No incoming calls CUCM
I am trying to configure a CUCM with a SIP trunk to a 2811 and a voice GW to my SIP trunk provider.
CUCM8.6 <SIP>2811<SIP> Callcentric.
I am able to make outgoing calls but am failing miserably with incoming.
I suspect it is my incoming dial peer.
The incoming calls hit my 2811 but do not seem to go to my CUCM.
I have attached an output from my "debug ccsip calls"
Anything help would be greatly appreciated.
Robert,
surely that is not all the sip debug information, I am missing the INVITES, TRYING etc SIP messages, can you re-attach and maybe also debug your dial peers to see what gets hit (if anything at all ) when making an inbound call.
Cheers
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Similar Messages
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CUCME Not Incoming Calls, Outgoing calls ok
Hello everybody,
i am configuring a CUCME with SIP trunk, i can make calls to outside but i can´t recieve any from outside, this is my second time a configure with SIP
i´ve used the command debug voice dialpeer all to check was going on, but i can´t find the problem.
this is my config:
ip host sip-server A.B.C.D
voice service voip
ip address trusted list
ipv4 A.B.C.D 255.255.255.252
voice translation-rule 1
rule 1 /325277\(\)/ /1\1/
voice translation-profile IN
translate called 1
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming IN
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
ephone-dn 1
number 100
description RECEPTION
ephone 2
mac-address AAAA.BBBB.CCCC
ephone-template 1
type 7942
keep-conference
button 1:1
NOTE: IP Address are hidden, just for security
These are the output of my debug/tests:
#test voice translation-rule 1 32527700
Matched with rule 1
Original number: 32527700 Translated number: 100
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=32527700, Called Number=32527700, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=32527700
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=32527700, Expanded String=32527700, Calling Number=32527700T
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=32527700, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=59513212, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=59513212
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=59513212T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=59513212
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=59513212T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:exit@6704
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Calling Number=59513212, Called Number=32527700, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_VIA_URI; URI=sip:A.B.C.D:5060
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected]:5060;user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected];user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected];user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=32527700
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=32527700, Expanded String=32527700, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=1 Is Matched
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:exit@6655
Can Anyone help me???
Thanks in Advance!!!Thanks, these are the output
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:32527700@(WAN):5060;user=phone SIP/2.0
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
From: ;tag=6e8b9968-CC-25
To:
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Max-Forwards: 70
Supported: 100rel,timer
User-Agent: Huawei SoftX3000 V300R601
Session-Expires: 300
Min-SE: 90
Contact:
Content-Length: 376
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)
s=Sip Call
c=IN IP4 (SIP_SERVER)
t=0 0
m=audio 11554 RTP/AVP 8 0 18 4 2 98 98 98
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:98 G726-32/8000
a=rtpmap:98 G726-24/8000
a=ptime:20
a=fmtp:18 annexb=no
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=32527700, Called Number=32527700, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=32527700
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=32527700, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=59513212, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpAssociateIncomingPeerCore:
Calling Number=59513212, Called Number=32527700, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 422 Session Timer too small
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
From: ;tag=6e8b9968-CC-25
To: ;tag=4CD1E84-2094
Date: Wed, 29 Jan 2014 22:53:19 GMT
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Min-SE: 1800
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Content-Length: 0
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:32527700@(WAN):5060;user=phone SIP/2.0
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
From: ;tag=6e8b9968-CC-25
To: ;tag=4CD1E84-2094
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
*Jan 29 16:53:31: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:(SIP_SERVER):5060 SIP/2.0
Via: SIP/2.0/UDP (WAN):5060;branch=z9hG4bK3B11F0F
From: ;tag=4CD4D7C-1634
To:
Date: Wed, 29 Jan 2014 22:53:31 GMT
Call-ID: 3017EE62-885411E3-80B4FEFC-CAA82B4A
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Max-Forwards: 70
Timestamp: 1391036011
CSeq: 66 REGISTER
Contact:
Expires: 3600
Supported: path
Content-Length: 0
*Jan 29 16:53:31: //973/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP (WAN):5060;branch=z9hG4bK3B11F0F
Call-ID: 3017EE62-885411E3-80B4FEFC-CAA82B4A
From: ;tag=4CD4D7C-1634
To: ;tag=f2056e8e
CSeq: 66 REGISTER
Content-Length: 0
I´ve replaced the IP Adress for (SIP_SERVER) / (WAN) / SIP_SERVER_INTERNAL
Thank you -
CUCM 7 - Route incoming call to a specific VM
Hello Forum,
Hope this is an easy one for you guys.
I am trying to accomplish the following and not sure if its possible.
Ex.
Incoming call to ext. 1111
Ext. 1111 gets added to 2 phones, then call rings simultaneously on said 2 phones
After 5 rings and no answer, it rolls over to a different already existing VM (ext 8888)
Any help would be appreciated as I am not sure where to change the settings to make this happen.
Thanks in advance,
LSAssuming x1111 does not need a dedicated voicemail, you could do the following:
Ensure that x1111 has the correct VM profile and set the CFB/CFNA settings to forward to VM as usual. On the mailbox for 8888, set an alternate extension of 1111.
Hailey
Please rate helpful posts! -
Hi,
I am in the process of upgrading a customer who is on 8.0.3. They have an ITSP terminating SIP Trunk directly on the CCM Server
I upgraded the system to 10.5.2. During cutover I was able to make outgoing calls but all incoming calls were failing.
After reverting back to the old system, everything is working fine again, and I dont understand what could be the possible issue that it doesnt work on 10.5.2 but it works well on 8.0.3.
I checked almost everything and dont find anything that stands out, which may be contributing to the issue.
Any idea what could be missing here?
ThanksThanks for all your tips.
It was turned out that, the URI was a FQDN and during the first install of the 8.0.3 (in the sandbox) I had not bothered to get the DNS Services replicated and then didnt check if the ITSP was sending the invite on URI based on FQDN or IP Address
Thanks -
Switched over to MGCP from H323, no incoming calls fast busy
Hello, I'm on the network side crossing over to the Voice side. We replaced a 3825 Voice Router at a branch office with a 2921. The 3825 was setup with a T1 and had a PRI connected to the FXO ports. The 2921 is now connected via fiber and TAC helped get the router registered to the Call Manager. I'm trying to match up the old dial peers on the new router. I can't make out going or receive incoming calls, I get a This Call Can't be completed at this time.
When it was on the T1, the branch office was using H323. Now that's connects to the same CUCM, it's on MGCP. Shouldn't the old Dial Peers work on the new router?I had to configure the FXO ports for the DN to route the incoming calls. I learned just because the Call Manager configured the Voice Gateway as a client, you still must configure the FXO ports to route the main DID to a DN on the LAN.
Great advice, learning alot about telephony and VoIP.
Wish I had more experience troubleshooting the CUCM and DID portion, also learned the phone compnay doesn't turn up their ISDN switch until you configure your PBX. And a PRI testing isn't the same as the data T1, it's about having the Signaling channel turned up and or configured.
Lessons learned, wished I would have crossed over to VoIP earlier. -
Agent Desktop Error When receive Incoming Call
Hi, I have CAD Ver 8.5.1.39 | CUCM Ver 8.5.1.10000-26 | UCCX Ver 8.5.1.10000-37 running on my network now.
there some problem occurs when I put an Agent into the Line Group with Top Down Distribution call(actualy all the Distribution call type face the same problem).
1. When the Agent Receive Incoming Call from outside(PSTN) sometimes the CAD don't Show Up and show no response but the CIPC work normaly. showed in Not Responded.jpg
2. When the Agent Receive Call and it is show up. I can't answer the call. When I press ctrl-A or i push the answer icon i got the error message show in error.jpg , sometimes when i want to make a call the same problem happens
3. When the Agent Pop Up and the caller end the Call before I can answer. The call got stuck in Agent and Agent cannot make a call. Agent must logout and login again to fix this problem, show on the stuck.jpg
4. When the problem 3 occurs and if the Agent don't fix this problem, the stack got more caller. show in stuck2.jpg
I dont know why this happen, all the call come from outside(PSTN). when I test call from internal the problem doesn't occur.
Please help me. Any help will be helpful
Thank you for your help and participation
Regards,
Yopie.Hi,
I have an auto attendant, for example like this:
Welcome to the XYZ Company,
Press 1 for purchasing (10 Agent)
Press 2 for Consult (5 Agent)
Or press 0 for operator help (3 Agent)
When the consumer press 1, I want the call distribute to 10 Agent in the Purchasing team.
To make scenario like this I use the hunt pilot and to use hunt pilot i must have line group first.
So I need to put the 10 Agent into the Line Group.
Thanks. -
Matching B-channel for dialpeer assigment (Incoming call)
Is it possible to manipulate a specific B Channel to use a dialpeer? For example on a T1 with 24 channels, I need the last 4 B-channels to choose an specific dialpeer, same concepts as matching incoming DID's but on a B-Channel slot.
If a call arrives on B-channel 20, I need that call to be sent to a specific DN at the CallManager.
I am using H.323 on the gateway side.
thanks in advanced.
OscarHi!
I'm stuck in a similar situation at the moment, and hope someone has solved this issue.
I have a "back to back connection" between two PBX'es that communicate using Q.931. I have had to replace my old hardware running a legacy CCS solution due to its incapability of understanding overlap signalling. I have replaced it with two Cisco routers running E1 (Q931). Between the units I have a high latency low bandwidth network.
The issue is that the PBX'es are configured in a way that requires calls between the PBX'es to use the same time slot at both ends. I'm running a VoIP network with CUCM between the two sites.
=============================================
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-21
trunk-group TS01 timeslots 1
trunk-group TS02 timeslots 2
trunk-group TS03 timeslots 3
trunk-group TS04 timeslots 4
trunk-group TS05 timeslots 5
trunk-group TS06 timeslots 6
trunk-group TS07 timeslots 7
trunk-group TS08 timeslots 8
trunk-group TS09 timeslots 9
trunk-group TS10 timeslots 10
trunk-group TS11 timeslots 11
trunk-group TS12 timeslots 12
trunk-group TS13 timeslots 13
trunk-group TS14 timeslots 14
trunk-group TS15 timeslots 15
trunk-group TS17 timeslots 17
trunk-group TS18 timeslots 18
trunk-group TS19 timeslots 19
trunk-group TS20 timeslots 20
trunk-group TS21 timeslots 21
dial-peer voice 81030001 pots
trunkgroup TS01
description ** PBX TS01 **
translation-profile incoming 61031401
translation-profile outgoing 25
destination-pattern 81030001T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
dial-peer voice 81030002 pots
trunkgroup TS02
description ** PBX TS02 **
translation-profile incoming 61031402
translation-profile outgoing 25
destination-pattern 81030002T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
voice translation-rule 25
rule 1 /^810300../ //
voice translation-rule 61031401
rule 1 // /61031401\1/
voice translation-rule 61031402
rule 1 // /61031402\1/
voice translation-profile 25
translate called 25
voice translation-profile 61031401
translate called 61031401
voice translation-profile 61031402
translate called 61031402
=====================================
The idea is that I do not know (or care) what numbers are used as SOURCE or DESTINATION of the original call. My network should be transparent to the PBX number plan. I need to add a prefix, and it should be based on the timeslot the call comes in on. I route the traffic between the routers using the prefix.
The configuration excerpt above should add 61031401 prefix to all calls entering on TS01, and 61031402 to all calls entering on TS02 etc. Calls from the remote should have corresponding prefixes 81030001 for TS01 and 81030002 for TS02 etc.
The outbound (from voip to pots) routing of the above configuration works.
However I have a challenge with the incoming prefixing.
All calls inbound end up using "dial-peer 81030001 pots".
I believe the reason this dial-peer "takes" all of the calls inbound from pots is due to the line "incoming called-number ."
Removing this makes no inbound pots call work as the "destination-pattern 8103001T" is never matched.
Removing "destination-pattern 8103001T" from the dial-peer is not working as it kills the voip to pots routing of inbound calls from the remote router.
Anyone got a good idea for me? -
Incoming called URI number manipulation in Call manager 10.5
Dear Experts,
can we manipulate the called URI number like we manipulate the digits (e.g Translation pattern) ?
can we have manipulate the incoming called number to match a route pattern
for eg. the called uri is 955XXXX@CUCM-address
route pattern 955XXXX ==> Voice Gateway
thanks for your help in advance
Anasbecause it should match a route pattern not Directory ? it always return 404 not found
it comes as a URI because it is use SIP trunk to reach the CUCM -
How can I evenly split incoming calls between 2 employees without a shared voicemail?
I need to load balance a low volume of calls between 2 DNs.
Incoming calls need to be evenly split between Employee1 and Employee2. If they don’t answer the call, it needs to go to their personal voicemail, not a common voicemail account.
I think the Hunt pilot and line group is what I want, and I can utilize a circular distribution, but my only hurdle is voicemail. I can get voicemail to go to a single account, but I need a method to simply split incoming calls to 2 employees, and (if no answer), drop into their personal voicemail.
I cannot figure this out, any suggestions?
I’m running CUCM 9.02, Unity Connection 9.02, and UCCX v10 (I mention UCCX as a tool I could leverage, Employee1 and Employee2 are not agents or part of a call center)
Thanks for any assistance.Queuing is not enabled. I have no desire to queue the calls.
In the Hunt Pilot settings:
When I chose “Do Not Forward Unanswered Calls”, calls drop after (what appears to be) the Line Group’s RNA Reversion Timeout is met. Never goes to voicemail.
The same when I select “Use Forward Settings of Line Group Member”.
If I select “Forward Unanswered Calls to” and manually select my VM pilot DN, missed calls finally go to Unity Connection, but not to any voicemail account. If I add the Hunt Pilot number as an alternate extension to Employee1’s unity account, then all unanswered calls will drop into Employee1’s personal account, even the calls missed by Employee2. This is not the desired solution. I need Employee2 ‘s voicemail account to answer the calls missed by Employee2 as well as Employee1’s voicemail account to answer the calls missed by Employee1.
This brings me back to my original question. Is there a way to get this to work with Hunt pilots and line groups, or should I be trying something else? I just need calls to DN:111 split 50/50 between 2 DNs (DN for Employee1, and DN for Employee2), and have missed calls drop into the personal voicemail of the DN that missed the call.
I’m hoping someone who has gotten this to work can share their Hunt Pilot and Line Group settings.
Thanks again. -
Incoming Calls via CUBE to IVR- intermittent garbled audio
Connecting via an ITSP and incoming calls are:
ITSP--SIP Trunk-- MyCUBE--SIP Trunk--CUCM--IVR(CTI Ports and RPs)
|
7900 Type A sccp Handsets
INbound calls to sccp phones direct sound OK.
Outbound calls sound OK
Inbound calls to IVR intermittent voice quality issues (outside caller hears garbled message)
QoS reservations are adequate and no issues with that (after hours with v low traffic and closed call centrestill have this issue)
CUBE is 3825 and tried 15.1M6, 7 and 2.
All devices and trunk in same region= HQ (tried SIP trunk in Hub_none with same results). intra and inter Region has 64k per call
CUBE has XCode, Soft and Hard MTPs, Conf resources registered with CUCM 8.6
When incoming call comes in via ISDN to CUBE, no issues.
Using G711alaw and G729 r8 in CUBE dial peers. DTMF works fine.No VAD.
Also, when some calls present to agent via IVR, sometimes the agent with 7900 hears silence whilst external caller hears the agent fine. Inbound calls direct to agent DID work 100%. I have noticed on the dead air calls to agent, the ptime = 0 on the handset for Rx... CUBE inbound/oubound Dial peers are matching g711a so I am not sure what is issue.The problem was solved with the following commands:
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
modem country v12 belgium
The connections are stable now.
Best regards
Thomas -
SIP incoming call with G722-64 codec not working
Hi, Guys.
Have setup cube sip trunk to ITSP, incoming and outgoing calls are working. Except for an incoming call with g722 codec and video h263 (just need voice call). The called number does not even ring. The caller informed that his using polycom phone.
Also, itsp provided 10 numbers for testing in which we can assigned to our phones but only the main number is working. When doing an incoming call, (dialing the other numbers except from the main number) can see always on the logs that itsp is always feeding the main number. I think it was because of the configuration under the sip-ua (register the maint number to a registrar) but itsp informed that it was also their setup for other clients and is working. Appreciate your help on these.
ThanksI have looked at your logs and here are my observations..
1. When you disabled fast start on CUCM, I asked you to enable early offer on your CUBE, however I dont see this in your logs..
This is the INVITE sent to your ITSP, as you can see, this doesnt contain any SDP, that suggest you are doing delayed offer..
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC862F
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 0011462194-3037647155-0083893506-2887478836
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1396510389
Contact: <sip:[email protected]:5060>
History-Info: <sip:[email protected]:5060>;index=1,<sip:[email protected]:5060>;index=2
Expires: 300
Allow-Events: telephone-event
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
Content-Length: 0
2. If you are doing DO, then your CUBE needs to send an answer to what your ITSP is offering in its ACK..but this is not happening
Here is what I see..Your CUBE sends SDP in its PRACK
Sent:
PRACK sip:[email protected]2.147.134.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC9232B
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
CSeq: 103 PRACK
RAck: 323009643 102 INVITE
Allow-Events: telephone-event
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]2.147.134.21:5060;transport=udp",response="a9d772d988ec971cdad556fd4a992bd0",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="58621262",qop=auth,algorithm=MD5,nc=00000002
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 293
v=0
o=CiscoSystemsSIP-GW-UserAgent 1079 7198 IN IP4 172.21.8.134
s=SIP Call
c=IN IP4 172.21.8.134
t=0 0
m=audio 17082 RTP/AVP 8 96 100
c=IN IP4 172.21.8.134
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=ptime:20
###Here is your ACK to the 200 OK from ITSP###
On the ACK...Your CUBE doesnt include any SDP in its ACK, hence your ITSP disconnected the call immediately
ACK sip:[email protected]2.147.134.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKCA1335
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
017151: Apr 3 07:33:11.089 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 202.147.134.21:5060;branch=z9hG4bKahsb3f108gbhq8pbn5k1sdj0gkbf0.1
From: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
To: <sip:[email protected]>;tag=AE7F464-1B0F
Call-ID: [email protected]
CSeq: 323009054 BYE
Max-Forwards: 9
Content-Length: 0
I have two suggestions..
1. Downgrade or upgrade your CUBE IOS. Something is not quite right with this behaviour
2. Send your full sh run
On your inbound call issue, you need to send me the logs for a call to another of your DDI.. -
Cisco Mobile 8.0 No incoming calls to iPhone
I am trying to use cisco mobile 8.0 iphone, I am able to register the phone with CUCM, also able to call other SCCP device connected with CUCM. However when call get connected, I can’t hear anything from SCCP device but the SCCP device can hear the voice from Iphone(SIP device). And also I can’t make outgoing call to iphone(SIP) number from SCCP device. Am I missing something?
I am using CUCM 8.0.2 and iphone 4 with 8.0.2
ThanksThe one-way loss of audio is a routing or firewall issue. Check to ensure that the path from the SCCP phone is not obstructed and routable to the iPhone.
As for incoming calls to the iPhone, is the program running in the foreground? The current release does not support multitasking. -
Dropped Incoming calls after 20 seconds, Reason: Q.850;cause=86
I have the following scenario for incoming calls:
PSTN ---- E1 ---> Digium Gateway --- SIP ---> Router 2921 ------SIP ----> CUCM
All incoming calls from the PSTN get dropped after 20 seconds. All outgoing calls to the PSTN work fine..
The Router 2921 generates a BYE message with Reason: Q.850;cause=86 to the Digium Gateway.
Attached is the debug, show run and a packet capture with all the SIP messages..
If someone could please give me to a solution.
Thanks!Hello Carlos,
Here is the analysis of the traces....
Timestamp Node / Interface Device IP Direction Protocol Message Name TCP Handle / From Tag Call Ref / ID
05/09/2014 16:52:36.608 172.17.0.6 172.16.50.10 In SIP INVITE as12f8f5d4 [email protected]
05/09/2014 16:52:36.620 172.17.0.6 172.17.0.3 Out SIP INVITE A424AFA0-1D7B [email protected]
05/09/2014 16:52:36.620 172.17.0.6 172.16.50.10 Out SIP 100 Trying as12f8f5d4 [email protected]
05/09/2014 16:52:36.624 172.17.0.6 172.17.0.3 In SIP 100 Trying A424AFA0-1D7B [email protected]
05/09/2014 16:52:36.632 172.17.0.6 172.17.0.3 In SIP 180 Ringing A424AFA0-1D7B [email protected]
05/09/2014 16:52:36.632 172.17.0.6 172.16.50.10 Out SIP 180 Ringing as12f8f5d4 [email protected]
05/09/2014 16:52:39.140 172.17.0.6 172.17.0.3 In SIP 200 OK A424AFA0-1D7B [email protected]
05/09/2014 16:52:39.144 172.17.0.6 172.17.0.3 In SIP SUBSCRIBE 844851~97a97412-a383-4ad3-9458-162b2737f313-47291175 [email protected]
05/09/2014 16:52:39.148 172.17.0.6 172.17.0.3 Out SIP ACK A424AFA0-1D7B [email protected]
05/09/2014 16:52:39.148 172.17.0.6 172.17.0.3 Out SIP 200 OK 844851~97a97412-a383-4ad3-9458-162b2737f313-47291175 [email protected]
05/09/2014 16:52:39.148 172.17.0.6 172.17.0.3 Out SIP NOTIFY A424AFA0-1D7B [email protected]
05/09/2014 16:52:39.152 172.17.0.6 172.17.0.3 In SIP 200 OK A424AFA0-1D7B [email protected]
05/09/2014 16:52:39.152 172.17.0.6 172.16.50.10 Out SIP 200 OK as12f8f5d4 [email protected]
05/09/2014 16:52:39.652 172.17.0.6 172.16.50.10 Out SIP 200 OK as12f8f5d4 [email protected]
05/09/2014 16:52:40.652 172.17.0.6 172.16.50.10 Out SIP 200 OK as12f8f5d4 [email protected]
05/09/2014 16:52:42.652 172.17.0.6 172.16.50.10 Out SIP 200 OK as12f8f5d4 [email protected]
05/09/2014 16:52:46.652 172.17.0.6 172.16.50.10 Out SIP 200 OK as12f8f5d4 [email protected]
05/09/2014 16:52:50.652 172.17.0.6 172.16.50.10 Out SIP 200 OK as12f8f5d4 [email protected]
05/09/2014 16:52:54.652 172.17.0.6 172.16.50.10 Out SIP 200 OK as12f8f5d4 [email protected]
05/09/2014 16:52:58.652 172.17.0.6 172.17.0.6 Out SIP BYE A424AFAC-1EC1 [email protected]
05/09/2014 16:52:58.652 172.17.0.6 172.17.0.3 Out SIP BYE A424AFA0-1D7B [email protected]
The 200 Ok is not being acknowledged by the Digium Gateway . Hence the call is dropped .
Please check why the "Digium" side is not sending an ACK for this transaction.
Hope this helps!
Regards,
Karthik Sivaram -
Calling ID is Unknown Number in Incoming Calls
Dear All
i have CUCM 7.1 and h323 gateway , my issue is :
Some times , not always , the calling Id is giving " Unknown Number " in incoming calls
what is the problem and how to solve it ?
thanksdo you see the calling number provided by ITSP in the GW for those faulty calls?
what is the connection to pstn? isdn pri? if so, collect debug isdn q931 to check whether or not the itsp provides the caller id? It could be issue with provider too.
Please rate all the useful posts -
Prefixing a 9 and 91 to incoming calls from SIP provider for callback
I am wondering what would be the best options for prefixing a 9 or 91 to incoming calls over a sip connection to allow callback from missed calls and recieved calls. The setup is
callmanager 7.1.5 >>>>sip trunk>>>>>>>>>>>>CUBE>>>>>>>>>>>sip to ITSP
I am thinking voice translation rules is the only option for this? any configuration examples for this would be greatly appreciated.
would this work?
voice-translation rule 1
rule 1 // /9/
voice-translation profile prefix_9
translate calling 1
dial-peer voice 101 voip
destination-pattern ???????...$
voice-class codec 1
session protocol sipv2
session target ipv4: to callmanager
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 1001 voip
translation profile incoming prefix_9
destination-pattern T
session protocol sipv2
session target ipv4: to sip provider
incoming called-number ???????...$
dtmf-relay rtp-nteYour config should work fine, except your profile is only applied to one dial-peer, make sure you apply it to the one that is used to redirect the call to CUCM.
Also, you did not mention what country you are in, but if this is US you may want to prefix 91 to national calls as carriers don't provide 9 as part of the CLID delivery, also what about your international calls, you may would be more explicit in your first rule to match for national digit string and then have another rule for international.
HTH,
Chris
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