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,
Seth
Hi 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
Similar Messages
-
Hi all.
i would like to know if it is possible to configure a dual sip trunk on a uc500, its like a dual redundant link to a sip server.
best regartsThen it would be better to ask on forum related to Cisco technology in question. You are off-topic here and your's chances to receive valuable reply here are rather low.
Please delete this threat here (big red [ Delete ] button) and recreate it in a more appropriate forum.
Thank you. -
AppFabric 1.1 with Windows 7 losing or abandone sessions and ignore the timeout settings
Hi,
We have a enterprise financial web solution using aspx and AppFabric 1.1 and we have an issue with Sessions as the get abandoned or appFabric is ignoring the timeout settings.
I tried almost everything: use sql and xml configuration, deactivate firewalls, ensure all sessions are set to 20 minutes expirations. We have the appFabric settings on the web.config and our project is multilayer but sessions are mainly
created/consumed at the web project level..
We use AppFabric 1.1 with Sql Server 2008R2 over a Windows7 x64 environment.
I also tried all the options using powerShell command: get-command -module DistributedCacheAdministration, I increased the TTL to 38 mins but still nothing makes any difference. Sessions are lost between ~~ 5 to 10 minutes no matter if
you are playing with the solution or you left aside..
Please help. I really appreciate any suggestion, idea, recommendation..
Hernan Bogantes
Florida.Note the provider type in AF 1.1 is
type="Microsoft.Web.DistributedCache.DistributedCacheSessionStateStoreProvider,Microsoft.Web.DistributedCache"
which is different from the type in 1.0:
type="Microsoft.ApplicationServer.Caching.DataCacheSessionStoreProvider"
https://msdn.microsoft.com/en-us/library/hh361709(v=azure.10).aspx -
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 -
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" -
SIP phone registering on SIP trunk
Hi,
i have a UC 500 connected to our phone provider using a SIP trunk.
All the phones are SPA508 G
All is working fine !
Then, some days ago i added a SIP phone (extention 350) on the UC500, that also worked fine, and then after some minutes all our incoming/outgoing calls were blocked.
I called my provider that told me that our IP was banned because they have seen to much registration attempt from a bad user that was "350"
I can confirm with a "sh sip-ua register status" command that i had two sip registration : my SIP trunk and the SIP phone
Then it seems that the UC 500 is trying to register the SIP phone on the SIP trunk ?
What am i doing wrong ?
Is there a command to avoid that ?
Bellow is how the SIP phone and the SIP trunk are configured
Many thanks for your help, i was unable to find anything about that, but i guess somebody already had this problem !
The SIP phone -------------------------------------------------------------------------
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
fax protocol none
modem passthrough nse codec g711ulaw
sip
registrar server expires max 3600 min 120
no update-callerid
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice register global
mode cme
source-address 10.1.1.1 port 5060
max-dn 20
max-pool 20
load 9971 sip9971.9-2-2
load 9951 sip9951.9-2-2
load 8961 sip8961.9-2-2
load 7971 term71.default
authenticate register
authenticate realm xxxxxx.com
timezone 13
hold-alert
mwi stutter
mwi reg-e164
create profile sync 0636240803635305
voice register dn 1
number 350
name Conference
label Conference
voice register pool 1
id mac 1234.1234.1234
number 1 dn 1
username 350 password 1234
codec g711ulaw
The SIP trunk ----------------------------------------------------------------------
sip-ua
credentials username user1234 password 1234 realm sipgw9.provider.com
authentication username user1234 password 1234 no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sipgw9.provider.com expires 3600
sip-server dns:sipgw9.provider.comI'm still searching on the forum, and maybe i found somthing related to my problem, not sure... any advice ?
Disable outbound proxy on voice register global as by default it will use the outbound proxy configured on the system which would not make sense
voice register global
no outbound-proxy
found there : https://supportforums.cisco.com/discussion/10760741/uc500-sip-server-and-sip-trunk -
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# -
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! -
SIP Inspection and dynamic port opening after re-invite
Platform: ASA 8.3(2)
Hello,
I have SIP devices along with SipTrunk and media endpoints. I am having issues with the ASA not dynamically opening (sip inspect enabled) UDP ports for RTP after a SIP re-invite causes the media endpoints to change within SDP.
The problem as below.
Initial SIP invite setups properly with ports dynamically opened between the media endpoints in the ASA
Re-inivite from the SIP device causes the media endpoints to change within the SDP
ASA blocks ports associated to the new media endpoints
I can resolve this by allowing the ports in the ACL, but suprised this is not working as re-invites to change media endpoints is to be expected in SIP conversation.
Regards,
AJBelow is the script you can use to reproduce this. Points worth mentioning.
Initial invite sets up the media between SIP Trunk and a media device ( 10.1.2.150) in the inside network, SIP signalling will be with 10.1.2.100. At this poit RTP flows freely between the SIP Trunk and the media device.
If the call is fax, a re-invite will occur and this will cause the IP address to change in the SDP. The new media endpoint becomes 10.1.2.151 (This device is SIP and Media (T38) capable).
For every SIP call we establish 10.1.2.150 will be used for media, we do not want to change this behaviour.
ASA 8.3 (2)
conf t
interface Ethernet0/0
nameif Inside_Voice
security-level 100
ip address 10.1.2.11 255.255.255.0 standby 10.1.2.12
exit
interface Ethernet0/1
nameif Outside_SIP_Trunk
security-level 0
ip address 10.1.60.254 255.255.255.0 standby 10.1.60.253
exit
object-group network SIP_trunks
network-object 1.2.3.0 255.255.255.0
exit
object-group service SIP_service
service-object tcp destination eq sip
service-object udp destination eq sip
exit
object-group network SIP_inside_servers
network-object host 10.1.2.100
exit
access-list Outside_SIP_in extended permit object-group SIP_service object-group SIP_trunks object-group SIP_inside_servers
access-group Outside_SIP_in in interface Outside_SIP_Trunk
route Outside_SIP_Trunk 0.0.0.0 0.0.0.0 10.1.60.1
class-map inspection_default
match default-inspection-traffic
exit
policy-map global_policy
class inspection_default
inspect dns preset_dns_map
inspect ftp
inspect h323 h225
inspect h323 ras
inspect ip-options
inspect netbios
inspect rsh
inspect rtsp
inspect skinny
inspect esmtp
inspect sqlnet
inspect sunrpc
inspect tftp
inspect sip
inspect xdmcp
inspect icmp
inspect icmp error
lass class-default
set connection decrement-ttl
exit
service-policy global_policy global
end -
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 -
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.
-
Unable to perform call transfer & call park through SIP Trunk (SKYPE)
The Scenario is:
I have set up a SIP trunk to SKYPE and we are able to make outbound call to a number via SIP Trunk.
After the call is established, when we tried to make call transfer, the call DROP and the phone at the other end shows error "Temp Fail".
I tried to "enable MTP" in SIP Trunk and We are able to perform call-transfer but it limits the call session to 1.
Anyone has facing the same issue?MTP is needed to invoke supplementary functions like hold, transfer etc. Make sure that the MTP is checked on SIP trunk, MTP is assigned to the MRGL of the device pool on SIP trunk and has sufficient resources.
HTH
Manish -
Hi!
I'm trying to register a SIP Trunk to a SIP server. The trunk registration is done, but not keep alive. The trunk register with SIP server when an outgoing call starts, but when this call ends, the SIP trunk closes the connection with SIP server. Then, the
outgoing calls work OK, but the incoming calls doesn't work because the SIP Trunk is unregistered while no active outgoing calls.
Then, can i keep alive the SIP Trunk registration with SIP Server?
Thanks!!!You need to talk to the SIP provider and get them enable OPTIONS on the SIP Trunk and enable OPTIONS on the PSTN Gateway. Check the registration interval of the SIP trunk on the Gateway and try increasing it to a higher value.
http://thamaraw.com -
Telco Messages via PRI (VG) connected to CUCM 9.X via SIP Trunk??
Hello
Questions:
Should I hear tel-co message on an IP Phone if a call that is meant to be long distance is sent to the gateway without a one (1). Currently these calls simply ring until disconnect, but work properly if the user dials a 1, the user expects to get a message from the provider and I am wondering if the SIP trunk between GW and CUCM is not allowing it?
Scenario:
IP Phone --> CUCM (SIP Trunk) --> ISR 2901(PRI) --> PSTN
In CUCM we have a local pattern 9.[2-9]XX[2-9]XXXXXX
If the IP Phone dials a number that is actually a long distance number, which would require a 1, but they dial it as a 10 digit local number, the call gets to the gateway and IP Phone hears ringing but it, rings until it eventually disconnects. At this point I believe the gateway is sending a cause code back to CUCM and I would expect error message back from the telco informing the caller of the issue, such as "Please dial 1 before a long distance number". Is there a specific setting on the SIP trunk and or Gateway to achieve this?
See Q931 Debug
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838F
Exclusive, Channel 15
Display i = 0xB1, 'London Hydro'
Calling Party Number
UCS5-GW-02# i = 0x2181, '5193334444'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '5191112222'
Plan:ISDN, Type:National
*Jul 23 15:33:28.363: ISDN Se0/0/1:23 Q931: RX <- STATUS pd = 8 callref = 0xC3FA
Cause i = 0x80AB28 - Access information discarded
Call State i = 0x01
*Jul 23 15:33:28.411: ISDN Se0/0/1:23 Q931: RX <- CALL_PROC pd = 8 callref = 0xC3FA
Channel ID i = 0xA9838F
Exclusive, Channel 15
*Jul 23 15:33:28.415: ISDN Se0/0/1:23 Q931: RX <- PROGRESS pd = 8 callref = 0xC3FA
Cause i = 0x80FF - Interworking error; unspecified
Progress Ind i = 0x8088 - In-band info or appropriate now available
Thanks
RichardYes, Early media needs to be turned on the SIP Profile that is associated to the SIP trunk. The setting is SIP Rel 1xx options. It needs to be set for Send PRACK for all 1XX messages. If you have CUCM 7.x , this setting is a service parameter.
-
CUCM 10.5.1 and Exchange 2010 Unified Messaging (UM) SIP Trunk Problem
This is more a comment if you're migrating from a lower version to 10.x. Hopefully Google will pickup this post so others don't spend too much time (I got lucky and found this in about 30 minutes).
There are many more SIP options than in the past. If you configured your integration as per the integration doc, all settings are relevant, however there are some new defaults that need to change.
SYMPTOM: Dialing another number or AA pilot and being redirected internally works, but the call drops on calls from external phones. Exchange logs an Event ID 1079 from UMCore and also informational 1084 and 1172 events. A capture yields a status 200 OK, then an ACK, two BYEs and a status 481 that the call leg does not exist. The call is then dropped.
RESOLUTION:
In the SIP Trunk, modifiy the following:
Device Information. . . Check Media Termination Point Required. I've had some that have needed this checked and some where I needed to leave unchecked. In this case going from 7.1 to 10.5 necessitated enabling.
Call Routing Information. . . SIP Privacy: Need to change from Default to None.
Any comments how the SIP Privacy might affect security and functionality would be appreciated.Just to inform everyone. I rebuilt the edge server from scratch.
Now everything is working as expected.
I cannot work out how the edge was not passing on calls to exchange or not communicating correctly with exchange.
Anyway it is resolved now.
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