Voip Configuration

Urgent...i'm newbe on cisco networking, I have 3 2811 with voice router in my network, internet and data networking configuration now is good work, but i have problem with voip configuration, i can't do it. Scenarion for voip: all extension phone in one SITE can connected dial call to other SITE and all extension phone all SITE can dial call to out (ISDN). This is my append attachments my network topology. My router 2811 site B (Router B) have include call manager and interface to ISDN on SITE B and interface to PABX on SITE C is VWIC-1MFT-G703(E1 trunk).
Please...any one can help me to give example configurations or completion configurations for my network???
Thanks a lot for your Help....

Help me, you can do it friends????

Similar Messages

Help - VOIP CONFIGURATION

Hi Experts,
We required VoIP configuration:
We have following hardware and network diagram is attached
(1)    Location A to B is connected through 2 mbps MPLS line – currently working OK through Cisco Routers.
(2)    We want to configure Total 4 Cisco Ip Phones – (2 in Location A + 2 in location B ) + 1 Cisco Analog telephone adopter.
(3)    We have 4 IP phones and 4 analog Phones.
ROUTER A T HEAD OFFICE .
(1) 2901 UC Bundle w/ PVDM3-16 FL-CME-SRST-25 UC License PAK
(2) Communication Manager Express or SRST - 25 seat license
(3) 16-channel high-density voice and video DSP module
(4) Cisco Communications Manager Express License
RUTER B
CISCO 1905/K9 – ROUTER.
WE WANT INTERNAL CALLING
EXTENSION TO EXTENSION FOR THIS 8 PHONES.
IF POSSIBLE THEY HAVE EPABX WITH THEM- WE WANT TO CONFIGURE OUTSIDE CALLING ALSO.
WE DON’T REQUIRE ANY MORE – HIGHER FACILITY.

Configure your 2901 as CME router and register all 8 IP phones and Cisco ATA with CME. You can have internal calling with all phones and for external PSTN calling configure dial-peer on CME and point it to E1/T1 or FXO or remote PBX ports.

VOIP Configuration - SRP547

I am unable to get any configuration parameters to work on my SRP547 for VOIP. ADSL2 works fine. Equipment replaces a ya.com (Arcadyan ARV4518PW) which worked ine albeit with significantly less features. Problems encountered due to the lack of equivalents for some parameters and an overabundance in the new unit which simply don't exist in the Arcadyan product.
The only items listed in the current equipment configuration are:
X = ID'ed in CRP547
Activated: X
User ID: X
Display Name: X
Password: X
Use Auth ID: X
Auth ID: X
SIP Domain: ????
Listen Port: assume SIP Port
Listen VC: ???
Proxy Server: X
Proxy Port: X
Use Outbound Proxy: X
Ourbound Proxy Port: X
Registar Expire: X
DTMF Mode Outband (RFC 2833) - can't find equivalent
DNS SRV: X
Registratuin Query: X
Available Codecs: X
I don't know if just identifying the above is sufficient to make it work or does the service provider need to supply more parameters.
Thanks,
Rolando

Hi Andrew,
First of all thank you for the fast response. I've tried what you say yet still can't get it to work. I should have put what they have instead of an x so below is the list. The reason is that some fields that were filled in the previous router have a configuration that is different to default on the 547. I've tried both with default on the 547 and changing the values as listed in the previous router. What should I try next?
Activated:YES
User ID: +34937928932
Display Name: +34937928932
Password: ******
Use Auth ID: YES
Auth ID: 937928932
SIP Domain: voipd.ya.com
Listen Port: assume SIP Port - 5060
Listen VC: Default VC Route
Proxy Server: proxy.voip.ya.com
Proxy Port: 5060
Use Outbound Proxy: NO
Ourbound Proxy Port: BLANK
Registar Expire: 4500 (default on SRP is 3600)
DTMF Mode - Outband (RFC 2833) - can't find equivalent
DNS SRV: ENABLE
Registratuin Query: ENABLE
Available Codecs: 726.16,726.24, 726.32, 726.40, 729, PCMA
Again, thanks for helping me sort this out.
Rolando

VoIP configure on edge switch 3750

Greeting
I am testing no cisco phone on 3750:
interface FastEthernet1/0/6
description testing
switchport access vlan 100
switchport mode access
switchport voice vlan 101
switchport port-security maximum 2
switchport port-security
switchport port-security aging time 2
switchport port-security violation restrict
switchport port-security aging type inactivity
srr-queue bandwidth share 10 10 60 20
srr-queue bandwidth shape 10 0 0 0
priority-queue out
mls qos trust dscp
auto qos voip cisco-phone
macro description cisco-phone
spanning-tree portfast
spanning-tree bpduguard enable
service-policy input AutoQoS-Police-CiscoPhone
end
and found that "switchport port-security"
will drop the phone's dhcp discovery packets.
When phone first time power on, it can get ip address from dhcp server; but, when you log out from current phone number, and the phone start to get ip address from dhcp again, the switch will drop the dhcp discover packets which the phone used to communicate with dhcp server.
I tried to increase max number to 6 (switchport port-security maximum 2) but it is not useful.
I did show port-security int command, and there is only one mac address on the interface.
I have also checked the mac address, and I can not see any violated to the security rules.
Could any one advice me:
- what the cause ?
- how can I debug it?
- if possible to fix it without disable the port security?
Any comments will be appreciated
thanks in advance

Great thanks for the reply, I have found the problem. the problem is the "switchport port-security aging time" has to be lower than 2. I have set it to 1 min.
Another question, we have been asked to set qos trust dscp, as:
mls qos trust dscp
can I get advice, if it will cause the workstation (PC/server) which plug into this port to get high priority treatment?
Please advice.
Many Regards

VOIP configuration over mpls link

Hi
we want to deploy voip for multiple location over the mpls lines we have between the multiple location can any body suggest the feasibility and how to do it and any related document and guidance for that
Thanks
Manish gaur

Hi,
what do you call a "mpls link"? Do you have an IP interface connecting to a MPLS network implemented by a provider or do you operate a MPLS network yourself?
In any case you might have a look at
1) Enterprise QoS Solution Reference Network Design Guide Version 3.3
http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmigration_09186a008049b062.pdf
2) Layer 3 MPLS VPN Enterprise Consumer Guide Version 2
http://www.cisco.com/application/pdf/en/us/guest/netsol/ns171/c649/ccmigration_09186a008077b19b.pdf
3) Various design guides for Unified communications
http://www.cisco.com/en/US/netsol/ns656/networking_solutions_design_guidances_list.html#anchor10
Hope this helps!
Regards, Martin

VOIP Configurations

I recently purchased a Linksys Phone Adapter with Router Model No: SPA2102. I have a USR9106 SureConnect ADSL Wireless Gateway.
I’m having problem in connecting my Linksys through the USR.
Please help.

check the thread below and some of the suggestions there...
http://forums.linksys.com/linksys/board/message?board.id=VoIP_Adapters&message.id=1679&query.id=9541...

Please help with SIP configuration on 2801 router

Hi All.
Please help me to setup a SIP account. I’m already struggling to do that for a few days, and can’t find out how to finish that. We have 2xISDN lines running, so I need to add a SIP trunk to existing config.
The information from our SIP provider:
We have issued the following DDI range: 018877000 – 99
There is no need to register the DDI’s as these will be offered to your PABX IP address provided to in the completed SIP trunking form.
Configuration details are as follows:
Our Primary Proxy:-        99.234.56.78
Codec supported:-             G711Alaw, G729 (G711Alaw is the preferred codec)
Fax Support:-                     T38 and G711Alaw
DTMF:-                                 RFC2833 and INFO
CLI Method:-                     Remote-Party-ID
Trunk doesn’t require registration; you just need to send Invite. In cisco this is done through Dial-peer session-target command. We are authenticating your IP address for outgoing calls and incoming calls we then forward to the IP mentioned in the sip form.
This is a SIP configuration on Cisco2801 router (I used outgoing calls only):
translation-rule 10
Rule 0 ^90 0
Rule 1 ^91 1
Rule 2 ^92 2
Rule 3 ^93 3
Rule 4 ^94 4
Rule 5 ^95 5
Rule 6 ^96 6
Rule 7 ^97 7
Rule 8 ^98 8
Rule 9 ^99 9
interface FastEthernet0/0.1
description ***DATA VLAN***
encapsulation dot1Q 1 native
ip address 10.1.1.101 255.255.255.0
interface FastEthernet0/0.2
description ***VOICE VLAN***
encapsulation dot1Q 2
ip address 192.168.22.1 255.255.255.0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
call start slow
sip
bind control source-interface FastEthernet0/0.2
bind media source-interface FastEthernet0/0.2
registrar server expires max 36000 min 600
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 1 pots
description ### External Dialling via BRI ###
preference 7
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/0
forward-digits all
dial-peer voice 2 pots
description ### External Dialling via BRI ###
preference 2
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/1
forward-digits all
dial-peer voice 9000 voip
description ** Outgoing calls to SIP **
preference 1
destination-pattern 9T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:99.234.56.78:5060
dtmf-relay rtp-nte
codec g711alaw
no vad
sip-ua
timers connect 100
sip-server ipv4:99.234.56.78
I used debugging commands to troubleshoot the calls.
2801(config-dial-peer)#
094509: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=211, Called Number=, Voice-Interface=0x65FA35B4,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
094510: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20018
094511: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
094512: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9
094513: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094514: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094515: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90, Peer Info Type=DIALPEER_INFO_SPEECH
094516: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90
094517: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094518: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094519: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908, Peer Info Type=DIALPEER_INFO_SPEECH
094520: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908
094521: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094522: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094523: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086, Peer Info Type=DIALPEER_INFO_SPEECH
094524: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086
094525: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094526: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094527: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862, Peer Info Type=DIALPEER_INFO_SPEECH
094528: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862
094529: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094530: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094531: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908621, Peer Info Type=DIALPEER_INFO_SPEECH
094532: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908621
094533: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094534: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094535: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086215, Peer Info Type=DIALPEER_INFO_SPEECH
094536: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086215
094537: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094538: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094539: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157, Peer Info Type=DIALPEER_INFO_SPEECH
094540: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157
094541: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094542: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094543: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908621577, Peer Info Type=DIALPEER_INFO_SPEECH
094544: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908621577
094545: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094546: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094547: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086215777, Peer Info Type=DIALPEER_INFO_SPEECH
094548: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086215777
094549: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094550: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094551: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094552: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094553: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094554: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094555: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
094556: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774T
094557: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094558: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094559: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094560: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094561: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094562: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094563: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
094564: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
094565: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094566: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094567: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094568: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094569: Jan 24 09:27:10.719: fb_get_reject_cause_code: ERROR cause_code NULL
094570: Jan 24 09:27:10.727: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397230
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094571: Jan 24 09:27:11.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397231
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094572: Jan 24 09:27:12.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:12 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397232
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094573: Jan 24 09:27:14.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam" <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:14 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397234
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I made some changes in the router configuration.
I removed FA0/0.2 Voice interface from Voice service voip configuration (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2). And now it’s using ip address 10.1.1.101 (data ip).
The debugging is changed now. I can send and receive a respond from SIP server. But It shows an error: SIP/2.0 404 Not Found
Then it moves to ISDN line, and use this line to make a call.
102988: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
102989: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774T
102990: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
102991: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
102992: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102993: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
102994: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
102995: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
102996: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
102997: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
102998: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102999: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
103000: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
103001: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
103002: Jan 24 14:45:47.298: fb_get_reject_cause_code: ERROR cause_code NULL
103003: Jan 24 14:45:47.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Seam" <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3989446920-1171263969-2466545983-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327416347
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 2438 9821 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 19412 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
103004: Jan 24 14:45:47.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
From: "Sam "<sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Call-ID: [email protected]
CSeq: 101 INVITE
Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK4875CB9
Content-Length: 0
103005: Jan 24 14:45:47.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
From: "Sam " <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
103006: Jan 24 14:45:47.374: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up
103007: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=211, Peer Info Type=DIALPEER_INFO_SPEECH
103008: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=211
103009: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
103010: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=20018
103011: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=0862157774, Peer Info Type=DIALPEER_INFO_SPEECH
103012: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=0862157774
103013: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
103014: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
   Result=NO_MATCH(-1)
103015: Jan 24 14:46:08.815: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down
2801(config-dial-peer)#
Then I removed SIP-UA as I was told there is no registration necessary, only Dial-peer configuration.
But it didn’t affect anything.
Then I add translate-outgoing called 10 command to dial-peer 9000, nothing happened.
Really stuck and don't know where to look at.
Any help will be highly appreciated.
Thanks.

Hi Dan.
Yes, I saw that RTP debugging, but what can I change there? Maybe I need to open more ports on ASA for RTP like 19412?
I use Cisco ASDM for ASA to make changes.
There are static NAT rules for: Server source IPs(10.1.1.100) to Outside(translated IPs, 88.99.77.44) for a few ports.
Also I added Security policy access rules for LAN: Any to SIP, and Outside: SIP to any.
For NAT:
I can't add this: for LAN: STATIC ROUTER IP 10.1.1.101 (AS SOURCE) UDP 5060 TO OUTSIDE IP 88.99.77.44
(AS TRANSLATED) UDP 5060
Because there is already translation for the Server.
Debugging looks like that now. There is no Received: SIP/2.0, but I can make an outside call with no audio.
116013: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
116014: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
116015: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
116016: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
116017: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
116018: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
116019: Jan 25 15:28:25.588: fb_get_reject_cause_code: ERROR cause_code NULL
116020: Jan 25 15:28:25.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:25 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505305
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116021: Jan 25 15:28:26.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:26 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505306
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116022: Jan 25 15:28:27.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:27 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505307
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116026: Jan 25 15:28:57.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam" ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389preference 1771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505337
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I'll add Incoming dial-peer now.
Not sure what kind of NAT rule should I put into ASA to allow in and out sip traffic.
Appretiate your help.
Thanks a mill.

Gizmo VoIP on E71/72

Hi, I would like to know please if anyone is using their Gizmo VoIP configurations to make VoIP calls. I am using Fring, set up with my Skype. There is definately a lag in the call. Even on WiFi there is a lag. I cannot notice any difference between the call quality between packet data 3G and WiFi for VoIP. Is it any better on the E72? Is the bottlneck the data connection? Should I wait for Nokia's 4G phones to come out? any comments would be appreciated. Thank you.
Nokia E65, E71-2 (30) [410.21.010 RM-357]

Hi ,I'm Using Gizmovoip as my voip calls for incoming and outgoing,through wifi, i believe you need to install sip voip setting 3.x first, and then under netsetting download gizmovo ip, plus check your router setting for port forwarding ,give your phone a static IP and put that IP on DMZ, would prolly work out,give it a shot my incoming calls works flawless & out going at times get lags, that's cause of jitters in connection through ISP

Voip set up

hi all...
i am new to voip configuration... and i have to configure the voip for our branch office... we have 2610 with the 2FXS and 2FXO prot... now i have one pulblic IP address which is used by my local lan user...and we have configured NAT/PAT on our VPN concerntrator... now we are looking to establish the VOIP phone (analogphone with the help of FXO and FXS) we have ASTRIK server at our main office and i want to register my local office analog phone with server at our main office... now what kind of configuration we need on our 2610 in order to configure voip...
connectivity:
ADSL connection form ISP---VPN 3005---D-Link nonmanagable swithch---LAN
at present we have above connectivity and now i want to add my 2610 router with analogphone connected to it... how can i connect and how can i configure it...?
regards
Devang

Dial-peer matching information:
http://www.cisco.com/warp/public/788/voip/in_dial_peer_match.html
Analog DID for Cisco 2600 and Cisco 3600 Series Routers:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/dt_did.htm

FXS and FXO configuration and design help

hi all...
i am new to voip configuration... and i have to configure the voip for our branch office... we have 2610 with the 2FXS and 2FXO prot... now i have one pulblic IP address which is used by my local lan user...and we have configured NAT/PAT on our VPN concerntrator... now we are looking to establish the VOIP phone (analogphone with the help of FXO and FXS) we have ASTRIK server at our main office and i want to register my local office analog phone with server at our main office... now what kind of configuration we need on our 2610 in order to configure voip...
connectivity:
ADSL connection form ISP---VPN 3005---D-Link nonmanagable swithch---LAN
at present we have above connectivity and now i want to add my 2610 router with analogphone connected to it... how can i connect and how can i configure it...?
regards
Devang

you need to configure pots and voip dialPeers for the incoming/outgoing legs of the calls.
pots dialPeers will be used for the FXO/FXS ports. voip dialPeers will be used for connection to ASTERISK voip pbx.
you need voice port configuration for your FXS to connect to your analogPhone.
you need voice port configuration for your FXO to connect to analog pstn. (if you have any)
see these links for more info:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml
http://www.cisco.com/en/US/tech/tk652/tk653/tech_configuration_examples_list.html
there are plenty of examples throughout these for reference.

Configure static NAT for range of ports

Hi,
I have a 2911 with a 3CX IP PBX behind it that needs to have a static NAT to the 3CX server for TCP/UDP 5060 and UDP 9000-9049. Do I have to create a static NAT entry for every single port in order for this to work, or can a range be defined in the NAT entries?
As an example, say my 3CX server has an internal IP of 192.168.1.25 and my external IP is 1.2.3.4. Would I have to create an entry for each port?
ip nat inside source static tcp 192.168.1.25 5060 1.2.3.4 5060
ip nat inside source static udp 192.168.1.25 5060 1.2.3.4 5060
ip nat inside source static udp 192.168.1.25 9000 1.2.3.4 9000
ip nat inside source static udp 192.168.1.25 9001 1.2.3.4 9001
and so on...
Is this the correct way to do it, or is there another better way?
Also, I only have one public IP to work with, and there are multiple other hosts on this network that need to have access to the internet. Right now I have NAT setup with overload so that the other hosts can get to the Internet. Here's my config for that:
ip nat pool PATPOOL 1.2.3.4 1.2.3.4 netmask 255.255.255.252
ip nat inside source list NAT_ACL pool PATPOOL overload
ip access-list standard NAT_ACL
remark PAT to outside
permit 192.168.1.0 0.0.0.255
exit
My question with this is will the static NAT work if I already have NAT overload configured as above?
Thanks for the help in advance.
Austin
PS here is 3CX documentation on this subject http://www.3cx.com/blog/voip-howto/cisco-voip-configuration/

I ended up creating a static NAT entry for each individual port mapping. This worked just as it was supposed to.
I have seen examples of people using route maps and ACLs to accomplish forwarding a range ports. I have yet to see official documentation from Cisco on this, and in some cases those examples did not seem to work correctly.
ASAs with the latest code have the ability to forward a range of ports, but based on my research IOS lacks this feature.
In my case, forwarding 50 ports wasn't so bad. However, if you have hundreds or thousands of ports to forward you may want to try the route map/ACL approach.
Hopefully this information useful to others.

Newbie Needs Help Please - VoIP and VPN

Hello,
I have a question about some voip/vpn configuration. I've got two sites that have 1760 routers with fxo/fxs cards that are going to be tieing the two phone systems together with a couple of voip trunks. These are secondary devices on the network with no data traffic or real LAN even, but addressed with a 2nd wan ip on the fa 0/0 port. The only traffic going through these routers is voice/voip.
My question is about this voip setup with vpn. First off, should I or shouldn't I, or even can I? If I configure a site-to-site vpn connection, will the voip traffic pass over that? How would I set up a VPN tunnel for only the fxo/fxs cards with no "LAN" behind it? Is there a need to have a vpn? Any benefits to sending the voip data across vpn, other than the obvious encryption of the "call"?
The thing is, I've only setup a VPN with Cisco a few couple of times, and once was from a pre-configured script I found and the other was from some software called SDM. I've only known about setting up VPN's for connecting the LAN behind router A to the LAN behind router B. This setup has NO LAN. The only interfaces are the fa 0/0 (getting static wan ip), and the fxo and fxs cards handling the voice. I have no clue on how to configure a site-to-site VPN for this, or if I even can, or if I should.
The next two replies will have my voip configuration as it sits on the bench in testing; I have both fa 0/0 interfaces connected to the local LAN just for a connection and testing, but each will later be getting it's own static WAN IP on the fa 0/0 interfaces:

SITE 1:
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname SITE 1
boot-start-marker
boot-end-marker
enable secret XXX
no aaa new-model
voice-card 2
voice-card 3
ip cef
interface FastEthernet0/0
ip address 192.168.254.30 255.255.255.0
speed auto
no shutdown
no ip http server
no ip http secure-server
control-plane
voice-port 2/0
connection plar opx 290
voice-port 2/1
connection plar opx 291
voice-port 2/2
voice-port 2/3
voice-port 3/0
connection plar 190
voice-port 3/1
connection plar 191
voice-port 3/2
voice-port 3/3
dial-peer voice 180 pots
destination-pattern 180
port 2/0
dial-peer voice 181 pots
destination-pattern 181
port 2/1
dial-peer voice 190 voip
destination-pattern 19
session target ipv4:192.168.254.40
line con 0
logging synchronous
line aux 0
line vty 0 4
password xxx
logging synchronous
login
transport input telnet
end

Nokia, VoIP and N86

Does the Nokia N86 have the Internet Tel App like the N95 does?
I'm still on my Nokia N95 because we use a VoIP system at work and i have my phone connected to it when in range, the Nokia N96 does not have it nokia thought it would be "funny" to take this feature out, i believe Nokia thought it would be really funny to take it out of the Nokia N97 too, will the Nokia N86 have this so i can make VoIP calls without having to install rubbish 3rd part applications?

quackers82 wrote:
I downloaded SIP_VoIP_3_x_Settings_v2_0_en.sis onto my N86, still cannot make internet calls. I have no Internet .Tel Application and "Internet Call" is not an option when dialing. Most i can get is "Registered" under SIP Settings, but no application to dial or recieve.
Am i missing somthing Amadir?
You are just missing the actual VoIP configuration profile which must be created with the above mentioned SIP VoIP Settings plugin. Version of the plugin mentioned above should work basically on any 3rd Edt. FP2 phone other than N96 and N78 of course which don't seem to support VoIP at all.
After you have installed the SIP VoIP Settings plugin you will have a new "Advanced VoIP Settings" module visible on the phones "Net Settings" application. I'm not sure about the default menu structure of the N86 but typically Net Settings entry can be found either under Settings -> Connectivity or it could be located directly in the "Control Panel" menu. Look around and you will find it somewhere on the N86 menus.
Go to the Net Settings application and select Advanced VoIP Settings -> Create New Service. Either select your your existing SIP settings or define new SIP profile and you will have a new VoIP profile available to be used.
Starting with VoIP rel. 3.x (3rd Edt. FP2) phones there is no way to manually define VoIP settings on the phone UI without installing the VoIP Settings plugin and using the it via Net Settings.
Also a separate "Internet Telephony" application (where VoIP registration status was controlled previously) does not exist anymore but instead VoIP activation/deactivation has been integrated in to the "Contacts" application, after you have created the VoIP settings and profile as mentioned above.
Your VoIP service should show up as an new "Tab" (named as your VoIP settings profile) on the Contacts and from this Tab you can activate and later de-activate (Option -> deactivate service) the VoIP registration. Once VoIP registration has been successfully established you should be able to call "Internet Call" type of calls to your contacts similarly to earlier generation VoIP enabled phones (E66, E71, N95 etc.).

Minimum hardware for 'voice' lab ? - Please answer.

Hi,
I've got (3) C2610's running IP Plus IOS 12.3. I also have (2) Cat 2912XL.
If I want to play with VoIP configurations, how much am I going to miss if I don't have a Layer 3 switch ? I mean, the Cat 3550 is kind of a little expensive for me and therefore I would like to play with VoIP without a layer 3 switch for now, if possible.
Please advise.

Thanks. It is a C2610. I put 64D and 16F and I am running in reality the Enterprise Basic image from the XM platform. The problem is that it is 12.3 and I see that XM's 12.4 don't run on C2610's.
How about the switch ?
Can I practice voice labs (at CVOICE level) and dispense the need for layer 3 switches ?

Adsl, modem, router & AEBS

I have introduced a Level One broadband router on gateway mode between my x modem and the AEBS. THis seems to have caused the AEBS to start dropping the signal - the effect on the Macs is that on the airport menu we get the message 'airport scanning'. Previously we had a solid signal
Any thoughts? Is it in the setup - should the airport be in bridge mode?

Thanks Duane.
I'm still after the same features I mentioned. ADSL modem for being an ADSL modem, VoIP and QoS, AEBS for everything else.
I tried leaving the AEBS as a bridge, but couldn't reach any ports on anything behind the AEBS, even after putting it in a DMZ on the ADSL modem.
I do now have it working, in case it helps anyone else:-
Billion - NAT and DHCP on, VoIP configured and QoS to throttle everything when a call is active. Assigning a static IP to the AEBS, and putting that IP in the DMZ.
AEBS - NAT and DHCP on (share a public IP address), static IP to my main mac for forwarding some ports, and NAT PMP on which seems to work by auto-opening other ports as needed (i.e. iChat video works with it enabled, and not with it off).
So there 'ya go. May not be conventional (?), but all seems hunky dory.
Cheers.

Voip Configuration

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