Double Attended Transfer Error / Blind + Attend Error

Asterisk Asterisk Dialplan
1

Hello everyone,

First of all, I would like to thank everyone who is interested. I have a problem like this, the status descriptions are below.

I have a problem like this on the Asterisk Ael side;

  1. 202 Extension 0546 Calls and Talks
  2. 202 Extension Makes BlindTransfer and Redirects 0546 to 201 Extension.
  3. Extension 201 is talking to 0546
  4. Extension 201 is making an AttendedTransfer and forwarding the number 0546 to Extension 203
  5. Extension 201 is making a Hangup and transferring the call to Extension 203
  6. But when 201 is making a Hangup, the Hangup packet is not coming and only the information below the CLI screen is written

– Stopped music on hold on SIP/TESTSIP-00000069
– Channel SIP/201-ABC-0000006a left ‘simple_bridge’ basic-bridge
– Channel Local/203@verified-00000027;1 left ‘simple_bridge’ basic-bridge – Channel Local/203@verified-00000027;1 joined ‘simple_bridge’ basic-bridge <1c01f1f3-2e21-4628-9dfd-fd79b2ac3205> – <Local/203@verified-000000 27;1> Playing ‘beep.ulaw’ (language ‘tr’) > 0x7f06900c14e0 – Strict RTP learning complete - Locking on source address 192.168.102.40:30418 Switchboard1*CLI> But Hangup Package Does Not Come

Below is Channel and Bridge Information (Another Call - Same Behavior)

-----201-202 While Making a Call from Attendant (Without Handing Over Yet)-----
switchboard1*CLI> bridge show b59745d6-d632-4d74-90e5-c0250e1198c5
Id: b59745d6-d632-4d74-90e5-c0250e1198c5
Type: basic
Technology: simple_bridge
Subclass: basic
Creator:
Name:
Video-Mode: none
Video-Source-Id:
Num-Channels: 1
Num-Active: 1
Duration: 00:04:13
Channel: SIP/TESTSIP-00000019

switchboard1CLI> bridge show 85ad0c45-4c93-4a52-aa24-9be524ab43f3 Id: 85ad0c45-4c93-4a52-aa24-9be524ab43f3 Type: basic Technology: simple_bridge Subclass: basic Creator: Name: Video-Mode: none Video-Source-Id: Num-Channels: 2 Num-Active: 2 Duration : 00:02:07 Channel: Local/202@verified-00000004;2 Channel: SIP/202-ABC-0000001b ----------------------------------------------------------------------------------------------------------------- switchboard1CLI> bridge show 6ade7237-dfd2-4a0e-ad5c-445d13e326a2 Id: 6ade7237-dfd2-4a0e-ad5c-445d13e326a2 Type: basic Technology: simple_bridge Subclass: basic Creator: Name: Video-Mode: none Video-Source-Id: Num-Channels: 2 Num-Active: 2 Duration: 00:02:39 Channel: SIP/201-ABC-0000001a Channel: Local/202@verified-00000004;1 When 201 Closes (Transferred to 202 Side) switchboard1CLI> bridge show 85ad0c45-4c93-4a52-aa24-9be524ab43f3 Id: 85ad0c45-4c93-4a52-aa24-9be524ab43f3 Type: basic Technology: simple_bridge Subclass: basic Creator: Name: Video-Mode: none Video-Source-Id: Num-Channels: 2 Num-Active: 2 Duration : 00:04:05 Channel: Local/202@verified-00000004;2 Channel: SIP/202-ABC-0000001b switchboard1CLI> bridge show b59745d6-d632-4d74-90e5-c0250e1198c5 Id: b59745d6-d632-4d74-90e5-c0250e1198c5 Type: basic Technology: simple_bridge Subclass: basic Creator: Name: Video-Mode: none Video-Source-Id: Num-Channels: 2 Num-Active: 2 Duration: 00:06:38 Channel: SIP/TESTSIP-00000019 Channel: Local/202 @verified-00000004;1

2

Howdy, welcome to the forums!

You might want to review some of the formatting options for your post. Specifically, there is a Discourse Markup dialect outlined more in the forum FAQ that you can take advantage of, to help others see more clearly what you shared.

That said, posting a SIP packet trace might help, as many phones do things differently when it comes to complicated transfer scenarios.

3

It’s not actually a markup problem. I think it is probably a screen scraping problem; the lack of newlines is present in the raw forum text.

In general, one should use log files, not screen scrapes.

1 Like
4

Good point, will add to the FAQ, thank you.