VoIP issue with 881

Hi,
I have a strange issue with outgoing calles when connected to cisco 881.
I got an IPphone (cisco SPA502G) which is registered to our SipServer (Public IP Server). in the past we had the 871 unit and everything was OK.
since we replaced it with the new 881, I cannot make outgoing calles only incoming works.
the configuration is the same and very simple with no special NAT for the IP Phone:
interface FastEthernet4
description ### WAN ###
ip address X.X.X.X 255.255.255.248
ip nat outside
interface Vlan1
ip address 192.168.10.1 255.255.255.0
ip nat inside
ip route 0.0.0.0 0.0.0.0 X.X.X.X
ip nat inside source list 1 interface FastEthernet4 overload
access-list 1 remark LAN NAT
access-list 1 permit 192.168.10.0 0.0.0.255
Is there something specific which is needed for the VoIP on the new 800 serias ?
Thanx,
   Arik

Hi,
I have a strange issue with outgoing calles when connected to cisco 881.
I got an IPphone (cisco SPA502G) which is registered to our SipServer (Public IP Server). in the past we had the 871 unit and everything was OK.
since we replaced it with the new 881, I cannot make outgoing calles only incoming works.
the configuration is the same and very simple with no special NAT for the IP Phone:
interface FastEthernet4
description ### WAN ###
ip address X.X.X.X 255.255.255.248
ip nat outside
interface Vlan1
ip address 192.168.10.1 255.255.255.0
ip nat inside
ip route 0.0.0.0 0.0.0.0 X.X.X.X
ip nat inside source list 1 interface FastEthernet4 overload
access-list 1 remark LAN NAT
access-list 1 permit 192.168.10.0 0.0.0.255
Is there something specific which is needed for the VoIP on the new 800 serias ?
Thanx,
   Arik

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    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:14343755E06711E4BB16E7FDC5486EA5

    We 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.

  • 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,
    Brett

    I 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!

  • Calling issue with Cisco 7937 conference station

    Hi Friends,
    I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
    When making calls from once remote site to another using Cisco 6921 phones calls working fine
    When making calls from once remote site to another using Cisco 7937 conference station to make call  any phone at remote office, calls are getting disconneted, remote phone rings when calls,  but its gets fast busy tone when other party picks up the phone and  not able to talk.
    I suspect the issue with Codec but we have configured transcoders  in VG and registered with CUCM
    Please help me if any one experience such issue earlier.
    Regards
    Siva

    hi Basant,
    1. Actually tow phones A and B are registerd with centralized CUCM, A and B are located in two different locations, RTP traffic between And B pass through service provider. 
    Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
    Show Run
    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
    sh run
    Building configuration...
    Current configuration : 12139 bytes
    ! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
    ! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
    ! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
    version 15.1
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    hostname eucamvgw01
    boot-start-marker
    boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
    boot-end-marker
    card type e1 0 0
    logging buffered 51200 warnings
    no logging console
    no aaa new-model
    no network-clock-participate wic 0
    no ipv6 cef
    ip source-route
    ip traffic-export profile cuecapture mode capture
    bidirectional
    ip cef
    ip multicast-routing
    ip domain name drreddys.eu
    ip name-server 10.197.20.1
    ip name-server 10.197.20.2
    multilink bundle-name authenticated
    stcapp ccm-group 2
    stcapp
    stcapp feature access-code
    stcapp feature speed-dial
    stcapp supplementary-services
    port 0/1/0
    fallback-dn 5428025
    port 0/1/1
    fallback-dn 5428008
    port 0/1/2
    fallback-dn 5421462
    port 0/1/3
    fallback-dn 5421463
    isdn switch-type primary-net5
    crypto pki token default removal timeout 0
    voice-card 0
    dsp services dspfarm
    voice call send-alert
    voice call disc-pi-off
    voice call convert-discpi-to-prog
    voice rtp send-recv
    voice service voip
    ip address trusted list
    ipv4 10.198.0.0 255.255.255.0
    ipv4 152.63.1.0 255.255.255.0
    address-hiding
    allow-connections sip to sip
    no supplementary-service h225-notify cid-update
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
    fax-relay ans-disable
    sip
    rel1xx supported "track"
    privacy pstn
    no update-callerid
    early-offer forced
    call-route p-called-party-id
    voice class uri 100 sip
    host 41.206.187.71
    voice class codec 10
    codec preference 1 g711alaw
    codec preference 2 g711ulaw
    codec preference 3 ilbc
    codec preference 4 g729r8
    codec preference 5 g729br8
    voice class codec 20
    codec preference 1 g729br8
    codec preference 2 g729r8
    voice moh-group 1
    moh flash:moh/Panjo.alaw.wav
    description MOH G711 alaw
    multicast moh 239.1.1.2 port 16384 route 10.198.2.9
    voice translation-rule 1
    rule 1 /^012237280\(..\)/ /54280\1/
    rule 2 /^012236514\(..\)/ /54214\1/
    rule 3 /^01223651081/ /5428010/
    rule 4 /^01223506701/ /5428010/
    voice translation-rule 2
    rule 1 /^00\(.+\)/ /+\1/
    rule 2 /^0\(.+\)/ /+44\1/
    rule 3 /^\([0-9].+\)/ /+\1/
    voice translation-rule 3
    rule 1 /^9\(.+\)/ /\1/
    rule 2 /^\+44\(.+\)/ /0\1/
    rule 3 /^\+\(.+\)/ /00\1/
    voice translation-rule 4
    rule 1 /^54280\(..\)/ /12237280\1/
    rule 2 /^54214\(..\)/ /12236514\1/
    rule 3 /^\+44\(.+\)/ /\1/
    rule 4 /^.54280\(..\)/ /12237280\1/
    rule 5 /^.54214\(..\)/ /12236514\1/
    voice translation-rule 9
    rule 1 /^\(....\)/ /542\1/
    voice translation-rule 10
    voice translation-rule 11
    rule 1 /^\+44122372\(....\)/ /542\1/
    rule 2 /^\+44122365\(....\)/ /542\1/
    voice translation-rule 12
    voice translation-rule 13
    rule 1 /^\([18]...\)/ /542\1/
    voice translation-rule 14
    voice translation-profile MPLS-incoming
    translate calling 10
    translate called 9
    voice translation-profile MPLS-outgoing
    translate calling 11
    translate called 12
    voice translation-profile PSTN-incoming
    translate calling 2
    translate called 1
    voice translation-profile PSTN-outgoing
    translate calling 4
    translate called 3
    voice translation-profile SRST-incoming
    translate calling 14
    translate called 13
    license udi pid CISCO2921/K9 sn FGL145110RE
    hw-module ism 0
    hw-module pvdm 0/0
    username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
    redundancy
    controller E1 0/0/0
    ip tcp path-mtu-discovery
    ip scp server enable
    interface Embedded-Service-Engine0/0
    no ip address
    shutdown
    interface GigabitEthernet0/0
    description internal LAN
    ip address 10.198.2.9 255.255.255.0
    duplex auto
    speed auto
    interface ISM0/0
    ip unnumbered GigabitEthernet0/0
    service-module ip address 10.198.2.8 255.255.255.0
    !Application: CUE Running on ISM
    service-module ip default-gateway 10.198.2.9
    interface GigabitEthernet0/1
    description to TATA NGN
    ip address 115.114.225.122 255.255.255.252
    duplex auto
    speed auto
    interface GigabitEthernet0/2
    description SIP Trunks external
    ip address 79.121.254.83 255.255.255.248
    ip access-group SIP-InBound in
    ip traffic-export apply cuecapture size 8000000
    duplex auto
    speed auto
    interface ISM0/1
    description Internal switch interface connected to Internal Service Module
    no ip address
    shutdown
    interface Vlan1
    no ip address
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 10.198.2.1
    ip route 10.198.2.8 255.255.255.255 ISM0/0
    ip route 41.206.187.0 255.255.255.0 115.114.225.121
    ip route 77.37.25.46 255.255.255.255 79.121.254.81
    ip route 83.245.6.81 255.255.255.255 79.121.254.81
    ip route 83.245.6.82 255.255.255.255 79.121.254.81
    ip route 95.223.1.107 255.255.255.255 79.121.254.81
    ip route 192.54.47.0 255.255.255.0 79.121.254.81
    ip access-list extended SIP-InBound
    permit ip host 77.37.25.46 any
    permit ip host 83.245.6.81 any
    permit ip host 83.245.6.82 any
    permit ip 192.54.47.0 0.0.0.255 any
    permit icmp any any
    permit ip host 95.223.1.107 any
    deny ip any any log
    control-plane
    voice-port 0/1/0
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/1
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/2
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    voice-port 0/1/3
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    no ccm-manager fax protocol cisco
    ccm-manager music-on-hold bind GigabitEthernet0/0
    ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
    ccm-manager sccp local GigabitEthernet0/0
    ccm-manager sccp
    mgcp profile default
    sccp local GigabitEthernet0/0
    sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
    sccp ccm 152.63.1.19 identifier 4 version 7.0
    sccp ccm 152.63.1.100 identifier 5 version 7.0
    sccp ccm 172.27.210.5 identifier 6 version 7.0
    sccp
    sccp ccm group 2
    bind interface GigabitEthernet0/0
    associate ccm 4 priority 1
    associate ccm 5 priority 2
    associate ccm 6 priority 3
    associate ccm 3 priority 4
    associate profile 1002 register CFB_UK_CAM_02
    associate profile 1001 register XCODE_UK_CAM_02
    associate profile 1000 register MTP_UK_CAM_02
    dspfarm profile 1001 transcode
    codec ilbc
    codec g722-64
    codec g729br8
    codec g729r8
    codec gsmamr-nb
    codec pass-through
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    maximum sessions 18
    associate application SCCP
    dspfarm profile 1002 conference
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 2
    associate application SCCP
    dspfarm profile 1000 mtp
    codec g711alaw
    maximum sessions software 200
    associate application SCCP
    dial-peer cor custom
    name SRSTMode
    dial-peer cor list SRST
    member SRSTMode
    dial-peer voice 100 voip
    description *** Inbound CUCM ***
    translation-profile incoming PSTN-incoming
    incoming called-number .
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 500 voip
    description *** Inbound TATA MPLS ***
    translation-profile incoming MPLS-incoming
    session protocol sipv2
    session target sip-server
    incoming called-number ....
    incoming uri from 100
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 510 voip
    description *** Outbound TATA MPLS ***
    translation-profile outgoing MPLS-outgoing
    destination-pattern 54[013-9]....
    session protocol sipv2
    session target ipv4:41.206.187.71
    session transport udp
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 520 voip
    description *** Outbound TATA MPLS ***
    translation-profile outgoing MPLS-outgoing
    destination-pattern 5[0-35-9].....
    session protocol sipv2
    session target ipv4:41.206.187.71
    session transport udp
    voice-class codec 20
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 200 voip
    description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 0122365....
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 201 voip
    description *** Inbound M12 *** 012237280XX
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 012237280..
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 202 voip
    description *** Inbound M12 *** 01223506701
    translation-profile incoming PSTN-incoming
    session protocol sipv2
    session target sip-server
    session transport udp
    incoming called-number 01223506701
    dtmf-relay rtp-nte
    codec g711ulaw
    no vad
    dial-peer voice 210 voip
    description *** Outbound M12 ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern +...T
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 211 voip
    description *** Outbound ISDN for SRST and emergency ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern 9.T
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 212 voip
    description *** Outbound ISDN for emergency ***
    translation-profile outgoing PSTN-outgoing
    destination-pattern 11[02]
    session protocol sipv2
    session target ipv4:83.245.6.81
    session transport udp
    dtmf-relay rtp-nte
    codec g711alaw
    no vad
    dial-peer voice 2000 voip
    description *** Outbound to CUCM Primary ***
    preference 1
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:152.63.1.19
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 2001 voip
    description *** Outbound to CUCM Secondary ***
    preference 2
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:152.63.1.100
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 2002 voip
    description *** Outbound to CUCM Teritiary ***
    preference 3
    destination-pattern 542....
    session protocol sipv2
    session target ipv4:172.27.210.5
    voice-class codec 10
    voice-class sip call-route p-called-party-id
    dtmf-relay rtp-nte
    no vad
    dial-peer voice 999010 pots
    service stcapp
    port 0/1/0
    dial-peer voice 999011 pots
    service stcapp
    port 0/1/1
    dial-peer voice 999012 pots
    service stcapp
    port 0/1/2
    dial-peer voice 999013 pots
    service stcapp
    port 0/1/3
    sip-ua
    no remote-party-id
    gatekeeper
    shutdown
    call-manager-fallback
    secondary-dialtone 9
    max-conferences 4 gain -6
    transfer-system full-consult
    ip source-address 10.198.2.9 port 2000
    max-ephones 110
    max-dn 400 dual-line no-reg
    translation-profile incoming SRST-incoming
    moh flash:/moh/Panjo.ulaw.wav
    multicast moh 239.1.1.1 port 16384 route 10.198.2.9
    time-zone 22
    time-format 24
    date-format dd-mm-yy
    line con 0
    login local
    line aux 0
    line 2
    no activation-character
    no exec
    transport preferred none
    transport input all
    transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
    stopbits 1
    line 131
    no activation-character
    no exec
    transport preferred none
    transport input all
    transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
    stopbits 1
    line vty 0 4
    session-timeout 60
    exec-timeout 60 0
    privilege level 15
    login local
    transport input all
    line vty 5 15
    session-timeout 60
    exec-timeout 60 0
    privilege level 15
    login local
    transport input all
    scheduler allocate 20000 1000
    ntp server 10.1.30.1
    end
    eucamvgw01#
    Sh SCCP
    =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
    SCCP Admin State: UP
    Gateway Local Interface: GigabitEthernet0/0
    IPv4 Address: 10.198.2.9
    Port Number: 2000
    IP Precedence: 5
    User Masked Codec list: None
    Call Manager: 10.198.2.9, Port Number: 2000
    Priority: 3, Version: 7.0, Identifier: 3
    Call Manager: 152.63.1.19, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 4
    Trustpoint: N/A
    Call Manager: 152.63.1.100, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 5
    Trustpoint: N/A
    Call Manager: 172.27.210.5, Port Number: 2000
    Priority: N/A, Version: 7.0, Identifier: 6
    Trustpoint: N/A
    MTP Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1000
    Reported Max Streams: 400, Reported Max OOS Streams: 0
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    TLS : ENABLED
    Transcoding Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1001
    Reported Max Streams: 36, Reported Max OOS Streams: 0
    Supported Codec: ilbc, Maximum Packetization Period: 120
    Supported Codec: g722r64, Maximum Packetization Period: 30
    Supported Codec: g729br8, Maximum Packetization Period: 60
    Supported Codec: g729r8, Maximum Packetization Period: 60
    Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
    Supported Codec: pass-thru, Maximum Packetization Period: N/A
    Supported Codec: g711ulaw, Maximum Packetization Period: 30
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: g729ar8, Maximum Packetization Period: 60
    Supported Codec: g729abr8, Maximum Packetization Period: 60
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    Conferencing Oper State: ACTIVE - Cause Code: NONE
    Active Call Manager: 152.63.1.19, Port Number: 2000
    TCP Link Status: CONNECTED, Profile Identifier: 1002
    Reported Max Streams: 16, Reported Max OOS Streams: 0
    Supported Codec: g711ulaw, Maximum Packetization Period: 30
    Supported Codec: g711alaw, Maximum Packetization Period: 30
    Supported Codec: g729ar8, Maximum Packetization Period: 60
    Supported Codec: g729abr8, Maximum Packetization Period: 60
    Supported Codec: g729r8, Maximum Packetization Period: 60
    Supported Codec: g729br8, Maximum Packetization Period: 60
    Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
    Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
    Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
    TLS : ENABLED
    Alg_Phone Oper State: ACTIVE - Cause Code: NONE
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    Supported Codec: g729r8, Maximum Packetization Period: 220
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    eucamvgw01#

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    BTCare Community Mod
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