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SIP Trunk - Calls not working - SIP Disconnect 400

Level 1
Level 1


I am trying to get a SIP trunk working but am unable to make or recieve calls. Our setup is:

SIP Trunk (Provider) ---> SIP Gateway (Me) -----> Cisco Gateway (Me) --- CUCM

The SIP gateway is a cisco 2900 device and is completely separate from our network. Its job is to send and recieve calls on the SIP trunk. There is a ISDN QSIG Pri line configured from this SIP gateway to another of our voice gateways then to deliver calls to the IP phones via MGCP.

Anyway, the SIP trunk is not working. Below is a copy of my config (some of it). I also attached a copy of the calls logs. Can someone help me interpret the SIP call logs to try to narrow down why the call fails?

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  registrar server
  asserted-id pai
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw



controller E1 0/1/0
framing NO-CRC4
pri-group timeslots 1-31
description :: E1 PRI link to VOICE GW

dial-peer voice 1000 voip
translation-profile outgoing OutBound_CallerID_SIP
destination-pattern .T
session protocol sipv2
session target
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 100 pots
destination-pattern 358.........
incoming called-number .
port 0/1/0:15
dial-peer voice 101 pots
description **Dial-Peer to match Incoming calls from SIP TRUNK**
destination-pattern 12345...
incoming called-number .
port 0/1/0:15
dial-peer voice 102 pots
description **Dial-Peer to match Incoming calls from SIP TRUNK**
destination-pattern 35812345...
incoming called-number .
port 0/1/0:15
dial-peer voice 1001 voip
translation-profile outgoing OutBound_CallerID_SIP

description "TEST DIAL PEER"
destination-pattern 0044.........
session protocol sipv2
session target
codec g711alaw

Here is the CCSIP call log failure:

The Call Setup Information is:
Call Control Block (CCB) : 0x40ADE120
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +358xxxxxxxxx
Called Number : 003xxxxxxxxxxx
Source IP Address (Sig ): 10.xx.xx.xx
Destn SIP Req Addr:Port : 10.xx.xx.214:5060
Destn SIP Resp Addr:Port : 10.xx.xx.214:5060
Destination Name : 10.xx.xx.214

*Dec 3 09:22:44 UTC: //12659/3E5136E58059/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): 10.xx.xx.xx
Source IP Port (Media): 17650
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

*Dec 3 09:22:44 UTC: //12659/3E5136E58059/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 127
Disconnect Cause (SIP) : 400

4 Replies 4

Vivek Batra
VIP Alumni
VIP Alumni

Is it the call going from gateway to service provider? If yes, there is something inline in INVITE message which your service provider didn't like and hence rejecting with cause 400 Bad Request. At this point of time, I can only doubt on the number you're sending in From field. Are you sure is that +358xxxxxxxxx the number starts with + which service provider is expecting. Please try removing + and see if it makes any difference.

- Vivek

Turns out it was a bug in IOS that was causing the SIP connection to fail to the provider:

Bug Search


Unexpected RTP PayloadType :255 in SDP Body



OPTIONS message create questionable error traces.

handling inbound options.


Customer Visible

Add Notification

Save Bug

Open Support Case

View Bug in CDETS

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How this bug is impacting your call flow as I could not see 255 PT in SDP.

- Vivek

Level 7
Level 7

Is your SIP provider on a Public IP address or can you reach it via a Private Address?

What Codecs do the support?

As Vivek says, are they expecting the From number to have the + at the start?