Forwarding with SIP Trunking and Retaining CID
We are using SIP trunking to get to the PSTN and CUCM 7.1. The SIP provider only permits calls originating from their own DIDs.
So how can we allow our users to forward all calls to say their cell phone. AND when a call comes in we want them to be able to see the original caller ID. Is it possible? What is the mojo? Thanks!
Hi,
for the SIP trunks there are no limits from the system. Check this out: https://supportforums.cisco.com/message/3795863#3795863
If you have different voice codecs for your phones and the trunk you need DSP ressources for transcoding.
As I know the UC520 have a PVDM2-64 with 4 DSP chips. You can use the DSP calculator from cisco to find out how many DSPs you need. But keep in mind that conferencing and transcoding can't share a DSP processor.
For example 1 DSP for conferencing and 3 for transcoding.
best regards
Christian
Similar Messages
-
Problem with sip trunk between CCM and Huawei through Cisco ASA5520
Hello,
I have a next problem
During SIP conversation ASA is changing the ip address of CCM to corresponding name in ASA configuration inside the SIP packet:
To: <sip:443230282@Server_CCM1;user=phone>
ASA name configuration:
name x.y.z.h Server_CCM1
But it should be without any changes like that: To: <sip:[email protected];user=phone>. Because of that session cant be established. Remote SIP peer gives an error "Bad Request - 'Malformed/Missing URL"
When name was deleted in ASA "no name x.y.z.h Server_CCM1" we have no any problem with SIP initialization and call proccesing.
We are going to upgrade ASA from 8.2 to 8.3 and it seems that we will have the same problem because object will be created automaticly in new version (we are using a NAT) and we will not be able to delete an object like we did in version 8.2.
What configuration in ASA version 8.3 should be done to avoid this issue.
P.S Detailed debug from Huawei in attachment.
Thank you.Hi.
depending on your config, you might be hitting CSCta16361, this is fixed in 8.2(4)
if you can confirm it's still happening in latest 8.2 release, then a TAC case needs to be opened so investigation is done and a new bug is opened.
if you've tested 8.2(4) already and it's still doing the same, then a TAC Service Request should be opened for more investigation and possibly opening a new defect.
Best regards,
Fadi.
does this answer your question? if yes please mark it resolved. -
PROBLEM WITH FORWARDING ALL - SIP TRUNK
Hello,
I'm experiencing the following problem:
I have this scenario: PSTN - SIP GW - CUCM6.1 - SIP TRUNK - CUM8.6
Phone A (extension 33476761834) is registered on CUCM8.6, with external forwarding to the PSTN.
If someone from CUCM8.6 calls - it works.
If someone from CUCM6.1 or from PSTN calls - fast busy. Error in CUCM traces:
19:20:47.056 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 145.245.235.201 on port 36615 index 3295
[1925279,NET]
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/TCP 145.245.235.201:5100;branch=z9hG4bK3744a3159aed8f
From: <sip:[email protected]>;tag=e1d37fe6-cb7b-46e7-a868-6fe81d6bb391-40319184
To: <sip:[email protected]>;tag=907605~2b367b5a-23ed-4193-a18b-e8c2f777615e-62999964
Date: Thu, 26 Sep 2013 17:20:46 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=100
Content-Length: 0
|2,100,63,1.580442^145.245.235.232^*
Any idea? Thanks a lot!
(Attaching the complete trace).Hi,
Making some tests regarding the problem we saw this:
Oct 1 23:02:22: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Date: Tue, 01 Oct 2013 23:02:22 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH
From: <sip:[email protected]>;tag=e1d37fe6-cb7b-46e7-a868-6fe81d6bb391-40468009
Allow-Events: presence
Supported: timer,replaces
Min-SE: 1800
Diversion: " Elodie" " <sip:[email protected]>;reason=unconditional;privacy=off;screen=yes
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM6.1
To: <sip:[email protected]>
Contact:
Expires: 180
Call-ID: [email protected]
Via: SIP/2.0/TCP 145.245.235.201:5060;branch=z9hG4bK3863af2fe80489
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 67
Oct 1 23:02:22: //350172/5D95E6C38917/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Malformed CC-Diversion/Diversion/CC-Redirect Header'
Via: SIP/2.0/TCP 145.245.235.201:5060;branch=z9hG4bK3863af2fe80489
From: <sip:[email protected]>;tag=e1d37fe6-cb7b-46e7-a868-6fe81d6bb391-40468009
To: <sip:[email protected]>;tag=DA70FE78-21AE
Call-ID: [email protected]
CSeq: 101 INVITE
Reason: Q.850;cause=100
Content-Length: 0
The problem is because in the redirect header, the comma "," is not a valid parameter so in the alerting name I removed the comma after the surname (i.e. "Elodie, Mary" to "Elodie Mary" and now is working.
We will change the alerting name by the moment and I will also investigate if there is a parameter to just not divert this name because is not needed this info in forwarded calls.
Anyway the problem is solved -
Problems between an UC520 and Asterisk with sip trunk
I have an UC520 and Asterisk with a sip trunk created between them, the calls from the UC520 to the Asterisk are ok, but the calls form de Asterisk to the UC520 are always busy.
Logs from the asterisk show that the first part of the call is ok, but the call is not complete, this means that the part where the extensions are with @ipuc520 doesn't appear
I created a sip trunk from de CCA 1.9 and it puts this for incoming calls for the dial peer, if I compare with a CCME, there is no configuration for incoming call there
/* Style Definitions */
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mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:Calibri;
mso-fareast-theme-font:minor-latin;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;
mso-fareast-language:EN-US;}
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:x.y.z.w
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
And there is no configurarion at all that could block the calls
The x.y.z.w was the sip server ip (asterisk ip)
The comminication between sip and h323 are allowed in the four ways
The allowed codecs are g711ulaw and g729r8
Asterisk is working now with other CCME and they are ok so I copied the configuration from those CCME to the UC520 and from the other sip trunks in asterisk the new trunk sip for uc520
The sip trunk created from the CCA was replaces for the one from the CCME that is working now
The routes are ok in Asterisk.
There is no translation profile in incoming calls.
There is no ACL applied in all configuration.
There is no log about callres incoming from the asterisk.
Could anyone halp me pls?Hi Rina,
Help me to try and understand what you are trying to do.
In this code snippet i see the following:
001808: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=7129, Called Number=7129, Peer Info Type=DIALPEER_INFO_SPEECH
001809: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=7129
001810: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
001811: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20036
This looks as though you have a call coming in from the Asterisk system to number 7129, which then leads to this according to the config file you provided.
number 7129
label 7129
description7129
name 7129
call-forward busy 6001
call-forward noan 6001 timeout 10
Which at this point I am going to assume this is ephone-dn 10 (Please confirm). If this is the case then the inbound call is being matched correctly to a DN (Which has its own dial-peer tag "Dial-peer Tag=20036".
But then i see this:
001817: 1w3d: //-1/55940098BA19/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
001818: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
001819: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
So the incoming call has been matched to Dial-peer 1000 which is an incoming VoIP dial-peer:
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
But then can see it has no where to go. So either I am reading this all wrong and the 7129 number is a result of another call taking place whilst you were debugging the system, or it is part of the debug and I am missing something here.
Rina, just so I understand this all. Are you trying to do WAN type calling from one system UC-500 (System "A") to the Asterisk system ( System B) and same? And so far calls going from the UC-500 to the Asterisk system are fine, but calls coming in from the Asterisk system to the UC-500 are not?
What happens on the Asterisk side when you try to call an Extension on the UC-500, do you get any ringing? Or is it a fast busy tone?
I am going to look over your configuration and debug a little further when I get home, maybe I am missing something here and can identify it.
Cheers,
David. -
Lync 2013 with SIP trunk with panasonic kx-tde200
Hi
My company has installed a panasonic ip-pbx kx-tde for multiline with 100 number range for telephone service.
Now my company is going to replace multiline by sip trunk . It will still work with Panasonic pbx box just need to reprogramme to be able to connected to the sip proxy which is managed by internet service provider.
For this scenario , would Lync 2013 voice work if I just add PSTN gateway which is the ip of panasonic pbx address to the frontend in topology ? Or I may need a mediation server as a must requirement to make lync voice work?
Thanks
WenFeiMedia bypass allow a call to basically skip the mediation server once it's established and go directly from gateway (in this case the PBX) and the endpoint (the telephone handset or Lync client) More information here: http://technet.microsoft.com/en-us/library/gg398719.aspx
By having this (if your PBX supports it) you reduce the load on the mediation servers. Before you go too far down the road also make sure that your PBX supports SIP trunks that are SIP over TCP (as Lync doesn't work with SIP over UDP)
Sort of, the easiest way is to add the .com as an additional SIP domain in Topology builder, you will need to create DNS records for it (both internal and external) and you will need to reissue the certs with additional SANs to support the second domain.
YOu will also need to update all the users to use the new suffix of xxx.com. So it's not a small task.
If this helped you please click "Vote As Helpful" if it answered your question please click "Mark As Answer" | Blog
www.lynced.com.au | Twitter
@imlynced -
NexVortex SIP trunk and UC500 default timeout settings?
Hey guys,
I'm doing a little SIP trunk testing to determine a good provider for my customer base, and had some general questions as I can't seem to get outgoing or incoming phone calls to work at all.
To keep things simple, I'm using an 8user UC540W with 3 IP phones - a 525G, a 524G, and a 7937 conference phone. I have a static IP on the UC540, have run through the telephony wizard and everything seems to be working on the LAN/PBX side of things. The big difference, and the major variable that we are working with (I believe), is that we're working with Satellite internet connectivity rather than terrestrial Internet connectivity. This is an Enterprise satellite connection, and we have run voice over the connection without problems, but this is our first attempts at SIP trunking from a UC500. Due to the latency involved inherent in satellite (ping times around 550-700ms), I believe that either UC540 or NexVortex server/switch is timing out. Is there any way to determine what the default setting is for a SIP acknowledgement on the UC540 and change this if it is too small?
Here is what I have found, if it is helpful:
Outgoing calls:
1. The SIP provider, NexVortex, says that they are seeing an invite from the UC540, but not on port 5060. On the two calls that we tested, it first saw an invite on 63452, and then on 51677. Is there any reason why this would not be sent out on 5060?
Incomign calls:
1. On incoming calls, Nexvortex is routing the calls to the proper IP, but is then receiving an "error 500 reason Q850" from the UC540. What does this error mean?
I am also attaching my config in the event that it helps. When I look at the SIP trunk status in CCA, it does not show that registration is working, so I assume that's a good place to start.
Lastly, the guys over at NexVortex don't seem to run across the UC500 very often. If anybody has setup their UC500 to work with NexVortex and wouldn't mind posting a screenshot from CCA (feel free to remove usernames and passwords), I'd appreciate it. I'm not certain that I have all of the information in the right places.
Thanks,
SethHi Steven,
Thanks for the continued help.
I was able to make the changes in the config. Here are snapshots from the current config:
dial-peer voice 1000 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3000 voip
description IncomingSIP
translation-profile incoming IncomingSIP_Called_4
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 14068906254$
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3001 voip
description IncomingSIP2
translation-profile incoming IncomingSIP2_Called_5
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 1406890624[2-3]$
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3002 voip
incoming called-number 14068906254$
no dial-peer outbound status-check pots
sip-ua
authentication username nomadgcs password 7 *removed*
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:66.23.129.253:5060 expires 3600
sip-server ipv4:66.23.129.253:5060
connection-reuse
host-registrar
We are calling from within the 406 area code, so when we dial the number with the leading 406, we get a message saying "You don't need the area code" from the telephone company. When we dial this from a cell, we get the following:
1. 4068906254 - "All circuits are busy, please try your call again..."
2. 8906254 - rings once, then no sound, then disconnects after about 10 seconds.
I don't know if this would factor in at all, but our NexVortex account is setup to deliver 14068906254 to the UC500, but would NexVortex deliver the entire string of characters if it is only receiving 4068906254 or 8906254?
Thanks,
Seth -
How to configure a port channel with VLAN trunking (and make it work..)
We're trying to configure a port channel group with trunked ports to connect a NetApp HA pair. We want to create two data LIFs and connect them to the switch stack. We are trying to create 2 data lifs, one for cifs and one for nfs that are on different vlans.
We want the same ports to be able to allow multiple vlans to communicate. (trunked)
These data lifs should be able to fail over to different nodes in the HA pair and still be able to communicate on the network.
What this means is that we have to connect 4 ports each for each node in the NetApp HA Pair to the switches and create a port channel of some type that allows for trunked vlans. When we configure the ports, the configuration is as follows (below):
We are only able to configure an IP on one of the vlans.
When we configure an IP from another vlan for the data lif, it does not respond to a ping.
Does anyone have any idea what I'm doing wrong on the Cisco switch?
interface GigabitEthernet4/0/12
description Netapp2-e0a
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
channel-protocol lacp
channel-group 20 mode active
end
interface GigabitEthernet4/0/13
description Netapp2-e0c
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
channel-protocol lacp
channel-group 20 mode active
end
interface GigabitEthernet6/0/12
description Netapp2-e0b
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
channel-protocol lacp
channel-group 20 mode active
end
interface GigabitEthernet6/0/13
description Netapp2-e0d
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
channel-protocol lacp
channel-group 20 mode active
end
interface Port-channel20
description Netapp2-NFS
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
spanning-tree portfast
spanning-tree bpduguard enable
endOur problem was fixed by the storage people. They changed the server end to trunk, and the encapsulation / etherchannel.
I like all the suggestions, and they probably helped out with the configuration getting this to work.
Thanks!
interface Port-channel20
description Netapp2-NFS
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
switchport mode trunk
interface GigabitEthernet4/0/12
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
switchport mode trunk
channel-protocol lacp
channel-group 20 mode active
interface GigabitEthernet4/0/13
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
switchport mode trunk
channel-protocol lacp
channel-group 20 mode active
interface GigabitEthernet6/0/12
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
switchport mode trunk
channel-protocol lacp
channel-group 20 mode active
interface GigabitEthernet6/0/13
switchport trunk encapsulation dot1q
switchport trunk allowed vlan 10,20,511,519
switchport mode trunk
channel-protocol lacp
channel-group 20 mode active -
Port Forwarding with Verizon-Br​anded Action GT704-WG
I am new to the forums, and I am already overjoyed with the 30 minutes I spent figuring out that the login mechanism does not even work with browsers other than IE7/8. Lots of fun to start.
I live in a residential building with the residential High Speed Internet Plan. I assume this is the case because it is a DSL line and they are not using business grade, or I do not think I would be having these problems.
I have spent many hours trying to figure out how to port forward with the above-mentioned router/modem/gateway/**bleep** extraordinnaire. I am in IT professionally, and I am tired of feeling **bleep**. I called Verizon tech, but they were very unhelpful unless I pay for a premium tech service to figure out port forwarding problems. And yes, I have read the info on port-forward.com numerous times for this device.
Essentially, I want to forward an arbitrary port, let's say 6336, to a server running Ubuntu 9.10. I have put this computer in the DMZ using the web interface. I have also set up a mapping rule to forward 6336 (all three entries pointed to 6336 in the Security/Applications/Forwarding section, following docs on port-forward.com for this specific model AND Verizon's own documentation). Occassionally, I can see the service is viewable from the likes of canyouseeme.org. Nonetheless, I cannot access the port: I get a connection refused error when using ssh on this port. Yes, I correctly configured ssh on the DMZ server/host to respond to the non-standard 6336 port, and I have tried it with the server's firewall system (ufw/iptables, for the curious) enabled and disabled. It NEVER works.
Some posts here have indicated I might need a static IP assignment. That can be done only client-side (meaning the Ubuntu server in the DMZ), if I understand people saying it is not possible to do static assignments from this **bleep** gateway, so I am not sure if that should matter (I wonder what happens with this **bleep** when two computers try to demand the same static IP; dare I guess that he asks first wins?). I cannot recall if I have uPnP enabled or disabled as I am at the office right now, but I believe it is off. Not sure if this matters either. If I am wrong, I would love if someone could let me know. I am at a loss and sick of dealing with such a simple problem. I would really appreciate the help.
Solved!
Go to Solution.Ok.
#1 On your computer setup a Static IP.
This means following the directions at http://www.cyberciti.biz/tips/howto-ubuntu-linux-convert-dhcp-network-configuration-to-static-ip-con...
#2 The Static IP must be outside of that DHCP range of the router. So, this means your IP has be above 1 but below 64.
#3 In the router forward to the IP Address that you setup on the computer.
OR if the router can give the computer the same IP Address each time, you could do that.
If you are the original poster (OP) and your issue is solved, please remember to click the "Solution?" button so that others can more easily find it. If anyone has been helpful to you, please show your appreciation by clicking the "Kudos" button. -
SIP Trunk - No voice with Single Number Reach
Hi Community.
I setup SIP Trunk with the CCA. Everything is working Call In and Call Out. Call Forward and so on.
But with Single Number reach is something wrong. The mobile phone is ringing and I can get the call, but I hear not any voice.
Can someone please help me out? Below the config.
version 15.1
parser config cache interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
service compress-config
service sequence-numbers
dot11 ssid cisco-data
vlan 1
authentication open
dot11 ssid cisco-voice
vlan 100
authentication open
ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.1.1 10.1.1.9
ip dhcp excluded-address 10.1.1.241 10.1.1.255
ip dhcp pool phone
network 10.1.1.0 255.255.255.0
default-router 10.1.1.1
option 150 ip 10.1.1.1
ip domain name site1.365873.trk.ipvoip.ch
ip name-server 8.8.8.8
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp router-traffic
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
isdn switch-type basic-net3
voice call send-alert
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
registrar server expires max 3600 min 3600
localhost dns:site1.365873.trk.ipvoip.ch
no update-callerid
voice class codec 1
codec preference 1 g711alaw
voice register global
mode cme
source-address 10.1.1.1 port 5060
load 9971 sip9971.9-2-2
load 9951 sip9951.9-2-2
load 8961 sip8961.9-2-2
timezone 23
voice source-group CCA_SIP_SOURCE_GROUP_CUE_CME
access-list 2
translation-profile incoming SIP_Incoming
voice source-group CCA_SIP_SOURCE_GROUP_EXTERNAL
access-list 3
voice translation-rule 9
rule 1 /0041449475090/ /90/
rule 2 /0041449475091/ /91/
rule 3 /0041449475092/ /92/
rule 4 /0041449475093/ /93/
rule 5 /0041449475094/ /94/
rule 6 /0041449475095/ /95/
rule 7 /0041449475096/ /96/
rule 8 /0041449475097/ /97/
rule 9 /0041449475098/ /98/
rule 10 /0041449475099/ /99/
voice translation-rule 410
rule 1 /^0\(.*\)/ /\1/
rule 15 /^..$/ /0041449475090/
voice translation-rule 411
rule 1 /^0\(.*\)/ /ABCD0\1/
voice translation-rule 412
rule 1 /^ABCD\(.*\)/ /\1/
voice translation-rule 422
rule 15 /^ABCD\(.*\)/ /\1/
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
rule 1 /^9\([1-9]\)$/ /004144947509\1/
rule 15 /^..$/ /0041449475090/
voice translation-rule 1112
rule 1 /^0/ //
voice translation-rule 2000
rule 1 /0041449475098/ /98/
voice translation-rule 2001
rule 1 /0041449475097/ /97/
voice translation-rule 2002
rule 1 /^6/ //
voice translation-rule 2222
voice translation-profile AA_Profile
translate called 2001
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PSTN_CallForwarding
translate redirect-target 410
translate redirect-called 410
voice translation-profile PSTN_Outgoing
translate calling 1111
translate called 1112
translate redirect-target 410
translate redirect-called 410
voice translation-profile SIP_Called_9
translate calling 3265
translate called 9
voice translation-profile SIP_Incoming
translate called 411
voice translation-profile SIP_Passthrough
translate called 412
voice translation-profile SIP_Passthrough_CallBlocking
translate called 422
voice translation-profile VM_Profile
translate called 2000
voice translation-profile XFER_TO_VM_PROFILE
translate redirect-called 2002
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
fax interface-type fax-mail
license udi pid UC540W-BRI-K9 sn FGL163220SL
archive
log config
logging enable
logging size 600
hidekeys
username admin privilege 15 secret xxx
username xxx password 0 ""
username xxx password 0 ""
ip tftp source-interface Loopback0
bridge irb
interface Loopback0
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
no ip address
ip inspect SDM_LOW out
ip virtual-reassembly in
ip verify unicast reverse-path
load-interval 30
shutdown
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
no ip address
macro description cisco-desktop
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport voice vlan 100
no ip address
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/8
no ip address
macro description cisco-desktop
spanning-tree portfast
interface BRI0/1/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
isdn static-tei 0
interface BRI0/1/1
no ip address
shutdown
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn sending-complete
isdn static-tei 0
interface Dot11Radio0/5/0
no ip address
ssid cisco-data
ssid cisco-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
antenna receive right
antenna transmit right
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
ip address 192.168.10.2 255.255.255.0
ip access-group 102 in
ip nat inside
ip virtual-reassembly in
interface BVI100
description $FW_INSIDE$
ip address 10.1.1.1 255.255.255.0
ip access-group 103 in
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip dns server
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 192.168.10.1
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
access-list 1 remark SDM_ACL Category=2
access-list 1 permit 10.1.1.0 0.0.0.255
access-list 1 permit 192.168.10.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
access-list 2 remark CCA_SIP_SOURCE_GROUP_ACL_INTERNAL
access-list 2 remark SDM_ACL Category=1
access-list 2 permit 192.168.10.2
access-list 2 permit 10.1.10.0 0.0.0.3
access-list 2 permit 192.168.10.0 0.0.0.255
access-list 2 permit 10.1.1.0 0.0.0.255
access-list 3 remark CCA_SIP_SOURCE_GROUP_ACL_EXTERNAL
access-list 3 remark SDM_ACL Category=1
access-list 3 permit 212.147.47.216
access-list 3 deny any
access-list 100 remark auto generated by SDM firewall configuration
access-list 100 remark SDM_ACL Category=1
access-list 100 deny ip 192.168.10.0 0.0.0.255 any
access-list 100 deny ip host 255.255.255.255 any
access-list 100 deny ip 127.0.0.0 0.255.255.255 any
access-list 100 permit ip any any
access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 101 remark SDM_ACL Category=1
access-list 101 permit tcp 10.1.1.0 0.0.0.255 eq 2000 any
access-list 101 permit udp 10.1.1.0 0.0.0.255 eq 2000 any
access-list 101 deny ip 10.1.1.0 0.0.0.255 any
access-list 101 deny ip 192.168.10.0 0.0.0.255 any
access-list 101 deny ip 192.168.1.0 0.0.0.255 any
access-list 101 deny ip host 255.255.255.255 any
access-list 101 deny ip 127.0.0.0 0.255.255.255 any
access-list 101 permit ip any any
access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
access-list 102 remark SDM_ACL Category=1
access-list 102 deny ip 10.1.10.0 0.0.0.3 any
access-list 102 deny ip 10.1.1.0 0.0.0.255 any
access-list 102 deny ip 192.168.1.0 0.0.0.255 any
access-list 102 deny ip host 255.255.255.255 any
access-list 102 deny ip 127.0.0.0 0.255.255.255 any
access-list 102 permit ip any any
access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 103 remark SDM_ACL Category=1
access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 deny ip 10.1.10.0 0.0.0.3 any
access-list 103 deny ip 192.168.10.0 0.0.0.255 any
access-list 103 deny ip 192.168.1.0 0.0.0.255 any
access-list 103 deny ip host 255.255.255.255 any
access-list 103 deny ip 127.0.0.0 0.255.255.255 any
access-list 103 permit ip any any
access-list 104 remark auto generated by SDM firewall configuration##NO_ACES_14##
access-list 104 remark SDM_ACL Category=1
access-list 104 deny ip 10.1.10.0 0.0.0.3 any
access-list 104 deny ip 10.1.1.0 0.0.0.255 any
access-list 104 permit ip any any
access-list 104 permit udp host 8.8.8.8 eq domain any
access-list 104 permit icmp any any echo-reply
access-list 104 permit icmp any any time-exceeded
access-list 104 permit icmp any any unreachable
access-list 104 deny ip 10.0.0.0 0.255.255.255 any
access-list 104 deny ip 172.16.0.0 0.15.255.255 any
access-list 104 deny ip 192.168.0.0 0.0.255.255 any
access-list 104 deny ip 127.0.0.0 0.255.255.255 any
access-list 104 deny ip host 255.255.255.255 any
access-list 104 deny ip host 0.0.0.0 any
access-list 104 deny ip any any
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone CH
station-id name FAX
station-id number 99
caller-id enable
voice-port 0/0/1
cptone CH
shutdown
caller-id enable
voice-port 0/0/2
cptone CH
shutdown
caller-id enable
voice-port 0/0/3
cptone CH
shutdown
caller-id enable
voice-port 0/1/0
compand-type a-law
cptone CH
bearer-cap Speech
voice-port 0/1/1
compand-type a-law
cptone CH
bearer-cap Speech
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.1.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register mtpa4934c6ee4e0
dspfarm profile 2 transcode
description CCA transcoding for SIP Trunk VTX
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application SCCP
dial-peer cor custom
name internal
name local
name local-plus
name international
name national
name national-plus
name emergency
name toll-free
dial-peer cor list call-internal
member internal
dial-peer cor list call-local
member local
dial-peer cor list call-local-plus
member local-plus
dial-peer cor list call-national
member national
dial-peer cor list call-national-plus
member national-plus
dial-peer cor list call-international
member international
dial-peer cor list call-emergency
member emergency
dial-peer cor list call-toll-free
member toll-free
dial-peer cor list user-internal
member internal
member emergency
dial-peer cor list user-local
member internal
member local
member emergency
member toll-free
dial-peer cor list user-local-plus
member internal
member local
member local-plus
member emergency
member toll-free
dial-peer cor list user-national
member internal
member local
member local-plus
member national
member emergency
member toll-free
dial-peer cor list user-national-plus
member internal
member local
member local-plus
member national
member national-plus
member emergency
member toll-free
dial-peer cor list user-international
member internal
member local
member local-plus
member international
member national
member national-plus
member emergency
member toll-free
dial-peer voice 1 pots
destination-pattern 99
port 0/0/0
no sip-register
dial-peer voice 2 pots
port 0/0/1
no sip-register
dial-peer voice 3 pots
port 0/0/2
no sip-register
dial-peer voice 4 pots
port 0/0/3
no sip-register
dial-peer voice 5 pots
description ** MOH Port **
destination-pattern ABC
port 0/4/0
no sip-register
dial-peer voice 6 pots
description tcatch all dial peer for BRI/PRIv
translation-profile incoming nondialable
incoming called-number .%
direct-inward-dial
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
direct-inward-dial
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
direct-inward-dial
port 0/1/1
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
translation-profile outgoing XFER_TO_VM_PROFILE
destination-pattern 98
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 97
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2012 voip
description ** cue prompt manager number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 96
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (VTX) **
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1001 voip
corlist outgoing call-local
description ** star code to SIP trunk (VTX) **
destination-pattern *..
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 1003 voip
description ** Passthrough Inbound Calls for PSTN from CUE **
translation-profile incoming SIP_Passthrough
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number ABCDT
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1005 voip
description ** Passthrough Inbound Calls for MWI from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number A80T
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1009 voip
description ** Passthrough Inbound Calls for Internal Extensions from CUE **
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number ^..$
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1033 voip
corlist outgoing call-local
description **CCA*Switzerland*Short Code Services**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0187
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
dial-peer voice 1042 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Ambulance / Poisioning**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0014[45]
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
dial-peer voice 1041 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 00333333333
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
dial-peer voice 1025 voip
corlist outgoing call-national
description **CCA*Switzerland*National Destination Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00[789]1.......
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
dial-peer voice 1020 voip
corlist outgoing call-national
description **CCA*Switzerland*Regional Announcement VM**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 01600
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
dial-peer voice 1040 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 000333333333
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
dial-peer voice 1043 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Ambulance / Poisioning**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 014[45]
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
dial-peer voice 1035 voip
corlist outgoing call-national
description **CCA*Switzerland*Mobile Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 007[46789].......
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
dial-peer voice 1024 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Personal Numbering**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00878......
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
dial-peer voice 1029 voip
corlist outgoing call-national
description **CCA*Switzerland*Voicemail Access**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00860.........
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
dial-peer voice 1036 voip
corlist outgoing call-national
description **CCA*Switzerland*VPN Access**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00869.............
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
dial-peer voice 1027 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Premium Rate (Business)**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00900......
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
dial-peer voice 1026 voip
corlist outgoing call-national
description **CCA*Switzerland*Test Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00868T
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
dial-peer voice 1034 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Shared Cost numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0084[0248]......
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
dial-peer voice 1038 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Emergency**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0011[278]
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
dial-peer voice 1037 voip
corlist outgoing call-toll-free
description **CCA*Switzerland*Toll Free Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00800......
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
dial-peer voice 1039 voip
corlist outgoing call-emergency
description **CCA*Switzerland*Emergency**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 011[278]
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
dial-peer voice 1032 voip
corlist outgoing call-national
description **CCA*Switzerland*National Destination Numbers**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 00[23456]........
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
dial-peer voice 1023 voip
corlist outgoing call-international
description **CCA*Switzerland*International Calls**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 000T
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
dial-peer voice 1031 voip
description **CCA*Switzerland*Premium Rate (Social)**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 0090[16]......
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
dial-peer voice 1030 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 014[0357]
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
dial-peer voice 1045 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA/Glaciers Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 0141[45]
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
dial-peer voice 1028 voip
corlist outgoing call-national-plus
description **CCA*Switzerland*Directory Enquiries**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 018[15].
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
dial-peer voice 1021 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 011[45].
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
dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*Switzerland*Short Code Services**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 01[67].
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
dial-peer voice 1044 voip
corlist outgoing call-emergency
description **CCA*Switzerland*REGA/Glaciers Air Rescue**
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 1
destination-pattern 00141[45]
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
dial-peer voice 2002 voip
description ** cue voicemail PSTN number **
translation-profile outgoing VM_Profile
destination-pattern xxx$
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 2003 voip
description ** cue auto attendant PSTN number **
translation-profile outgoing AA_Profile
destination-pattern xxx$
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1110 pots
preference 9
destination-pattern xxx
port 0/0/0
no sip-register
dial-peer voice 3006 voip
description SIP
translation-profile incoming SIP_Called_9
session protocol sipv2
session target sip-server
incoming called-number xxx.
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
no dial-peer outbound status-check pots
sip-ua
keepalive target dns:site1.365873.trk.ipvoip.ch
authentication username xxx password 7 xxx
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
registrar dns:site1.365873.trk.ipvoip.ch expires 3600
sip-server dns:site1.365873.trk.ipvoip.ch
host-registrar
telephony-service
sdspfarm units 5
sdspfarm transcode sessions 10
sdspfarm tag 2 mtpa4934c6ee4e0
video
fxo hook-flash
max-ephones 40
max-dn 300
ip source-address 10.1.1.1 port 2000
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone ehookenable 1
service phone ehookEnable 1
service dnis overlay
service dnis dir-lookup
service dss
timeouts interdigit 5
system message SwissT.Net
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
cnf-file location flash:
cnf-file perphone
user-locale U4 load CME-locale-de_DE-German-8.1.2.2.tar
network-locale U4
load 521G-524G cp524g-8-1-17
load 525G spa525g-7-5-4
load 501G spa50x-30x-7-5-2b
load 502G spa50x-30x-7-5-2b
load 504G spa50x-30x-7-5-2b
load 508G spa50x-30x-7-5-2b
load 509G spa50x-30x-7-5-2b
load 525G2 spa525g-7-5-4
load 301 spa50x-30x-7-5-2b
load 303 spa50x-30x-7-5-2b
time-zone 23
time-format 24
date-format dd-mm-yy
keepalive 30 auxiliary 4
voicemail 98
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh flash:/media/music-on-hold.au
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 xxx
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern .T
transfer-pattern 0.T
transfer-pattern 6.. blind
secondary-dialtone 0
night-service day Sun 17:00 09:00
night-service day Mon 17:00 09:00
night-service day Tue 17:00 09:00
night-service day Wed 17:00 09:00
night-service day Thu 17:00 09:00
night-service day Fri 17:00 09:00
night-service day Sat 17:00 09:00
fac standard
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-template 1
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
service phone webAccess 0
softkeys remote-in-use Newcall
softkeys idle Redial Pickup Mobility Newcall Cfwdall Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Trnsfer Mobility TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-template 15
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-template 16
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
ephone-template 17
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
ephone-template 18
url services 1 http://10.1.10.1/voiceview/common/login.do VoiceviewExpress
softkeys remote-in-use CBarge Newcall
softkeys idle Redial Newcall Mobility Cfwdall Pickup Gpickup Dnd Login
softkeys seized Cfwdall Endcall Redial Pickup Gpickup Callback
softkeys connected Hold Endcall Mobility Trnsfer TrnsfVM Confrn Acct Park
button-layout 7931 2
ephone-dn 9
number BCD no-reg primary
description MoH
moh out-call ABC
ephone-dn 292
number xxx
description SIP Main Number registration
preference 10
ephone-dn 293 dual-line
number 90 secondary xxx no-reg both
label Zentrale
description 90
name Zentrale
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 294 dual-line
number 94 secondary xxx no-reg both
label LL
description Lehrling Lehrnende
name Lehrling Lehrnende
mobility
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 295 dual-line
number 93 secondary xxx no-reg both
label CM
description
name
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 10
ephone-dn 296 dual-line
number 92 secondary xxx no-reg both
label EE
description
name
mobility
call-forward busy 98
call-forward noan 98 timeout 20
ephone-dn 297 dual-line
number 91 secondary xxx no-reg both
label RS
description
name
mobility
snr xxx delay 1 timeout 30 cfwd-noan 98
snr ring-stop
call-forward busy 98
call-forward noan 98 timeout 10
ephone-dn 298
number 6.. no-reg primary
description ***CCA XFER TO VM EXTENSION***
call-forward all 98
ephone-dn 299
number A801.. no-reg primary
mwi off
ephone-dn 300
number A800.. no-reg primary
mwi on
ephone 1
device-security-mode none
mac-address A44C.11A0.B648
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:296 2:293 3m297 4m295
button 5m294
ephone 2
device-security-mode none
mac-address A44C.11A0.B566
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:297 2:293 3m296 4m295
button 5m294
ephone 3
device-security-mode none
mac-address A44C.11A0.B5C4
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:295 2:293 3m297 4m296
button 5m294
ephone 4
device-security-mode none
mac-address A44C.11A0.B67A
ephone-template 1
max-calls-per-button 2
username "xxx" password xxx
type 525G2
button 1:294 2:293 3m297 4m296
button 5m295
alias exec cca_voice_mode PBX
alias exec cca_vm_notification schedule from_time=00 to_time=24
alias exec clid-ALL_BRI ;1:0-4;1:0-9;1:0-9;1:1-9
alias exec clid-SIP ;1:1-9;1:1-9;1:1-9
banner login ^CCisco Configuration Assistant. Version: 3.2 (3). Fri Jul 04 13:18:33 CEST 2014^C
line con 0
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
transport preferred none
transport input all
line vty 5 100
transport preferred none
transport input all
ntp master
ntp server 91.240.0.5 prefer
enHi Patrick
I am working on this one as well. I have a UC560 with SIP Trunk provider Les.NET.
It was working fine until a few weeks ago when something changed on the provider end and broke it. My hunch it is something to do with the SIP REFER.
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html
Here is an excerpt from the above page:
Call Transfer
When a call comes in on an SIP trunk to an SCCP Phone or CUE AutoAttendant (AA) and is transferred, the CME by default will send a SIP REFER message to the SP proxy. Most SP Proxy Servers do not support the REFER method. This needs to be configured in order to force the CME to hairpin the call:
Router(config)#voice service voip
Router(conf-voi-serv)#no supplementary-service sip refer
Figure 3 shows the behavior of the CME system with the REFER method disabled. -
Issue with instant ringback when using sip trunk to SP
Hi all,
We use CUCM 8.0.2.
We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
c2900-universalk9-mz.SPA.150-1.M3.bin
Cisco CISCO2911/K9 (revision 1.0)
Technology Package License Information for Module:'c2900'
Technology Technology-package
Current Type
ipbase ipbasek9 Permanent
security securityk9 Permanent
uc uck9 Permanent
data None None
We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
We use 7945 and CIPC for our phones.
We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
Any ideas why this happens and how to stop it?
I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
voice service voip
address-hiding
mode border-element
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
header-passing error-passthru
early-offer forced
midcall-signaling passthru
interface GigabitEthernet0/0
ip address x.x.x.x 255.255.255.252
ip access-group acl.SIP-IN in
no ip redirects
no ip unreachables
ip verify unicast reverse-path
ip virtual-reassembly
duplex full
speed 100
no cdp enable
gateway
timer receive-rtp 1200
sip-ua
connection-reuse
gatekeeper
shutdown
dial-peer voice 1 voip
description *** INBOUND CALLS FROM CARRIER ***
translation-profile incoming SIPTRUNK-INCOMING
session protocol sipv2
incoming called-number #blah blah#
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 61 voip
description **** WA, SA AND NT NUMBERS ****
destination-pattern 0[8]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[8]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 81 voip
description **** MOBILE NUMBERS ****
destination-pattern 0[4]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[4]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 500 voip
description *** INBOUND SIP TRUNK TO CUCM PUB ***
translation-profile outgoing SIPTRUNK-CALLING-ADD-0
preference 1
destination-pattern 5[12]..
session protocol sipv2
session target ipv4:<OUR CUCM PUBLISHER IP>
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
Any help or a point in the right direction would be greatly appreciated.
Cheers,
BrettI ended up resolving this issue as follows:
In CUCM, under Device > Device Settings > SIP Profile.
I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
Now, I get the expected delay before hearing ringback.
Solved! -
Best Practice to Integrate CER with RedSky E911 Anywhere via SIP Trunk
We are trying to integrate CER 9 with RedSky for V911 using a SIP trunk and need assistance with best practice and configuration. There is very little documentation regarding "best practice" for routing these calls to RedSky. This trunk will be handling the majority of our geographically dispersed company's 911 calls.
My question is: should we use an IPsec tunnel for this? The only reference I found was this: http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/virtual-office/deployment_guide_c07-636876.htmlm which recommends an IPsec tunnel for the SIP trunk to Intrado. I would think there are issues with an unsecure SIP trunk for 911 calls. Looking for advice or specifics on how to configure this. Does the SIP trunk require a CUBE or is a CUBE only required for the IPsec tunnel?
Any insight is appreciated.
Thank you.you can use Session Trace in RTMT to check who is disconnecting the call and why.
-
SIP trunk incoming and outgoing calls issue
Hi Everyone,
We recently installad a SIP trunk and terminated on CUBE and CUCM but we have issues on incoming and outgoing calls, When someone dial in from outside he keeps listening the dailing ring even after we pick up the phone and at the end the callers time exipres and call gets disconnected.
For Dailing out, the dialed number rings and caller hear the dailing ring as well but if someone pick the phone it apprears that call is connected but no audio in it, dead air.
Our call flow is as
IP Phones => CUCM --->SIPTRUNK--->CUBE=>SIPTRUNK=>SP
I have attached the config for CUBE and debug ccsip messages output for both incoming and outgoing calls.
Please if some help in sorting out this issue, Thanks in Advance
TasneemInbound call>>>>
The reason you are experiencing this is that your CUBE is requesting PRACK and your provider is not responding to it..
Here we have your cube sending 180 ringing with "Require 100rel"..This was sent several times and your ITSP didnt respond probably because they do not support 100rel...(It is Huwaei after all, they do what they like)
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Date: Thu, 16 Jan 2014 13:31:34 GMT
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Require: 100rel
RSeq: 2507
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "TEST STC" ;party=called;screen=yes;privacy=off
Contact:
Record-Route:
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Length: 0
AFter the CUBE didnt get any response, it then replied with Gateway Timeout...
Jan 16 13:31:54.550: //31880/5A4406E48184/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 504 Gateway Timeout
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Reason: Q.850;cause=102
Content-Length: 0
I suggest you disable this parameter..and test again
voice service voip
sip
rel1xx disable
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay" -
Confused by basic SIP Trunk configuration.
I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing. I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
If someone can make some sense of this, I'd truly appreciate it!Hello Aditya and thanks for the consideration!
I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production.
I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
==============================================
Branch_SIP#show run
Building configuration...
Current configuration : 3529 bytes
! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Branch_SIP
boot-start-marker
boot-end-marker
! card type command needed for slot/vwic-slot 0/2
enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 10.0.10.1 10.0.10.10
ip dhcp excluded-address 10.0.30.1 10.0.30.10
ip dhcp pool Data
network 10.0.10.0 255.255.255.0
default-router 10.0.10.254
option 150 ip 192.168.5.250
dns-server 192.168.5.200
ip dhcp pool Voice
network 10.0.30.0 255.255.255.0
default-router 10.0.30.254
dns-server 192.168.5.200
option 150 ip 172.16.0.1
ip dhcp pool data
option 150 ip 172.16.0.2
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
bind media source-interface Loopback1
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2851 sn FTX1031A2FM
redundancy
interface Loopback1
ip address 2.2.2.2 255.255.255.255
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 10.0.10.254 255.255.255.0
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address 10.0.30.254 255.255.255.0
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/3/0
no ip address
shutdown
clock rate 2000000
interface Serial0/3/1
ip address 172.16.0.1 255.255.255.0
clock rate 250000
interface Internal-Service-Module0/0
no ip address
shutdown
!Application: CUE Running on AIM2
hold-queue 512 out
router eigrp 1
network 0.0.0.0
network 2.2.2.2 0.0.0.0
network 10.0.0.0
network 10.0.10.0 0.0.0.255
network 10.0.30.0 0.0.0.255
network 172.16.0.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 172.16.0.2
tftp-server flash:term45.default.loads
tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
tftp-server flash:cnu45.8-5-3TH1-6.sbn
tftp-server flash:apps45.8-5-3TH1-6.sbn
tftp-server flash:dsp45.8-5-3TH1-6.sbn
tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
control-plane
voice-port 0/0/0
voice-port 0/0/1
mgcp profile default
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
codec g711ulaw
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
sip-server ipv4:192.168.5.250
telephony-service
codec g711ulaw
max-ephones 24
max-dn 48
ip source-address 172.16.0.1 port 2000
system message SIP Branch Site
cnf-file location flash:
load 7960-7940 P00308010200.bin
max-conferences 8 gain -6
transfer-system full-consult
ephone-dn 1
number 4008
ephone-dn 2
number 4005
ephone 1
device-security-mode none
mac-address 001D.A21A.2065
button 1:1
line con 0
exec-timeout 0 0
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
speed 115200
line vty 0 4
password cisco
login
transport input all
line vty 5 15
password cisco
login
transport input all
scheduler allocate 20000 1000
end
Branch_SIP#show debug
TFTP:
TFTP Event debugging is on
CCSIP SPI: SIP Call Statistics tracing is enabled (filter is OFF)
Branch_SIP#
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4B6C5C28
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 4008
Called Number : 5005
Source IP Address (Sig ): 172.16.0.1
Destn SIP Req Addr:Port : 192.168.5.250:5060
Destn SIP Resp Addr:Port : 192.168.5.250:5060
Destination Name : 192.168.5.250
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 2.2.2.2
Source IP Port (Media): 19472
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 63
Disconnect Cause (SIP) : 503
Branch_SIP# -
Hi,
I have a requirement to move from a h323 environment to a SIP environment. I am looking for best practises especially around security. I have 2 CUCM servers (8.5) located in separate cities for redundancy. I also have 2 voice gateways which at the moment are h323 to the PSTN, each located at different cities.
My requirements are:
1. Creat a sip trunk to the provider instaed of using PRI.
2. If the Wan link fails on one gateway to provider, the alternate router in the other location should be able to receive the setup messages and if a user logs on via extension mobility, should be able to answer the call.
Are there any simplified design docos about for this? I am hesitant to create a SIP trunk straight to the provider for security, so thinking of terminating the call on the voice routers with CUBE. I'm pretty sure this is run of the mill and would appreciate some input.
Cheers!
Pieter+5 to Chris..Always use CUBE...
Here are more ideas..
1. Create two sip trunks, first one to Cube 1, second to CUBE 2
on your sip trunk set DTMF as no preference
2. Assign the trunks to CUCM group with your two servers in it
3. Configure route groups with Circular algorithm distribution (this way you have load balancing on your two cubes)
4. Configure dial-peers on your CUBE gateway to point in preferential order to your cucm servers
5. Use dtmf-relay rtp-nte on your dial-peer (ensure you have sip as the protocol on your dial-peers)
6. configure your codec selection properly on your dial-peers
7. configure your region settings properly between your sip trunk and phones.
8. Evaluate if you need xcoders, provide one if you do and ensure you set your region correctly between your xcoder and CUBE
9. Is there voicemail involved? Ensure you set your region settings between cube and voicemail correctly, otherwise calls to voicemail may invoke xcoder (from experience)
10. Provision adequate bandwidth for your calls. Once you move to sip, you loose the luxury of e1 channels. You are solely relying on Bandwidth. Ensure you have adequate bandwidth for your concurrent calls
11. Provision QoS for your calls
12. Is there fax involved? You need to think carefully on this one. What fax method do you want to use. T.38/pass through. Does your provider support T.38?
13. Have a thorough test plan. Test call transfers, call on hold, call forward etc
Just a few pointers..Careful planning and implementation is required for a successful SIP implementation -
LYNC 2010 with Multiple Sites and Pools
We have LYNC 2010 Enterprise Pool in North America with 5 Front End servers, 2 Edge Servers. No Director servers deployed. We want to enable Dial-In Conferencing feature for all users across the Globe (to avoid 3<sup>rd</sup> party conferencing
cost…) and proposing 2 additional pools ( one in EMEA and another in APAC regions). I am looking for best practices to deploy multiple pools on multiple central sites.
Do we need to deploy Director Servers? (Microsoft recommend Director, however is this mandatory for multiple pools?)
How can we handle Simple URLs and DNS entries in multiple Edge pools (edge pool in each frontend pool)?
Do we need to publish different ISA firewall rules for simple URLs and mobility?
As we are not planning to replace PBX now, and only looking to provide dial-in Conferencing Bridge with SIP trunking, what is the best practice to provide site resilience in case one central site is down?
Thanks in advance for your help…
Tek-NerdHello.
1. This is a recommendation and it's not mandatory even with multiples pools. The only point if you don't deploy director is that for the first connection, all sig wil go through one pool (the one behind the srv and dns record) and after this pool redirect
the user to the good one (and also do the authentification) .Also director is made to avoid this type of load on one pool (connection from other pool and from edge authentification), but still not mandatory.
2.Simple URL point only on one entry (or pool of RP) somewhere in the world (after there are some unofficial workaround) but don't forget that behind that there is the external web service URL (for each pool) and also an edge association to one pool. And
the user is really using thoses URL and svc (edge associated to the pool as the ext web url)
For the external web svc & edge association, you can create a local edge pool closer to your internal pool so for those user the media will be closer and don't go trhough the world. (exemple you add one emea pool, the media that come from internet goes
through the emea edge and emea RP handling Lync external web services). But the simple urls still point to your main edge/RP pool.
3. Like point before, not mandatory, recommendation if you don't want your media going around the world and coming back. Be carefull the association is still fixed (depending of the user's pool parameter), so if the user travel still using original edge/rp
settings.
4. You can create multiple trunk with lync so have backup in case of one is no more working. After this is hard to say what is the best, this is depending of your infrastructure.
David
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