Redirect SIP Trunk calls to FXO port
Hi,
This is the scenario. There are 3 branches, two of them are Cisco Call Manager Express and one of them is Elastix-based.
So, as the image explains, the three branches have SIP trunks fully operational. The branches are in different cities, so the numbers structure changes. In city A it begins with 2, in B begins with 3 and in C begins with 4. Every POTS number is a 7 digit number (2XXXXXX, 3XXXXXX, 4XXXXXX). And every user, in every branch, have a 4 digit number beginning with the city code (2XXX, 3XXX, 4XXX).
But, every time city A wants to make a call to a POTS number in city B, it goes across the A´s FXO line. So it charges a inter-city cost to the call.
The client wants that every time a city A user wants to call a POTS number in city B, goes over the SIP trunk to city B and use the FXO on the city B call manager.
I have made a pattern for city A. So, everytime the user dials 3XXXXXX, it does not use the city A´s FXO, but it goes to the branch in city B.
What do I have to do now in branch B´s Call Manager Express to redirect that call to a local FXO?
Thanks in advanced!
Regards
PS. There is a diagram of the topology. Want to do what the red line is doing.
In this situation I would do an answer-address based on ANI so you are specifically identifying your site A and then just piggy back off the local FXO out.
So assuming you are sending just 4 digits over the SIP for each site:
Dial-peer voice X voip
answer-address "blah"
protocol sipv2
...(whatever else you need to configure in these dots)
At this point your CME at site B will take the call see that it is destined for a POTs line and it should send it out whatever local dial-peer you have setup for that site when they dial out to the PSTN locally.
EDIT:
Then again, you probably already have a general incoming dial-peer, the above design would just be specific for your site A and isn't really needed.
Similar Messages
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SIP trunk call to CTI port media endpoint problem
Hi, I have the following scenario:
CUCM ---SIP-----CUBE----SIP----ITSP
CUCM cluster has two servers 10.1.9.5 and 10.1.9.6 (subscriber), we have a CTI application running on 10.1.9.8, we got a bunch of SIP DIDs from ITSP and those SIP DID numbers are mapped to CUCM internal DNs on CUBE through translation, inbound SIP calls to mapped SCCP phones work fine, media stream of internal call leg is established directly between SCCP phone and CUBE as expected. However, when inbound SIP calls to the number which maps to CTI extention, there is no audio, debug on CUBE shows that the media stream of internal call leg is between one of the CUCM servers. I want to understand why CUCM is telling CUBE to use the IP address which is NOT CTI port is registered from (in this case 10.1.9.8)? why would CUCM treat a CTI extention differently from a regular SCCP phone extension. For regular inbound call to SCCP phones codec between CUCM-CUBE and between CUBE-ITSP are both g11ulaw.
Thanks,"debug on CUBE shows that the media stream of internal call leg is between one of the CUCM servers" ==>
debug on CUBE shows that the media stream of internal call leg is between 10.1.9.5 (happens to be MTP) and CUBE -
CME - Sending outbound calls to FXO port
Hi Guys,
Need your help for the below scenario.
Our customer has a CME where 4 FXO ports are already connected and working. Customer has added 2 more FXO port and few IP phones.
The requirement is when ever an outbound call is made from the newly configured IP phones, the call should go through the newly added FXO lines.
For eg ext 3001 , the outbound call should go through port 0/1/0
Already the prefix 9 is used for dialing the number and I guess only one prefix number can be used in CME.
I tried translation rule , cor list but none worked , the call is default going through the old fxo port and not to the new fxo port.
Can you guys help me with the configuration.
Regards
SathyaPrevious post on similar issue might be helpful -
https://supportforums.cisco.com/discussion/11431746/h323-choose-outbound-fxo-port-based-calling-number
Thnx -
I can't make any calls out this FXO port. Any ideas? Is it a telco issue?
IOS is 124-24.T4.bin
.Nov 18 06:12:29: htsp_timer_stop3
.Nov 18 06:12:29: [0/1/1] htsp_stop_caller_id_rx. message length 0htsp_setup_req
.Nov 18 06:12:29: htsp_process_event: [0/1/1, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup
.Nov 18 06:12:29: [0/1/1] set signal state = 0xC timestamp = 0
.Nov 18 06:12:29: htsp_timer - 1300 msec
.Nov 18 06:12:29: [0/1/1] htsp_dsm_close_done
.Nov 18 06:12:30: htsp_process_event: [0/1/1, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_0110]fxols_disc_clear
.Nov 18 06:12:30: htsp_timer_stop2
.Nov 18 06:12:30: htsp_timer - 1300 msec
.Nov 18 06:12:31: htsp_process_event: [0/1/1, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER]fxols_wait_dial_timer htsp_dial
.Nov 18 06:12:33: htsp_process_event: [0/1/1, FXOLS_WAIT_DIAL_DONE, E_DSP_DIALING_DONE]fxols_wait_dial_done htsp_progress
.Nov 18 06:12:33: htsp_timer - 350 msec
.Nov 18 06:12:33: htsp_call_bridged invoked
.Nov 18 06:12:33: htsp_process_event: [0/1/1, FXOLS_WAIT_CUT_THRU, E_HTSP_VOICE_CUT_THROUGH]fxols_handle_cut_thru
.Nov 18 06:12:33: htsp_timer_stop
.Nov 18 06:12:34: htsp_process_event: [0/1/1, FXOLS_OFFHOOK, E_DSP_SUP_DISCONNECT]fxols_outgoing_sup_disc
.Nov 18 06:12:34: htsp_timer - 3000 msec
.Nov 18 06:12:37: htsp_process_event: [0/1/1, FXOLS_OFFHOOK, E_HTSP_EVENT_TIMER]fxols_disc_confirm
.Nov 18 06:12:37: htsp_timer_stop
.Nov 18 06:12:37: htsp_timer_stop2
.Nov 18 06:12:37: htsp_timer_stop3
.Nov 18 06:12:37: htsp_timer_stop3
.Nov 18 06:12:37: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 3FE4209F37B14654000008A43C0E8F, SetupTime .06:12:29.778 GTM Tue Nov 18 2014, PeerAddress 605281980020, PeerSubAddress , DisconnectCause 11 , DisconnectText user busy (17), ConnectTime .06:12:33.158 GTM Tue Nov 18 2014, DisconnectTime .06:12:37.298 GTM Tue Nov 18 2014, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 102, TransmitBytes 4080, ReceivePackets 203, ReceiveBytes 4060
.Nov 18 06:12:37: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:11/18/2014 06:12:29.782,cgn:605281980020,cdn:080006099,frs:0,fid:45246,fcid:3FE4209F37B14654000008A43C0E8F,legID:10EF5,bguid:003FE4209F37B14654000008A43C0E8F
.Nov 18 06:12:37: htsp_process_event: [0/1/1, FXOLS_REMOTE_RELEASE, E_HTSP_RELEASE_REQ]fxols_offhook_release
.Nov 18 06:12:37: htsp_timer_stop
.Nov 18 06:12:37: htsp_timer_stop2
.Nov 18 06:12:37: htsp_timer_stop3
.Nov 18 06:12:37: [0/1/1] set signal state = 0x4 timestamp = 0
.Nov 18 06:12:37: htsp_timer - 2000 msec
.Nov 18 06:12:37: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 3FE4209F37B14654000008A43C0E8F, SetupTime .06:12:29.790 GTM Tue Nov 18 2014, PeerAddress 080006099, PeerSubAddress , DisconnectCause 11 , DisconnectText user busy (17), ConnectTime .06:12:33.160 GTM Tue Nov 18 2014, DisconnectTime .06:12:37.280 GTM Tue Nov 18 2014, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 102, TransmitBytes 4080, ReceivePackets 203, ReceiveBytes 5684
.Nov 18 06:12:37: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:11/18/2014 06:12:29.786,cgn:605281980020,cdn:080006099,frs:0,fid:45247,fcid:3FE4209F37B14654000008A43C0E8F,legID:10EF6,bguid:003FE4209F37B14654000008A43C0E8F
.Nov 18 06:12:37: htsp_process_event: [0/1/1, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
.Nov 18 06:12:39: htsp_process_event: [0/1/1, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
.Nov 18 06:12:39: fxols_dsp_prealloc_clid_wait. Line reversal alerting DSP preallocation done
.Nov 18 06:12:39: [0/1/1] htsp_start_caller_id_rx:BELLCORE
.Nov 18 06:12:39: htsp_start_caller_id_rx create dsp_stream_manager
.Nov 18 06:12:39: [0/1/1] htsp_dsm_create_success returns 1
.Nov 18 06:12:39: htsp_process_event: [0/1/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
voice-port 0/1/1
supervisory disconnect dualtone mid-call
cptone MX
timeouts interdigit 4
timeouts call-disconnect 3
timeouts wait-release 3
caller-id enable
caller-id alerting dsp-pre-allocateHi Jason,
Have you tried disabling the command "supervisory disconnect dualtone mid-call"?
Did you have DSP installed?
Regards -
Hi all,
i have implement a Sip trunk with Alcatel PABX, and it is working perfectly. The Voice traffic flow as below:
IP Phone(HK)----Alcatel Omnipcx PBX(HK)<---->IP WAN(MPLS)<----->CUCM7.1 cluster(MY)----29xxGW------PSTN(Telco PSTN)-----External/Mobile phone.
the sip trunk is build from CM to Alcatel. The situation are as:
1. Call from local extension to HK local extension is working
2. Call from local extension to HK outside number (eg: mobile, fixed line) via mpls are working fine
3. Call from HK local extension to MY outside number (eg: mobile, fixed line) via mpls is NOT working.
Example:
HK (ext no: 2000) dial access code + destination mobile number (8859 + 0123456789), 8859 is to tell the call to route thru sip trunk to MY.
I have run the real time monitor in CM sdl trace log, i can see the number already arrive my CM (which is 0123456789), and i have the route patten match the number, but then i cannot see the CM passing the number to the PSTN gateway.
anyone can advice?
ThanksI have logged the case with CIsco and provide me a command which disables all the UIPDATE messages to initiate from VG side to Soft switch side and ask Soft switch guys to change their Session-Expires to 3600 from 1800. Now issue seems solved.
here is a command " no no update-callerid" configured in voice service voip in SIP.
voice service voip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
min-se 90
no update-callerid -
UC520 Resets every time when making a SIP trunk Call.
I have a UC520 and i just upgraded to the 8.0.2 software pack. I also added SIP for Skype SIP trunk - when i make a call the router resets. When i do a show ver is see this line - System returned to ROM by error - a SegV exception.
never heard of that...thats not cool...open a case asap.
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SIP Trunk Question - Outbound Calls Fail
Hi Folks,
I am using a Cisco 2821 as a router that will convert a SIP trunk to an E1 PRI. Si my setup is:
SIp-Trunk > 2821 Router > E1 port on 3900> CUCM
Inbound calls are working fine, but outbound are failing. I am starting to think its due to transcoding issue on the SIP-GW maybe (there is nothing configured on it for XCODE etc).
I think my configuration is fine as I am able to recieve calls inbound. Just outbound fail.
Here are the debugs from the SIP-GW:
"Debug CCSIP calls"
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4BB4F194
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 1528xxxx
Called Number : 909
Source IP Address (Sig ): 172.29.x.xxx
Destn SIP Req Addr:Port : 10.200.7.157:5060
Destn SIP Resp Addr:Port : 10.200.7.157:5060
Destination Name : 10.200.7.157
*Nov 26 18:50:50 UTC: //929/F9E88693801B/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): 172.29.5.210
Source IP Port (Media): 16786
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 28
Disconnect Cause (SIP) : 484
Can anyone shed some light on the area that I need to focus on? This is my first attempt at SIP and I am confused :)
Thanks.Hi
909 is what I meant to dial as that is the help desk for the telco.
I tried mobile numbers as well getting the same error codes. And international numbers.
If it's based on the called number being wrong then I guess I will have to play with the calling party ID and call type as well... Maybe this is causing it to fail?? -
Dear all,
Am using cisco 2921 CME 8.6 with ios version c2900-universalk9-mz.SPA.151-4.M4.bin..
Also am using a dedicated one fxo port for a specific user to send and receive calls,, the user always faces busy tone when initiate outgoing call, the fxo port doesnt disconnect from the previous call.
voice-port 0/3/2
supervisory disconnect dualtone mid-call
no battery-reversal
cptone SA
timeouts interdigit 3
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 4015
impedance complex2
caller-id enable
Any Idea to solve the issue, please???
#sh voice port 0/3/2
Foreign Exchange Office 0/3/2 Slot is 0, Sub-unit is 3, Port is 2
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is plar opx
Connection Number is 4015
Initial Time Out is set to 15 s
Interdigit Time Out is set to 3 s
Call Disconnect Time Out is set to 1 s
Power Denial Disconnect Time Out is set to 1000 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 1 s
Companding Type is u-law
Region Tone is set for SA
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to complex2 Ohm
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming):
Translation profile (Outgoing):
lpcor (Incoming):
lpcor (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is disabled
Number Of Rings is set to 1
Supervisory Disconnect is dualtone mid-call
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 65 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
Secondary dialtone is disabledHi again Lan,
Strange issue occur, the voice ports seem to be ok, in on-hook status and incoming calls are fine, the strange issue is outgoing calls can`t be initiated despite the dial-peer are configured. this issue happens to me suddenly before changing any thing. As soon i dial the first digit after dialing 9, i receive a fast busy tone, like below as soon i dial 7 i receive a fast busy tone from an ip phone with no restriction...
dial-peer voice 2002 pots
trunkgroup 2
corlist outgoing CDMA
destination-pattern 977.......
forward-digits 9
voice-port 0/2/0
trunk-group 2
supervisory disconnect dualtone mid-call
no battery-reversal
cptone SA
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 4446
impedance complex2
caller-id enable
I restart the router, boot from another ios, remove and configure the dial-peers, shut and unshut the voice port, nothing solve the issue. it seems it doesn`t recognize outside digits,,, local calls are working fine..
Any ideas to solve the issue please...
the below is info you requested while the ports are hanging:
VoiceGW#sh voice call status
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers
0x11 121D 0x2B326244 0/2/1 No DSP 7444624261 None 0/1015
0x17 121D 0x2B334ADC 0/3/3 No DSP *7444624261 None 1015/0
1 active call found
VoiceGW#sh voice call status
Enter configuration commands, one per line. End with CNTL/Z.
VoiceGW(config)#exdo sh voice port summ
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
=============== == ============ ===== ==== ======== ======== ==
0/1/0 -- fxs-ls up dorm on-hook idle y
0/1/1 -- fxs-ls up dorm on-hook idle y
0/1/2 -- fxs-ls up dorm on-hook idle y
0/1/3 -- fxs-ls up dorm on-hook idle y
0/2/0 -- fxo-ls up dorm idle on-hook y
0/2/1 -- fxo-ls up up idle off-hook y
0/2/2 -- fxo-ls up dorm idle on-hook y
0/2/3 -- fxo-ls up dorm idle on-hook y
0/3/0 -- fxo-ls up dorm idle on-hook y
0/3/1 -- fxo-ls up dorm idle on-hook y
0/3/2 -- fxo-ls up dorm idle on-hook y
0/3/3 -- fxo-ls up up idle off-hook y
50/0/1 1 efxs up up on-hook idle y
50/0/1 2 efxs up up on-hook idle y
VoiceGW(config-voiceport)#no sh
Jul 31 12:13:57.923: htsp_process_event: [0/2/1, FXOLS_CONNECT, E_HTSP_IF_OOS]htspm_mgt_statehtspm_disc_ind
Jul 31 12:13:57.923: htsp_timer_stop
Jul 31 12:13:57.923: htsp_timer_stop2
Jul 31 12:13:57.923: htsp_timer - 1000 msec
Jul 31 12:13:57.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_RELEASE_REQ]act_disc_conf
Jul 31 12:13:57.923: htsp_process_event: [0/3/3, FXOLS_OFFHOOK, E_HTSP_RELEASE_REQ]fxols_offhook_release
Jul 31 12:13:57.923: htsp_timer_stop
Jul 31 12:13:57.923: htsp_timer_stop2
QNB_VoiceGW(config-voiceport)#Jul 31 12:13:57.923: htsp_timer_stop3
Jul 31 12:13:57.923: [0/3/3] set signal state = 0x4 timestamp = 0
Jul 31 12:13:57.923: htsp_timer - 2000 msec
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_EVENT_TIMER]htspm_disc_conf
Jul 31 12:13:58.923: htsp_timer_stop
Jul 31 12:13:58.923: htsp_timer_stop2
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_IF_OOS_CONF]
Jul 31 12:13:58.923: htsp_timer_stop
Jul 31 12:13:58.923: htsp_timer_stop2
Jul 31 12:13:58.923: htsp_timer_stop3
Jul 31 12:13:58.923: htsp_timer_stop_mlpp
Jul 31 12:13:58.923: %LINK-3-UPDOWN: Interface Foreign Exchange Office 0/2/1, changed state to Administrative Shutdown
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_DSP_INTERFACE_INFO]
Jul 31 12:13:59.339: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_IF_INSERVICE]
Jul 31 12:13:59.343: %LINK-3-UPDOWN: Interface Foreign Exchange Office 0/2/1, changed state to up
Jul 31 12:13:59.343: Foreign Exchange Office 0/2/1 rx_signal_map:
F F F F
5 F F F
F F F F
F F F F
Jul 31 12:13:59.343: [0/2/1] set signal state = 0xC timestamp = 0
Jul 31 12:13:59.343: Foreign Exchange Office 0/2/1 tx_signal_map:
0 4 4 4
4 4 6 4
C C C C
C C C C
Jul 31 12:13:59.343: htsp_process_event: [0/2/1, S_OPEN_PEND, E_HTSP_GO_UP]
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_NULL, E_HTSP_INIT]fxols_null_init
Jul 31 12:13:59.455: [0/2/1] set signal state = 0xC timestamp = 0
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_INIT, E_HTSP_INSERVE]fxols_init_inserve
Jul 31 12:13:59.455: [0/2/1] set signal state = 0x4 timestamp = 0
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
Jul 31 12:13:59.923: htsp_process_event: [0/3/3, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Jul 31 12:13:59.927: htsp_process_event: [0/3/3, FXOLS_ONHOOK, E_DSP_SIG_0100] -
Issue with SPA525g registation and FXO port call calls are not disconnecting properly
Hi,
I have a UC540 and updated it to the latest IOS version with the latest firmware to my phones and i am having registration problems with SPA525g IP Phones. I updated the firmware of the phones as well and create manual tftp bindings with but still it is not registering. I run a couple of debugs (debug tftp events and debug ephone registration) I can see from the logs and in the phone that it is taking the proper VLAN and being discovered via CDP and being pointed to the TFTP server and still wont register. I can see that it is also taking its own .cnf file properly then the output sccp token regected invalid devices error is shown I have a SPA502G and it is working fine. Also there is a previous issue that all the voice port are shown as engage or offhook even the calls are disconnected thus make the main PSTN number busy am based in UAE and our service provider is etisalat I have check with them about the proper disconnection values but still it the same. That's why I have arrived in the conclusion to just update everything including the IOS and the phones firmware. I have put my config in this post, I am also trying to take the CCNA Voice exam on the 2nd week of april and I think that if i don't know how fix this issue for our customer then I would probably fail that exam. any suggestion and help is greatly appreciated cisco experts.
! Last configuration change at 13:36:42 ZP4 Thu Sep 13 2012 by Nick
! NVRAM config last updated at 13:45:41 ZP4 Thu Sep 13 2012 by Nick
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
hostname UC540
boot-start-marker
boot system flash:uc500-advipservicesk9-mz.151-2.T4
boot-end-marker
logging buffered 64000
enable secret 5 $1$3CIf$.rXyHeJQrwd97X/f2dS0M1
no aaa new-model
clock timezone ZP4 4 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-3558175224
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-3558175224
revocation-check none
crypto pki certificate chain TP-self-signed-3558175224
certificate self-signed 01 nvram:IOS-Self-Sig#3.cer
dot11 syslog
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 10.1.3.1 10.1.3.10
ip dhcp pool phone
network 10.1.3.0 255.255.255.0
default-router 10.1.3.1
option 150 ip 10.1.3.1
ip name-server 213.42.20.20
ip name-server 195.229.241.222
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW cuseeme
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
stcapp supplementary-services
port 0/0/0
fallback-dn 301
port 0/0/1
fallback-dn 302
port 0/0/2
fallback-dn 303
port 0/0/3
fallback-dn 304
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
trunk group ALL_FX0
voice call send-alert
voice rtp send-recv
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
sip
no update-callerid
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
voice class dualtone-detect-params 1
freq-max-deviation 50
freq-max-power 0
freq-min-power 13
freq-power-twist 4
cadence-variation 6
voice class custom-cptone UAE-CUSTOM
dualtone disconnect
frequency 406
cadence 398 344 237 527 400
voice class custom-cptone CCAjointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
voice class custom-cptone CCAleavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
voice class cause-code 1
no-circuit
voice register global
voice hunt-group 1 parallel
list 301,302,303
timeout 24
pilot 511
voice translation-rule 4
rule 15 // //
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
voice translation-rule 1112
rule 1 /^9/ //
rule 3 /^0/ //
voice translation-rule 2222
voice translation-rule 3265
rule 1 /\(^..........$\)/ /9\1/
rule 2 /\(^.........$\)/ /9\1/
rule 15 /\(^ABCD$\)/ /ABCD\1/
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 3265
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PROFILE_ALL_FXO
translate calling 4
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
license udi pid UC540W-FXO-K9 sn FHK143074G6
archive
log config
logging enable
logging size 600
hidekeys
username cisco privilege 15 secret 5 $1$vjNa$OFKLhupqR8al6x2b8Xmcj/
username adminac privilege 15 secret 5 $1$NDC.$PtD0y4YGIj5SqI1gghxWE1
username Nick privilege 15 secret 5 $1$iAmL$tsg7Jf2TEND1NN.h8z2dy/
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 $FW_OUTSIDE$
ip address 192.168.101.2 255.255.255.252
ip nat outside
ip virtual-reassembly in
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
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport access vlan 20
spanning-tree portfast
interface FastEthernet0/1/8
switchport access vlan 100
macro description cisco-switch
interface Dot11Radio0/5/0
no ip address
shutdown
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
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 Vlan20
ip address 10.10.10.1 255.255.255.0
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
no ip address
ip nat inside
ip virtual-reassembly in
shutdown
interface BVI100
description $FW_INSIDE$
ip address 10.1.3.1 255.255.255.0
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 0.0.0.0 192.168.101.1
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
logging esm config
access-list 1 remark SDM_ACL Category=2
access-list 1 permit 192.168.10.0 0.0.0.255
access-list 1 permit 10.1.3.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
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 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 permit udp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 deny ip 10.1.3.0 0.0.0.255 any
access-list 101 deny ip 192.168.10.0 0.0.0.255 any
access-list 101 deny ip 192.168.101.0 0.0.0.3 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 10.1.3.0 0.0.0.255 any
access-list 102 deny ip 192.168.101.0 0.0.0.3 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 102 permit ip 192.168.101.0 0.0.0.3 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 192.168.101.0 0.0.0.3 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 105 permit ip any any
snmp-server community public RO
tftp-server flash:/phones/521_524/cp524g-8-1-17.bin alias cp524g-8-1-17.bin
tftp-server flash:/phones/5x5/spa5x5-7-1-3c.bin alias spa5x5-7-1-3c.bin
tftp-server flash:/phones/525/spa525g-7-4-8.bin alias spa525g-7-4-8.bin
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone GB
station-id name Cordless
station-id number 329
caller-id enable
voice-port 0/0/1
cptone AE
caller-id enable
voice-port 0/0/2
cptone AE
caller-id enable
voice-port 0/0/3
cptone AE
caller-id enable
voice-port 0/1/0
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4FXO-0/1/0-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/1
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/1-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/2
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
supervisory dualtone-detect-params 1
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/2-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/3
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/3-Custom-BG
bearer-cap Speech
caller-id enable
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 10.1.3.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confprof1
dspfarm profile 1 conference
description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec729
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
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
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 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/1/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/1/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/1/2
no sip-register
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 388
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 6 pots
description "catch all dial peer for BRI/PRI"
translation-profile incoming nondialable
incoming called-number .%
direct-inward-dial
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/1/3
no sip-register
dial-peer voice 69 pots
destination-pattern 329
port 0/0/0
dial-peer voice 300 pots
trunkgroup ALL_FX0
description Local Numbers
destination-pattern 9T
forward-digits 9
dial-peer voice 301 voip
destination-pattern 2..
session target ipv4:192.168.201.2
dial-peer voice 303 pots
trunkgroup ALL_FXO
trunkgroup ALL_FX0
description **InternationalCall**
destination-pattern 88T
dial-peer voice 304 pots
trunkgroup ALL_FX0
description *EM1*
destination-pattern 9[1-9]T
forward-digits 3
dial-peer voice 302 pots
trunkgroup ALL_FX0
description **Mobiles**
destination-pattern 9.[0-9].[0-9]......
dial-peer voice 305 pots
trunkgroup ALL_FX0
description **800-**
destination-pattern 9[0-9][0-9][0-9]T
no dial-peer outbound status-check pots
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 5
sdspfarm tag 1 confprof1
conference hardware
video
fxo hook-flash
max-ephones 40
max-dn 300
ip source-address 10.1.3.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone webAccess 0
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
system message American Center
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.2/CCMCIP/authenticate.asp
load 521G-524G cp524g-8-1-17
load 525G spa525g-7-4-8
load 501G spa5x5-7-1-3c
load 502G spa5x5-7-1-3c
load 504G spa5x5-7-1-3c
load 508G spa5x5-7-1-3c
load 509G spa5x5-7-1-3c
time-zone 35
date-format dd-mm-yy
voicemail 388
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh MOH2.wav
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 $1$iDgA$MKNi2RWfsO0KjuC82kgLJ1
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
secondary-dialtone 9
fac standard
create cnf-files version-stamp 7960 Aug 29 2012 12:00:04
line con 0
privilege level 15
logging synchronous
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
exec-timeout 0 0
logging synchronous
login local
transport input all
line vty 5 100
login local
transport input all
ntp master
end
Some of the output are not shown becaus it is to long I have attach the whole config for reference and any advice on how could I optimize and resolve my issues is greatly appreciated. ThanksNicolo - First off this stuff gets crazy sometimes. No worries about the exam. Sometimes when FXO ports go crazy it is due to battery reversal. If you go to the FXO port settings try turning battery reversal on and or off... depending on its current setting. See if that helps.
As for the 525s not registering.. These are inside the network correct? Are you connecting one directly to the UC500 with a Cat5E or Cat6 patch cable and the same thing happens? Does the MAC address on the phone match a MAC address under the EPHONE settings?
If you telnet into the UC500 can you execute a "dir" command at the CLI prompt and "CD" (change directory) into the phones folder and then the spa525g folder? Do files exist in there?
Also I only see an IP address under BVI100? This is the voice side of things what happened to the IP address under BVI1 (Data VLAN). Can you give us some information about the internal network? Cna you PING this phone system from the network? What IP address does it have? -
Transfer called number from FSX port SPA8000 to FXO port analog PBX.
Hello all!
I have CUCM 8.6 which connecting with SPA8000 by SIP trunk.
FXS port SPA8000 connect to FXO port analog PBX.
When I call from CUCM to analog PBX through SPA8000, SPA8000 does not pass called number in line.
Anybody have ideas?Here is what I would recomend, I have set this up at several installations and it works well. The only difference would be that the ones that I have setup had more than 64K between sites but that is still doable so long as you keep the traffic on that link well regulated.
Anyhow. At each site install a VIC-2FXO and a VIC-2FXS into your 1760. You can use the higer density modules if you need more paths. Tie the FXOs to analog station ports on your PBX and the FXSs to analog CO ports. This may seem backwards but it is correct. Cisco's FXS is expecting to provide dialtone and the analog CO ports on the PBX are expecting to receive dial tone. Depending on what you are using for call processing ITS/CCME/CCM configure your gateway and your route pattern, etc. You will also need to be able to somehow access those fxs ports for outbound calls. You could have direct line access buttons on your phones and instruct users that when calling the remote site the must use lines 7 and 8 as an example. You could also setup a line pool code for these new lines.
Now when site A wants to call Site B, they dial whatever you wish them to dial. You do whatever translations you may wish prior to sending the call. Here is how the call flow works. User at site A accesses the line either by hitting a line button or dialing the line pool code. This grabs the analog line tied to one of the FXs ports on the router. The call is then sent over the 64K link. The router at Site B then uses one of the FXO ports to pass the call to an analog station port on the other PBX. To the PBX at site B it will appear as though someone pickedu picked up an analog station and dialed a local extension.
Hope this helps. If you need any further clarification on this let me know and I will try to clear it up further for you. -
UC520 SIP trunk unable to make outgoing calls, incoming calls are ok
I have an new SIP trunk set on an UC520 and the incoming calls are ok, but the outgoing calls are getting an busy tone(not working).
The bellow trace is showing that the cause is "No route to destination (3) ". The question is this route has to do with the firewall(ip routing) or with the voice translation rules?
001866: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x843CE50C
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 0777777777 <- main sip number
Called Number : 0888888888 <- called number
Source IP Address (Sig ): 0.0.0.0
Destn SIP Req Addr:Port : 0.0.0.0:0
Destn SIP Resp Addr:Port : 0.0.0.0:0
Destination Name :
001867: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 3
Disconnect Cause (SIP) : 200Hi Emil,
I've added bind control and media interface but outgoing calls are the same blocked, strange thing is that the cause is still no route to destination (3)
but
UC_520#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP User Agent bind status(media): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl -
SIP Trunk not accepting inbound calls
I have a CME setup using Engin as a SIP provider
I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
My SIP registration is OK and the number is configured as the primary DN on one of my phones
Router#sh sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
038682XXXX -1 1124 yes
101 20001 45 no
102 20003 18 no
103 20005 45 no
104 20006 45 no
I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
Router#
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
Sep 8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4C39C570
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0417XXXXXX
Called Number : 038682XXXX
Source IP Address (Sig ): 211.30.48.136
Destn SIP Req Addr:Port : 203.161.164.69:5060
Destn SIP Resp Addr:Port : 203.161.164.69:5060
Destination Name : 203.161.164.69
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 211.30.48.136
Source IP Port (Media): 17768
Destn IP Address (Media): 203.161.164.69
Destn IP Port (Media): 18314
Orig Destn IP Address:Port (Media): [ - ]:0
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 21
Disconnect Cause (SIP) : 403
Any Ideas
DougHi Tapan,
Firstly the topology is as follows
ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
The VM is provided by the ISP
debug ccsip messages
Sep 9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Contact:
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: multipart/mixed,application/media_control+xml,application/sdp
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 317
v=0
o=BroadWorks 18275729 1 IN IP4 203.161.164.69
s=-
c=IN IP4 203.161.164.69
t=0 0
m=audio 18128 RTP/AVP 18 8 0 101
c=IN IP4 203.161.164.69
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=bsoft: 1 image udptl t38
Sep 9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Sep 9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0
Sep 9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
CSeq: 633854439 ACK
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
Max-Forwards: 9
Content-Length: 0
Voice Config
Router#
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mel.byo.engin.com.au
no call service stop
voice class codec 1
codec preference 1 g711ulaw
voice translation-rule 10
rule 1 /^0/ //
voice translation-rule 11
rule 1 /^.*/ /0386821234/
voice translation-profile PSTN_Outgoing
translate calling 11
voice-card 0
dsp services dspfarm
mgcp profile default
sccp local Vlan100
sccp ccm 10.1.100.1 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface Vlan100
associate ccm 1 priority 1
associate profile 1 register confdsp
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
dial-peer voice 99 voip
translation-profile outgoing PSTN_Outgoing
destination-pattern .T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 100 voip
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number 0386821234
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 110 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number .%
dtmf-relay rtp-nte
no vad
dial-peer voice 90 voip
description Melbourne 03 Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [89].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 91 voip
description National Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 0[278]........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 92 voip
description Vic/Tas 03 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [56].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 93 voip
description Mobile numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 04........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 94 voip
description 13XXXX numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 13[1-9]...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 96 voip
description 1300/1800 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 1[38]00......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 98 voip
description Emergency 000
translation-profile outgoing PSTN_Outgoing
destination-pattern 000
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
sip-ua
credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
authentication username 0386821234 password 7 XXXX
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
registrar dns:mel.byo.engin.com.au expires 3600
sip-server dns:mel.byo.engin.com.au
connection-reuse
telephony-service
sdspfarm conference mute-on #1 mute-off #2
sdspfarm units 2
sdspfarm tag 1 confdsp
conference hardware
max-ephones 42
max-dn 144
ip source-address 10.1.100.1 port 2000
calling-number initiator
service phone videoCapability 1
service phone displayOnDuration 00:01
service phone displayOnTime 08:30
service phone displayOffTime 17:30
service phone displayIdleTimeout 00:01
service phone displayOnWhenIncomingCall 1
system message Cisco CME
load 7941 SCCP41.8-4-2S
load 7942 SCCP42.8-4-2S
load 7945 SCCP45.8-4-2S
load 7961 SCCP41.8-4-2S
load 7962 SCCP42.8-4-2S
load 7965 SCCP45.8-4-2S
load ata ATA030204SCCP090202A
time-zone 48
date-format dd-mm-yy
voicemail 90125200
mwi relay
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 0.T
create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
ephone-dn 1 dual-line
number 038682XXXX
label 101
name 7965
mwi sip
ephone-dn 2 dual-line
number 102
label 102
name 7941
ephone-dn 3 dual-line
number 103
label 103
name 7920
ephone 1
device-security-mode none
video
mac-address 0023.5EB8.6E4E
type 7965
button 1:2 2:1
ephone 3
device-security-mode none
mac-address 0019.0633.A933
max-calls-per-button 2
type 7920
button 1:3
ephone 10
device-security-mode none
mac-address 0019.E7B7.BAB3
max-calls-per-button 2
type ata
button 1:1 -
Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP
Hi Cisco Community,
I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
Below is an example of a call that is connected with the current setup:
Note:
IP: 10.18.81.2 (CUBE)
IP: 10.18.81.11 (CUCM SUB)
IP: 10.111.111.254 (ITSP SBC)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1417347869
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
s=SIP Call
c=I
PM-HO-VG-01#N IP4 10.18.81.2
t=0 0
m=audio 22256 RTP/AVP 18 0 8 101
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf9
PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,application/xml
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 236
v=0
o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.80.40
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
PM-HO-VG-01#
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
PM-HO-VG-01#sh sip
PM-HO-VG-01#sh sip-ua call
PM-HO-VG-01#sh sip-ua calls
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 27218091323
Called Number : 0862000000
Bit Flags : 0xC04018 0x10000100 0x0
CC Call ID : 64511
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.111.111.254]:5060
Destn SIP Resp Addr:Port: [10.111.111.254]:5060
Destination Name : 10.111.111.254
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64511
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22256
Media Dest IP Addr:Port : [10.111.111.254]:20074
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 0218091323
Called Number : 0862000000
Bit Flags : 0xC0401E 0x10000100 0x80004
CC Call ID : 64510
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.18.81.11]:5060
Destn SIP Resp Addr:Port: [10.18.81.11]:5060
Destination Name : 10.18.81.11
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64510
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22350
Media Dest IP Addr:Port : [10.18.80.40]:21928
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
PM-HO-VG-01#
PM-HO-VG-01#
PM-HO-VG-01#
As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22256 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 102 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 360
v=0
o=BroadWorks 316169737 2 IN IP4 10.111.111.254
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=inactive
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22350 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Length: 0
Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 103 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 306
v=0
o=BroadWorks 316169737 3 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 2
PM-HO-VG-01#00 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 213
v=0
o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.81.10
t=0 0
m=audio 4000 RTP/AVP 18
a=X-cisco-media:umoh
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=sendonly
Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 104 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Race Condition
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
Timestamp: 1417347889
CSeq: 104 BYE
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 101 BYE
Content-Length: 0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 200
PM-HO-VG-01#Hi Manish,
Again, excellent feedback. Much appreciated.
I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
I will be doing some intensive test again later on this week and will send the logs.
Here is my question to both of you:
Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
Thanks again for your support fellows. -
Incorrect Caller ID on calls from outside line via FXO port.
Have a public phone line connected to my CUCME 2801 router VIC2-2FXO card. All inbound calls are passed to DN-5001 (group number). Can receive and send calls without a problem, but incoming calls all show "911" for caller ID. Think this is simply an issue with the out bound dial-peer, of which the lowest numbered out bound dial-peer is for 911 services. Not sure how to correct this so inbound calls show the proper caller ID?
Below is a copy of my CUCME show run output from the FXO port config thru all the dial-peers. Any pointers is greatly appreciated.
Thanks.
Kirk E.
voice-port 0/0/0
connection plar opx immediate 5001
voice-port 0/0/1
voice-port 0/2/0
station-id name POTS
station-id number 7000
voice-port 0/2/1
ccm-manager config
dial-peer voice 7000 pots
destination-pattern 5006
port 0/2/0
dial-peer voice 90 pots
description Emergency Services
destination-pattern 911
port 0/0/0
forward-digits 3
dial-peer voice 91 pots
description 10 Digit local dialing
destination-pattern [234].........
port 0/0/0
forward-digits 10
dial-peer voice 92 pots
description 11 Digit local/long distance dialing
destination-pattern 1[2348].........
port 0/0/0
forward-digits 11
dial-peer voice 93 pots
description Long Distance
destination-pattern 011T
port 0/0/0
prefix 011
dial-peer voice 94 pots
description Backup bench POTS phone
destination-pattern 7000
port 0/2/0
dial-peer voice 2 voip
destination-pattern 51..
session protocol sipv2
session target ipv4:172.16.2.155
dtmf-relay sip-notify
codec g711ulaw
no vadHi
Can you find the below:-
Hi
1- Please find the below table as the following link http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800b53c7.shtml
Caller ID Requires VIC-2FXO-M1, VIC-2FXO-M2, VIC-4FXO-M1, VIC2-2FXO, VIC2-4FXO, or MRP3-8FXOM1
under voice-port
caller-id enable
2-If above configure and still have no caller id , please add the below commannds to the voice-port
caller-id alerting line-reversal
cptone ? "based on your"
caller-id alerting ring 2 "the default is 1" maximum number of rings to be detected before a call is answered over an FXO voice port.
4-Do debug to make sure all ok
"debug vpm signal "
[0/3/0] get_fxo_caller_id:Caller ID received. Message type=128 length=31 checksum=74
Thank you
please rate all useful information -
Has anyone set up Lync server 2010 to use the Gamma SIP trunks, that dont require the use of a gateway?
No requirement for an additional gateway device, with direct MS Lync connectivity
The trouble is i cannot get Lync to connect to the trunks. We have purchased the SIP trunks from a gamma supplier(we didnt now they were a supplier, until recently when we asked for support and they went 'duhhhhhh me no know, we just
sell things dunow how to set things up' what a PAIN IN THE A***), and they say that the SIP trunks are pointed at our EFM IP address. which also has the DDIs assigned to it.
So, i setup a PSTN gateway on lync topology using IP of EFM, Listening port 5060 using TCP. Are these ports and protocol okay?
The VoIP phones seem to want to call, they just lack any sound, no ringing tone, no dissconnected tone. Just says calling "+44157322****" So the dial plan is changing 22**** to the correct local code and whatever the +44 thing
is.
Any advice on how i can find the problem, or how to setup the trunks up would be hugely appreciated.
P.S We initially tried to use an audiocodes mediant 1000, which was what we asked our trunk supplier about, and then they informed us about being a gamma supplier, and that the gamma trunks do not require a gateway. Followed setting
up guide for mediant 1000 with gamma trunks through audiocodes blah, to no success. I think thats because it was changing the coders, which was not needed if the trunks are directly compatable.Hi,
Please review the SIP trunk topology.
http://technet.microsoft.com/en-us/library/gg398720.aspx
To
implement SIP trunking, you must route the connection through a Mediation Server, which proxies communications sessions between Lync Server 2010 clients and the service provider and transcodes media when necessary. Each Mediation Server has
an internal and an external network interface. The internal interface connects to the Front End Servers. The external interface is commonly called the gateway interface because it has traditionally been used to connect the Mediation Server to a PSTN gateway
or an IP-PBX. To implement a SIP trunk, you connect the external interface of the Mediation Server to the external edge component of the ITSP. The external edge component of the ITSP could be a Session Border Controller (SBC), a router, or a gateway.
Generally the gateway is not required in your organization. You need to configure Mediation Server setting. For the details about
the SIP trunk configuration of ITSP side, you need to contact Gamme Support for further assistance.
Regards,
Kent Huang
TechNet Community Support ************************************************************************************************************************ Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a
marked post does not actually answer your question.
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