[asterisk-bugs] [JIRA] (ASTERISK-29253) Incorrect bridging on transfer

Sat Apr 16 11:47:57 CDT 2022

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29253?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258804#comment-258804 ] 

Yury Kirsanov edited comment on ASTERISK-29253 at 4/16/22 11:47 AM:
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Joshua,
I've contacted Sangoma sales via email and phone - nobody wanted to discuss a possibility of fixing this issue on a paid basis with me. I've asked multiple times if anyone from Sangoma will be able to have a look at the issue - and got an answer 'Maybe some day, but no timeframes'. That's where my 'not too interested' is coming from. The issue is here for more than a year and I don't agree that it was a minor issue as it was a major issue for our business here in Australia. So I've done all the steps in sorting out this issue for our company and the patch did help to resolve it. I can't guarantee it's a correct patch or it will resolve same issue for everyone as I'm not a developer, again - we had to hire someone to resolve it for us in our particular case. I'm thankful for your support and appreciate that you're replying to us on Saturday, but it's my weekend too and also we have a long weekend due to public holidays in Australia - it's Easter time. But I stiil found some time to present the solution to this problem even though nobody wanted to help us in resolving this. So please understand me too - I don't want to go through the whole process of submitting a proper Git patch due to the lack of response from Sangoma. I was very frustrated nobody wanted to help us even on a paid basis. I do understand that Asterisk is a free software but still it's hard to understand why new features in Asterisk are given priority over fixing old bugs. And this bug comes from at least Asterisk 11. As you can see by comments above - a lot of people are affected by it. Again - my apologies if I offended you or somebody else and hope that the patch is correct for this issue.

I can also add that I was able to apply it to Asterisk 18.11.2 today and it is working fine under our circumstances. Thanks.

was (Author: lt_flash):
Joshua,
I've contacted Sangoma sales via email and phone - nobody wanted to discuss a possibility of fixing this issue on a paid basis with me. I've asked multiple times if anyone from Sangoma will be able to have a look at the issue - and got an answer 'Maybe some day, but no timeframes'. The issue is here for more than a year and I don't agree that it was a minor issue as it was a major issue for our business here in Australia. So I've done all the steps in sorting out this issue for our company and the patch did help to resolve it. I can't guarantee it's a correct patch or it will resolve same issue for everyone as I'm not a developer, again - we had to hire someone to resolve it for us in our particular case. I'm thankful for your support and appreciate that you're replying to us on Saturday, but it's my weekend too and also we have a long weekend due to public holidays in Australia - it's Easter time. But I stiil found some time to present the solution to this problem even though nobody wanted to help us in resolving this. So please understand me too - I don't want to go through the whole process of submitting a proper Git patch due to the lack of response from Sangoma. I was very frustrated nobody wanted to help us even on a paid basis. I do understand that Asterisk is a free software but still it's hard to understand why new features in Asterisk are given priority over fixing old bugs. And this bug comes from at least Asterisk 11. As you can see by comments above - a lot of people are affected by it. Again - my apologies if I offended you or somebody else and hope that the patch is correct for this issue.

I can also add that I was able to apply it to Asterisk 18.11.2 today and it is working fine under our circumstances. Thanks.

> Incorrect bridging on transfer
> ------------------------------
>
>                 Key: ASTERISK-29253
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29253
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Bridges/bridge_simple
>    Affects Versions: 16.15.0, 18.1.1
>         Environment: Ubuntu Linux 18.04.5 LTS
>            Reporter: Yury Kirsanov
>            Assignee: Unassigned
>         Attachments: bridge_simple.tar.gz, bridge_softmix.tar.gz, call_flow.txt
>
>
>  We have an Asterisk server and one SIP device registered with it and one SIP trunk to another system. Also we another SIP device to call for test purposes.
> Here's a simplified diagram of call flow with attended transfer we're trying to achieve:
> SIP Device A -> Asterisk PBX -> SIP Trunk -> External User -> Attended transfer -> Asterisk PBX -> SIP Device B.
> SIP Device A originates a call to some pre-defined number that's routed into SIP trunk (TLS+SRTP). Remote party behind SIP trunk ("External user") answers this call and then starts attended transfer to SIP Device B on Asterisk PBX. That SIP trunk uses Re-INVITE with Replaces header in order to complete transfer. During transfer SIP Device A can hear MOH after External User initiated attended transfer to SIP Device B. Then External User tries to complete the transfer connecting SIP Device A with SIP Device B. Call connects but SIP Device A continues to hear Music On Hold while SIP Device B can hear what SIP Device A says.
> Now, if we unload module 'bridge_simple' Asterisk PBX starts to use 'bridge_softmix' module and connect calls correctly, SIP Device A can establish two way communication with SIP Device B. But during attended transfer no MOH is played at all even though Asterisk shows messages like 'Starting music on hold'. And without 'bridge_simple' no Music On Hold is played at all even if we set up just a local extension that plays MOH, like this:
> exten=>100,1,Answer()
> exten=>100,n,MusicOnHold()
> If we load bridge_simple then MOH is played fine but during transfer calls are bridged incorrectly.
> I'm happy to provide full logs upon request as I don't want to edit them.
> Also, if we use SIP REFER method for transferring calls there's another issue - when External User tries to finalize call transfer Asterisk drops established call to SIP Device B and immediately re-dials it. I believe this happens because SIP Trunk is not passing Replaces in Refer-To header, but that's another issue.

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