SIP Trunk trouble

Unanswered Question
Apr 25th, 2007

I need help with getting a sip trunk working. I can get it to working use g711ulaw. When I change the dialpeer to only use g729. I get the following message on my debug ccapi errors.

Apr 25 21:56:14.444: //237880/9F0FD67CA1BE/SIP/Error/sipSPI_ipip_copy_sdp_to_ch

nnelInfo:

ailed to update call entry

Apr 25 21:56:14.448: //237881/9F0FD67CA1BE/SIP/Error/sipSPI_ipip_copy_channelIn

o_to_sdp:

ilter mis-match, failing call

Apr 25 21:56:14.448: //237881/9F0FD67CA1BE/SIP/Error/sipSPIAddSDPMediaPayload:

all Origination Failed: None of the selected codec from CLI is supported by SIP

Apr 25 21:56:14.448: //237881/9F0FD67CA1BE/SIP/Error/sipSPIOutgoingCallSDP: Err

r with codec types on media line

Apr 25 21:56:14.448: //237881/9F0FD67CA1BE/SIP/Error/sipSPICreateOutboundSDP: E

ror in creating an SDP for the outbound call - Check for supported codecs

Apr 25 21:56:14.448: //237881/9F0FD67CA1BE/SIP/Error/preprocessSetup: Error dur

ng outbound SDP creation

Apr 25 21:56:14.448: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_spi_process_ccapi_event:

CCAPI Event Preprocessor Failure

I've tried this configuration two different ways and the result is the same. The first way is that I have a SIP Trunk configured from Call Manager to my Cisco2851. The Cisco2851 has the sip-ua statements configured to authenicate to the sip provider. I have a dialpeer that forwards the call to the sip provider. The second method was using a h323 gateway configured in call manager that hands to the calls off to the router. It uses the same dialpeers as the sip trunk. Like I stated the results is the same no matter what I do. If I use g711 it works fine both in and out. I have transcoding configured on my router, but its like its not kicking it up. I have MTP required on both device configurations in CM. I also have the correct Media resources enabled on both devices.

The SIP Trunk is connected to a MetaSwitch.

Here is something else I don't understand. Configured under g711 like I said calls work fine, but I still receive the following messages. Not sure if it is related to my g729 problem.

*Apr 25 22:08:04.592: //237975/4657D074A224/SIP/Error/sipSPI_ipip_copy_sdp_to_ch

annelInfo:

failed to update call entry

*Apr 25 22:08:04.596: //237976/4657D074A224/SIP/Error/sipSPIProcessRtpSessions:

No active streams; returning FAIL

*Apr 25 22:08:05.788: //237976/4657D074A224/SIP/Error/sipSPI_ipip_copy_sdp_to_ch

annelInfo: Unable to find the proper instance for FMTP

SIP: fmtp attribute, level 1 instance 0 not found.

*Apr 25 22:08:05.788: //237976/4657D074A224/SIP/Error/sipSPI_ipip_copy_sdp_to_ch

annelInfo: Unable to acquire event mask for rfc2833 dtmf relay

*Apr 25 22:08:05.788: //237975/4657D074A224/SIP/Error/sipSPISetStreamInfo: Numbe

r of active streams is zero (0)!

*Apr 25 22:08:13.924: //237976/4657D074A224/SIP/Error/sipSPI_ipip_copy_sdp_to_ch

annelInfo: Unable to find the proper instance for FMTP

SIP: fmtp attribute, level 1 instance 0 not found.

*Apr 25 22:08:13.924: //237976/4657D074A224/SIP/Error/sipSPI_ipip_copy_sdp_to_ch

annelInfo: Unable to acquire event mask for rfc2833 dtmf relay

*Apr 25 22:08:14.480: //237975/4657D074A224/SIP/Error/sipSPIUdeleteccCallIdFromT

able: Entry not found for ccCallId

Can some one shoot me a configuration that is done to support rfc2833. I want to make sure I did it right.

Thanks in advance

I have this problem too.
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Paolo Bevilacqua Wed, 04/25/2007 - 14:21

Hi,

Which IOS are you using? is this an _isv image ?

The callmanager can only generate g711 so that you must transcoding ? If it can generate g.729, does it work with codec transparent on the 2811 ?

Finally can you post the relevant config please ?

kelvin.blair Wed, 04/25/2007 - 15:05

IOS=12.4.9t1 Advanced Enterprise Version. If I set my dialpeer to codec transparent it does work, but its neg. codec is g711. Here is the gateway config for sip and transcode.

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

sip

transport switch udp tcp

voice class codec 622

codec preference 1 g729r8

codec preference 2 g729br8

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register mtp0019e7657be8

associate profile 1 register cfb0019e7657be8

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec gsmfr

codec g729r8

codec g729br8

maximum sessions 13

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 6

associate application SCCP

dial-peer voice 6220231 voip

destination-pattern 601622XXXX <--Outbound

session protocol sipv2

session target sip-server

session transport udp

voice-class codec 622

dtmf-relay rtp-nte digit-drop

!

dial-peer voice 3260301 voip

destination-pattern 601326XXXX <---INbound

session protocol sipv2

session target ipv4:10.9.5.100

session transport udp

voice-class codec 622

dtmf-relay sip-notify rtp-nte

no vad

sip-ua

authentication username XXXXX password XXXX

sip-server ipv4:x.x.x.x

User Masked Codec list: None

Call Manager: x.x.x.x, Port Number: 2000

Priority: N/A, Version: 3.1, Identifier: 1

Call Manager: x.x.x.x, Port Number: 2000

Priority: N/A, Version: 3.1, Identifier: 2

Conferencing Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: X.X.X.X, Port Number: 2000

TCP Link Status: CONNECTED, Profile Identifier: 1

Reported Max Streams: 48, Reported Max OOS Streams: 0

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: g729r8, Maximum Packetization Period: 60

Supported Codec: g729br8, Maximum Packetization Period: 60

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Transcoding Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: x.x.x.x , Port Number: 2000

TCP Link Status: CONNECTED, Profile Identifier: 2

Reported Max Streams: 26, Reported Max OOS Streams: 0

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: gsmfr, Maximum Packetization Period: 20

Supported Codec: g729r8, Maximum Packetization Period: 60

Supported Codec: g729br8, Maximum Packetization Period: 60

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

kelvin.blair Wed, 04/25/2007 - 21:00

Ok.. I figured out my problem, but now having another issue. Calling from Call Manager-->SIP Trunk--->SIP Dialpeer--->MetaSwitch--->PSTN it rings the phone, but if I hang up the ipphone, the PSTN phone still rings for about 5 seconds. If I call from the PSTN in via the sip trunk and disconnect. The call is disconnected quickly. It seems to be outbound calls to PSTN phones is having this problem. Does anyone know what may be causing this? Is it some type of timer? I see the cancel message being sent to SIP provider, but not sure if it is my end or theres.

MARK HOLLOWAY Wed, 04/25/2007 - 22:22

The BYE message isn't being received from one of your end points. The call termination is most likely delayed for 5 seconds because after one end terminates with BYE, the feature server (IP PBX?) assumes the call is dead. It only happens VoIP to PSTN, but not VoIP to VoIP? When you call VoIP to PSTN, and PSTN to VoIP, are you hanging up the same phone each time? Or hanging up the phone on the device you made the call from? I'm guessing you are hanging up the call from the device you originated dialing from. If this is true, then the BYE message from the device you are hanging up when making VoIP to PSTN calls is not gettings its BYE message to the feature server. Could be a NAT traversal, fixup, or firewall problem?

kelvin.blair Thu, 04/26/2007 - 04:48

No NAT is involved in this configuration. It is strickly ethernet. Yes, If I hang up the device that orginates the call, the other end keeps ringing..

MARK HOLLOWAY Thu, 04/26/2007 - 06:11

The Metaswitch is on a routable network back to the Cisco router? You don't have one device on a private IP and the other one on a public IP? Also, you don't have any one-way audio problems? What are the other devices (routers, firewalls) between your router and the Metaswitch?

kelvin.blair Thu, 04/26/2007 - 06:18

Metaswitch is connected to a 3750 Switch that has a Fiber connection to a 3750 switch stack where call manager is connected. These switches are doing intervlan routing.

Aaron Dhiman Thu, 04/26/2007 - 12:01

I remember trying to do something similar with a SIP Trunk last year. At the time, Cisco said MTP did not support G.729a for RFC 2833. Might still be the same now.

kelvin.blair Thu, 04/26/2007 - 12:02

Yep.. been working working on this all day and figured that one out. I have in fact found a device this may work. It is Nextone SBC which does a h323 to sip conversion and vice versa. It also supports the g729 straigh through.

MARK HOLLOWAY Thu, 04/26/2007 - 12:10

If you go that route I'd recommend the Acme Packet Session Director Net-Net 4000.

Paolo Bevilacqua Thu, 04/26/2007 - 12:12

Hi Kevin,

If you have solved the issue, why are you evaluating another device?

What was causing the problem in the first place ?

kelvin.blair Thu, 04/26/2007 - 12:14

I've figured out what the problem is. Not solved it.. It is a limitation to the current version of CM 4.2.3.

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