DTMF Digits passing the Voice Gateway
Hi,
iam using Cisco 3825 router where E1 circuits are connected. Need to check the DTMF digits passing when i am trying to connect to an IVR number and the options digits pressed.
How can I get that.
Help required.
Thanks in Advance.
Which VOIP protocol are you using?
If using MGCP check:
debug mgcp packet and look for NOTIFY messages. Check
mgcp dtmf-relay voip codec all mode out-of-band command is configured.
If H225 check dtmf-relay alphanumeric under VOIP dial-peers.
For checking debugs run debug h245 asn1
For TDM run:
debug voip vtsp tone
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Blocking incoming collect calls in the voice gateway
Hello
I am using a C3825 router and I want to block incoming collect calls. I tried the command "double-answer" under cas-custom but it is not working. Does anyone have an alternative? I am using an E1 R2 digital.
Thank you
MarcosJonathad,
OP is not in the US, and does not have ISDN, has E1 R2 instead
E1 R2 has a method to block collect calls called double-answer. This method is supported and documented by Cisco.
But for some reason it doesn't work for OP.
In these case, it is necessary to involve an experienced consultant, if not TAC escalation directly. -
Configuring TNT in IOS Voice Gateway
Hello all
1. Is TNT, Take Back And Transfer, supported in IOS voice gateway in general?
2. If yes, is there any specific IOS configuration needed at the voice gateway?
(I do not see any need as the DTMF digits(-tones) will be sent in-band as in normal voice stream to the PSTN (which supported/subscribe to TNT).
Please advise. thanks
PaulNo need to configure anything for TNT as the transfer DTMF sequence is sent inband.
Chris -
Changing CAS e&m-wink-start to a PRI on voice gateway
Cisco IOS Software, C3900 Software (C3900-UNIVERSALK9-M), Version 15.1(4)M4, RELEASE SOFTWARE (fc1)
I want to change a CAS e&m-wink-start to a PRI on controller T1 0/1/0 using mgcp below. I am posting the new configuration below the current e&m-wink-start configuration on the voice gateway router.
Current
controller T1 0/1/0
cablelength long 0db
ds0-group 1 timeslots 1-24 type e&m-wink-start
description
New configuration
Router(config)# no contoller T1 0/1/0
Router(config)# no interface Serial0/1/0
Router(config)#controller t1 0/1/0
Router(config-controller)#cablelength long 0db
Router(config-controller)#framing esf
Router(config-controller)#linecode b8zs
Router(config-controller)#clock source line
Router(config-controller)#pri-group timeslots 1-24 service mgcp
Router(config-controller)#description circuit ID
Router(config-if)# interface serial0/1/0
Router(config-if)# no ip address
Router(config-if)# encapsulation hdlc
Router(config-if)# isdn switch-type primary-4ess
Router(config-if)# isdn incoming-voice voice
Router(config-if)# isdn bind-l3 ccm-managerHi,
You need additional configs for this to work..You also need to configure the isdn bind on the D-channel not as you have it..
Router(config-if)# interface serial0/1/0:23
Router(config-if)# no ip address
Router(config-if)# encapsulation hdlc
Router(config-if)# isdn switch-type primary-4ess
Router(config-if)# isdn incoming-voice voice
Router(config-if)# isdn bind-l3 ccm-manager
Additional configs....change ip to suit your needs
ccm-manager redundant-host 192.168.103.114
ccm-manager mgcp
ccm-manager music-on-hold bind xxx--put relevant interface
mgcp
mgcp call-agent 192.168.103.115 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp modem passthrough voip codec g711alaw
mgcp modem passthrough voip redundancy
mgcp ip qos dscp af31 media
mgcp ip qos dscp cs3 signaling
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
mgcp default-package dtmf-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
mgcp tse payload 100
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interfacexx -----------------------put relevant interfcae here
mgcp bind media source-interface xxx--------------------same
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared" -
Hello everyone;
I do have an analog line that i want to affect to (04) four IP Phones, so that any one of these four ip phones users can use this same analog line, and when we call that analog line number, any one of these four ip phone users can answer.
Is this possible to do on the Voice gateway?
I think about configuring many connection plar under one voice port, like this:
voice port 1/0/0
connection plar opx 1000
connection plar opx 1001
connection plar opx 1002
connection plar opx 1003
And about, puting many answer adress under dial-peer voice this way:
Dial-peer voice
answer adress 1000
answer adress 1001
answer adress 1002
answer adress 1003
but i don't know if it will work , and don't see how it could be done for the voice translation rule and the voice translation profile.
Ragards.
CaméliaThis will be done in only specific Dial-peer and Voice port.
A voice Translation rule is not need because once the voice-port number is dialed for example 011-27272, the plar command forward it 2001 and the dial-peer 10 is matched, you only need to configure a hunt-pilot in CUCM.
Voice-Port 1/0/0
connection plar opx 2001
2.) You will need to match incoming calls from PSTN, you can do this by use the below commands to match all incoming calls from PSTN.
Dial-peer voice 1 Pots
Incoming called-number .
3.) Finally you will need to send calls to 2001 via CUCM to match the Hunt-Pilot.
Dial-peer voice 10 voip
destination-pattern 2001
session-target ipv4:<<CUCM-IP-Address>>
Please Rate. -
Etherchannel on a Voice Gateway
We're considering etherchannel on a 2851 voice gateway. The goal is to use both of the dual Gig LAN interfaces for redundancy. Anyone with recommendations please let me know.
ThanksHi,
Instead of Etherchannel, how about using the backup interface command on the second NIC of the router.
The benefit is we can use the same IP on the second NIC as the first/primary NIC.
The backup interface remains in standby mode and becomes alive only when the primary NIC is down.When the primary NIC comes up the secondary automatically goes in standby mode.
Advantages : "Backup interface" command is independent of routing protocols. That is, it does not depend on routing protocol convergence, route stability and so on.
Each of the voice gateway NICs can be conencted physically to LAN swiches.
The Voice gateway protocol which can be easily used with this setup is H323 or SIP, not sure on MGCP though.
Only thing is the calls active calls will drop when the primary interface goes down, or comes back again(the secondary NIC goes to standby mode) which is the normal behaviour.
Can you please feedback on the usage of "backup interface" command for configuring NIC resilience on the voice gateway? -
Voice Gateway & Gatekeeper On a single Router?
Hello All,
Has anyone had experience with putting voice gateway and voice gatekeeper functionality into one router? We will have three clusters initially (with about 1,000 users worldwide though) and was considering one gatekeeper per cluster for CAC and tail end hop off routing for LD/International calls.
Any guidance on what sort of platform these should be running on if kept separate? DSP resources in them, etc.?
The voice gateways will be a mixture of 2851 and 3800 series routers. Thanks in advance for the help.
Thanks,
DaveHi,
I have done this many times in lab, dont know if that will be ok in production if you have a good powerfull problem i dont think that it will be a problem.
About the DSP's maybe you can use the DSP calculator - anyway this refers to many parameters on the network.
BR,
Teo -
Troubleshooting no ringback voice gateway
Hi all,
I'd like to ask about voice gateway that installed on my customer site. below is the topology:
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mso-padding-alt:0in 5.4pt 0in 5.4pt;
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PSTN – Voice Gateway – C2960-48PST-L – Callmanager and IP Phone
|
|(FO)
Analog phone – PBX.A – Voice Gateway –––– Voice Gateway – PBX.B – Analog phone
the problem is that there's no ring back tone when calling to another site analog phone from ip phone/analog phone. The PBX (NEC) are using E1 connection to the router. Does the ringback provided by PBX or the Voice gateway?
Thanks in advance
DiasYes, we did find a solution. It ended up it was on the carrier's end. I believe that they were generating ring back inbound to calls we originated, but not outbound to calls originated from the carriers. They switched that so that they did not send us ring back, but did send carriers ring back for inbound calls and that appeared to solve the issue.
Here's my carrier's explanation in case that wasn't confusing enough: "We made in-band tone available to the SIP trunk via an egress profile change on the SONUS Gateway. Early-media is determined / initiated by
the called or far-end switch." -
Call Manager register fxs port with voice gateway- problem
I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
I have no dial tone.
If i write no shut down on the voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
I've entered no mgcp and mgcp commands and i've reset the voice gateway.
How can i call from the pots to the voip phone?
The ios version on the voice gateway is Version 12.4(22)T4.
Here is an outghtput from the Voice gateway.
ccm-manager mgcp
ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.1.1.33
ccm-manager config
mgcp
mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp validate domain-name
mgcp rtp payload-type g726r16 static
mgcp profile default
timeout tone busy 600
timeout tone dial 600
dial-peer voice 999223 pots
service mgcpapp
port 2/23
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999888 pots
service mgcpapp
port 2/23
The CUCM 6 is registered with the voice gateway.Is your campaign using CPA? If so, what's the behavior if CPA is not enabled?
I think the best thing to do is to run a trace...
Call Manager > Cisco Unified Serviceability > Trace > Configurations
Select a CUCM server - any subscriber would work.
Service Group - CM Services
Cisco CallManager (Inactive)
Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here.
Also, make sure your phone is in the correct CSS in Call Manager -
Where can I learn the structure of Voice gateway ?
I'm making a essay about Structure of Voice gateway: hardware and software construction. I can not find any books or any manual deal with it . So,can someone give me more information about structure of voice gateway or give me some useful resources please?
I'm waiting for your replying.Cisco doesn't make public the software structure of their products; to a certain extent it can be inferred from documentation and familiarity with the product, but probably not enough for an in-deep analysis.
For your task, I suggest you focus on some open-source development of GW, that has no secrets.
Hope this helps, please rate post if it does! -
What versions of Call Manager support the new VG3XX Voice Gateways?
What versions of Call Manager support the new VG3XX Voice Gateways?
http://www.cisco.com/c/en/us/td/docs/routers/access/vg350/software/configuration/guide/vg350_scg/scgapp_platforms_ORIG.html#64565
-
CALL DOES NOT ROUTE OUT THE LOCAL GATEWAY
Local calls will not route out the local Gateway of branch1 to the PSTN or from the PSTN back to branch1, however they will route out either CorpHQ or branch2 backup gateways. When I go into the route group configuration for branch1, and remove the backup gateways, I get a fast busy tone when I dial the local number. I know the MGCP Gateway at branch1 is functioning because when I dial 911 and run debug ISDN Q931, the call routes properly through branch1, so I have a call routing problem. I ran DNA and it came back as ROUTE THIS PATTERN and all of the number translations looked accurate, so I didn't have to check for any block patterns. I'm not getting any errors on the calling party phone display. When I deleted the route pattern for the branch1 site and forced it to use the global route pattern, I received a debug output on branch1. I do not know a debug command (such as debug voip dial-peer or debug ccsip messages) to use for an MGCP Gateway to see if the call is actually reaching the Gateway.
I have checked the following:
the route pattern configuration
the translation pattern configuration
the called party transformation pattern configuration
the route list configuration to make sure the correct route group for branch1 was selected
the route group configuration to make sure that the branch1 Gateway was first in the order of selected devices
the route pattern configuration to make sure the correct route list for branch1 ist selected
the Gateway configuration to make sure it's using the device pool for branch1 and to make sure the called party transformation CSS for the branch1 Gateway is applied
the device pool configuration to make sure it's using the route group branch1
Any assistance would be greatly appreciated
Regards,
RonHi Nishant:
Please see the attachments for the Gateway pages
The significant digits for inbound calls for all 3 gateways is '4'
Please see the running-configs of the 3 gateways and the PSTN
Please see the debugs for the INBOUND calls
Many Thanks,
Ron
The following INBOUND call from the PSTN to 2065011001 is now working, however it is supposed to be routing through CorpHQ and is instead routing through Branch1. Please see 'DEBUG VOIP CCAPI INOUT' & 'DEBUG ISDN Q931'
Branch1#
ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x0096
Cause i = 0x8290 - Normal call clearing
//22/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x16, digit_event=0x0, enable=FALSE, consume=FALSE)
//22/5A001212800B/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=22
//22/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x16, disp=0, digit_event=0x0, enable=FALSE, consume=FALSE)
//22/5A001212800B/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=22
//22/5A001212800B/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms))
//22/5A001212800B/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Network, Params=0x0, Call Id=22
//23/5A001212800B/CCAPI/ccGetCallStatistics:
Call Stats=0x4A5346FC, Call Id=23
//22/5A001212800B/CCAPI/ccConferenceDestroy:
Conference Id=0xC, Tag=0x0
//22/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:
Conference Id=0xC, Source Interface=0x49E07FD4, Source Call Id=22,
Destination Call Id=23, Disposition=0x0, Tag=0x0
//23/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:
Conference Id=0xC, Source Interface=0x495BABA4, Source Call Id=23,
Destination Call Id=22, Disposition=0x0, Tag=0x0
//22/5A001212800B/CCAPI/cc_generic_bridge_done:
Conference Id=0xC, Source Interface=0x495BABA4, Source Call Id=23,
Destination Call Id=22, Disposition=0x0, Tag=0x0
//22/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//22/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
//22/5A001212800B/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
//23/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//23/5A001212800B/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
//23/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x495BABA4, Tag=0x0, Call Id=23,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
//23/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 4821DDE8
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 1
//22/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x49E07FD4, Tag=0x0, Call Id=22,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
//22/5A001212800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 4821DEC8
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 0
ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x8096
ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0096
ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0097
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18381
Preferred, Channel 1
Progress Ind i = 0x8183 - Origination address is non-ISDN
Display i = 'Seattle US Phone'
Calling Party Number i = 0x4180, '2065015111'
Plan:ISDN, Type:Subscriber(local)
Called Party Number i = 0xC1, '2065011001'
Plan:ISDN, Type:Subscriber(local)
//-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x49E07FD4, Interface Type=6, Destination=, Mode=0x9,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=D000000002f5368f000000F580000097)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 1
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1210179280,feature_id:24
//24/74820328800C/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1
//24/74820328800C/CCAPI/ccCallSetContext:
Context=0x4A524790
//-1/xxxxxxxxxxxx/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x495BABA4, Interface Type=9, Destination=0.0.0.0, Mode=0x9,
Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=, FinalDestinationFlag=FALSE, Outgoing Dial-peer=0, Call Count On=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=D000000002f5368f000000F580000097)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 2
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1210179056,feature_id:25
//25/74820328800C/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=9, FlowMode=1
//25/74820328800C/CCAPI/ccCallSetContext:
Context=0x4A524580
//25/74820328800C/CCAPI/cc_api_call_connected:
Interface=0x495BABA4, Data Bitmask=0x0, Progress Indication=NULL(0),
Connection Handle=0
//25/74820328800C/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
//24/74820328800C/CCAPI/cc_api_call_proceeding:
Interface=0x49E07FD4, Progress Indication=NULL(0)
//24/74820328800C/CCAPI/cc_api_call_connected:
Interface=0x49E07FD4, Data Bitmask=0x1, Progress Indication=DESTINATION IS NON ISDN(2),
Connection Handle=0
//24/74820328800C/CCAPI/cc_api_call_connected:
Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
//24/74820328800C/CCAPI/ccCallModify:
Nominator=0x1000, Params=0x4A2E7368, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x18, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x18, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=15000(ms), Inter Digit Timeout=10000(ms))
//24/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID1=0x18, callID2=0x19, tag=0x0)
//24/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID1=0x18, gcid=0-0-0-0, tag=0x0)
//25/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:
(confID=0x4A2E757C, callID2=0x19, gcid=0-0-0-0, tag=0x0)
//24/74820328800C/CCAPI/ccConferenceCreate:
Conference Id=0x4A2E757C, Call Id1=24, Call Id2=25, Tag=0x0
//24/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:
Conference Id=0xD, Source Interface=0x49E07FD4, Source Call Id=24,
Destination Call Id=25, Disposition=0x0, Tag=0xFFFFFFFF
//25/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:
Conference Id=0xD, Source Interface=0x495BABA4, Source Call Id=25,
Destination Call Id=24, Disposition=0x0, Tag=0x0
//24/74820328800C/CCAPI/cc_generic_bridge_done:
Conference Id=0xD, Source Interface=0x495BABA4, Source Call Id=25,
Destination Call Id=24, Disposition=0x0, Tag=0x0
//24/74820328800C/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0xD, Destination Call Id=25)
//25/74820328800C/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0xD, Destination Call Id=24)
//24/74820328800C/CCAPI/cc_api_caps_ind:
Destination Interface=0x495BABA4, Destination Call Id=25, Source Call Id=24,
Caps(Codec=0x1, Fax Rate=0x1, Vad=0x1,
Modem=0x2, Codec Bytes=20, Signal Type=3)
//24/74820328800C/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
//25/74820328800C/CCAPI/cc_api_caps_ind:
Destination Interface=0x49E07FD4, Destination Call Id=24, Source Call Id=25,
Caps(Codec=0x4, Fax Rate=0x2, Vad=0x1,
Modem=0x0, Codec Bytes=20, Signal Type=2)
//25/74820328800C/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
//25/74820328800C/CCAPI/cc_api_caps_ack:
Destination Interface=0x49E07FD4, Destination Call Id=24, Source Call Id=25,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=OFF(0x1),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=9314)
//24/74820328800C/CCAPI/cc_api_caps_ack:
Destination Interface=0x495BABA4, Destination Call Id=25, Source Call Id=24,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=OFF(0x1),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=9314)
//24/74820328800C/CCAPI/cc_api_call_modify_done:
Result=0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_voice_mode_event:
Call Id=24
//24/74820328800C/CCAPI/cc_api_voice_mode_event:
Call Entry(Context=0x4A524790)
//24/74820328800C/CCAPI/cc_process_notify_bridge_done:
Conference Id=0xD, Call Id1=24, Call Id2=25
//24/74820328800C/CCAPI/ccSetDigitTimeouts:
Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms)
//24/74820328800C/CCAPI/ccSetDigitTimeouts:
Call Entry(Inter Digit Timeout=4000(ms), Initial Digit Timeout=4000(ms))
//24/74820328800C/CCAPI/ccRestartDigitTimeoutMsec:
Digit Timeout=0, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x18, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=24
//24/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x49E07FD4, callID=0x18, disp=0, digit_event=0x1, enable=TRUE, consume=FALSE)
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0x0, Interface=0x49E07FD4, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=4000(ms), Inter Digit Timeout=4000(ms))
ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x8097
Channel ID i = 0xA98381
Exclusive, Channel 1
//24/74820328800C/CCAPI/ccCallModify:
Nominator=0x1000, Params=0x4A2E6E68, Call Id=24
//24/74820328800C/CCAPI/cc_api_call_modify_done:
Result=0, Interface=0x49E07FD4, Call Id=24
ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x8097
Progress Ind i = 0x8088 - In-band info or appropriate now available
//24/74820328800C/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Ring Back,
Tone Direction=Network, Params=0x0, Call Id=24
//24/74820328800C/CCAPI/cc_handle_inter_digit_timer:
Generate inter-digit timeout CC_EV_CALL_DIGIT_END event
The following INBOUND call from the PSTN to 5126022001 fails and is supposed to be routing through Branch1 and is instead routing through CorpHQ. Please see 'DEBUG VOIP CCAPI INOUT'
CorpHQ#
//-1/A31ADF52800B/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=5126026222
cisco-anitype=4
cisco-aniplan=1
cisco-anipi=0
cisco-anisi=0
dest=5126022001
cisco-desttype=4
cisco-destplan=1
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=-1
cisco-rdnplan=-1
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
//-1/A31ADF52800B/CCAPI/cc_api_call_setup_ind_common:
Interface=0x49F42894, Call Info(
Calling Number=5126026222,(Calling Name=)(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed),
Called Number=5126022001(TON=Subscriber, NPI=ISDN),
Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
Incoming Dial-peer=1, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
//-1/A31ADF52800B/CCAPI/ccCheckClipClir:
In: Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed)
//-1/A31ADF52800B/CCAPI/ccCheckClipClir:
Out: Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed)
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:cc_get_feature_vsa malloc success
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
cc_get_feature_vsa count is 1
//-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
:FEATURE_VSA attributes are: feature_name:0,feature_time:1241383960,feature_id:13
//13/A31ADF52800B/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=5126026222(TON=Subscriber, NPI=ISDN, Screening=Not Screened, Presentation=Allowed),
Called Number=5126022001(TON=Subscriber, NPI=ISDN))
//13/A31ADF52800B/CCAPI/cc_process_call_setup_ind:
Event=0x497D0010
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 5126022001
//13/A31ADF52800B/CCAPI/ccCallSetContext:
Context=0x4A131A54
//13/A31ADF52800B/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 13 with tag 1 to app "_ManagedAppProcess_Default"
//13/A31ADF52800B/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
//13/A31ADF52800B/CCAPI/ccCallDisconnect:
Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
//13/A31ADF52800B/CCAPI/ccCallDisconnect:
Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
//13/A31ADF52800B/CCAPI/cc_api_get_transfer_info:
Transfer Number Is Null
//13/A31ADF52800B/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x49F42894, Tag=0x0, Call Id=13,
Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
//13/A31ADF52800B/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
:cc_free_feature_vsa freeing 49FE0410
//-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
vsacount in free is 0
PSTN#sh run
Building configuration...
Current configuration : 13975 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname PSTN
boot-start-marker
boot-end-marker
card type e1 0 0
card type t1 0 1
logging message-counter syslog
no aaa new-model
clock timezone EST -5
clock summer-time EST recurring
network-clock-participate wic 0
network-clock-participate wic 1
no network-clock-participate aim 0
dot11 syslog
ip source-route
ip cef
no ip domain lookup
ip domain name att.com
ip name-server 177.1.100.110
ip multicast-routing
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-ni
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol cisco
sip
bind control source-interface Loopback10
bind media source-interface Loopback10
header-passing
voice translation-rule 101
rule 1 /^\+.*/ //
rule 2 /^501.*/ //
rule 3 /^1206.*/ //
rule 4 /^00.*/ //
rule 5 /^0011.*/ //
rule 6 /^206/ /1206/
rule 7 /^1512.*/ /\0/
rule 8 /^011\(.*\)/ /\1/
voice translation-rule 102
rule 1 /^1\(2065015111\)$/ /\1/ type any subscriber plan any isdn
rule 2 /^1\(2065015555\)$/ /\1/ type any subscriber plan any isdn
rule 3 /^1\(2065015151\)$/ /\1/ type any subscriber plan any isdn
rule 4 /^1\(5126026222\)$/ /\1/ type any national plan any isdn
rule 5 /^31670357575$/ /&/ type any international plan any isdn
rule 6 /^31207037333$/ /&/ type any international plan any isdn
rule 7 /^31107047444$/ /&/ type any international plan any isdn
rule 8 /^911$/ /&/ type any unknown plan any unknown
rule 9 /^15126022.../ /&/ type any unknown plan any unknown
rule 10 /^31207033.../ /&/ type any unknown plan any unknown
rule 11 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 103
rule 1 /^206.*/ /&/ type any subscriber plan any isdn
rule 2 /^1/ // type any national plan any isdn
rule 3 /^00/ // type any international plan any isdn
voice translation-rule 201
rule 1 /^\+.*/ //
rule 2 /^602.*/ //
rule 3 /^1512.*/ //
rule 4 /^00.*/ //
rule 5 /^0011.*/ //
rule 6 /^512/ /1&/
rule 7 /^1206.*/ /&/
rule 8 /^011\(31.*\)/ /\1/
voice translation-rule 202
rule 1 /^1\(5126026222\)$/ /\1/ type any subscriber plan any isdn
rule 2 /^1\(2065015555\)$/ /\1/ type any national plan any isdn
rule 3 /^1\(2065015151\)$/ /\1/ type any national plan any isdn
rule 4 /^1\(2065015111\)$/ /\1/ type any national plan any isdn
rule 5 /^31670357575$/ /&/ type any international plan any isdn
rule 6 /^31207037333$/ /&/ type any international plan any isdn
rule 7 /^31107047444$/ /&/ type any international plan any isdn
rule 8 /^911$/ /&/ type any unknown plan any unknown
rule 9 /^12065011.../ /&/ type any unknown plan any unknown
rule 10 /^31207033.../ /&/ type any unknown plan any unknown
rule 11 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 203
rule 1 /^512.*/ /&/ type any subscriber plan any isdn
rule 2 /^1/ // type any national plan any isdn
rule 3 /^00/ // type any international plan any isdn
voice translation-rule 301
rule 1 /^\+.*/ //
rule 2 /^20.*/ //
rule 3 /^0\([1-8].*\)/ /31\1/
rule 4 /^011/ //
rule 5 /^0031/ //
rule 6 /^703..../ /3120&/
rule 7 /^00\(1.*\)/ /\1/
voice translation-rule 302
rule 1 /^31207037333$/ /7037333/ type any subscriber plan any isdn
rule 2 /^7033\(...\)$/ /0207033\1/
rule 3 /^911$/ /112/ type any unknown plan any unknown
rule 4 /^31\(670357575\)$/ /0\1/ type any national plan any isdn
rule 5 /^31\(107047444\)$/ /0\1/ type any national plan any isdn
rule 6 /^12065015555$/ /&/ type any international plan any isdn
rule 7 /^12065015151$/ /&/ type any international plan any isdn
rule 8 /^12065015111$/ /&/ type any international plan any isdn
rule 9 /^15126026222$/ /&/ type any international plan any isdn
rule 10 /^12065011...$/ /&/ type any unknown plan any unknown
rule 11 /^15126022...$/ /&/ type any unknown plan any unknown
rule 12 /^....$/ /&/ type any unknown plan any unknown
voice translation-rule 303
rule 1 /^703.*/ /&/ type any subscriber plan any isdn
rule 2 /^010/ // type any national plan any isdn
rule 3 /^1/ // type any international plan any isdn
voice translation-rule 1000
rule 1 /.*\(1...$\)/ /206501\1/
rule 2 /.*\(2...$\)/ /512602\1/
rule 3 /.*\(45..$\)/ /020757\1/
voice translation-rule 1001
rule 1 /^1206...5...$/ /+&/
rule 2 /^1512...6...$/ /+&/
rule 3 /^31.0...7...$/ /+&/
voice translation-profile 1-HQ-Change_DNIS-Check_ANI
translate called 101
voice translation-profile 1-HQ-Proper_Types
translate calling 102
translate called 103
voice translation-profile 2-BR1-Change_DNIS-Check_ANI
translate called 201
voice translation-profile 2-BR1-Proper_Types
translate calling 202
translate called 203
voice translation-profile 3-BR2-Change_DNIS-Check_ANI
translate called 301
voice translation-profile 3-BR2-Proper_Types
translate calling 302
translate called 303
voice translation-profile SIP-NORMALIZE-DNIS-ANI
translate calling 1001
translate called 1000
voice-card 0
dspfarm
archive
log config
hidekeys
controller E1 0/0/0
clock source internal
pri-group timeslots 1-3,16
description == Voice Circuit to Branch2
controller T1 0/1/0
clock source internal
cablelength long 0db
pri-group timeslots 1-3,24
description == Voice Circuit to CorpHQ
controller T1 0/1/1
clock source internal
cablelength long 0db
pri-group timeslots 1-3,24
description == Voice Circuit to Branch1
interface Loopback0
ip address 177.1.254.254 255.255.255.255
interface Loopback10
ip address 177.1.254.250 255.255.255.255
interface Loopback11
ip address 177.1.254.251 255.255.255.255
interface FastEthernet0/0
description ==TO INTERNET==
ip address 192.168.1.150 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description === To HQ
ip address 177.1.19.1 255.255.255.0
duplex auto
speed auto
interface Serial0/0/0:15
description == PRI Circuit to R3-BR2
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no isdn outgoing ie network-facility
isdn outgoing display-ie
no cdp enable
interface Serial0/1/0:23
description == PRI Circuit to R1-HQ
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
isdn negotiate-bchan
isdn outgoing display-ie
no cdp enable
interface Serial0/1/1:23
description == PRI Circuit to R2-BR1
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn protocol-emulate network
isdn incoming-voice voice
isdn supp-service name calling
isdn negotiate-bchan resend-setup
isdn outgoing ie network-facility
no cdp enable
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.1.1
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
control-plane
voice-port 0/0/0:15
translation-profile incoming 3-BR2-Change_DNIS-Check_ANI
description == Voice PRI to Branch2
voice-port 0/1/0:23
translation-profile incoming 1-HQ-Change_DNIS-Check_ANI
description == Voice PRI to CorpHQ
voice-port 0/1/1:23
translation-profile incoming 2-BR1-Change_DNIS-Check_ANI
description == Voice PRI to Branch1
dial-peer voice 1 pots
description == All inbound calls from HQ BR1 BR2 into PSTN
incoming called-number .
direct-inward-dial
dial-peer voice 101 pots
description == Subscriber Calls from PSTN into CorpHQ
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^2065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 102 pots
description == National Calls from PSTN into CorpHQ
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^12065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 103 pots
description == International Calls into CorpHQ from PSTN Coming from NL Ph
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern ^0012065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 104 pots
description == + Calls into CorpHQ from PSTN Coming from Mobiles
translation-profile outgoing 1-HQ-Proper_Types
preference 1
destination-pattern +12065011...$
direct-inward-dial
port 0/1/0:23
forward-digits 10
dial-peer voice 201 pots
description == Subscriber Calls from PSTN into Branch1
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^5126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 202 pots
description == National Calls from PSTN into Branch1
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^15126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 203 pots
description == International Calls into Branch1 from PSTN Coming from NL Ph
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern ^0015126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 204 pots
description == + Calls into Branch1 from PSTN Coming from Mobiles
translation-profile outgoing 2-BR1-Proper_Types
preference 1
destination-pattern +15126022...$
direct-inward-dial
port 0/1/1:23
forward-digits 10
dial-peer voice 301 pots
description == Subscriber Calls from PSTN into Branch2
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^7033...$
direct-inward-dial
port 0/0/0:15
forward-digits 7
dial-peer voice 302 pots
description == National Calls from PSTN into Branch2
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^0207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 10
dial-peer voice 303 pots
description == International Calls into Branch2 from PSTN Coming from US Ph
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^01131207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 304 pots
description == International Calls into Branch2 from PSTN Coming from US Ph
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern ^31207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 305 pots
description == + Calls into Branch2 from PSTN Coming from Mobiles
translation-profile outgoing 3-BR2-Proper_Types
destination-pattern +31207033...$
direct-inward-dial
port 0/0/0:15
forward-digits 9
prefix 0
dial-peer voice 1000 voip
description == Calls into AT&T SIP ITSP for VC Week1 Lab1
rtp payload-type nse 99
rtp payload-type nte 100
voice-class sip localhost dns:sip1.att.com
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 5000 voip
service aa
destination-pattern A5000
session target ipv4:177.1.254.254
incoming called-number A5000
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
num-exp 1888....... 911
num-exp 1900....... 911
num-exp 1976....... 911
num-exp 1777....... 911
num-exp 1444....... 911
num-exp 0800....... 911
num-exp 0900....... 911
sip-ua
telephony-service
no auto-reg-ephone
max-ephones 1
max-dn 10
ip source-address 177.1.254.254 port 2000
caller-id block code *67
system message You WILL PASS this Exam!
voicemail A5000
max-conferences 8 gain -6
call-forward pattern .T
dn-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Sep 01 2012 15:29:37
ephone-dn 1 dual-line
number 12065015111 secondary +12065015111
label Seattle, US +1 206 501 5111
description INE PSTN Phone
name Seattle US Phone
ephone-dn 2 dual-line
number 15126026222 secondary +15126026222
label Austin, US +1 512 602 6222
name Austin TX Phone
ephone-dn 3 dual-line
number 31207037333 secondary +31207037333
label Amsterdam, NL +31 20 703 73 33
name Amsterdam NL Phone
ephone-dn 4 dual-line
number 12065015555 secondary +12065015555
label Hurley Mobile +1 206 501 5555
name Hurley's Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 5 dual-line
number 12065015151 secondary +12065015151
label Hurley's Home +1 206 501 5151
name Hurley's Home
call-forward busy A5000
call-forward noan A5000 timeout 12
ephone-dn 6 dual-line
number 31670357575 secondary +31670357575
label Sawyer's Mobile +31 6 70357575
name Sawyer's Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 7 dual-line
number 911 secondary 112
label US/EU Emer/FreePhone/Prem
name Emergency Services
ephone-dn 8 dual-line
number 15126026262 secondary +15126026262
label BLinus Mobile +1 512 602 6262
name Benjamin Linus Mobile
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 9 dual-line
number 31207037373 secondary +31207037373
label DHume Home +31 20 703 73 73
name Desmond Hume Home
call-forward busy A5000
call-forward noan A5000 timeout 16
ephone-dn 10 dual-line
number 31107047444 secondary +31107047444
label Rotterdam, NL +31 10 704 74 44
name Rotterdam NL Phone
ephone 1
device-security-mode none
mac-address A456.3040.0DAA
type 7975
button 1:1 2:2 3:3 4:10
button 5:6 6o7,8,5,4
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 10
ntp server 64.90.182.55
end
CorpHQ#sh run
Building configuration...
Current configuration : 6353 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname CorpHQ
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone PST -8
clock summer-time PDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 177.1.11.1 177.1.11.14
ip dhcp excluded-address 177.1.11.21 177.1.11.254
ip dhcp excluded-address 177.2.11.1 177.2.11.14
ip dhcp excluded-address 177.2.11.21 177.2.11.254
ip dhcp pool CorpHQ-Phones
network 177.1.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.1.11.1
dns-server 177.1.100.110
ip dhcp pool Branch1-Phones
network 177.2.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.2.11.1
dns-server 177.1.100.110
no ip domain lookup
ip multicast-routing
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-ni
voice service voip
allow-connections h323 to h323
fax protocol cisco
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
no update-callerid
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice translation-rule 1
rule 1 // // type any subscriber plan any isdn
voice translation-rule 2
rule 1 // // type any national plan any isdn
voice translation-rule 3
rule 1 // // type any international plan any isdn
voice translation-rule 10
rule 1 /^[2-9].........$/ /9&/
rule 2 /^1[2-9].........$/ /9&/
rule 3 /^011/ /9&/
voice translation-profile MakeInternational
translate called 3
voice translation-profile MakeNational
translate called 2
voice translation-profile MakeSubscriber
translate called 1
voice translation-profile Prefix9_InFrom_CUCM
translate called 10
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller T1 0/0/0
pri-group timeslots 1-3,24
description == Voice Circuit to PSTN
interface Loopback0
ip address 177.1.254.1 255.255.255.255
ip pim dense-mode
interface FastEthernet0/0
description == To CorpHQ-Switch
no ip address
duplex auto
speed auto
interface FastEthernet0/0.10
description == Server VLAN
encapsulation dot1Q 10
ip address 177.1.10.1 255.255.255.0
ip pim dense-mode
interface FastEthernet0/0.11
description == Voice VLAN
encapsulation dot1Q 11
ip address 177.1.11.1 255.255.255.0
ip helper-address 177.1.10.10
ip nbar protocol-discovery
ip pim dense-mode
interface FastEthernet0/0.12
description == Data VLAN
encapsulation dot1Q 12
ip address 177.1.12.1 255.255.255.0
interface FastEthernet0/0.13
description == PSTN PHONE VLAN
encapsulation dot1Q 13
ip address 177.1.13.1 255.255.255.0
interface FastEthernet0/1
description === To PSTN
ip address 177.1.19.254 255.255.255.0
duplex auto
speed auto
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 36
cdp enable
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To BR1
bandwidth 384
ip address 177.0.101.1 255.255.255.0
ip pim dense-mode
snmp trap link-status
frame-relay interface-dlci 101
ip rsvp bandwidth 136
interface Serial0/1/0.2 point-to-point
description == FR To BR2
ip address 177.0.201.1 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 102
ip rsvp bandwidth 136
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 177.1.19.1
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0.10
no ip http server
no ip http secure-server
control-plane
voice-port 0/0/0:23
voice-port 0/3/0
voice-port 0/3/1
ccm-manager music-on-hold
sccp local Loopback0
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 1 register CorpHQ-729-MTP
associate profile 2 register CorpHQ-711-MTP
associate profile 3 register CorpHQ-HW-Xcode
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec ilbc
maximum sessions 2
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
translation-profile outgoing MakeSubscriber
destination-pattern 911
no digit-strip
port 0/0/0:23
dial-peer voice 11 pots
translation-profile outgoing MakeSubscriber
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
dial-peer voice 12 pots
translation-profile outgoing MakeNational
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
dial-peer voice 13 pots
translation-profile outgoing MakeInternational
destination-pattern 9011T
port 0/0/0:23
prefix 011
dial-peer voice 100 voip
description == Inbound/Outbound SIP PSTN GW From/To CUCM Pub
translation-profile incoming Prefix9_InFrom_CUCM
destination-pattern ^2065011...$
voice-class codec 1
session protocol sipv2
session target ipv4:177.1.10.10
incoming called-number .
ip qos dscp cs3 signaling
dial-peer hunt 1
sip-ua
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 2
ntp server 177.1.254.254
end
Branch1#sh run
Building configuration...
Current configuration : 3838 bytes
! Last configuration change at 01:19:02 CDT Thu Oct 10 2013
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname Branch1
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone CST -6
clock summer-time CDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
dot11 syslog
ip source-route
ip cef
ip multicast-routing
no ipv6 cef
ntp update-calendar
ntp server 177.1.254.1
multilink bundle-name authenticated
isdn switch-type primary-ni
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller T1 0/0/0
pri-group timeslots 1-3,24 service mgcp
interface Loopback0
ip address 177.1.254.2 255.255.255.255
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.11
description == Voice VLAN
encapsulation dot1Q 11
ip address 177.2.11.1 255.255.255.0
ip helper-address 177.1.254.1
ip pim dense-mode
interface FastEthernet0/0.12
description == Data VLAN
encapsulation dot1Q 12
ip address 177.2.12.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn supp-service name calling
isdn bind-l3 ccm-manager
isdn outgoing ie facility
isdn outgoing display-ie
isdn outgoing ie redirecting-number
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 37
cdp enable
no frame-relay inverse-arp
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To HQ
ip address 177.0.101.2 255.255.255.0
ip pim dense-mode
snmp trap link-status
frame-relay interface-dlci 101
ip rsvp bandwidth 136
interface Serial0/1/1
no ip address
shutdown
clock rate 2000000
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
no ip http server
no ip http secure-server
control-plane
voice-port 0/0/0:23
ccm-manager fallback-mgcp
ccm-manager redundant-host 177.1.10.10
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
mgcp
mgcp call-agent 177.1.10.20 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp fax t38 ecm
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
mgcp profile default
sccp local Loopback0
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 3 register Br1-HW-Xcode
associate profile 1 register Br1-729-MTP
associate profile 2 register Br1-711-MTP
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
scheduler allocate 20000 1000
end
Branch2#sh run
Building configuration...
Current configuration : 5789 bytes
! No configuration change since last restart
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
hostname Branch2
boot-start-marker
boot system flash:c2800nm-advipservicesk9-mz.124-24.T7.bin
boot system flash:
boot-end-marker
card type e1 0 0
logging message-counter syslog
no aaa new-model
clock timezone CEST 1
clock summer-time CEDT recurring
network-clock-participate wic 0
no network-clock-participate aim 0
dot11 syslog
ip source-route
ip cef
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
no supplementary-service h225-notify cid-update
fax protocol cisco
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice class custom-cptone JOIN-TONE
dualtone conference
frequency 300 3600
cadence 150 100 500
voice class custom-cptone LEAVE-TONE
dualtone conference
frequency 300 3600
cadence 500 100 150
voice translation-rule 1
rule 1 /^7033...$/ /020&/
voice translation-rule 10
rule 1 /^0/ /0&/
voice translation-rule 200
rule 1 /^206501...$/ /1&/
voice translation-profile 7DigitDNIS-to-10Digit
translate called 1
voice translation-profile Prefix0_InFrom_CUCM
translate called 10
voice translation-profile Prefix1-toCorpHQ-ANI
translate calling 200
voice-card 0
dsp services dspfarm
archive
log config
hidekeys
controller E1 0/0/0
pri-group timeslots 1-3,16
description == Voice Circuit to PSTN
controller E1 0/0/1
interface Loopback0
ip address 177.1.254.3 255.255.255.255
h323-gateway voip bind srcaddr 177.1.254.3
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.11
encapsulation dot1Q 11
ip address 177.3.11.1 255.255.255.0
ip helper-address 177.1.10.10
interface FastEthernet0/0.12
encapsulation dot1Q 12
ip address 177.3.12.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bchan-number-order ascending
no cdp enable
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 37
cdp enable
no frame-relay inverse-arp
frame-relay lmi-type ansi
ip rsvp bandwidth
interface Serial0/1/0.1 point-to-point
description == FR To HQ
ip address 177.0.201.2 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 102
ip rsvp bandwidth 136
interface Serial0/1/1
no ip address
shutdown
clock rate 2000000
interface Service-Engine1/0
no ip address
shutdown
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
control-plane
voice-port 0/0/0:15
translation-profile incoming 7DigitDNIS-to-10Digit
ccm-manager music-on-hold
sccp local Loopback0
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 4 register Br2-HW-Conf
associate profile 3 register Br2-HW-Xcode
associate profile 2 register Br2-711-MTP
associate profile 1 register Br2-729-MTP
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 4 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 1
conference-join custom-cptone JOIN-TONE
conference-leave custom-cptone LEAVE-TONE
associate application SCCP
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
dial-peer voice 10 pots
destination-pattern 112
no digit-strip
port 0/0/0:15
dial-peer voice 11 pots
destination-pattern 00[1-9]T
port 0/0/0:15
prefix 0
dial-peer voice 12 pots
translation-profile outgoing Prefix1-toCorpHQ-ANI
destination-pattern 000T
port 0/0/0:15
prefix 00
dial-peer voice 100 voip
description == Inbound/Outbound H323 PSTN GW From/To GK and CUCM Pub
translation-profile incoming Prefix0_InFrom_CUCM
destination-pattern 0207033...$
voice-class codec 1
session target ipv4:177.1.10.10
incoming called-number .
ip qos dscp cs3 signaling
dial-peer voice 101 voip
description == Outbound H323 PSTN GW To CUCM Sub
destination-pattern 0207033...$
voice-class codec 1
session target ipv4:177.1.10.20
ip qos dscp cs3 signaling
dial-peer hunt 1
telephony-service
max-ephones 1
max-dn 1
ip source-address 177.1.254.3 port 2000
max-conferences 8 gain -6
moh test.au
multicast moh 239.2.1.1 port 16384 route 177.1.254.3 177.3.11.1
transfer-system full-consult
create cnf-files version-stamp 7960 Sep 13 2013 18:55:27
line con 0
exec-timeout 0 0
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
login
scheduler allocate 20000 1000
ntp source Loopback0
ntp update-calendar
ntp server 177.1.254.1
end -
Dears,
I would like to know if is possible send data as DTMF Digits through an CVP application. We have an UCCE infraestructure installed on our customer, but they have a new project that will need transfer the call and data for an Avaya System. Is It possible? How can I do this?
Best regards!
AlessandroAddition to the UUI, You can also have a look at TakeBack-and-Transfer Section in the CVP Srnd which talks about your requirement, Below is the snippet i copied from the CVP SRND.
TNT (also known as Transfer Connect) is a transfer mechanism offered by some U.S. PSTN service
providers (such as AT&T and Verizon). With this transfer method, inband DTMF tones are outpulsed to
the PSTN by Unified CVP. These inband tones act as a signaling mechanism to the PSTN to request a
transfer to be completed. A typical DTMF sequence is *8xxxx, where xxxx represents a new routing label
that the PSTN understands. Upon detection of a TNT DTMF sequence, the PSTN drops the call leg to
the ingress gateway port and then re-routes the caller to a new PSTN location (such as a TDM ACD
location).
This behavior might be necessary for a customer with existing ACD site(s) but no IVR, who wants to
use Unified CVP initially as just an IVR. Over time, the customer might want to transition agents from
the TDM ACD(s) to Cisco Unified CCE and use Unified CVP as an IVR, queueing point, and transfer
pivot point (thus eliminating the need for TNT services).
In Unified CVP deployments with the ICM, the DTMF routing label outpulsed could have been a Unified
ICM translation routing label to enable passing of call data to another Unified ICM peripheral (such as
a TDM ACD). In this scenario, Unified CVP views the call as completed, and Unified CVP call control
is ended. With TNT, if the transfer to the termination point fails, there is nothing Unified CVP can do to
re-route the call. While some TNT services do have the ability to re-route the call back to Unified CVP,
Unified CVP sees this call as a new call.
Regards,
Senthil -
Cisco 2911 Voice Gateway SIP PSTN Calls Fail
Hello All,
I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway. 2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy. Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below). does anyone have any insight on how to correct this? Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call. Thanks in advance for any help!!
From: <sip:[email protected]>:tag=6166CDC4-882
To: <sip:[email protected]>
Shawn C. Smithi have same problem my cucm ip is 192.168.200.53
my Voice Gateway is SIP by ip 192.168.200.86 for internal
and 172.29.7.94
and my SIP Server is 10.208.9.69
if its oky can yuo take a look at my problem please
this is the syslog from debug
May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
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
Supported: Geolocation
Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
Session-Expires: 1800
P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=2217156
----- ccCallInfo IE subfields -----
cisco-ani=2217156
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=90555769123
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
Interface=0x30CF41D4, Call Info(
Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=90555769123(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 1
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=90555769123(TON=Unknown, NPI=Unknown))
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
Event=0x2B82D890
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 90555769123
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
Context=0x2ABC2E44
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=0555769123(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Aysar Mohamed
Account Number=2217156, Final Destination Flag=TRUE,
Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=2217156
----- ccCallInfo IE subfields -----
cisco-ani=2217156
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=0555769123
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 2
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
Context=0x2ABC1984
May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=802
May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
Interface=0x30CF41D4, Progress Indication=NULL(0)
May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1401481174
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Contact: <sip:[email protected]:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 328
Content-Type: application/sdp
v=0
o=- 17192647 17192647 IN IP4 10.208.9.69
s=SBC call
c=IN IP4 10.208.9.69
t=0 0
m=audio 39910 RTP/AVP 8 0 102 102 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=fmtp:116 0-15
a=fmtp:18 annexb=yes
May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
Modem=0x0, Codec Bytes=160, Signal Type=2)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event=170, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event=98, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
Cause Value=0
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
(confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
(confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
delay media to slow start case, codec negotation is not done
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0x16, Destination Call Id=466)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0x16, Destination Call Id=465)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
Conference Id=0x16, Call Id1=465, Call Id2=466
May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 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
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 233
v=0
o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
s=SIP Call
c=IN IP4 192.168.200.86
t=0 0
m=audio 18288 RTP/AVP 8 0 19
c=IN IP4 192.168.200.86
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
Cause Value=41, Interface=0x30CF41D4, Call Id=466
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=466
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
Conference Id=0x16, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: vsacount in free is 1
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=41
Content-Length: 0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: vsacount in free is 0
May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.7.94:5060 SIP/2.0
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>;tag=739BBC-1CE2
Date: Fri, 30 May 2014 20:19:36 GMT
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 446
v=0
o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
s=SIP Call
c=IN IP4 172.29.7.94
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.7.94
m=image 0 udptl t38
c=IN IP4 172.29.7.94
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
My SIP GW internal ip address is 192.168.200.86
and the Public IP is : 172.29.7.94
My CUCM is 192.168.200.53
my GW Config is :
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9\(\)/ /\1/
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
voice translation-profile OUT
translate called 3
voice translation-profile REDIAL
translate calling 5
voice translation-profile SIP-NEW
translate called 4
application
service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
license udi pid CISCO2921/K9 sn FCZ164960G0
hw-module pvdm 0/0
hw-module pvdm 0/1
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.200.86 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 172.29.7.94 255.255.255.252
duplex auto
speed auto
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 192.168.200.1
ip route 10.208.9.0 255.255.255.0 172.29.7.93
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register NAGHI-MTP
dspfarm profile 2 mtp
codec g711alaw
maximum sessions hardware 25
associate application SCCP
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 813 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 814 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 815 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 816 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 817 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 818 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Please i need ur help ASAP -
How does a voice gateway handle a call received from CUCM?
For example, I have a voice gateway configured via a SIP trunk as a device in CUCM. When our users dial the international pattern (8011!) and CUCM forwards that to the device(gateway) over the SIP trunk, how does the gateway handle that request in terms of matching dial-peers? Is it handled the same way a call incoming from the PSTN would be? By matching the destination pattern?
Ultimately what I'm trying to do is figure out whether our international calling is configured properly but that's difficult to do If I don't know what the gateway is doing with the call once it receives it from the CUCM.
Thanks in advance!Thanks, we have a conference call Monday and I'll try some debugging to see what's going on. I am a little more confused now if that's possible because my understanding is that they can make INTL calls but based off of the configs I've seen and what you've told me they should be able to. Below are my configs. There is a translation profile applied but it points to a blank rule. I also didn't see any transformations in the route pattern.
mgcp profile default
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
port 0/0/0:23
dial-peer voice 100 voip
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad
dial-peer voice 101 voip
destination-pattern 5...
session protocol sipv2
session target ipv4:x.x.x.x
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad
dial-peer voice 2 pots
destination-pattern 911
port 0/0/0:23
forward-digits all
dial-peer voice 3 pots
destination-pattern 9[94]11
port 0/0/0:23
forward-digits 3
dial-peer voice 5 pots
translation-profile outgoing NATL
destination-pattern 9[2-9]......
port 0/0/0:23
forward-digits 7
dial-peer voice 6 pots
translation-profile outgoing NATL
destination-pattern 9707[2-9]......
port 0/0/0:23
dial-peer voice 7 pots
translation-profile outgoing NATL
destination-pattern 91[2-9]..[2-9]......
port 0/0/0:23
forward-digits 11
dial-peer voice 8 pots
translation-profile outgoing INTL
destination-pattern 9011T
port 0/0/0:23
prefix 011
voice translation-rule 10
rule 1 // // type any international plan any unknown
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