Sip/iax trunk call refuse problem

Asterisk Asterisk Support
1

I try to connect asterisk server to grandstream UCM6102 device using sip trunk and also try iax trunk as well. In case of sip trunk, it shows that sip peer are connected in both site. but call can’t forward through trunk . It give error message Everyone is busy/congested at this time (1:0/0/1) same as for IAX trunk.
Here are my configuration in asterisk server.

* Name       : fightvaw
  Description  : SIP Trunk from asterisk server
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : test1
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     : 
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Named Callgr : 
  Nam. Pickupgr: 
  MOH Suggest  : 
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Max forwards : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : port,invite
  Force rport  : Yes
  Symmetric RTP: Yes
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 202.166.200.154
  Addr->IP     : 202.166.200.154:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 
  SIP Options  : (none)
  Codecs       : (gsm|ulaw|alaw)
  Codec Order  : (ulaw:20,alaw:20,gsm:20)
  Auto-Framing :  No 
  Status       : OK (3 ms)
  Useragent    : 
  Reg. Contact : 
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   : 
  Use Reason   : No
  Encryption   : No

And debug code

<--- SIP read from UDP:202.166.198.142:5060 --->
OPTIONS sip:103.233.58.74 SIP/2.0
Via: SIP/2.0/UDP 202.166.198.142:5060;branch=z9hG4bK3e7d202c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@103.233.58.74>;tag=as082c57d3
To: <sip:103.233.58.74>
Contact: <sip:asterisk@202.166.198.142:5060>
Call-ID: 512-7d1044a86fb866443b7cb3ef7b55c1f9@103.233.58.74
CSeq: 102 OPTIONS
User-Agent: Grandstream UCM6102V1.5A 1.0.9.25
Date: Fri, 21 Aug 2015 05:18:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 202.166.198.142:5060 (no NAT)
Looking for s in Default (domain 103.233.58.74)

<--- Transmitting (no NAT) to 202.166.198.142:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 202.166.198.142:5060;branch=z9hG4bK3e7d202c;received=202.166.198.142
From: "asterisk" <sip:asterisk@103.233.58.74>;tag=as082c57d3
To: <sip:103.233.58.74>;tag=as4f0ed9cf
Call-ID: 512-7d1044a86fb866443b7cb3ef7b55c1f9@103.233.58.74
CSeq: 102 OPTIONS
Server: Asterisk PBX 11.6-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '512-7d1044a86fb866443b7cb3ef7b55c1f9@103.233.58.74' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:116.68.210.20:5060 --->
OPTIONS sip:192.168.85.51:5060 SIP/2.0
Via: SIP/2.0/UDP 116.68.210.20:5060;branch=z9hG4bK+dd8f8ba8f5dd88d9a002d8ca6d1d2b501+sip+1+48ef3953
Max-Forwards: 70
Call-ID: B88A7A3C-1@116.68.210.20:5060
From: <sip:116.68.210.20:5060;lr>;tag=sip+1+9f80e34+5bb6d52d
CSeq: 123882957 OPTIONS
Content-Length: 0
To: sip:192.168.85.51:5060
Contact: <sip:116.68.210.20:5060;lr>
Accept: application/sdp, application/dtmf-relay

<------------->
--- (10 headers 0 lines) ---
Sending to 116.68.210.20:5060 (no NAT)
Looking for s in Default (domain 192.168.85.51)

<--- Transmitting (no NAT) to 116.68.210.20:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 116.68.210.20:5060;branch=z9hG4bK+dd8f8ba8f5dd88d9a002d8ca6d1d2b501+sip+1+48ef3953;received=116.68.210.20
From: <sip:116.68.210.20:5060;lr>;tag=sip+1+9f80e34+5bb6d52d
To: sip:192.168.85.51:5060;tag=as0b448a64
Call-ID: B88A7A3C-1@116.68.210.20:5060
CSeq: 123882957 OPTIONS
Server: Asterisk PBX 11.6-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'B88A7A3C-1@116.68.210.20:5060' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '12239A95-1@116.68.210.20:5060' Method: OPTIONS

<--- SIP read from UDP:116.68.211.20:5060 --->
OPTIONS sip:192.168.85.51:5060 SIP/2.0
Via: SIP/2.0/UDP 116.68.211.20:5060;branch=z9hG4bK+9530bbadc0f70a1418618513d8fb255c1+sip+1+2440da8e
Max-Forwards: 70
Call-ID: 61512D8C-1@116.68.211.20:5060
From: <sip:116.68.211.20:5060;lr>;tag=sip+1+21980265+7fe3a856
CSeq: 306824087 OPTIONS
Content-Length: 0
To: sip:192.168.85.51:5060
Contact: <sip:116.68.211.20:5060;lr>
Accept: application/sdp, application/dtmf-relay

<------------->
--- (10 headers 0 lines) ---
Sending to 116.68.211.20:5060 (no NAT)
Looking for s in Default (domain 192.168.85.51)

<--- Transmitting (no NAT) to 116.68.211.20:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 116.68.211.20:5060;branch=z9hG4bK+9530bbadc0f70a1418618513d8fb255c1+sip+1+2440da8e;received=116.68.211.20
From: <sip:116.68.211.20:5060;lr>;tag=sip+1+21980265+7fe3a856
To: sip:192.168.85.51:5060;tag=as3143086c
Call-ID: 61512D8C-1@116.68.211.20:5060
CSeq: 306824087 OPTIONS
Server: Asterisk PBX 11.6-cert11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '61512D8C-1@116.68.211.20:5060' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '2BCFDD94-1@116.68.211.20:5060' Method: OPTIONS

<--- SIP read from UDP:202.166.198.142:5030 --->
jaK
<------------->

<--- SIP read from UDP:202.166.198.142:5030 --->
jaK
<------------->

<--- SIP read from UDP:202.166.198.142:5030 --->
jaK
<------------->

<--- SIP read from UDP:202.166.198.142:5030 --->
jaK
<------------->

<--- SIP read from UDP:202.166.198.142:5030 --->
jaK
<------------->

<--- SIP read from UDP:202.166.198.142:5030 --->
jaK
<------------->

similarly for IAX trunk configuration.

Name       : server-voiceinn
  Description  : 
  Secret       : <Set>
  Context      : test
  Parking lot  : 
  Mailbox      : 
  Dynamic      : No
  Callnum limit: 0
  Calltoken req: No
  Trunk        : Yes
  Encryption   : (aes128,keyrotate)
  Callerid     : "" <>
  Expire       : -1
  ACL          : No
  Addr->IP     : 202.166.200.154 Port 4569
  Defaddr->IP  : 0.0.0.0 Port 0
  Username     : server-voiceinn
  Codecs       : (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|siren7|siren14)
  Codec Order  : (none)
  Status       : OK (22 ms)
  Qualify      : every 25000ms when OK, every 10000ms when UNREACHABLE (sample smoothing Off)

and Debug:

IAX2 Debugging Enabled
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: REGREQ 
   Timestamp: 00015ms  SCall: 00053  DCall: 00000 [202.166.200.154:4569]
   USERNAME        : server_office
   REFRESH         : 60

Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: CTOKEN 
   Timestamp: 00015ms  SCall: 00001  DCall: 00053 [202.166.200.154:4569]
   CALLTOKEN       : 51 bytes

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: REGREQ 
   Timestamp: 00018ms  SCall: 00053  DCall: 00000 [202.166.200.154:4569]
   USERNAME        : server_office
   REFRESH         : 60
   CALLTOKEN       : 51 bytes

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: REGAUTH
   Timestamp: 00012ms  SCall: 15607  DCall: 00053 [202.166.200.154:4569]
   AUTHMETHODS     : 3
   CHALLENGE       : \x31\x33\x33\x31\x37\x36\x36\x32\x31
   USERNAME        : server_office

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass: REGREQ 
   Timestamp: 00021ms  SCall: 00053  DCall: 15607 [202.166.200.154:4569]
   USERNAME        : server_office
   REFRESH         : 60
   MD5 RESULT      : 530d070d16d11e1015a349d3cf0c0eb1

Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass: ACK    
   Timestamp: 00021ms  SCall: 15607  DCall: 00053 [202.166.200.154:4569]
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass: REGREJ 
   Timestamp: 00015ms  SCall: 15607  DCall: 00053 [202.166.200.154:4569]
   CAUSE           : Registration Refused
   CAUSE CODE      : 29

Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: ACK    
   Timestamp: 00015ms  SCall: 00053  DCall: 15607 [202.166.200.154:4569]
younginnov*CLI> 
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: POKE   
   Timestamp: 00017ms  SCall: 12212  DCall: 00000 [202.166.200.154:4569]

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: PONG   
   Timestamp: 00017ms  SCall: 00001  DCall: 12212 [202.166.200.154:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass: ACK    
   Timestamp: 00017ms  SCall: 12212  DCall: 00001 [202.166.200.154:4569]

In Grandstream device site i create sip trunk named voiceinn and IAX trunk server-office with inbound and outbound call route. then i create sip extension in both site and try to make call. i got error message. and not possible to call to remote extension.

I cant figure out the problem. how can i solve this issue?

2

There are no INVITE transactions in your SIP log; it is useless for debugging this problem.

3

i get follow log when i start debug the specific trunk.

Reliably Transmitting (NAT) to 202.166.200.154:5060:
OPTIONS sip:202.166.200.154 SIP/2.0
Via: SIP/2.0/UDP 103.233.58.74:5060;branch=z9hG4bK3bf8ef76;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@103.233.58.74>;tag=as5661ede6
To: <sip:202.166.200.154>
Contact: <sip:asterisk@103.233.58.74:5060>
Call-ID: 0da4dc8f551eb0f20db2561c28879b5e@103.233.58.74:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert11
Date: Mon, 24 Aug 2015 07:39:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:202.166.200.154:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.233.58.74:5060;branch=z9hG4bK3bf8ef76;received=103.233.58.74;rport=5060
From: "asterisk" <sip:asterisk@103.233.58.74:5060>;tag=as5661ede6
To: <sip:202.166.200.154>;tag=as7b0d698e
Call-ID: 0da4dc8f551eb0f20db2561c28879b5e@103.233.58.74:5060
CSeq: 102 OPTIONS
Server: Grandstream UCM6102V1.5A 1.0.9.25
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:202.166.200.154:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '0da4dc8f551eb0f20db2561c28879b5e@103.233.58.74:5060' Method: OPTIONS
[Aug 24 13:24:45] WARNING[350]: chan_sip.c:4037 retrans_pkt: Retransmission timeout reached on transmission e10d6f1d0255092122f8354f345ffb79 for seqno 1 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
younginnov*CLI> 
younginnov*CLI> 
Reliably Transmitting (NAT) to 202.166.200.154:5060:
OPTIONS sip:202.166.200.154 SIP/2.0
Via: SIP/2.0/UDP 103.233.58.74:5060;branch=z9hG4bK71b6515f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@103.233.58.74>;tag=as0fdcfab9
To: <sip:202.166.200.154>
Contact: <sip:asterisk@103.233.58.74:5060>
Call-ID: 7552484c1ae281690ff788c16d41a72b@103.233.58.74:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert11
Date: Mon, 24 Aug 2015 07:40:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:202.166.200.154:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 103.233.58.74:5060;branch=z9hG4bK71b6515f;received=103.233.58.74;rport=5060
From: "asterisk" <sip:asterisk@103.233.58.74:5060>;tag=as0fdcfab9
To: <sip:202.166.200.154>;tag=as2684170f
Call-ID: 7552484c1ae281690ff788c16d41a72b@103.233.58.74:5060
CSeq: 102 OPTIONS
Server: Grandstream UCM6102V1.5A 1.0.9.25
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:202.166.200.154:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '7552484c1ae281690ff788c16d41a72b@103.233.58.74:5060' Method: OPTIONS
[Aug 24 13:25:56] WARNING[350]: chan_sip.c:4037 retrans_pkt: Retransmission timeout reached on transmission bfcca0cbdeea1b28aa02032da5dc819b for seqno 1 (Non-critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response

is there any command to find invite response only. Is this a problem of firewall ? I can’t figure out.