CME & Asterisk SIP Trunk (Only Works One Way)

Arash Barakizade · ‎02-11-2017

Hi all,

i must say im new with cisco voip solution,but this is the situation

i have a Cisco 2800 router CME in one site,and one Elastix on the other site,which these two different networks and sites can comminucate through an IPsec tunnel.

i managed to create a Sip trunk in elastix with these configuration.

host=192.168.56.65
type=peer
context=192.168.56.65
allow=g729&ulaw
nat=no
qualify=yes
disallow=all

and this is my CME Configuration

dial-peer voice 891 voip
description To-CME
destination-pattern 4..
session protocol sipv2
session target ipv4:192.168.1.6:5060
dtmf-relay rtp-nte
codec g711ulaw
no vad
!

elastix extensions as you can figure out start with 4,and CME extensions start with 1 and 2.

the point is with this configuration elastix extensions can dial and call CME extensions with no problem,but this works only one way

#csim start 407
csim: called number = 407, loop count = 1 ping count = 0

csim err csimDisconnected recvd DISC cid(32539)
csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0

this is when i try to call elastix extensions

Call 1
SIP Call ID : 00308420412d7b8e49710a0d7b4b2a41@192.168.1.6:5060
State of the call : SIP_STATE_OPTIONS_WAIT (27)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : Unknown
Called Number :
Bit Flags : 0x40000C 0x104 0x10
CC Call ID : 32542
Source IP Address (Sig ): 192.168.56.65
Destn SIP Req Addr:Port : [192.168.1.6]:0
Destn SIP Resp Addr:Port: [192.168.1.6]:5060
Destination Name : 192.168.1.6
Number of Media Streams : 1
Number of Active Streams: 0
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_IDLE
Stream Call ID : -1
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: [192.168.56.65]:0

Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1

and this is when i call from elastix to CME

Call 1
SIP Call ID : 17af6dea02cea03c6a1b7f3b672523a7@192.168.1.6:5060
State of the call : STATE_SENT_ALERTING (14)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : AutoPorsche
Called Number : 219
Bit Flags : 0xC0001C 0x300 0x404
CC Call ID : 32548
Source IP Address (Sig ): 192.168.56.65
Destn SIP Req Addr:Port : [192.168.1.6]:5060
Destn SIP Resp Addr:Port: [192.168.1.6]:5060
Destination Name : 192.168.1.6
Number of Media Streams : 1
Number of Active Streams: 0
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ADDING
Stream Call ID : -1
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [192.168.56.65]:18608
Media Dest IP Addr:Port : [192.168.1.6]:12718

Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1

any ideas?

Mohammed al Baqari · ‎02-11-2017

Hi,

One 2800 router can you enable 'debug ccsip mess' then initiate a test call. Please share the output to analyze further.

Arash Barakizade · ‎02-12-2017

sh logging
Syslog logging: enabled (0 messages dropped, 45 messages rate-limited,
148 flushes, 0 overruns, xml disabled, filtering disabled)

No Active Message Discriminator.

Inactive Message Discriminator:

80.85.10 msg-body drops 80.85.100.38

Console logging: level debugging, 161425 messages logged, xml disabled,
filtering disabled
Monitor logging: level debugging, 0 messages logged, xml disabled,
filtering disabled
Buffer logging: level debugging, 160095 messages logged, xml disabled,
filtering disabled
Logging Exception size (4096 bytes)
Count and timestamp logging messages: disabled
Persistent logging: disabled

No active filter modules.

ESM: 0 messages dropped

Trap logging: level informational, 3239 message lines logged
Logging to 192.168.50.4 (udp port 514, audit disabled,
authentication disabled, encryption disabled, link up),
3239 message lines logged,
0 message lines rate-limited,
0 message lines dropped-by-MD,
xml disabled, sequence number disabled
filtering disabled

Log Buffer (4096 bytes):
ers:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20035
2: Dial-peer Tag=20054
Feb 12 08:04:14.079: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=141, Peer Info Type=DIALPEER_INFO_SPEECH
Feb 12 08:04:14.079: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=141
Feb 12 08:04:14.079: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Feb 12 08:04:14.079: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20035
2: Dial-peer Tag=20054
Feb 12 08:04:16.483: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.168.56.65 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK3f95aa54
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.6>;tag=as79b85cb0
To: <sip:192.168.56.65>
Contact: <sip:Unknown@192.168.1.6:5060>
Call-ID: 2beac08b0a77e00774b80bd90ffe09d4@192.168.1.6:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Sun, 12 Feb 2017 09:05:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Feb 12 08:04:16.487: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Feb 12 08:04:16.487: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
Feb 12 08:04:16.487: //-1/AEE6C723B68D/DPM/dpAssociateIncomingPeerCore:
Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Feb 12 08:04:16.487: //-1/AEE6C723B68D/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
Feb 12 08:04:16.491: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Date: Sun, 12 Feb 2017 08:04:16 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: "Unknown" <sip:Unknown@192.168.1.6>;tag=as79b85cb0
Allow-Events: telephone-event
Supported: 100rel,resource-priority,replaces,sdp-anat
Content-Length: 172
To: <sip:192.168.56.65>;tag=16D63984-433
Content-Type: application/sdp
Call-ID: 2beac08b0a77e00774b80bd90ffe09d4@192.168.1.6:5060
Accept: application/sdp
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK3f95aa54
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS

v=0
o=CiscoSystemsSIP-GW-UserAgent 8394 7363 IN IP4 192.168.56.65
s=SIP Call
c=IN IP4 192.168.56.65
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.56.65

Feb 12 11:34:22: %ISDN-6-CONNECT: Interface Serial0/2/1:30 is now connected to 09112336572 N/A
Feb 12 08:04:22.435: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=00, Peer Info Type=DIALPEER_INFO_SPEECH
Feb 12 08:04:22.435: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=00
Feb 12 08:04:22.435: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Feb 12 08:04:22.435: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=NO_MATCH(-1)
Feb 12 08:04:22.435: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=9809112336572, Peer Info Type=DIALPEER_INFO_SPEECH
Feb 12 08:04:22.435: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=9809112336572
Feb 12 08:04:22.435: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Feb 12 08:04:22.435: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=402
2: Dial-peer Tag=401
3: Dial-peer Tag=409
Feb 12 08:04:24.759: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
jaK
Feb 12 08:04:24.779: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
jaK

Shri2 · ‎02-12-2017

Arash,

Logs seems to be incorrect.

Can you make test call during off business hour and collect ccsip and ccapi logs with below step.

enter below command.

debug ccsip messages

debug voip ccapi inout

term mon

make a test call and collect the logs. Once done attach the logs along with running config and calling/called number.