SIP going straight to voicemail and not accepting calls

General Help
1

I cannot get a extension to ring, I think it is a problem with freepbx, the extension is registered and I can call out, I have tried the “follow me” module as well to see if I can override the problem somehow, I went to the asterisk group and they said the logs show it isnt event trying the extension (log below) this is a fresh install of asterisk 1.2 and freepbx 2.3 … I cannot find any error anywhere?

  1. – Executing Macro(“SIP/101-0818b4e0”, “exten-vm|100|100”) in new stack

  2. -- Executing Macro("SIP/101-0818b4e0", "user-callerid") in new stack

  3. -- Executing NoOp("SIP/101-0818b4e0", "user-callerid: device 101") in new stack

  4. -- Executing Set("SIP/101-0818b4e0", "AMPUSER=101") in new stack

  5. -- Executing GotoIf("SIP/101-0818b4e0", "0?report") in new stack

  6. -- Executing GotoIf("SIP/101-0818b4e0", "0?start") in new stack

  7. -- Executing Set("SIP/101-0818b4e0", "REALCALLERIDNUM=101") in new stack

  8. -- Executing NoOp("SIP/101-0818b4e0", "REALCALLERIDNUM is 101") in new stack

  9. -- Executing Set("SIP/101-0818b4e0", "AMPUSER=101") in new stack

  10. -- Executing Set("SIP/101-0818b4e0", "AMPUSERCIDNAME=Doug") in new stack

  11. -- Executing GotoIf("SIP/101-0818b4e0", "0?report") in new stack

  12. -- Executing Set("SIP/101-0818b4e0", "AMPUSERCID=101") in new stack

  13. -- Executing Set("SIP/101-0818b4e0", "CALLERID(all)="Doug" <101>") in new stack

  14. -- Executing Set("SIP/101-0818b4e0", "REALCALLERIDNUM=101") in new stack

  15. -- Executing NoOp("SIP/101-0818b4e0", "TTL:  ARG1: 100") in new stack

  16. -- Executing GotoIf("SIP/101-0818b4e0", "0?continue") in new stack

  17. -- Executing Set("SIP/101-0818b4e0", "__TTL=64") in new stack

  18. -- Executing GotoIf("SIP/101-0818b4e0", "1?continue") in new stack

  19. -- Goto (macro-user-callerid,s,23)

  20. -- Executing NoOp("SIP/101-0818b4e0", "Using CallerID "Doug" <101>") in new stack

  21. -- Executing Set("SIP/101-0818b4e0", "FROMCONTEXT=exten-vm") in new stack

  22. -- Executing Set("SIP/101-0818b4e0", "VMBOX=100") in new stack

  23. -- Executing Set("SIP/101-0818b4e0", "EXTTOCALL=100") in new stack

  24. -- Executing Set("SIP/101-0818b4e0", "CFUEXT=") in new stack

  25. -- Executing Set("SIP/101-0818b4e0", "CFBEXT=") in new stack

  26. -- Executing Set("SIP/101-0818b4e0", "RT=15") in new stack

  27. -- Executing Macro("SIP/101-0818b4e0", "record-enable|100|IN") in new stack

  28. -- Executing GotoIf("SIP/101-0818b4e0", "0?2:4") in new stack

  29. -- Goto (macro-record-enable,s,4)

  30. -- Executing AGI("SIP/101-0818b4e0", "recordingcheck|20070928-154413|1191019453.7") in new stack

  31. -- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck

  32. -- AGI Script recordingcheck completed, returning 0

  33. -- Executing NoOp("SIP/101-0818b4e0", "No recording needed") in new stack

  34. -- Executing Macro("SIP/101-0818b4e0", "dial|15|tr|100") in new stack

  35. -- Executing GotoIf("SIP/101-0818b4e0", "1?dial") in new stack

  36. -- Goto (macro-dial,s,3)

  37. -- Executing AGI("SIP/101-0818b4e0", "dialparties.agi") in new stack

  38. -- Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi

  40. – Unregistered SIP ‘101’

  41. -- Registered SIP '101' at 192.168.0.103 port 48036 expires 3600

  42. -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 101

  44. swig*CLI>

    • Name : 101

  47. Secret :

  48. MD5Secret :

  49. Context : from-internal

  50. Subscr.Cont. :

  51. Language :

  52. AMA flags : Unknown

  53. CallingPres : Presentation Allowed, Not Screened

  54. Callgroup : 0

  55. Pickupgroup : 0

  56. Mailbox : 101@device

  57. VM Extension : asterisk

  58. LastMsgsSent : 0/0

  59. Call limit : 0

  60. Dynamic : Yes

  61. Callerid : “device” <101>

  62. Expire : 3404

  63. Insecure : no

  64. Nat : Always

  65. ACL : No

  66. CanReinvite : No

  67. PromiscRedir : No

  68. User=Phone : No

  69. Trust RPID : No

  70. Send RPID : No

  71. DTMFmode : rfc2833

  72. LastMsg : 0

  73. ToHost :

  74. Addr->IP : 192.168.0.103 Port 48036

  75. Defaddr->IP : 0.0.0.0 Port 5060

  76. Def. Username: 101

  77. SIP Options : (none)

  78. Codecs : 0xc (ulaw|alaw)

  79. Codec Order : (ulaw,alaw)

  80. Status : OK (108 ms)

    • Name : 100

  86. Secret :

  87. MD5Secret :

  88. Context : from-internal

  89. Subscr.Cont. :

  90. Language :

  91. AMA flags : Unknown

  92. CallingPres : Presentation Allowed, Not Screened

  93. Callgroup : 0

  94. Pickupgroup : 0

  95. Mailbox : 100@default

  96. VM Extension : asterisk

  97. LastMsgsSent : 0/0

  98. Call limit : 0

  99. Dynamic : Yes

  100. Callerid : “device” <100>

  101. Expire : 2493

  102. Insecure : no

  103. Nat : Always

  104. ACL : No

  105. CanReinvite : No

  106. PromiscRedir : No

  107. User=Phone : No

  108. Trust RPID : No

  109. Send RPID : No

  110. DTMFmode : rfc2833

  111. LastMsg : 0

  112. ToHost :

  113. Addr->IP : 192.168.0.110 Port 5060

  114. Defaddr->IP : 0.0.0.0 Port 5060

  115. Def. Username: 100

  116. SIP Options : (none)

  117. Codecs : 0xc (ulaw|alaw)

  118. Codec Order : (ulaw,alaw)

  119. Status : OK (7 ms)

  120. Useragent : Grandstream HT287 1.0.8.23

  121. Reg. Contact : sip:[email protected]

  123. Useragent : X-Lite release 1011s stamp 41150

  124. Reg. Contact : sip:[email protected]:48036;rinstance=3c6de34aa5575492

2

the most common cause for this is the extension’s astdb objects got corrupted. Go into FreePBX and resubmit the extension you can’t call and then try it again. That process will re-create that extension’s object in astdb. You may have to do it with each extension on the system.

Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx

2 Likes
3

Verbosity was 3 and is now 20
– Starting simple switch on ‘Zap/4-1’
== Starting Zap/4-1 at from-zaptel,s,1 failed so falling back to exten ‘s’
== Starting Zap/4-1 at from-zaptel,s,1 still failed so falling back to context ‘default’
– Hungup ‘Zap/4-1’
– Starting simple switch on ‘Zap/4-1’
== Starting Zap/4-1 at from-zaptel,s,1 failed so falling back to exten ‘s’
== Starting Zap/4-1 at from-zaptel,s,1 still failed so falling back to context ‘default’
– Hungup ‘Zap/4-1’

After adding the extension back I get this error, I have tried it several times

4

If anyone can help solve this riddle, it would be greatly appreciated.

5

Here’s what I did:

http://wiki.ppckernel.org/w/Setting_up_Asterisk_and_FreePBX_2.3_in_a_Debian_etch_chroot

(That doesn’t come from any authoritative source, I’m just trying to
write it.)

As for phones, I have several softphones on the same subnet and a
grandstream that connects remotely. None of the phones can call any
other phone. Furthermore, followme numbers are never rung. Calls
always just go straight to voicemail.

Status shows OK w/correct IP in sip show peers, and I’ve tried both nat
yes and nat never for the exts. Also, I don’t have any server nat
configuration set.

I’ve been trying to fix this for over a week now. The only thing I see that might be out of order is chan_zap:

Nov 12 12:52:49 WARNING[1745] loader.c: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call
Nov 12 12:52:49 WARNING[1745] loader.c: Loading module chan_zap.so failed!

Asterisk doesn’t start when it’s enabled, but I have no idea whether or not it’s necessary. Everything BUT ringing phones works fine without it.

When are we going to see FreePBX take a step forward and get some working binary packages? Not everyone wants to run bloated appliances like Elastix and Trixbox, nor does everyone have a dedicated PBX box. Nor does everyone run CentOS. Yes, I’ve tried virtual machines in both Xen and VMware, but the timing issues there make this a questionable alternative, at best. Get with the program and start developing your software for more than one distro, and generate some usable binary package while you’re at it. The complexity and number of things that can go wrong in your INSTALL instructions are untenable. Let’s see some activity in this bug:

http://www.freepbx.org/trac/ticket/1167

6

same issue here. yesterday it was a snom 320 phone stuck in some type of DND state when the phone is idol. This happened before and i deleted the extension and rebuilt it and it cleared up. So I did that and then rebooted the system and the phone started ringing again. Same customer calls me today and another phone is doing the same thing bu thistime it a linksys 941 phone. I did the same procedure as the Snom yesterday but this Linksys will not clear the UNKNOW DND state.

It does not appear to be a registration issue because the phone dials out just fine. All calls internal or external go straight to the busy voicemail greeting.

Any Suggestions would be greatly appreciated, I started a post in the trixbox forums but not getting to far.

Many Thanks

7

get into the asterisk cli, to check the status of a extension and it’s DND in the database type:

database get DND ext

where ext is the extension number.

If the extension has DND enabled you will get a reply of Value: YES
If the extension does not have DND defined you will get: Database entry not found.

8

I know this is a really old thread but I felt it necessary to necro-post and thank Philippe for the solution as it solved the issue I was having with calls not being accepted by registered SIP clients. Such a simple solution but it took almost 4 hours of searching, debugging, searching, poking, and swearing until I came across this posting. Thanks!

9

I know this thread is really old but I have a very similar situation. I have an extension returning YES when doing a query on the database. I can’t seem to locate where and why it is set to DND. Any suggestions on where to look or what to post for you guys to look at?