UCCE 9 Outbound SIP problem
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
Similar Messages
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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 -
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 -
SIP REFER with UCCE and CUPS SIP Proxy
I am running UCCE 8.0.1 with CVP and CUPS as the SIP proxy. I am looking to transfer calls to PSTN and release from CVP to free CVP ports. I am using the rfxxxxxxx method in the ICM script, which seems to work fine from CVP. However the SIP proxy send an Invite to our SBC instead of a REFER. Is there a way to configure SIP Proxy to send the REFER instead of the invite? I would like to release the call from our SBC as well.
The other idea was to insert a custom header in CVP I could then pull out at the SBC and replace with a REFER. Does anyone have any links to documentation on this?
Thanks
TChow about check "enable send calls to originator" for the refer label routing in CVP? this would bypass proxy.
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UCCX 7 outbound dialing, problem with area codes
Hello all,
I'm just testing the outbound dialing feature in a lab, and have run into the following problem:
I have following:
outbound configuration:
general:
Customer dialing time: start (9:00) end (21:00)
dialing prefix - 9
international prefix - 00
Local Area Code: 44
Area Codes:
44 -> Europe/London
34 -> Europe/Madrid
86 -> GMT-10 (for testing purposes, does not correspond to real China time).
Then, I upload a .CSV file into a campaign with the following information:
Jane,Smith,00447111113007
Jacky,Chan,0086611115002
Juan,Pablo,0034611114001
The dialing works, the agent receives the call, accepts it, and then the call is made.
if anyone is interested, although it has nothing to do with the question itself, the calls are translated to other extension on CUCM to be able to make a call.
The problem is that, they are all treated as in the same AREA CODE, although I have clearly defined area codes for each of those countries.
when the call appears it shows on the CAD as BAAreaCode: +00060
Therefore the calls are made through to China even when it's midnight there!!!???
I know I must be doing something wrong, but I don't know what?
Many thanks for your help,
Cheers,Hi Aaron,
Sorry for being so late to answer you. But I've done quite a few tests last weekend and have couple traces as well.
Apparently, the first time I added area codes, I did something incorrect, although I do not know what exactly, because now I've added area codes and it partially worked.
Area code: 4420 - Europe/London with Daylight saving
Area code: 3498 - Europe/Madrid with Daylight saving
Area code: 8661 - GMT/+7 NO Daylight saving
The first two Area codes (Spain and London) were correctly identified, In traces I could even see:
this one is for Spain:
type=com.cisco.executor.QueuedExecutor,Thread=MIVR_CFG_MGR_INVOKE_NOTIFICATION-33-189,Thread priority=7,Original Thread=null,Original thread priority=5,Time=null,Exception=null 358035:
Jul 03 10:56:40.390 BST %MIVR-CFG_MGR-7-UNK:configStubImpl-replace() notified with DialingListConfig,time=2010-07-03 10:56:40.373,recordId=0,impl=class com.cisco.crs.outbound.DialingListConfig,desc=,key[137],columns[137,1,,Jacky,Chan,0086611115002,,,377,0,424,1,424,1,,1278150825843,6,0,0,0,0,0,0,0,0] 358036:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getPhoneNumber: callStatus=2callResult=0lastNumDialed=0 358037:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getUnformattedPhoneNumber: dlcID:138 358038:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getFormattedPhoneNumber: phoneNum=0034981114001 0034981114001 358039:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getFormattedPhoneNumber: intPrefix=00 localAreaCode=004420 lenAreaCode=6 include lac=true 358040:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:numToDial=0034981114001 0034981114001 358041:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getUnformattedPhoneNumber: dlcID:138 358042:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: timezone=sun.util.calendar.ZoneInfo[id="Europe/Madrid",offset=3600000,dstSavings=3600000,useDaylight=true,transitions=165,lastRule=java.util.SimpleTimeZone[id=Europe/Madrid,offset=3600000,dstSavings=3600000,useDaylight=true,startYear=0,startMode=2,startMonth=2,startDay=-1,startDayOfWeek=1,startTime=3600000,startTimeMode=2,endMode=2,endMonth=9,endDay=-1,endDayOfWeek=1,endTime=3600000,endTimeMode=2]] 358043:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: DST observed=true 358044:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: localTimeMins=716 globalStartTime=480 globalEndTime=1410 358045:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: isValidTime=true 358046:
Jul 03 10:56:40.390 BST %MIVR-SS_OB-7-UNK:CampaignMgr.getContactsFromMemory: no more in-memory contacts left for campaignID 1 358047:
As we can see, localTimeMins is 716 which is: 11:56AM. That was correct time, and in between min/max globalstart times.
And this one is for London:
getPhoneNumber: callStatus=2callResult=0lastNumDialed=0 276065:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getUnformattedPhoneNumber: dlcID:130 276066:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getFormattedPhoneNumber: phoneNum=00442081114002 276067:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getFormattedPhoneNumber: intPrefix=00 localAreaCode=004420 lenAreaCode=6 include lac=true 276068:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:numToDial=00442081114002 276069:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getUnformattedPhoneNumber: dlcID:130 276070:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: timezone=sun.util.calendar.ZoneInfo[id="Europe/London",offset=0,dstSavings=3600000,useDaylight=true,transitions=242,lastRule=java.util.SimpleTimeZone[id=Europe/London,offset=0,dstSavings=3600000,useDaylight=true,startYear=0,startMode=2,startMonth=2,startDay=-1,startDayOfWeek=1,startTime=3600000,startTimeMode=2,endMode=2,endMonth=9,endDay=-1,endDayOfWeek=1,endTime=3600000,endTimeMode=2]] 276071:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: DST observed=true 276072:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: localTimeMins=627 globalStartTime=480 globalEndTime=1410 276073:
Jul 03 10:27:41.687 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: isValidTime=true 276074:
As we see, localTimeMins is 627 which is: 10:27AM which is correctly between min/max globalstart times too.
But the problem started with GMT+7.
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CampaignMgr.getContactsFromMemory: for campaignID=1 280255:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CampaignMgr.getContactsFromMemory: campaignID=1;#of DLC=2 280256:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getPhoneNumber: callStatus=2callResult=0lastNumDialed=0 280257:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getUnformattedPhoneNumber: dlcID:131 280258:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getFormattedPhoneNumber: phoneNum=0086611115002 280259:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getFormattedPhoneNumber: intPrefix=00 localAreaCode=004420 lenAreaCode=6 include lac=true 280260:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:numToDial=0086611115002 280261:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getUnformattedPhoneNumber: dlcID:131 280262:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: timezone=sun.util.calendar.ZoneInfo[id="Etc/GMT+7",offset=-25200000,dstSavings=0,useDaylight=false,transitions=0,lastRule=null] 280263:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: DST observed=false 280264:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: localTimeMins=148 globalStartTime=480 globalEndTime=1410 280265:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: isValidTime=false 280266:
Jul 03 10:28:41.859 BST %MIVR-SS_OB-7-UNK:CampaignMgr.getContactsFromMemory: cannot dial at this time, will be marked RETRY
It correctly identifies Etc/GMT+7 but then it calculates the localtime wrongly... It came up with 148 which is 2:28AM... and the time I was making a call GMT+0 time was around 9:28AM so GMT+7 should've been 4:28PM...
Then I tried GMT-7 too:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:CampaignMgr.getContactsFromMemory: for campaignID=1 433346:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:CampaignMgr.getContactsFromMemory: campaignID=1;#of DLC=2 433347:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getPhoneNumber: callStatus=2callResult=0lastNumDialed=0 433348:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getUnformattedPhoneNumber: dlcID:140 433349:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getFormattedPhoneNumber: phoneNum=0086611115002 433350:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getFormattedPhoneNumber: intPrefix=00 localAreaCode=004420 lenAreaCode=6 include lac=true 433351:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:numToDial=0086611115002 433352:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:CMgrUtil: getUnformattedPhoneNumber: dlcID:140 433353:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:OutboundContactInfo:isRequestingContacts() for CSQ 2 returns true 433354:
Jul 03 11:25:40.843 BST %MIVR-SS_OB-7-UNK:OutboundContactInfo:setNrGetContactsReqSent() for CSQ 2 to 1 433355:
Jul 03 11:25:40.859 BST %MIVR-SS_OB-7-UNK:OutboundContactsRequestor before sleep 433356:
Jul 03 11:25:40.859 BST %MIVR-SS_OB-7-UNK:Dialer:printLicenses() working_resources for CSQ:2 is 0 433357:
Jul 03 11:25:40.859 BST %MIVR-SS_OB-7-UNK:Dialer:printLicenses() total_working_resources:0, OB_contacts_in_progress:0 433358:
Jul 03 11:25:40.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: timezone=sun.util.calendar.ZoneInfo[id="Etc/GMT-7",offset=25200000,dstSavings=0,useDaylight=false,transitions=0,lastRule=null] 433359:
Jul 03 11:25:40.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: DST observed=false 433360:
Jul 03 11:25:40.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: localTimeMins=1045 globalStartTime=480 globalEndTime=1410 433361:
Jul 03 11:25:40.859 BST %MIVR-SS_OB-7-UNK:CMgrUtil: isValidLocalTime: isValidTime=true 433362:
Jul 03 11:25:40.859 BST %MIVR-SS_OB-7-UNK:CampaignMgr.getContactsFromMemory: remaining # of in-memory contacts for campaignID 1:1 433363:
and this time it did all the way around, instead of early morning, it was 1045: 5:25PM...
I do not understand what am I doing bad... but I'm starting to think it's some kind of a bug...
and the other problem is: outbound dialer does not even take contacts out of the campaign for calling if the server's local time is not between global start time and global end time. which should not be the case, as I understand campaign times should be applied to area code local time not to server local time.
Many thanks to all,
Cheers,
George -
UC500- Outbound SIP Routing Question
We have a UC540 that is receiving its trunking from a cloud based sip server. They have two trunks on the server, one for voice traffic and one for fax traffic. Inbound this works fine but all outbound traffic goes over the 'voice' trunk. Is it possible to route outbound traffic, from the fxs ports that the fax machines are connected to, onto the 'fax' trunk of the sip server? The sip server only provides one ip address, with unique registration to each trunk.
Hey guys,
Thanks for the response. Let me clarify my question a bit. I get the translation routing part and that is very helpful but I am still stuck on how to point a dial peer to the 'fax' trunk on the sip server.
A typical dial peer on the switch looks like this:
dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*Generic Locale*Long Distance**
translation-profile outgoing SIP-Trunk-Out
preference 1
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
and the sip-ua looks like this:
sip-ua
credentials username 7xxxxx5 password xxxxx realm xxx.xxx.xxx.236 (voice trunk/user ID on the sip server)
credentials username 7xxxxx6 password xxxxx realm xxx.xxx.xxx.236 (fax trunk/user ID on the sip server)
keepalive target ipv4:xxx.xxx.xxx.236:5060
authentication username 7xxxxx5 password xxxxxx
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
registrar ipv4:xxx.xxx.xxx.236 expires 3600
sip-server ipv4:xxx.xxx.xxx.236:5060
connection-reuse
host-registrar
Thanks,
Chad -
JDBC Outbound Adapter Problems
Hello
We have a problem with our Outbound JDBC Adapter. This Adapter writes IDOC's into a ORACLE-Database.
After a certain amount of records the procedure breaks down with the error: ORA-08177: can't serialize access for this transaction.
Here is the configuration of the adapter out of the adapter-engine:
jdbc adapter java class
classname=com.sap.aii.messaging.adapter.ModuleXMB2DB
mode=XMB2DB_XML
##Adress for XMB endpoint
XMB.httpPort=8210
XMB.httpService=/db/Receiver
##DB Adapter specific parameters (example for SQL-Server, see docu)
db.jdbcDriver=oracle.jdbc.driver.OracleDriver
db.connectionURL=jdbc:oracle:thin:sapxi/[email protected]:1521:gdmgbbwHi !
I have make some test's today. I copy the adapter from our productive system P15 SP3 to the developers D15 SP6. Then we take the same message and send this from sap p42 to sap xi d15 sp5 to the adapter d15 (copy from p15 sp3) to the same Oracle database and it's working.
SAP P42 - Orders - XI D15 SP6 - Orders - Adapter D15 (copy from P15SP3) = it's working Record > ca. 800
I think we have a problem in the patch sp5.
We found also some different size in the adapter files.
This are the jar- files with different size:
SAPAdapterService.exe
adapter.properties
aii_msg_adapter.jar
aii_msg_runtime.jar
aii_rfcadapter.jar
aii_util_grmg.jar
aii_util_misc.jar
aii_util_rb.jar
inqmyxml.jar
lcrclient.jar
logging.jar
I make tomorow, a test with a adapter sp4.
Regards Thomas Neuhaus -
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 -
UCCX 10.5 Outbound- CPA problem
Hi, I can't seem to get CPA back from the gateway. Does CPA only work with POTS legs? The phone I'm calling is registered via CME on the SIP gateway (2911 with 15.4.1-T). How are UCCX Outbound labs typically setup? Do I have the right IOS feature set (see attached screen shot of Feature Navigator)
I'm so close! Please help. Thanks.
Feb 17 12:13:12.902 PST %MIVR-SS_OB-7-UNK:GWCallStateMonitorThread-dlcID-89-GWCallStateMonitorThread:waitForCPACompleteState. Timed out
Feb 17 12:13:02.883 PST: //10/4FE940BC4DDD/HPI/[]/hpi_call_progress_detect:
CALL_ERROR; CPA is disabled globally or DSP_CPA_NOT_SUPPORTED
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
no ip address trusted authenticate
cpa timing live-person 2501
cpa timing timeout 3001
cpa timing term-tone 15500
cpa threshold active-signal 18db
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
signaling forward unconditional
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
options-ping 60
dspfarm profile 15 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
call-progress-analysis
maximum sessions 4
associate application CUBEIt only worked for me on POTS legs, as Edward says. I did try the
Call Progress Analysis Over IP-to-IP Media Session feature, but didn't get it to work. The name implies it should work, maybe I didn't configure it right.
I also have an IP provider, I added another router with a T1 crossover.
UCCX---SIP Gateway---T1---SIP Gateway----SIP provider -
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 -
Maill.app constant outbound traffic problem
Hi Guys
strange issue where mail.app is just sending lots of outbound traffic and saturating my upload bandwidth...
I can also see its the top ranker in littlesnitch for outbound traffic....
My desktop appears unaffected...its only my MBP showing this problem. Could this be some sort of syncing issue? It has now been doing it for a while.
TIA
Neilhmm now its stopped and now i have constant inbound traffic? It obviously a sync of some kind but it took ages and i'm not really sure why it even needed such a large sync?
Ill see how long it takes to download whatever its downloading and then post the graph for future ref....
As a note I did do a quick packet capture and it seems to be only communicating with IP addresses registered to apple and it appears to be encrypted using SSL....so im not expecting anything too fishy....such as a virus or spyware -
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!
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