed with this? If plese make an step by step instruction! I think I have spend at least 14 hour trying to fix this....:-((
I had always trouble with nokia SIP and voip settings , and getting it to work.
Use a sip client application, like Nimbuzz or fring.
Much easier setup and works straight.
Haikal
Similar Messages
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E61: sip problem with Gizmo, cannot make call
I register for Gizmo few days ago. I cannot make any call out (to 411, 1-747xxx numbers, any landline, any mobile )
I cannot hear the other side, but the other side hears my voice
what's wrong?
I'm using the latest v3 firmwareI have some problems with Gizmo as well. Not exactly the same problem as yours. In my case, initiating a call takes too long (~1 minute). I would recommend trying Truphone. It is another SIP-based app; and in my opinion, much better than Gizmo.
You can also check my blog posting here, http://www.s60tips.com/2007/06/28/which-voip-applications-to-use-part-v/Message Edited by antonypranata on 05-Sep-200709:22 AM
Antony Pranata
Visit S60Tips.com for tips, tricks and tutorials of using S60 phones -
Dear All,
I'm implementing the SIP Outbound Dialer, the architecture is below
1. ROGGER + Campaign Manager : (version 9.0)
2. CMPG + CTI Server + CTIOS + MRPG + SIP Dialer : (version 9.0)
3. CUCM ver 8.6
4. Gateway + E1 trunk + IOS 15.1(3)T
The problem is SIP Dialer doesn't call out after successfully loaded calling list into the CampaignManager (checked the log in baImport). I checked the status of SIP Dialer that it Active all (please, see the attach file). But when I check the log on MRPG it said that "Failed an attempt to ACTIVATE the Peripheral's Routing Client ". I don't sure that this is the cause of the problem or not, I've re-checked configuration on MRPG manay times but nothing strange. Anyone found this problem before please, suggest.
BR.
Winai K.Hi,
Except for 15.1(x)T train other IOS versions do not have the capability of doing the CPA analysis (determining voice \answering machine \ fax etc) effectively.
As a result it is recomended to use above code.
Thank you
Anuj -
Hi,
I'm trying to send a REGISTER Sip Message to Asterisk with credentials (following the instructions from Javadoc of JSR180). The code is the follow:
try {
// open listener in application specific port 5080
sipNotifier = (SipConnectionNotifier)Connector.open("sip:5080");
// build the contact URI
contact = new String("sip:emanuele@"+sipNotifier.getLocalAddress()+":"+sipNotifier.getLocalPort()+";transport=UDP");
// open client connection to the SIP registrar in this case "host.com"
sipConnection = (SipClientConnection) Connector.open("sip:89.89.89.89");
sipConnection.setListener(this);
// initialize REGISTER with appropriate headers
sipConnection.initRequest("REGISTER", null);
sipConnection.setCredentials("ciccio", "ciccio", "mydomain.com");
sipConnection.setHeader("From", "sip:[email protected]");
sipConnection.setHeader("To", "sip:[email protected]");
sipConnection.setHeader("Expires", "600");
sipConnection.setHeader("Contact", "<"+contact+">");
sipConnection.send();
boolean handled = false;
int scode = 0;
while(!handled) {
// wait max 30 secs for response
sipConnection.receive(30000);
scode = sipConnection.getStatusCode();
switch(scode)
case 200:
// handle OK response
handled = true;
break;
default:
// handle other responses
handled = true;
// wait maximum 15 seconds for response
} catch ( IOException e ) {
e.printStackTrace( );
voipmidlet.addMessage( "Error: " + e.getMessage( ) );
finally
if (sipConnection != null)
try {
sipConnection.close();
catch ( IOException e ) {
e.printStackTrace();
Analyzing the traffic I can see the server receives the first register message, answers with unauthorized but my j2me app doesn't catch this response.
The server respond on the same port that is the source port the client send message from and that port is chosen randomly at the moment of the send and it's different from the port my listener is where.
Does anyone know a solution?
Thanks
EmanueleSorry I have the lines:
sipConnection.initRequest("REGISTER", sipNotifier);
and not
sipConnection.initRequest("REGISTER", null);
in my code....the code in the previous post has that error but also with sipNotifier it doesn't work due the same problem. -
SIP problem with Huawei b593....router or EE4G PAYG?
I'm struggling to get any SIP service to work on my Huawei b593s-22 / EE4G Data PAYG sim. The SIP/VOIP service hangs on 'registering' and I can't make outgoing calls. If I put in my old Three data sim, the SIP/VOIP go to UP status immediately, so it must be EE - right? A quick search tells me VOIP isn't blocked on EE PAYG?? Freespeech support told me "This is the source of your problem, the IP address 100.108.207.89 is not a public IP. This means your provider EE 4G is placing a NAT (to translate the private IP 100.108.207.89 to a shared public one) between you and the public internet. This is OK for web surfing but will not work for VoIP. You need to have a public IP address assigned. Talk to EE 4G and see if they have a solution for you." Any suggestions?
Hi,
I'm also having difficulties. I have enabled the UPnp, but the Back to My Mac details box is still stating that on my Huawei LTE CPE B593, NAT-PMP is turned off.
Any ideas?
Thanks
John -
N95 V20.0.15 - SIP Problem
Sip registration fais after upgrade to v20.0.15. Worke fine with previous version v12.
Anyone has similar problem?
Is their any solution or we have to wait for the new versionRegistration works fine for me, BUT if I receive a SIP call from a client with enabled video the phone hangs completely (I have to remove battery, no workaround).
This didn't happen with v12.
This is very annoying, because since I have no control over the settings of the softphones used by the people that call me, having my phone registered to Internet Telephony service offers a great risk of leaving me aout of service.
Is there any solution to this?
Thanks
Jorge -
Error -8, SIP problems...
Hello everyone!
I am having some problems trying to video chat with iChat.
Me and the other end are using iChat AV 3.1.8 (v448) on OS X Tiger 10.4.9.
We both can connect to the Apple test bot (AIM Users appleu3test01, appleu3test02, appleu3test03) but when we try to connect to each other we get the -8 error, where sometimes the SIP (Seesion Initiation Protocol) comes.
My question is: Is there any solution for this error?
Thanks anyway,
Andre TanigutiHi
First have you both set the Quicktime streaming setting, goto sys prefs/quicktime/streaming/streaming speed, set to 1.5mbps(dont use automatic)
In ichats prefs click on video and change bandwidth limit to NONE.
Restart iChat.
Tony -
Hi GUYS,
Please help me..
I have experiencing problems with SIP phones behind firewall running on CIsco 887 VA-M.
I got these messages :
5 02:43:37.439: %AIC-4-SIP_PROTOCOL_VIOLATION: SIP protocol violation (Mandatory header field missing) - dropping udp session 192.168.33.120:5061 203.111.37.20:5060 on zone-pair in-out-zone class cmap-in-out-base
Jul 5 02:43:40.035: %AIC-4-SIP_PROTOCOL_VIOLATION: SIP protocol violation (Mandatory header field missing) - dropping udp session 192.168.33.117:5060 203.111.37.20:5060 on zone-pair in-out-zone class cmap-in-out-base
I have downgraded software to 151-4.M6 and greated the policy to skip those checkings but no any improvements
My config is
boot-start-marker
boot system flash:c880data-universalk9-mz.151-4.M6.bin
boot-end-marker
no aaa new-model
memory-size iomem 10
crypto pki token default removal timeout 0
ip source-route
ip dhcp excluded-address 192.168.33.1 192.168.33.99
ip dhcp excluded-address 192.168.33.150 192.168.33.254
ip dhcp pool 1
network 192.168.33.0 255.255.255.0
default-router 192.168.33.1
dns-server 8.8.8.8
ip dhcp pool `
ip cef
ip domain name ues
ip name-server 8.8.8.8
no ipv6 cef
license udi pid CISCO887VA-M-K9 sn FGL171725DT
controller VDSL 0
class-map type inspect match-all cmap-manage
match access-group 23
class-map type inspect match-any cmap-in-out-ALL_allowed
match access-group 150
class-map type inspect match-any cmap-in-out-base
match protocol https
match protocol http
match protocol dns
match protocol ftp
match protocol pop3
match protocol citrix
match protocol citriximaclient
match protocol icmp
match protocol smtp
match protocol pptp
match protocol gopher
match protocol sip
match protocol h323
match protocol sip-tls
policy-map type inspect allow_all
class type inspect cmap-in-out-ALL_allowed
pass
class class-default
drop
policy-map type inspect pmap-out-in-manage
class type inspect cmap-manage
pass
class class-default
drop
policy-map type inspect pmap-in-out
class type inspect cmap-in-out-base
inspect
class type inspect cmap-in-out-ALL_allowed
pass
class class-default
drop
zone security in
zone security out
zone-pair security in-out-zone source in destination out
service-policy type inspect pmap-in-out
zone-pair security out-self-zone source out destination self
service-policy type inspect pmap-out-in-manage
zone-pair security out-in-zone source out destination in
service-policy type inspect allow_all
interface Ethernet0
no ip address
shutdown
no fair-queue
interface ATM0
no ip address
no ip route-cache
load-interval 30
no atm ilmi-keepalive
pvc 8/35
encapsulation aal5mux ppp dialer
dialer pool-member 1
interface FastEthernet0
switchport access vlan 100
no ip address
interface FastEthernet1
switchport access vlan 100
no ip address
interface FastEthernet2
switchport access vlan 100
no ip address
interface FastEthernet3
switchport access vlan 100
no ip address
interface Vlan1
no ip address
interface Vlan100
ip address 192.168.33.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
zone-member security in
interface Dialer0
ip address negotiated
no ip redirects
no ip unreachables
no ip proxy-arp
ip mtu 1492
ip flow ingress
ip nat outside
ip virtual-reassembly in
zone-member security out
encapsulation ppp
ip tcp adjust-mss 1350
dialer pool 1
ppp authentication chap pap callin
ppp chap hostname
ppp chap password 0 673569
ppp pap sent-username
no cdp enable
ip forward-protocol nd
no ip http server
no ip http secure-server
ip nat inside source list FOR_NAT interface Dialer0 overload
ip route 0.0.0.0 0.0.0.0 Dialer0
ip access-list extended FOR_NAT
permit ip 192.168.33.0 0.0.0.255 any
ip access-list extended KILL-TFTP
deny udp any eq tftp any
permit ip any any
access-list 150 permit ip any any
access-list 150 remark TEMP
line con 0
no modem enable
line aux 0
line vty 0 4
login local
transport input ssh
end
Thanks a lot!Try to do disable inspection of protocol-violation for sip, using this config:
class-map type inspect sip SIP_VIOLATION_CLASS
match protocol-violation
policy-map type inspect sip SIP_VIOLATION_POLICY
class type inspect sip SIP_VIOLATION_CLASS
allow
policy-map type inspect pmap-in-out
class type inspect cmap-in-out-base
inspect
service-policy sip SIP_VIOLATION_POLICY -
Hello,
We are a SIP provider in France. More and more of our customers are using the WIFI/SIP features of Nokia mobile phones. They can register without problem, as well as they can get SIP calls on their mobile.
Yet, many of them have problems to place calls. As far as we can see in the SIP traces, it looks like the N95 answers by a CANCEL to a 180 Ringing message.
We have done some tests with a customer with the following configuration :
Nokia N95 8GB
V 20.0.16 28-02-08 RM-320
Is this a known problem ? Would it be possible to get in touch with some developers of the SIP stack to trace this problem ?
Thanks and regards,
Guillaume
Solved!
Go to Solution.I wouldn't be so sure of that. I have an N95-1 registered to my own Asterisk server and I can place calls no problem.
This said, if you want to get hold of Nokia you've come to the wrong place. This is just a forum for users of Nokia products to share information. You should be able to contact a Nokia customer service representative on 0811.004567 and they should be able to pass the message on.
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you! -
SIP Problem - Can make, but not receive calls
I two E-Series phones (E51 and E71). The E71 is configured to connect over my WLAN to my corportate phone system. this works perfectly for making and receiving internal and external calls, however, my E51 can only make calls. both phones have identical settings with regard to my phone system but i wonder if there is a known bug and more importantly a workaround? i have also tried to configure a spare E65 and this presents the same problem as the E51.
Any ideas? starting to get a headache over this....
Many thanks
BenI am sorry I didn't get back to you but I have been away for a while. I have discovered that when I turn the firewall off on my imac I can receive Facetime calls - turn it back on and I can't. Seems a little silly that Apple's own firewall blocks Facetime.
I wonder if anyone else has this problem. I am a little concerned about having to keep my firewall turned off.
Thank you very much for your help anyway. -
I've just purchased my new N79 and configured the SIP setting using Tools > Settings >Connection > SIP settings.
everything went smoothly and my sip account is registered successfully. After this i tried to find out the Internet Tel option as nokia E65 has this option but it is not there in N79. I don't get the option intenet call when i select options>call.
Kindly anyone knows how to do internet call afer SIP settings are successful??
Regards
InamHi Amit here,
Go to this website http://www.forum.nokia.com/info/sw.nokia.com/id/b1c361a2-7eb2-4853-8c0c-d2f54e184237/SIP_VoIP_3_1_Se...
First you have to make your account on nokia forum.
Download version 3 software and install on your phone.
Then go to menu-tools-connectivity-net settings-advance VOIP setting and make a new profile using your SIP profile.
Then it will go to contact automaticaly or if now going go to cantact and you will find there Activate service.
After activating this service go to any contact and option-call-internet call
Thank you -
I have installed 2 RV220w routers and 5 RV120w routers on a 7 site vpn ( all with the latest 10.5.8 furnware)
I am running an Elastix 2.4 asterisk based pbx
The issue I am having is this:
With the sip alg disabled and the port forward set for 5060 udp and 10000-20000 udp I cannont recieve sip calls
With the sip alg enabled and the port forward set for 5060 udd and 10000-20000 upd I can reiceive calls BUT
the rtp packets are coming at any random port 9000 to 50000
Aparently the sip alg is rewriting the sip header causing this
Normally I would run this with the sip alg turned off but in that mode I cannot get the sip port to pass calls at all
I am thinking of downgrading the firmware to the 10.4.17
Does anyone have experience with this
ThanksPlease see the following thread:
https://supportforums.cisco.com/thread/2269123
- Marty -
Psychomania.. ! Help ! SIP problems
Hello...
Here is one for Psychomania. Please help.
I am trying to set up 'VoipTalk' on my E71 running in offline mode only (I have no SIM in it).
As standard, I set uo the account on a softphone on my PC and its all running well. I even tried the auto-message when I heard the confirmation that that the account is set up... great !!
So, then I went to my E71 (offline mode) and fed into the SIP settings all the VoipTalk details and the damn thing will not register !!! Message saying on my E71 'Registration Failed'.
Any ideas... ?
I double checked but all is according to the instructions of VoipTalk 'E70' settings (they dont have a guide for E71)
here it is.... https://www.voiptalk.org/products/nokia-e70-voip-setup.html
So, could there be a difference between the E71 and E71 settings ????
Psycomanis - please help me.... !!
Could it be coz I am in Offline mode ?
Please tell me..somebody... its driving me craaazy.
Harry"A port conflict was detected in the server configuration. The server is configured to listen on two ports that have the same port number and IP address.
Channel "sip" address "sip://fe80:0:0:0:fcff:ffff:feff:ffff:5060" conflicts with channel "sip" address "sip://fe80:0:0:0:fcff:ffff:feff:ffff:5060""
Check your config.xml file and see how many times the 5060 port is configured. -
Hi,
I have a client with a CUCM 9.1 an Alcatel OXE R10.1. They are interconnect by trunk SIP. All the E1 "T2" access are on the Cisco.
He has a phone on the OXE that is redirected to a phone on the Cisco after 4 ringings. It's working when the call is from the E1 or the OXE but isn't working with the call from the Cisco. The call crash after the 4 ringings.
Can you help me ?
Thank's
PierreEnable the option "redirect by application" from the SIP profile applied to the CUCM sip trunk.
Thanks
Manish -
How I solved my audio/video ichat problem
Hope to be useful to somebody.
I just had to disable the internet sharing and voila, everything is working well now.Thank you very, very much. I tryed everything to solve this Problem, but forgot i put internet sharing on to work on a college's computer wich hadn't wireless. I put it of now and voilá it works agian. Wil proberly work for everyone who had this, connection error with this SIP problem in it.
Thanks, thanks, thanks.
Hans
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