Redirecting Trunk SIP problem

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
Pierre

Enable the option "redirect by application" from the SIP profile applied to the CUCM sip trunk.
Thanks
Manish

Similar Messages

HTTP Authentication Digest for SIP messages in a trunk SIP CUCME

Hello,
we would like to implement HTTP Authentication Digest for SIP messages in a trunk SIP between a Cisco 2851 and an Asterisk server.
We are using CUCM Express with 15.1(4)M (CME 8.6) as voice gateway to connect to PSTN.
According to Cisco documentation:
"To configure a gateway to use HTTP Authentication Digest, give the following command in each dial peer or SIP-UA configuration mode:
authentication username username password password [realm realm]."
The problem is that when call is from CISCO to ASTERISK, Asterisk sends a challenge to Cisco to do Authentication:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.70.11:5060;branch=z9hG4bK3E205D
Remote-Party-ID: "DN1001" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "DN1001" <sip:[email protected]>;tag=5317D4-2271
To: <sip:[email protected]>
Date: Thu, 20 Feb 2014 10:55:56 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1679566433-2572423651-2156454406-1292596908
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: 1392893756
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 208
<--- Reliably Transmitting (no NAT) to 10.0.70.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.70.11:5060;branch=z9hG4bK3E205D;received=10.0.70.11
From: "DN1001" <sip:[email protected]>;tag=5317D4-2271
To: <sip:[email protected]>;tag=as665c9410
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="559bd1d2"
Content-Length: 0
However, when call is for ASTERISK to Cisco, there is no challenge sent.
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
Max-Forwards: 70
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Thu, 20 Feb 2014 09:58:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
<--- SIP read from UDP:10.0.70.11:60829 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>
Date: Thu, 20 Feb 2014 10:58:27 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>;tag=556830-757
Date: Thu, 20 Feb 2014 10:58:27 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "DN1001" <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
My configuration in Cisco device is:
dial-peer voice 1 voip
description **Calls to ASTERISK **
destination-pattern 9T
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
keepalive target ipv4:10.1.32.70
authentication username CCME password 7 070E234F4A realm asterisk
sip-server ipv4:10.1.32.70:5060
To avoid that the ASTERISK is blocked by Cisco TOLLFRAUD_APP I have added:
voice service voip
ip address trusted list
ipv4 10.1.32.70 255.255.255.255
allow-connections sip to sip
sip
registrar server
The issue is that I would like that Cisco also send a challenge to asterisk server to authenticate SIP messages.
Any ideas?.
Regards.

Hello,
yes, but credentials command configure credentials that are used when Cisco UA must register in a server.
I do not need register Cisco into Asterisk server. What I want is that Cisco authenticate SIP messages that receive. I know
that can be enough with TOLLFRAUD_AP where remote IP is checked, but I want to do something like others routing
protocols (as OSPF, BGP) where every message must be authenticated.
Thanks.
Regards.

Calling to Trunk SIP, I not hear the first 10 seconds of audio

Hi guys,
I have problem with the calls to a trunk SIP. In some calls, i not hear the first 10 seconds of audio.
Scenario
PSTN--->GW MGCP---->CCM--->Trunk SIP---PBX SIP with IVR
The trunk SIP is using MTP (MTP checked).
The capture show a parameter a:inactive
Please your help
Thanks a lot

Hi,
MTP should be checked since the third party SIP phone support only RFC 2833 DTMF method and DTMF RFC method should be RFC 2833 OR OOB & RFC 2833.
(SCCP Phone support both inband and out of band)
MGCP would be configured to support OOB.
From the SIP profile configuration you can select the early offer mode for all calls.
For Early media it requires a PRACK message (SDP answer) as response to 183 session in progress (containing SDP offer).
I think as from the setup as you presented it should work fine with MTP checked, DTMF method as RFC 2833 and from SIP profile it should have Early offer.
I would like to see the SDI or SDL traces for that particular call.
Even I have the same setup for one of my client (CUCM with Genesys Contact center).
I can provide you the SIP Trunk working configuration.
One query from your attached text file, if you say MTP unchecked then the content length should had been 0 (i.e no SDP parameter).
Please clarify.
Kind Regards,
Raaj.

SPA 8000 configuration trunk SIP

Hello everybody,
I need your help with my first configuration of my SPA 8000.
Before that equipement i used a Diguim asterisk for created 4 analogicals lines using a trunk sip which provide to me 4 phones numbers.
I try to configurate this SPA to make the same thing, I have registration on both four lines, I can make outcoming call but i can't receive incoming call.
I would like to start again my configuration and i need you help.
I don't find where i should put the following informations:
- My provider for the SIP is WideVoip.
- I have a unical accompt number fo the 4 lines
- I have the password for this 4 lines
- A proxy adress
- 4 phones numbers
Thanks for your help.
Best regards

Hi Hameed
I just received complaints of this exact issue with a SPA122....did you ever find out if there's to cut off the voltage when the lines are unregistered or no link on WAN?
Thanks,
George

Passing Information from UCCX call variables through trunk SIP to Astersik

Hi All,
We need to pass some informations from our UCCX 8.5SU3/CUCM 8.6.2a to our Asterisk Server.
This two PBX are connected by a trunk sip.
Is it possible to do it?I've read about sip header,but i've never work on it.
Is it possible with a javascript?
Could you please help us?
Thanks
Stefano

No. CCX uses JTAPI (CTI/QBE) to integrate with CUCM, not SIP. As such there is no mechanism for it to manipulate or add extra SIP headers. You would need to use one of the native scripting options (e.g. ODBC, HTTP GET/POST, SMTP) or write a custom Java class that can interface natively with the other application. Examples of this exist such as the excellent documents on SFTP, CIFS, and LDAP.
Please remember to rate helpful responses and identify helpful or correct answers.

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

UC320W - FW2.3.2 - General SIP Trunk - SIP domain setting missing

Hello,
I would like to create a SIP Trunk for the german Telecom.
I am using a UC320W with the latest Firmware 2.3.2 and the region is set to germany.
The SIP setting for the german Telecom requires to set the SIP domain.
Should not be a problem, but when I try to create a new SIP Trunk based on a general provider the SIP domain field is missing in the form.
So I am not able to set this value and therefore the connection cannot be established with a blank domain.
I have tried to downgrade the Firmware to 2.2.2 with the same result.
In some documents of the UC320W I have found screenshots, where a field for the SIP domain is available.
Does someone has the same issue and knows how to fix it?
Is it related to the german region? Or is it possible to set the SIP domain in different way?
Any help is more than welcome!!!!
Thanks in advance...
Rene

Do you have "term mon" with debugs ?
Also, you will need to register some meaningful number, not just '240'.

ZBFW and SIP problems

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

Redirect jsp page problem

          I have problem when I want to redirect a page, inputparam2test.jsp back to the perious
          page, inputparam1test.jsp. Everything works until I
          add a "if" statement in the inputparam1test.jsp.
          The error is "Error 500--Internal Server Error".
          Can anybody help me out? Thanks a lot!
          inputparam1test.jsp:
          <html>
          <head>
          <title>create a patameter input form</title>
          </head>
          <body>
          <%!
          String dataset;
          String modeltype;
          %>
          <%
          dataset = request.getParameter("dataset");
          modeltype = request.getParameter("modeltype");
          %>
          <center>
          <form action="inputparam2test.jsp" method="post">
          <table cellspacing=10>
          <tr>
          <td colspan=2>
                    Please enter model parameters
               </td>
               </tr>
          <tr>
               <td>
                    Dell LRI
               </td>
               <td>
                    <select name="DellLRI">
                    <option value="1">1
                    <option value="0">0
                    </select>
               </td>
               </tr>
               <%
               if(modeltype.equals("Direct")) {
          out.println("in if");
          %>
               <tr>
               <td>
               </td>
               <td>
                    <input type="submit" value="Next>" name="Next>">
               </td>
               </tr>
          </form>
          </center>
          </body>
          </html>
          inputparam2test.jsp:
          <html>
          <head>
          <title>continue input parameters </title>
          </head>
          <body>
          <%
          response.sendRedirect("inputparam1test.jsp");
          %>
          </body>
          </html>


You do not check if the modelType is null and
          if(modeltype.equals("Direct")) {
          out.println("in if");
          will result in a NullPointerException when it is null.
          Almond Huang <[email protected]> wrote:
          > I have problem when I want to redirect a page, inputparam2test.jsp back to the perious
          > page, inputparam1test.jsp. Everything works until I
          > add a "if" statement in the inputparam1test.jsp.
          > The error is "Error 500--Internal Server Error".
          > Can anybody help me out? Thanks a lot!
          > inputparam1test.jsp:
          > <html>
          > <head>
          > <title>create a patameter input form</title>
          > </head>
          > <body>
          > <%!
          > String dataset;
          > String modeltype;
          > %>
          > <%
          > dataset = request.getParameter("dataset");
          > modeltype = request.getParameter("modeltype");
          > %>
          > <center>
          > <form action="inputparam2test.jsp" method="post">
          > <table cellspacing=10>
          > <tr>
          > <td colspan=2>
          >           Please enter model parameters
          >      </td>
          >      </tr>
          > <tr>
          >      <td>
          >           Dell LRI
          >      </td>
          >      <td>
          >           <select name="DellLRI">
          >           <option value="1">1
          >           <option value="0">0
          >           </select>
          >      </td>
          >      </tr>
          >      <%
          >      if(modeltype.equals("Direct")) {
          > out.println("in if");
          >      }
          > %>
          >      <tr>
          >      <td>
          >      </td>
          >      <td>
          >           <input type="submit" value="Next>" name="Next>">
          >      </td>
          >      </tr>
          > </form>
          > </center>
          > </body>
          > </html>
          > inputparam2test.jsp:
          > <html>
          > <head>
          > <title>continue input parameters </title>
          > </head>
          > <body>
          > <%
          > response.sendRedirect("inputparam1test.jsp");
          > %>
          > </body>
          > </html>
          Dimitri

[Cisco ISE 1.2 with 3850 - Trunk AP] Problem with MAB

Hi everyone,
After reading some documentation about using MAB in a trunk port with the 3850 I would like to know if someone has implemented ISE policies with a 3850 interface in trunk mode. My problem is that when I try using MAB in a trunk port the mac address of the AP it´s no visible in the "show mac address interface" and because of that the AP is not authenticated in ISE. The thing is that if I use a 2960 everything goes smoothly with no problems!
Let me show you what I have,
interface GigabitEthernet1/0/3
description AP
switchport trunk native vlan 999
switchport mode trunk
trust device cisco-phone
authentication event fail action next-method
authentication host-mode multi-host
authentication order mab dot1x
authentication priority dot1x mab
authentication port-control auto
mab
snmp trap mac-notification change added
snmp trap mac-notification change removed
dot1x pae authenticator
dot1x max-req 4
auto qos voip cisco-phone
service-policy input AutoQos-4.0-CiscoPhone-Input-Policy
service-policy output AutoQos-4.0-Output-Policy
############################################# switch model - 3850 ##################################################
SW1#sh mac address-table interface GigabitEthernet1/0/3
          Mac Address Table
Vlan    Mac Address       Type        Ports
SW1#sh dot1x interface Gi1/0/3
Dot1x Info for GigabitEthernet1/0/3
PAE                       = AUTHENTICATOR
QuietPeriod               = 60
ServerTimeout             = 0
SuppTimeout               = 30
ReAuthMax                 = 2
MaxReq                    = 4
TxPeriod                  = 30
Switch Ports Model              SW Version        SW Image              Mode
*    1 56    WS-C3850-48P       03.03.03SE        cat3k_caa-universalk9 INSTALL
############################################# Different switch model - 2960 ##################################################
interface GigabitEthernet1/0/1
description AP
switchport trunk native vlan 999
switchport mode trunk
srr-queue bandwidth share 1 30 35 5
priority-queue out
authentication event fail action next-method
authentication host-mode multi-host
authentication order mab dot1x
authentication priority dot1x mab
authentication port-control auto
mab
snmp trap mac-notification change added
snmp trap mac-notification change removed
mls qos trust device cisco-phone
mls qos trust cos
dot1x pae authenticator
dot1x max-req 4
auto qos voip cisco-phone
service-policy input AUTOQOS-SRND4-CISCOPHONE-POLICY
SW1#$cation sessions interface GigabitEthernet1/0/1
            Interface: GigabitEthernet1/0/1
          MAC Address: xxxx.xxxx.4a38
           IP Address: 172.18.1.170
            User-Name: xx-xx-xx-xx-4A-38
               Status: Authz Success
               Domain: DATA
       Oper host mode: multi-host
     Oper control dir: both
        Authorized By: Authentication Server
          Vlan Policy: N/A
      Session timeout: N/A
         Idle timeout: N/A
    Common Session ID: 0A18129D000060E39DAE8A8A
      Acct Session ID: 0x0000725D
               Handle: 0x0F00028C
Runnable methods list:
       Method   State
       mab      Authc Success
       Switch Ports Model              SW Version            SW Image
     1 28    WS-C2960X-24PS-L   15.0(2)EX5            C2960X-UNIVERSALK9-M
SW2#sh dot1x interface Gi1/0/1
Dot1x Info for GigabitEthernet1/0/1
PAE                       = AUTHENTICATOR
QuietPeriod               = 60
ServerTimeout             = 0
SuppTimeout               = 30
ReAuthMax                 = 2
MaxReq                    = 4
TxPeriod                  = 30
Am I doing something wrong?
BR,

I know what you mean and I agree with what you are saying :) Nonetheless, at the moment, the official stance from Cisco on this is that 802.1x is not supported on trunk ports. Now one can argue that MAB is different but I think we are just splitting hairs here :)
Like I said, I have gotten stuff to work before but always had some goofy things happening so in general I have stayed away from doing it.
Now in your situation, if your configuration is working fine on the 2960 but not on the 3850, then most likely the issue is with the XE code running on the 3850s. The XE code has been very problematic until recently so you are probably hitting some sort of a defect. As a result, I recommend that you upgrade the switch(es) to 3.3.5 or 3.6.1. Version 3.7.x is also out but it just came out 8 days ago so I would not recommend going to it.
Thank you for rating helpful posts!

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 firmware

I 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

Dial plan: Can we change the redirect number (sip-sip)?

Hello Scott Page/Alexei/Dirk Anyone.
I have a question on dial plan:
Call flow: A---cucm----pgw----voicemail
              B-----|
Its a sip to sip call, what i need to do is change the redirect number or rather add a digit to the redirect number at the PGW side.
For instance:
A number: 902228990
B number: 902228996
Redirect number: 92228996 --->Here i want to change it to 902228996
pgw patch: 9.8.2    "Patch:"CSCOgs014/CSCOnn014""
bash-3.00$ uname -a
SunOS tcs-tza1 5.10 Generic_127127-11 sun4u sparc SUNW,Netra-440
I checked from the cisco website but i think its only for isup sip calls only but not sure.
numan-add:fullnumbertrans:svcname="Service Name",numbtype="Number Type", digstring="Original Digits",
translatednum="Translated Digits"
Where numtype can be choose 3 for redirect number.
numtype—Identifier for the number type (1-5), it is one of the following values: 1—called party number 2—calling party number 3—redirecting number 4—calling party number and redirecting number 5—original called number
Can some help me out with the dial plan configuration?.
Regards,
Aby

Guys,
This configuration would work?, please let me know
1. mml> numan-add:service:custgrpid="DP00",name="BATMANredirect"
2. mml> numan-add:fullnumbertrans:svcname="BATMANredirect",digstring="92228996", translatednum="902228996",numtype="3"
numtype=3-->redirect number
3. Adding a result type of NUM_TRANS:
mml>   numan-add:resulttable:custgrpid="DP00",name="results",resulttype="NUM_TRANS", dw1="BATMANredirect",dw2="3",dw3="3",setname="setname3"
dw2--->number type hence i put it 3
dw3--->NOA--->hence i put it as national ie 3,
dw4--->Dialplan?-- can i avoid this?
4. mml> numan-add:resulttable:custgrpid="DP00",name="noar",resulttype="R_NUMBER_TYPE", dw1="4",setname="setname3"
dw1===>NOA again, i put it as dw1="4" as national.
Can any of you help me if this would do the trick or anything else needs to be added?.
Thanks and appreciate your time.
Regards,
Aby

Trunk port problem

Dear
I have problem, I need to config trunk port on SLM2048 but I can not see any options in web gui
How can I do this problem my switch informations are on below
Model Name
SLM2048
Hardware Version
00.03.00
Boot Version
1.0.1
Firmware Version
1.0.1

The router does not like the combination of cards you have in the chassis.
In order for a PRI configuration to work, the controller slot should support both voice and data capabilities. This is possible only on slot 0 and slot 1 in the 1760.
Make sure you are inserting the VWIC module into slot 0 or slot 1 in order for the PRI to work properly. MAke sure you have a PVDM in the chassis by looking for the "PVDM OK" LED on the back of the chassis.
Anther reason you are seeing this is that you don't have enough DSP resources to support 30 B channels.
See this doc on CCO for details:
http://www.cisco.com/en/US/products/hw/routers/ps221/products_tech_note09186a0080094a66.shtml

Library Folder Redirect causing MCX Problems for groups

Hi there,
My problem is related to this topic:
http://discussions.apple.com/thread.jspa?threadID=1685965&tstart=15
I have found that the only way to stop my afp server freaking out, and slowing down (drastically) the end users experience is to redirect the Library folder from their home folder to /tmp/user/Library so that it's being read and written locally.
The only problem with this of course is that all preferences are reset to default each time a user logs in. I'm not so bothered about that but it seems that my group MCX preferences are also not being created. Or at least some of them aren't, the dock seems fine but the preferences for the apple mouse for instance aren't being picked up.
I'm wondering if this is because the MCXRedirect isn't creating the Library folder quick enough for certain preferences to be copied!?
I've tried redirecting sub folders within the Library folder except Preferences but I still get the slow down (for instance Safari bounces about 7 times on launch without the Library folder redirected, It bounces just once with the redirect.
I'm wondering if there's a way to set the defaults locally on each machine, so that even if MCX doesn't work, the machines own default preferences could be replaced by mine!?
I'm running out of ideas, I have four brand new iMac suites, a very fast new server with plenty of RAM, running on a gigabit network and my users all complain of crippling slowness, constant beach-balling.
Thanks in advance!!

It didn't do anything to Front Row on my computer. I doubt it did anything on yours, either.
1) Update wouldn't have done that. There is no code in the update to replace that string with 10.5.8.
2) Your boot files are messed up or non-existent. Very rarely this can be fixed without reinstalling the system.
3) There are always a lot of permissions that can't be repaired by DU because they don't need to be repaired. They're not broken, just different. Did you check permissions before you applied the update?
4) Sounds like you have serious hardware problems. With the Snow Leopard disk installed, can you restart while holding down Option key and get to the startup manager? If so, pick the Snow Leopard disk and boot from it. Use Disk Utility in the Utilities menu (available after selecting a language) and repair the disk. Don't repair permissions from that disk.
If it finds errors and fixes them, try booting from the hard drive. You may have to re-install Snow Leopard.

SIP problem port

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
Emanuele

Sorry 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.

Redirecting Trunk SIP problem

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