Aastra 6731i blind "Transfer failed"

I’m having trouble getting blind transfers to work between two Aastra 6731i.

Process–

  1. Dial Special Extension 222. Hear, “Helloo this is Lenny”
  2. Press Transfer button.
  3. Dial 120
  4. Press Transfer button
  5. Display shows, “Transfer failed”

The call is held and I can try again, this time with an attended transfer. Attended transfer works fine.

Now I’ve done some searching on this forum and found similar issues. I learned that the issues is likely related to pjsip and a conflict on Aastra and Snom phones. There is a helpful workaround on the issue tracker which shows how to disable the refer_blind_progress setting for specific extensions in FreePBX.

I tried the above workaround, however I don’t see any change in behavior. Blind transfers still fail. Perhaps there is a specific handset firmware I should be running? Perhaps I didn’t make the change properly?

The handset firmware version is identical on both. Firmware Version 3.3.1.4358
Running FreePBX distro 14.0.3.6.
Asterisk version 13.18.4

pjsip.endpoint_custom_post.conf contains the following–

[129](+)
refer_blind_progress=no

[120](+)
refer_blind_progress=no

After applying changes to the above config file, I verified via the CLI that the changes were applied–

[[email protected] ~]# asterisk -x "pjsip show endpoint 120" | grep refer
refer_blind_progress               : false
[[email protected] ~]# asterisk -x "pjsip show endpoint 129" | grep refer
refer_blind_progress               : false

Does anything look off?

Edit: I just tried on Asterisk 14.7.5 and saw no difference.
Edit: Note: I am using pjsip

Whatcha think, folks? Is this an issue I should just use CHAN_SIP to solve?

So I tried CHAN_SIP, no improvement there.

Here’s the output of asterisk -x "sip show peer 129"


  * Name       : 129
  Description  :
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : from-internal
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     : en
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox      : [email protected]
  VM Extension : *97
  LastMsgsSent : 0/0
  Call limit   : 2147483647
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "IT Department" <129>
  MaxCallBR    : 384 kbps
  Expire       : 30
  Insecure     : no
  Force rport  : No
  Symmetric RTP: No
  ACL          : Yes
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : Yes
  Send RPID    : Yes
  Path support : No
  Path         : N/A
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 192.168.100.50:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 129
  SIP Options  : (none)
  Codecs       : (ulaw|alaw|gsm|g726|g722)
  Auto-Framing : No
  Status       : OK (29 ms)
  Useragent    : Aastra 6731i/3.3.1.4358
  Reg. Contact : sip:[email protected]:5060;transport=udp
  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
  RTCP Mux     : No

The one thing that stuck out to me was Transfer mode although the configuration for this doesn’t seem to be exposed in FreePBX GUI, and I’m not sure what it does anyway. I couldn’t find a mention of it in the Asterisk Wiki. I did find something in the source code, but it doesn’t seem related to call transfers–

* @param transferMode Information transfer mode.(circuit/packet modes).

I’m rather stumped.

I fired up our old vanilla Asterisk server to make a comparison in the peer config. On this server, blind transfers succeeded without fail. It was version Asterisk 11.13.1~dfsg-2+b1 built by buildd @ x86-csail-01 on a i686 running Linux on 2015-01-06 03:30:17 UTC

  * Name       : <redacted>
  Description  : IT Closet
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : LocalSets
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : LocalSets
  Language     :
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox      : [email protected]
  VM Extension : asterisk
  LastMsgsSent : 0/1
  Call limit   : 2147483647
  Max forwards : 0
  Busy level   : 2
  Dynamic      : Yes
  Callerid     : "IT Closet" <120>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Force rport  : Yes
  Symmetric RTP: Yes
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : (null)
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: <redacted>
  SIP Options  : (none)
  Codecs       : (ulaw|alaw)
  Codec Order  : (ulaw:20,alaw:20)
  Auto-Framing : No
  Status       : OK (12 ms)
  Useragent    :
  Reg. Contact : sip:<redacted>@192.168.100.57:5060;transport=udp
  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

Good to know, the Transfer mode directive in this config matches the FreePBX peer config. So now what should I check next? I’m short of ideas.

Try with a normal extension, does it happen as well?
Monitor asterisk logs, see what happens when the transfer fails.

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Oh… A normal extension worked! This explains why I wasn’t getting complaints about call transfers :rofl: :nerd_face:

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