10-26-2021 08:42 AM
I am new to the sip trunk configuration.
I have a sip trunk for FreePBX to make calls to PSTN via CUCME
I am having issues connecting to PSTN from my config despite matching dial-peers. See debug below
Debug
Oct 26 14:55:49.997: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=07032356321, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Oct 26 14:55:49.997: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=220
Oct 26 14:55:49.997: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
Oct 26 14:55:49.997: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=07032356321, Peer Info Type=DIALPEER_INFO_SPEECH
Oct 26 14:55:50.001: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=07032356321
Oct 26 14:55:50.001: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Oct 26 14:55:50.001: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=07032356321, saf_enabled=0, saf_dndb_lookup=1, dp_result=0
Oct 26 14:55:50.001: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=6
2: Dial-peer Tag=220
Oct 26 14:55:50.001: //-1/A4A620DD9623/DPM/dpMatchPeersCore:
Calling Number=, Called Number=07032356321, Peer Info Type=DIALPEER_INFO_SPEECH
Oct 26 14:55:50.001: //-1/A4A620DD9623/DPM/dpMatchPeersCore:
LEGACY_VOICE_ROUTER#debug voip dialpeer Match Rule=DP_MATCH_DEST; Called Number=07032356321
Oct 26 14:55:50.001: //-1/A4A620DD9623/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Oct 26 14:55:50.001: //-1/A4A620DD9623/DPM/dpMatchSafModulePlugin:
dialstring=07032356321, saf_enabled=1, saf_dndb_lookup=1, dp_result=0
Oct 26 14:55:50.001: //-1/A4A620DD9623/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=6
2: Dial-peer Tag=220
LEGACY_VOICE_ROUTER#no debug voip dialpeer
Oct 26 14:55:52.121: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=asterisk, Peer Info Type=DIALPEER_INFO_SPEECH
Oct 26 14:55:52.121: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=asterisk
Oct 26 14:55:52.121: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Oct 26 14:55:52.121: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
Oct 26 14:55:52.121: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Your help will be highly apprecated.
dial-peer voice 6 pots
corlist outgoing LOCAL-CALL
translation-profile outgoing PSTN-outgoing
destination-pattern 0..........
direct-inward-dial
port 0/0/0:15
forward-digits 11
dial-peer voice 220 VOIP
description incoming to CME
destination-pattern 0..........
session protocol sipv2
session target ipv4:172.16.80.24
dtmf-relay rtp-nte
codec g711ulaw
no vad
Solved! Go to Solution.
10-26-2021 09:55 AM
What is this dial-peer voice 220 VOIP used for ?
Why are you using same destination pattern for both pots and voip dial-peer. ?
If dial-peer 220 is for incoming remove destination and use incoming called command.
10-26-2021 01:41 PM - edited 10-26-2021 03:14 PM
dial-peer 220 is incoming from FreePBX
EGACY_VOICE_ROUTER#debug isdn q931
debug isdn q931 is ON.
LEGACY_VOICE_ROUTER#
Oct 26 21:33:38.260: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num asterisk
Oct 26 21:33:38.260: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x01F4 callID = 0x8175 switch = primary-net5 interface = User
Oct 26 21:33:38.260: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x01F4
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0180, 'asterisk'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '07032356321'
Plan:ISDN, Type:Unknown
LEGACY_VOICE_ROUTER#
Oct 26 21:33:38.364: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x81F4
Channel ID i = 0xA9839F
Exclusive, Channel 31
LEGACY_VOICE_ROUTER#
Oct 26 21:33:40.380: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x81F4
Progress Ind i = 0x8288 - In-band info or appropriate now available
LEGACY_VOICE_ROUTER#
Oct 26 21:33:55.492: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x81F4
Cause i = 0x829C - Invalid number format (incomplete number)
Oct 26 21:33:55.492: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x01F4
Oct 26 21:33:55.608: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x81F4
Oct 26 21:33:55.640: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num asterisk
Oct 26 21:33:55.640: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x01F5 callID = 0x8176 switch = primary-net5 interface = User
LEGACY_VOICE_ROUTER#
Oct 26 21:33:55.640: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x01F5
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0180, 'asterisk'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '07032356321'
Plan:ISDN, Type:Unknown
Oct 26 21:33:55.744: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x81F5
Channel ID i = 0xA9839F
Exclusive, Channel 31
LEGACY_VOICE_ROUTER#
Oct 26 21:33:57.744: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x81F5
Progress Ind i = 0x8288 - In-band info or appropriate now available
LEGACY_VOICE_ROUTER#
Oct 26 21:34:12.891: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x81F5
Cause i = 0x829C - Invalid number format (incomplete number)
Oct 26 21:34:12.891: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x01F5
Oct 26 21:34:13.011: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x81F5
Oct 26 21:34:13.067: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num asterisk
Oct 26 21:34:13.067: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x01F6 callID = 0x8177 switch = primary-net5 interface = User
LEGACY_VOICE_ROUTER#
Oct 26 21:34:13.067: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x01F6
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0180, 'asterisk'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '07032356321'
Plan:ISDN, Type:Unknown
Oct 26 21:34:13.195: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x81F6
Channel ID i = 0xA9839F
Exclusive, Channel 31
LEGACY_VOICE_ROUTER#
Oct 26 21:34:15.175: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x81F6
Progress Ind i = 0x8288 - In-band info or appropriate now available
LEGACY_VOICE_ROUTER#debug isdn q931
Oct 26 21:34:30.315: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x81F6
Cause i = 0x829C - Invalid number format (incomplete number)
Oct 26 21:34:30.315: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x01F6
Oct 26 21:34:30.411: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x81F6
Oct 26 21:34:30.623: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num asterisk
Oct 26 21:34:30.623: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x01F7 callID = 0x8178 switch = primary-net5 interface = User
10-26-2021 05:41 PM
ISP is rejecting the call, because you are not sending right calling informations. Most ISP's reject the call or translate if calling number is other than your DID block.
Calling Party Number i = 0x0180, 'asterisk'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '07032356321'
Plan:ISDN, Type:Unknown
LEGACY_VOICE_ROUTER#
Oct 26 21:33:38.364: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x81F4
Channel ID i = 0xA9839F
Exclusive, Channel 31
LEGACY_VOICE_ROUTER#
Oct 26 21:33:40.380: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x81F4
Progress Ind i = 0x8288 - In-band info or appropriate now available
LEGACY_VOICE_ROUTER#
Oct 26 21:33:55.492: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x81F4
Cause i = 0x829C - Invalid number format (incomplete number)
10-26-2021 09:55 AM
What is this dial-peer voice 220 VOIP used for ?
Why are you using same destination pattern for both pots and voip dial-peer. ?
If dial-peer 220 is for incoming remove destination and use incoming called command.
10-26-2021 01:41 PM - edited 10-26-2021 03:14 PM
dial-peer 220 is incoming from FreePBX
EGACY_VOICE_ROUTER#debug isdn q931
debug isdn q931 is ON.
LEGACY_VOICE_ROUTER#
Oct 26 21:33:38.260: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num asterisk
Oct 26 21:33:38.260: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x01F4 callID = 0x8175 switch = primary-net5 interface = User
Oct 26 21:33:38.260: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x01F4
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0180, 'asterisk'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '07032356321'
Plan:ISDN, Type:Unknown
LEGACY_VOICE_ROUTER#
Oct 26 21:33:38.364: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x81F4
Channel ID i = 0xA9839F
Exclusive, Channel 31
LEGACY_VOICE_ROUTER#
Oct 26 21:33:40.380: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x81F4
Progress Ind i = 0x8288 - In-band info or appropriate now available
LEGACY_VOICE_ROUTER#
Oct 26 21:33:55.492: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x81F4
Cause i = 0x829C - Invalid number format (incomplete number)
Oct 26 21:33:55.492: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x01F4
Oct 26 21:33:55.608: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x81F4
Oct 26 21:33:55.640: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num asterisk
Oct 26 21:33:55.640: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x01F5 callID = 0x8176 switch = primary-net5 interface = User
LEGACY_VOICE_ROUTER#
Oct 26 21:33:55.640: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x01F5
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0180, 'asterisk'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '07032356321'
Plan:ISDN, Type:Unknown
Oct 26 21:33:55.744: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x81F5
Channel ID i = 0xA9839F
Exclusive, Channel 31
LEGACY_VOICE_ROUTER#
Oct 26 21:33:57.744: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x81F5
Progress Ind i = 0x8288 - In-band info or appropriate now available
LEGACY_VOICE_ROUTER#
Oct 26 21:34:12.891: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x81F5
Cause i = 0x829C - Invalid number format (incomplete number)
Oct 26 21:34:12.891: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x01F5
Oct 26 21:34:13.011: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x81F5
Oct 26 21:34:13.067: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num asterisk
Oct 26 21:34:13.067: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x01F6 callID = 0x8177 switch = primary-net5 interface = User
LEGACY_VOICE_ROUTER#
Oct 26 21:34:13.067: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x01F6
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0180, 'asterisk'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '07032356321'
Plan:ISDN, Type:Unknown
Oct 26 21:34:13.195: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x81F6
Channel ID i = 0xA9839F
Exclusive, Channel 31
LEGACY_VOICE_ROUTER#
Oct 26 21:34:15.175: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x81F6
Progress Ind i = 0x8288 - In-band info or appropriate now available
LEGACY_VOICE_ROUTER#debug isdn q931
Oct 26 21:34:30.315: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x81F6
Cause i = 0x829C - Invalid number format (incomplete number)
Oct 26 21:34:30.315: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x01F6
Oct 26 21:34:30.411: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x81F6
Oct 26 21:34:30.623: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num asterisk
Oct 26 21:34:30.623: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x01F7 callID = 0x8178 switch = primary-net5 interface = User
10-26-2021 05:41 PM
ISP is rejecting the call, because you are not sending right calling informations. Most ISP's reject the call or translate if calling number is other than your DID block.
Calling Party Number i = 0x0180, 'asterisk'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '07032356321'
Plan:ISDN, Type:Unknown
LEGACY_VOICE_ROUTER#
Oct 26 21:33:38.364: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x81F4
Channel ID i = 0xA9839F
Exclusive, Channel 31
LEGACY_VOICE_ROUTER#
Oct 26 21:33:40.380: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x81F4
Progress Ind i = 0x8288 - In-band info or appropriate now available
LEGACY_VOICE_ROUTER#
Oct 26 21:33:55.492: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x81F4
Cause i = 0x829C - Invalid number format (incomplete number)
10-26-2021 10:38 PM
From the shared information it would seem that you use your dial peers for both inbound and outbound direction. This is absolutely a valid thing, but IMHO is not really the best option to use as it makes it harder to see the direction of calls. For this it’s better to have a clear differentiation between inbound and outbound dial peers, where one would be used for the inbound call leg and another would be used for the outbound call leg per service type, aka pots or voip.
For your problem as such I agree with @Nithin Eluvathingal that you get a rejection of the call from your service provider. Fix the calling number sent by voice translation rule(s) on your gateway or by other means available in CM for this if you prefer that.
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