Unusual Issue with Dial-peers
To Start it off, I am running CUCM 8.6.2 The head router is a 2951. All my branch locations are 2901s running 15.2 with an FXO and FXS card. On all the 2901s I have dial peers set as follows
voice-port 0/1/0
trunk-group SRST-OUT
timing hookflash-out 50
connection plar opx 4042
caller-id enable
voice-port 0/1/1
trunk-group SRST-OUT
timing hookflash-out 50
connection plar opx 4042
caller-id enable
voice-port 0/1/2
trunk-group SRST-OUT
timing hookflash-out 50
connection plar opx 4042
caller-id enable
voice-port 0/1/3
trunk-group SRST-OUT
no echo-cancel enable
timing hookflash-out 50
dial-peer voice 1 pots
trunkgroup SRST-OUT
destination-pattern 9[2-9].........
dial-peer voice 999022 pots
service mgcpapp
port 0/2/2
dial-peer voice 2 pots
trunkgroup SRST-OUT
destination-pattern 911
dial-peer voice 3 pots
trunkgroup SRST-OUT
destination-pattern 9911
These rules should make it so that when a user is trying to get an outside line, that it will use the pots lines for external calls and 911 calls. it works at every other location except for the new one I am trying to open up. No matter what number is dialed, it is routed over my MPLS and back to my call manager in another city and going out the pri there rather than the POTS lines local to where I am. I can't seem to figure out why this particular one is misbehaving. I thought it might be a bad config, so I wiped it and re-programmed it all from scratch rather than a reload. I would appreciate any assistance.
Thanks
David
Hi David.
I guess you are using Route Groups and Route Lists for external dialplan, please check if callmanager is selecting the right gateway/ route list to route outgoing calls for the new site.
If you are using local route groups, remember to create a separate device pool for the new site associating the correct route group as local route group.
HTH
Regards
Carlo
Sent from Cisco Technical Support iPhone App
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hairpin_block:
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See the following URL. http://voiceonbits.com/2010/06/29/moh-issues-and-resolution/
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Inc. Multicast on IP Address
Inc. Multicast on Port Number
Audio Stream
Codec
Dst. IP Address
Dst. Port
Dst. IP Address
Dst. Port
1
G.711 ulaw
239.1.1.1
16384
239.1.1.1
16384
1
G.711 Alaw
239.1.1.2
16384
239.1.1.1
16386
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G.729
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16384
239.1.1.1
16388
1
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239.1.1.4
16384
239.1.1.1
16390
2
G.711 ulaw
239.1.1.5
16384
239.1.1.1
16392
2
G.711 Alaw
239.1.1.6
16384
239.1.1.1
16394
2
G.729
239.1.1.7
16384
239.1.1.1
16396
2
Wideband
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member 911
member fac-int
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member user-fac
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trunkgroup ALL_FXO
corlist outgoing call-911
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dial-peer voice 901 pots
trunkgroup ALL_FXO
corlist outgoing call-911
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dial-peer voice 9051 pots
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forward-digits 10
dial-peer voice 19 pots
trunkgroup ALL_FXO
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service clid_authen_collect
destination-pattern 900T
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dtmf-relay h245-alphanumeric
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ephone-dn 1
number 4121
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ephone 1
lpcor type local
lpcor incoming endusers
mac-address E0D1.730A.21DE
ephone-template 2
type 7942
button 1:1
voice register dn 33
number 4163
call-forward b2bua busy 5000
call-forward b2bua noan 5000 timeout 20
call-forward b2bua unregistered 5000
allow watch
name Joseph
mwi
voice register pool 33
busy-trigger-per-button 4
id mac BC67.1C31.C8AA
type 7821
number 1 dn 33
cor incoming fac-int 1 4163
dtmf-relay rtp-nte
codec g711ulaw
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DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
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Apr 12 16:22:39.825: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:22:35.609,cgn:4130,cdn:,frs:0,fid:2599,fcid:F692C420E06611E4BB0CE7FDC5486EA5,legID:284C,bguid:F692C420E06611E4BB0CE7FDC5486EA5mon
DAMAC-CME-ANOUD#terminal imon
^
% Invalid input detected at '^' marker.
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
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lpcor endusers
Apr 12 16:22:44.089: //-1/xxxxxxxxxxxx/LPCOR/lpcor_get_index_by_name:
lpcor endusers index 10
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
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debug voip lpcor error call informational is ON (filter is OFF)
debug voip lpcor error software is ON
debug voip lpcor error software informational is ON
debug voip lpcor detail is ON (filter is OFF)
debug voip lpcor function is ON (filter is OFF)
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DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
DAMAC-CME-ANOUD#
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lpcor endusers index 10
DAMAC-CME-ANOUD#
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DAMAC-CME-ANOUD#un all
All possible debugging has been turned off
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DAMAC-CME-ANOUD#!ok just send me these logs
DAMAC-CME-ANOUD#!i have to move from here
Apr 12 16:24:02.153: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 14343755E06711E4BB16E7FDC5486EA5, SetupTime 16:23:25.323 UTC Sun Apr 12 2015, PeerAddress 4130, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:43.393 UTC Sun Apr 12 2015, DisconnectTime 16:24:02.153 UTC Sun Apr 12 2015, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 930, ReceiveBytes 148800
Apr 12 16:24:02.153: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:Tnow
DAMAC-CME-ANOUD#\WC,ft:04/12/2015 16:23:25.321,cgn:4130,cdn:,frs:0,fid:2602,fcid:14343755E06711E4BB16E7FDC5486EA5,legID:2850,bguid:14343755E06711E4BB16E7FDC5486EA5
Apr 12 16:24:02.169: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 14343755E06711E4BB16E7FDC5486EA5, SetupTime 16:23:39.169 UTC Sun Apr 12 2015, PeerAddress 0097150107659, PeerSubAddress , DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime 16:23:43.389 UTC Sun Apr 12 2015, DisconnectTime 16:24:02.169 UTC Sun Apr 12 2015, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 930, TransmitBytes 156240, ReceivePackets 937, ReceiveBytes 149920
Apr 12 16:24:02.169: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:04/12/2015 16:23:39.169,cgn:4130,cdn:0097150107659,frs:0,fid:2603,fcid:14343755E06711E4BB16E7FDC5486EA5,legID:2851,bguid:14343755E06711E4BB16E7FDC5486EA5We have come across this issue today in 10.9.5 (so affects 10.9.4 as well) but it was occurring in Sydney as well with a client and for me in Melbourne.
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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!
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