Jump to content

DTFM issue after Trunk change


global_s
 Share

Recommended Posts

Hello,

I have setup 2 domain in my Snom blue

 

Originally I setup an inbound trunk to go directly to a conference room and it worked.

 

Then I setup the same inbound trunk to the second domain and suddenly, it stopped working for both.

So I removed the second trunk, made the first global and check the result.

Calls are received by the pbx, I can hear the welcome message to the conference, but DTFM are not received, so I can't enter a room. If I call from an extension everything works.

I disable the only trunk from being global with no success. Not even bringing it back to the original configuration solves the issue.

 

I recreated the trunk with no luck

 

Please help

 

Here's my log from a test call. I ended it because the system couldn't understand my digits.

 

[5] 2012/09/15 00:03:18: SIP Rx udp:81.201.83.45:5060:

INVITE sip:8877@sips14.s.tw SIP/2.0

Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

CSeq: 102 INVITE

From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899

To: <sip:8877@sips14.s.tw>

Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKce77d15175f9ec73a2dcd59c38df0a0b

Max-Forwards: 69

Content-Type: application/sdp

Contact: <sip:39333333333@81.201.83.45:5060;transport=udp>

User-Agent: Vox Callcontrol

Content-Length: 240

 

v=0

o=root 20488 20488 IN IP4 81.201.83.53

s=session

c=IN IP4 81.201.83.53

t=0 0

m=audio 11516 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

[8] 2012/09/15 00:03:18: Allocating for call port 11, SIP call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

[9] 2012/09/15 00:03:18: UDP(IPv4): Opening socket on 0.0.0.0:10084

[9] 2012/09/15 00:03:18: UDP(IPv4): Opening socket on 0.0.0.0:10085

[9] 2012/09/15 00:03:18: UDP(IPv6): Opening socket on [::]:10084

[9] 2012/09/15 00:03:18: UDP(IPv6): Opening socket on [::]:10085

[5] 2012/09/15 00:03:18: Identify trunk (IP address match) 12

[9] 2012/09/15 00:03:18: Resolve 256: aaaa udp 81.201.83.45 5060

[9] 2012/09/15 00:03:18: Resolve 256: a udp 81.201.83.45 5060

[9] 2012/09/15 00:03:18: Resolve 256: udp 81.201.83.45 5060

[5] 2012/09/15 00:03:18: SIP Tx udp:81.201.83.45:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKce77d15175f9ec73a2dcd59c38df0a0b

From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899

To: <sip:8877@sips14.s.tw>;tag=8828815886

Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

CSeq: 102 INVITE

Content-Length: 0

 

[7] 2012/09/15 00:03:18: Set packet length to 20

[6] 2012/09/15 00:03:18: Call-leg 11: Sending RTP for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 to 81.201.83.53:11516, codec not set yet

[8] 2012/09/15 00:03:18: Incoming call: Request URI sip:8877@sips14.s.tw, To is <sip:8877@sips14.s.tw>

[8] 2012/09/15 00:03:18: Call from a trunk 12

[8] 2012/09/15 00:03:18: Trunk Voxbone@sipmsk.of.tw has country code not set, area code not set

[9] 2012/09/15 00:03:18: Incoming: formatted From is = "39333333333" <sip:39333333333@voxbone.com;user=phone>

[9] 2012/09/15 00:03:18: Incoming: formatted To is = <sip:8877@sips14.s.tw;user=phone>

[9] 2012/09/15 00:03:18: Incoming: formatted URI is = sip:8877@sipmsk.of.tw;user=phone

[8] 2012/09/15 00:03:18: To is <sip:8877@sips14.s.tw;user=phone>, user 31, domain 1

[8] 2012/09/15 00:03:18: From number 39333333333 is configured as cell phone for extension 43402@sips14.s.tw, display name Francesco Silvestri

[8] 2012/09/15 00:03:18: To user 8877

[8] 2012/09/15 00:03:18: Send call to extension is not set. Route the call based on global number 8877

[5] 2012/09/15 00:03:18: Global trunk Voxbone@sipmsk.of.tw sends call to 8877 in domain sips14.s.tw

[8] 2012/09/15 00:03:18: From number 39333333333 is configured as cell phone for extension 43402@sips14.s.tw, display name Francesco Silvestri

[8] 2012/09/15 00:03:18: Set the To domain based on To user 8877@sips14.s.tw

[8] 2012/09/15 00:03:18: Call state for call object 12: idle

[7] 2012/09/15 00:03:18: Call port 11: set_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 codecs "", codec_preference count 7

[8] 2012/09/15 00:03:18: Call state for call object 12: connected

[8] 2012/09/15 00:03:18: Play space10, caching false

[8] 2012/09/15 00:03:18: call port 11: state code from 0 to 200

[7] 2012/09/15 00:03:18: Set packet length to 20

[9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: adding codec pcmu/8000 to available list

[9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec pcma/8000

[9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec g722/8000

[9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec g729/8000

[9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec g726-32/8000

[9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec gsm/8000

[9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: codec_preference size 7, available codecs size 2

[6] 2012/09/15 00:03:18: Call-leg 11: Codec pcmu/8000 is chosen for call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

[5] 2012/09/15 00:03:18: set codec: codec pcmu/8000 is set to call-leg 11

[9] 2012/09/15 00:03:18: Resolve 257: aaaa udp 81.201.83.45 5060

[9] 2012/09/15 00:03:18: Resolve 257: a udp 81.201.83.45 5060

[9] 2012/09/15 00:03:18: Resolve 257: udp 81.201.83.45 5060

[5] 2012/09/15 00:03:18: SIP Tx udp:81.201.83.45:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKce77d15175f9ec73a2dcd59c38df0a0b

From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899

To: <sip:8877@sips14.s.tw>;tag=8828815886

Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

CSeq: 102 INVITE

Contact: <sip:8877@174.16.25.5:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/4.5.0.1075 Delta Aurigids

Content-Type: application/sdp

Content-Length: 249

 

v=0

o=- 512677875 512677875 IN IP4 174.16.25.5

s=-

c=IN IP4 174.16.25.5

t=0 0

m=audio 10084 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[6] 2012/09/15 00:03:18: Call-leg 11: Codec pcmu/8000 is chosen for call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

[5] 2012/09/15 00:03:18: set codec: codec pcmu/8000 is set to call-leg 11

[5] 2012/09/15 00:03:18: SIP Rx udp:81.201.83.45:5060:

ACK sip:8877@174.16.25.5:5060;transport=udp SIP/2.0

Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

CSeq: 102 ACK

From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899

To: <sip:8877@sips14.s.tw>;tag=8828815886

Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bK81ebc6e5fe28edb493f1bf6c44b4a660

Max-Forwards: 69

Contact: <sip:39333333333@81.201.83.45:5060;transport=udp>

User-Agent: Vox Callcontrol

Content-Length: 0

 

[5] 2012/09/15 00:03:20: SIP Rx udp:188.40.65.170:27010:

REGISTER sip:174.16.25.5 SIP/2.0

Via: SIP/2.0/UDP 188.40.65.170:27010;branch=z9hG4bK0bf989e7;rport

From: <sip:924385@174.16.25.5>;tag=as5ba10c39

To: <sip:924385@174.16.25.5>

Call-ID: 5d05e6ae2909f0c12505dc292323cada@188.40.65.170

CSeq: 209 REGISTER

User-Agent: PBX

Max-Forwards: 70

Expires: 3600

Contact: <sip:Snomone@188.40.65.170:27010>

Event: registration

Content-Length: 0

 

[9] 2012/09/15 00:03:20: Resolve 258: aaaa udp 188.40.65.170 27010

[9] 2012/09/15 00:03:20: Resolve 258: a udp 188.40.65.170 27010

[9] 2012/09/15 00:03:20: Resolve 258: udp 188.40.65.170 27010

[5] 2012/09/15 00:03:20: SIP Tx udp:188.40.65.170:27010:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 188.40.65.170:27010;branch=z9hG4bK0bf989e7;rport=27010

From: <sip:924385@174.16.25.5>;tag=as5ba10c39

To: <sip:924385@174.16.25.5>;tag=8e52c4dad1

Call-ID: 5d05e6ae2909f0c12505dc292323cada@188.40.65.170

CSeq: 209 REGISTER

Content-Length: 0

 

[5] 2012/09/15 00:03:20: SIP Rx udp:81.26.212.150:6060:

OPTIONS sip:2.228.101.190:24942 SIP/2.0

Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0

From: sip:pinger@81.26.212.150;tag=7f105f66

To: sip:2.228.101.190:24942

Call-ID: 4f7b936-3b5e9ec8-4d604e4@81.26.212.150

CSeq: 1 OPTIONS

Content-Length: 0

 

[9] 2012/09/15 00:03:20: Resolve 259: aaaa udp 81.26.212.150 6060

[9] 2012/09/15 00:03:20: Resolve 259: a udp 81.26.212.150 6060

[9] 2012/09/15 00:03:20: Resolve 259: udp 81.26.212.150 6060

[5] 2012/09/15 00:03:20: SIP Tx udp:81.26.212.150:6060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0

From: <sip:pinger@81.26.212.150>;tag=7f105f66

To: <sip:2.228.101.190:24942>;tag=e5b876add8

Call-ID: 4f7b936-3b5e9ec8-4d604e4@81.26.212.150

CSeq: 1 OPTIONS

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Content-Length: 0

 

[5] 2012/09/15 00:03:20: SIP Rx udp:81.26.212.150:6060:

OPTIONS sip:2.228.101.190:24942 SIP/2.0

Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0

From: sip:pinger@81.26.212.150;tag=8f105f66

To: sip:2.228.101.190:24942

Call-ID: 4f7b936-4b5e9ec8-4d604e4@81.26.212.150

CSeq: 1 OPTIONS

Content-Length: 0

 

[9] 2012/09/15 00:03:20: Resolve 260: aaaa udp 81.26.212.150 6060

[9] 2012/09/15 00:03:20: Resolve 260: a udp 81.26.212.150 6060

[9] 2012/09/15 00:03:20: Resolve 260: udp 81.26.212.150 6060

[5] 2012/09/15 00:03:20: SIP Tx udp:81.26.212.150:6060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0

From: <sip:pinger@81.26.212.150>;tag=8f105f66

To: <sip:2.228.101.190:24942>;tag=6dedcc0421

Call-ID: 4f7b936-4b5e9ec8-4d604e4@81.26.212.150

CSeq: 1 OPTIONS

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Content-Length: 0

 

[5] 2012/09/15 00:03:20: SIP Rx udp:81.26.212.150:6060:

OPTIONS sip:2.228.101.190:24942 SIP/2.0

Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0

From: sip:pinger@81.26.212.150;tag=9f105f66

To: sip:2.228.101.190:24942

Call-ID: 4f7b936-5b5e9ec8-4d604e4@81.26.212.150

CSeq: 1 OPTIONS

Content-Length: 0

 

[9] 2012/09/15 00:03:20: Resolve 261: aaaa udp 81.26.212.150 6060

[9] 2012/09/15 00:03:20: Resolve 261: a udp 81.26.212.150 6060

[9] 2012/09/15 00:03:20: Resolve 261: udp 81.26.212.150 6060

[5] 2012/09/15 00:03:20: SIP Tx udp:81.26.212.150:6060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0

From: <sip:pinger@81.26.212.150>;tag=9f105f66

To: <sip:2.228.101.190:24942>;tag=2a5c9a957d

Call-ID: 4f7b936-5b5e9ec8-4d604e4@81.26.212.150

CSeq: 1 OPTIONS

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Content-Length: 0

 

[5] 2012/09/15 00:03:20: SIP Rx udp:81.26.212.150:6060:

OPTIONS sip:2.228.101.190:24942 SIP/2.0

Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0

From: sip:pinger@81.26.212.150;tag=af105f66

To: sip:2.228.101.190:24942

Call-ID: 4f7b936-6b5e9ec8-4d604e4@81.26.212.150

CSeq: 1 OPTIONS

Content-Length: 0

 

[9] 2012/09/15 00:03:20: Resolve 262: aaaa udp 81.26.212.150 6060

[9] 2012/09/15 00:03:20: Resolve 262: a udp 81.26.212.150 6060

[9] 2012/09/15 00:03:20: Resolve 262: udp 81.26.212.150 6060

[5] 2012/09/15 00:03:20: SIP Tx udp:81.26.212.150:6060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0

From: <sip:pinger@81.26.212.150>;tag=af105f66

To: <sip:2.228.101.190:24942>;tag=3b8e58f455

Call-ID: 4f7b936-6b5e9ec8-4d604e4@81.26.212.150

CSeq: 1 OPTIONS

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Content-Length: 0

 

[8] 2012/09/15 00:03:20: Play audio_en/co_welcome_conference.wav, caching false

[8] 2012/09/15 00:03:20: call port 11: state code from 200 to 200

[8] 2012/09/15 00:03:22: Play audio_en/co_enter_pin.wav, caching false

[8] 2012/09/15 00:03:22: call port 11: state code from 200 to 200

[5] 2012/09/15 00:03:31: SIP Rx udp:84.199.103.82:5060:

REGISTER sip:sips14.s.tw SIP/2.0

Via: SIP/2.0/UDP 172.16.2.10:5060;branch=z9hG4bK4b47edca8435D139

From: "7000" <sip:7000@sips14.s.tw>;tag=2EA9980C-B2C1BC0B

To: <sip:7000@sips14.s.tw>

CSeq: 2667 REGISTER

Call-ID: 7ecc7eb7-adc047ce-d29eb85d@172.16.2.10

Contact: <sip:7000@172.16.2.10:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.4.0267

Accept-Language: en

Authorization: Digest username="7000", realm="sips14.s.tw", nonce="642af809e14ae35f6f2b0fe0803a921f", uri="sip:sips14.s.tw", response="69ff23db49333866fb866c7c189d8995", algorithm=MD5

Max-Forwards: 70

Expires: 3600

Content-Length: 0

 

[9] 2012/09/15 00:03:31: Resolve 263: udp 84.199.103.82 5060

[5] 2012/09/15 00:03:31: SIP Tx udp:84.199.103.82:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 172.16.2.10:5060;branch=z9hG4bK4b47edca8435D139;rport=5060;received=84.199.103.82

From: "7000" <sip:7000@sips14.s.tw>;tag=2EA9980C-B2C1BC0B

To: <sip:7000@sips14.s.tw>;tag=370d21a9d7

Call-ID: 7ecc7eb7-adc047ce-d29eb85d@172.16.2.10

CSeq: 2667 REGISTER

Contact: <sip:7000@172.16.2.10:5060>;expires=59

Server: snomONE/4.5.0.1075 Delta Aurigids

Content-Length: 0

 

[5] 2012/09/15 00:03:40: SIP Rx udp:81.201.83.45:5060:

BYE sip:8877@174.16.25.5:5060;transport=udp SIP/2.0

Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

CSeq: 103 BYE

From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899

To: <sip:8877@sips14.s.tw>;tag=8828815886

Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bK234e3dd4eeaa0cef7584328a76c4a662

Max-Forwards: 69

User-Agent: Vox Callcontrol

Content-Length: 0

 

[9] 2012/09/15 00:03:40: Resolve 264: aaaa udp 81.201.83.45 5060

[9] 2012/09/15 00:03:40: Resolve 264: a udp 81.201.83.45 5060

[9] 2012/09/15 00:03:40: Resolve 264: udp 81.201.83.45 5060

[5] 2012/09/15 00:03:40: SIP Tx udp:81.201.83.45:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bK234e3dd4eeaa0cef7584328a76c4a662

From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899

To: <sip:8877@sips14.s.tw>;tag=8828815886

Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

CSeq: 103 BYE

Contact: <sip:8877@174.16.25.5:5060;transport=udp>

User-Agent: snomONE/4.5.0.1075 Delta Aurigids

Content-Length: 0

 

[8] 2012/09/15 00:03:40: Clearing call port 11, SIP call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

[8] 2012/09/15 00:03:40: Remove leg 12: call port 11, SIP call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45

[8] 2012/09/15 00:03:40: Hangup: Call 11 not found

Link to comment
Share on other sites

Hmm. The log looks "beautiful". Sometimes the provider also change things, which can drive you nuts trying to figure out if any change on your side is the problem.

 

The PBX also allows that in addition to listening to RFC2833 (RFC4733 now) you can also do inband DTMF detection. Maybe this is the way to go for you now.

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

 Share

×
×
  • Create New...