Re: Cisco Gateway does not accept an invite from astriesk

yarob.fathi · ‎08-10-2018

Dears,

My cisco gateway is not allowing a call from astriesk FPBX to pass to PSTN due to:

SIP/2.0 400 Bad Request - 'Malformed/Missing '

Although the first invite looks good to me, It sends a 100 trying then a 500 internal error,then SIP/2.0 400 Bad Request - 'Malformed/Missing URL'

Is there a way to manipulate the invite or what could be a possible solution.

yarob.fathi · ‎08-10-2018

Received:

INVITE sip:0911111135@10.198.199.1 SIP/2.0

Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK2ed821ba

Max-Forwards: 70

From: "Simplify" <sip:107@10.198.199.2>;tag=as5a972df5

To: <sip:0911111135@10.198.199.1>

Contact: <sip:107@10.198.199.2:5060>

Call-ID: 1c4366e523b2462910f1105657e41880@10.198.199.2:5060

CSeq: 102 INVITE

User-Agent: FPBX-14.0.3.1(13.19.1)

Date: Thu, 09 Aug 2018 23:27:29 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 323

v=0

o=root 1292608259 1292608259 IN IP4 10.198.199.2

s=Asterisk PBX 13.19.1

c=IN IP4 10.198.199.2

t=0 0

m=audio 14328 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv

Then the cisco gateway trys:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK2ed821ba

From: "Simplify" <sip:107@10.198.199.2>;tag=as5a972df5

To: <sip:0911111135@10.198.199.1>

Date: Thu, 09 Aug 2018 22:41:32 GMT

Call-ID: 1c4366e523b2462910f1105657e41880@10.198.199.2:5060

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M3

Content-Length: 0

Then oops:

Sent:

SIP/2.0 500 Internal Server Error

Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK2ed821ba

From: "Simplify" <sip:107@10.198.199.2>;tag=as5a972df5

To: <sip:0911111135@10.198.199.1>;tag=3C0AABE0-F

Date: Thu, 09 Aug 2018 22:41:32 GMT

Call-ID: 1c4366e523b2462910f1105657e41880@10.198.199.2:5060

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M3

Reason: Q.850;cause=6

Content-Type: multipart/mixed;boundary=uniqueBoundary

Mime-Version: 1.0

Content-Length: 281

--uniqueBoundary

Content-Type: application/x-q931

Content-Disposition: signal;handling=optional

Content-Length: 11

.Z

--uniqueBoundary

Content-Type: application/gtd

Content-Disposition: signal;handling=optional

RLC,

PRN,isdn*,,NET5*,

--uniqueBoundary--

Received:

ACK sip:0911111135@10.198.199.1 SIP/2.0

Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK2ed821ba

Max-Forwards: 70

From: "Simplify" <sip:107@10.198.199.2>;tag=as5a972df5

To: <sip:0911111135@10.198.199.1>;tag=3C0AABE0-F

Contact: <sip:107@10.198.199.2:5060>

Call-ID: 1c4366e523b2462910f1105657e41880@10.198.199.2:5060

CSeq: 102 ACK

User-Agent: FPBX-14.0.3.1(13.19.1)

Content-Length: 0

Received:

INVITE sip:10.198.199.1 SIP/2.0

Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK42c0a200

Max-Forwards: 70

From: "Simplify" <sip:107@10.198.199.2>;tag=as19a72c16

To: <sip:10.198.199.1>

Contact: <sip:107@10.198.199.2:5060>

Call-ID: 59241e8e291c6ef31cad42d33492282b@10.198.199.2:5060

CSeq: 102 INVITE

User-Agent: FPBX-14.0.3.1(13.19.1)

Date: Thu, 09 Aug 2018 23:27:29 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 321

v=0

o=root 227366592 227366592 IN IP4 10.198.199.2

s=Asterisk PBX 13.19.1

c=IN IP4 10.198.199.2

t=0 0

m=audio 10728 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv

Sent:

SIP/2.0 400 Bad Request - 'Malformed/Missing URL'

Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK42c0a200

From: "Simplify" <sip:107@10.198.199.2>;tag=as19a72c16

To: <sip:10.198.199.1>;tag=3C0AABE8-E6B

Date: Thu, 09 Aug 2018 22:41:32 GMT

Call-ID: 59241e8e291c6ef31cad42d33492282b@10.198.199.2:5060

CSeq: 102 INVITE

Allow-Events: telephone-event

Reason: Q.850;cause=100

Server: Cisco-SIPGateway/IOS-15.2.4.M3

Content-Length: 0

Sent:

SIP/2.0 400 Bad Request - 'Malformed/Missing URL'

Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK42c0a200

From: "Simplify" <sip:107@10.198.199.2>;tag=as19a72c16

To: <sip:10.198.199.1>;tag=3C0AABE8-E6B

Call-ID: 59241e8e291c6ef31cad42d33492282b@10.198.199.2:5060

CSeq: 102 INVITE

Reason: Q.850;cause=100

Content-Length: 0

ryanticer · ‎08-11-2018

Could you attach the gateway config? (sanitize passwords and any public IPs out)

Ryan

YAARUB F. MILAD · ‎08-11-2018

I copied the voice related config,should you need anything else please ask, thank you for your concern

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

signaling forward unconditional

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

bind control source-interface GigabitEthernet0/1.30

bind media source-interface GigabitEthernet0/1.30

!

voice-port 0/1/1:15

connection plar 107

!

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

no mgcp explicit hookstate

!

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

port 0/1/1:15

forward-digits all

!

dial-peer voice 2 voip

destination-pattern 1..

session protocol sipv2

session target ipv4:10.198.199.2

session transport udp

incoming called-number .

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3 pots

destination-pattern 09........

port 0/1/1:15

!

gateway

timer receive-rtp 1200

!

sip-ua

no remote-party-id

retry invite 3

retry response 3

retry bye 3

retry cancel 3

timers trying 1000

sip-server ipv4:10.198.199.2

ryanticer · ‎08-11-2018

Nothing major looks off to me in the config. It's one of the URLs it isn't happy about - I think it might be the contact header, but not 100% certain. Next step would be to enable some debugging to see what the gateway doesn't like:

no logging monitor

debug ccsip messages

debug ccsip error

logging buffered 1000000 debug

clear log

undebug all

show log

Post the show log output here.

YAARUB F. MILAD · ‎08-11-2018

*Aug 11 20:06:32.204: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0911111135@10.198.199.1 SIP/2.0
Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK3bb682c2
Max-Forwards: 70
From: "107" <sip:107@10.198.199.2>;tag=as13fb16c7
To: <sip:0911111135@10.198.199.1>
Contact: <sip:107@10.198.199.2:5060>
Call-ID: 4cdf282a5cbcf327601bc9be1c34c1fb@10.198.199.2:5060
CSeq: 102 INVITE
User-Agent: FPBX-14.0.3.1(13.19.1)
Date: Sat, 11 Aug 2018 20:52:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 352

v=0
o=root 253679241 253679241 IN IP4 10.198.199.2
s=Asterisk PBX 13.19.1
c=IN IP4 10.198.199.2
t=0 0
m=audio 13644 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

*Aug 11 20:06:32.204: //-1/E022D3B89F58/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:
MF: Not a Forked SIP leg..
SIP: (6753) Attribute mid, level 1 instance 1 not found.
*Aug 11 20:06:32.204: //6753/E022D3B89F58/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry
*Aug 11 20:06:32.204: //6753/E022D3B89F58/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry
*Aug 11 20:06:32.204: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count:
Unable to set CHANNEL_COUNT for callid 6753
*Aug 11 20:06:32.204: //6753/E022D3B89F58/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo:
Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.
*Aug 11 20:06:32.208: //6753/E022D3B89F58/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK3bb682c2
From: "107" <sip:107@10.198.199.2>;tag=as13fb16c7
To: <sip:0911111135@10.198.199.1>
Date: Sat, 11 Aug 2018 20:06:32 GMT
Call-ID: 4cdf282a5cbcf327601bc9be1c34c1fb@10.198.199.2:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Content-Length: 0

*Aug 11 20:06:32.208: ISDN Se0/1/1:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num 107
*Aug 11 20:06:32.208: ISDN Se0/1/1:15 Q931: Sending SETUP callref = 0x00A3 callID = 0x8024 switch = primary-net5 interface = User
*Aug 11 20:06:32.208: ISDN Se0/1/1:15 Q931: TX -> SETUP pd = 8 callref = 0x00A3
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, '107'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '0911111135'
Plan:ISDN, Type:Unknown
*Aug 11 20:06:32.228: ISDN Se0/1/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x80A3
Cause i = 0x8286 - Channel unacceptable
*Aug 11 20:06:32.232: //-1/xxxxxxxxxxxx/SIP/Msg/sipDisplayBinaryData:
Sending: Binary Message Body
*Aug 11 20:06:32.232: Content-Type: application/x-q931
08 02 00 A3 5A 08 02 82 86 0D 0A
*Aug 11 20:06:32.232: //6753/E022D3B89F58/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK3bb682c2
From: "107" <sip:107@10.198.199.2>;tag=as13fb16c7
To: <sip:0911111135@10.198.199.1>;tag=45C97E04-163A
Date: Sat, 11 Aug 2018 20:06:32 GMT
Call-ID: 4cdf282a5cbcf327601bc9be1c34c1fb@10.198.199.2:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Reason: Q.850;cause=6
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 281

--uniqueBoundary
Content-Type: application/x-q931
Content-Disposition: signal;handling=optional
Content-Length: 11
.#Z

--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional

RLC,
PRN,isdn*,,NET5*,

--uniqueBoundary--

*Aug 11 20:06:32.232: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0911111135@10.198.199.1 SIP/2.0
Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK3bb682c2
Max-Forwards: 70
From: "107" <sip:107@10.198.199.2>;tag=as13fb16c7
To: <sip:0911111135@10.198.199.1>;tag=45C97E04-163A
Contact: <sip:107@10.198.199.2:5060>
Call-ID: 4cdf282a5cbcf327601bc9be1c34c1fb@10.198.199.2:5060
CSeq: 102 ACK
User-Agent: FPBX-14.0.3.1(13.19.1)
Content-Length: 0

*Aug 11 20:06:32.236: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:10.198.199.1 SIP/2.0
Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK3376609c
Max-Forwards: 70
From: "107" <sip:107@10.198.199.2>;tag=as61c8de24
To: <sip:10.198.199.1>
Contact: <sip:107@10.198.199.2:5060>
Call-ID: 4733837e3b5a139716e7f1cb134b8cc8@10.198.199.2:5060
CSeq: 102 INVITE
User-Agent: FPBX-14.0.3.1(13.19.1)
Date: Sat, 11 Aug 2018 20:52:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 352

v=0
o=root 888189949 888189949 IN IP4 10.198.199.2
s=Asterisk PBX 13.19.1
c=IN IP4 10.198.199.2
t=0 0
m=audio 10138 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

*Aug 11 20:06:32.236: //-1/E027B5919F5E/SIP/Error/sipSPIGetUrlInfo:
Unknown/unsupported URL schema
*Aug 11 20:06:32.236: //-1/E027B5919F5E/SIP/Error/sact_idle_new_message_invite:
Invalid URL in incoming INVITE
*Aug 11 20:06:32.236: //-1/E027B5919F5E/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:
MF: Not a Forked SIP leg..
*Aug 11 20:06:32.236: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK3376609c
From: "107" <sip:107@10.198.199.2>;tag=as61c8de24
To: <sip:10.198.199.1>;tag=45C97E08-223F
Date: Sat, 11 Aug 2018 20:06:32 GMT
Call-ID: 4733837e3b5a139716e7f1cb134b8cc8@10.198.199.2:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Content-Length: 0

*Aug 11 20:06:32.736: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK3376609c
From: "107" <sip:107@10.198.199.2>;tag=as61c8de24
To: <sip:10.198.199.1>;tag=45C97E08-223F
Call-ID: 4733837e3b5a139716e7f1cb134b8cc8@10.198.199.2:5060
CSeq: 102 INVITE
Reason: Q.850;cause=100
Content-Length: 0

*Aug 11 20:06:32.736: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:10.198.199.1 SIP/2.0
Via: SIP/2.0/UDP 10.198.199.2:5060;branch=z9hG4bK3376609c
Max-Forwards: 70
From: "107" <sip:107@10.198.199.2>;tag=as61c8de24
To: <sip:10.198.199.1>;tag=45C97E08-223F
Contact: <sip:107@10.198.199.2:5060>
Call-ID: 4733837e3b5a139716e7f1cb134b8cc8@10.198.199.2:5060
CSeq: 102 ACK
User-Agent: FPBX-14.0.3.1(13.19.1)
Content-Length: 0

ryanticer · ‎08-11-2018

The 500 error appears to be caused by the next hop circuit refusing the call. The call is being refused due to a "channel unacceptable" error, which roughly translates to a 500 error via SIP back to source:

*Aug 11 20:06:32.228: ISDN Se0/1/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x80A3
Cause i = 0x8286 - Channel unacceptable

Once that error is resolved, we can take another look at the 400 error (if it still exists). I recommend checking the settings (or working with corresponding admin to check settings) of the system on the other end of the ISDN circuit to make sure settings are aligned.

Ryan

R0g22 · ‎08-13-2018

What is the ISDN channel selection set to ? It should typically be default as in bottom up. Telcos usually do top-down to prevent race around conditions. Also, ensure you don't have a full PRI configured when only a partial is provisioned by your Telco or vice-versa.

YAARUB F. MILAD · ‎11-16-2018

This issue was solved, It was a provider issue.