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

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SIP ITSP on CUCM 10.5.2 (No CUBE) Incoming calls fail, outgoing are fine

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?
Thanks

Thanks 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

My phone will not allow me to receive incoming or outgoing calls

My phone will allow me to text and do everything else, accept allow me to answer incoming calls, I can phyically anser it' but there is no sound on either ends. Then when I try to make a call it dials the number- but I am not able to hear the person on the other line- nor are they able to hear me. Any advice would be greatly appreciated- I already turned it on and off.
Thank you

This almost always is a hardware failure that requires replacement. See all the suggesetions in this article, especially the last.
iOS: Wi-Fi settings grayed out or dim

Microphone not working on my iphone 4s unless i'm on speaker or using headphones during calls both incoming and outgoing?

During calls incoming and outgoing I can hear the other person but they cannot hear me unless I am on speaker phone or am using headphones?

Basic troubleshooting steps clearly outlined in the User Guide are restart, resest, restore from backup, restore as new.
If you still have problems after going through ALL the recommended troubleshooting steps, then you likely have a hardware issue. You'll need to bring your phone into Apple for evaluation.

SIP trunk incoming and outgoing calls issue

Hi Everyone,
We recently installad a SIP trunk and terminated on CUBE and CUCM but we have issues on incoming and outgoing calls, When someone dial in from outside he keeps listening the dailing ring even after we pick up the phone and at the end the callers time exipres and call gets disconnected.
For Dailing out, the dialed number rings and caller hear the dailing ring as well but if someone pick the phone it apprears that call is connected but no audio in it, dead air.
Our call flow is as
IP Phones => CUCM --->SIPTRUNK--->CUBE=>SIPTRUNK=>SP
I have attached the config for CUBE and debug ccsip messages output for both incoming and outgoing calls.
Please if some help in sorting out this issue, Thanks in Advance
Tasneem

Inbound call>>>>
The reason you are experiencing this is that your CUBE is requesting PRACK and your provider is not responding to it..
Here we have your cube sending 180 ringing with "Require 100rel"..This was sent several times and your ITSP didnt respond probably because they do not support 100rel...(It is Huwaei after all, they do what they like)
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Date: Thu, 16 Jan 2014 13:31:34 GMT
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Require: 100rel
RSeq: 2507
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "TEST STC" ;party=called;screen=yes;privacy=off
Contact:
Record-Route:
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Length: 0
AFter the CUBE didnt get any response, it then replied with Gateway Timeout...
Jan 16 13:31:54.550: //31880/5A4406E48184/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 504 Gateway Timeout
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Reason: Q.850;cause=102
Content-Length: 0
I suggest you disable this parameter..and test again
voice service voip
sip
rel1xx disable
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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.
Thanks

I 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..

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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I've installed ios 7.0.6 on my iphone 4 and lost voice on outgoing and incoming calls. My simm is ok, i've checked it, I've done a full clean and restore. Still no calls. Please help.

Hi there,
I've just installed ios 7.0.6 on my iphone 4 and now lost voice on outgoing and incoming calls.It rings on incoming and I'm told rings on the recipient's 'phone, but then no vocal communication...its just dead. My simm is ok, i've checked it, I've done a full clean and restore of the 'phone. Still no calls.
I've not caused any physical or water damage to the 'phone. I would welcome any help/suggestions to get my calls back asap.
Please help.
Miss liberty

Have you tried restarting or resetting your iPhone?
Restart: Press On/Off button until the Slide to Power Off slider appears, select Slide to Power Off and, after It shuts down, press the On/Off button until the Apple logo appears.
Reset: Press the Home and On/Off buttons at the same time and hold them until the Apple logo appears (about 10 seconds).
Also consider deleting and reinstalling the Mail Account in question.

CUCME 8.6 Call not forwarding Voicemail

Hi frieds,
     In our office we are using CUCME 8.6 on Cisco 2951 and unity express 8.5 in ISM module. As per our configuration whenever user is busy or not answering , the call will forward to voicemail. Totally we have 24 PSTN line. So we have an additional gateway 2901. The Issue I’m facing is that, when a PSTN incoming call coming through the second gateway(2901), if the extension is busy or not answering the call is disconnecting instead of forwarding to voicemail.
My 2951 configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
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
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
registrar server.
Dial peer we are using for voice mail:
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad.
2901 Configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
ipv4 172.16.19.82
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
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad
============================
Debug CCSIP Calls
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0xAF40FD8
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 5000
Called Number            : 1099
Source IP Address (Sig ): 172.16.19.80
Destn SIP Req Addr:Port :
Destn SIP Resp Addr:Port :
Destination Name         :
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 172.16.19.80
Source IP Port    (Media): 25364
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 47
Disconnect Cause (SIP)   : 200
For your reference I here attach a network diagram
What the command which I missed?

Check License status on your CUE, I had same issue.. Finally figured out its about license.. sh license status
Sent from Cisco Technical Support iPhone App

I just bought this iPhone 4S second hand and during incoming and outgoing calls I can't hear the other person talk unless it's on speaker or if I have headphones in. I don't know what to do I've turned up my volume and checked my settings!! Need help

I Just bought a second hand iPhone 4S I am having issues with the hearing during incoming and outgoing calls, I have turned the volume up and checked my settings but I still have trouble hearing the other person speak. It works if it's on speaker or if I have headphones plugged in. I need help desperately!!!!! Any suggestions on what to do???

Hey Itsanaasaunders,
Thanks for the question. I understand that you are experiencing issues hearing sound while on a call (receiver wise, not speakerphone). The following resource provides troubleshooting steps for this symptom:
If you can't hear a person on a call or voicemail or if the sound quality is poor on iPhone - Apple Support
http://support.apple.com/en-us/HT203800
Clear the receiver
The iPhone receiver won't work properly if it's blocked or dirty. Here's where the receiver is:
Follow these steps, testing after each:
- Make sure nothing is blocking the receiver, such as a case or film.
If you have a new iPhone, remove the plastic film on the front and back of the device.
- Check the receiver opening to see if it's blocked or dirty. If necessary, clean the receiver opening with a clean, small, dry, soft-bristled brush.
- While you're on a call, turn on speakerphone. If you still can't hear, then your network or reception could be the issue. Try calling again later or from a different location.
Restart and update
1. Restart your iPhone and test again.
2. If there's still no sound or poor sound quality, update to the latest version of iOS.
Get more help
If you've tried these steps and there's still no sound or poor sound quality from your iPhone, contact Apple Support.
Thanks,
Matt M.

CUCME Not Incoming Calls, Outgoing calls ok

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