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Summary:ASTERISK-23757: One way audio and MOH at the Background for destination while transferring the calls
Reporter:Siva (siva.percipia)Labels:
Date Opened:2014-05-19 09:25:44Date Closed:2014-06-25 08:57:29
Priority:MajorRegression?
Status:Closed/CompleteComponents:
Versions:1.8.12.2 Frequency of
Occurrence		Constant
Related
Issues:
Environment:OS-CentOS release 5.8, Asterisk 1.8 with Freepbx 2.9Attachments:
Description:We are experiencing the problem with couple of sites.Below is our setup

Setup 1 Freepbx(Asterisk) - Sip Trunk - Call Manager
Setup 2 Freepbx(Asterisk) - Call Transfer over T1 trunk - Call Manager

Note:It's working fine if we call the extension directly.Issue occurs only during the call transfer

We are receiving calls to asterisk from outside and we are transferring the calls to another pbx cisco call manager over sip trunk.While transferring the calls they unable hear and they hear only MOH at the background.
{noformat}
FreePBX-CUCM
host=CUCM IP
type=friend
disallow=all
allow=ulaw
qualify=yes
dtmfmode=rfc2833
nat=no

FreePBX-Digium T1 GW(G200)
host=G200IP
username=xxxxxx
secret=xxxxxx
type=peer
allow=all
qualify=yes
nat=no
insecure=invite,port
context=from-trunk
{noformat}
I already opened a case with digium case # is 00397744.Since i created ticket for gateway they are unable to provide more assistance on this.Please check the below email from digium support.

I apologize for the delay, getting to the bottom of this issue has taken a fair amount of research. The good news is that this issue does not involve your G200 gateway. The bad news is that this is a bug with Asterisk itself. The offending parties are actually the two PBXs themselves.

The problem stems from the way Cisco Call Manager formats its SDP reinvite, and how Asterisk interprets the request to re-activate an inactive media session when an offer does not contain a=sendrecv.

Developers are aware of this issue, and are actively working to address it. It is a somewhat rare occurrence to see this issue, but luckily the information from your case can be added to the others to enhance development of the fix. I do not have a time frame on when a resolution will be released.

I am sorry that I could not currently provide more assistance. As this ticket deals with your gateway device and not Asterisk, I will be closing the ticket.

I recommend keeping an eye on the Asterisk mailing list if you'd like to keep abreast of development.
Comments:By: Michael L. Young (elguero) 2014-05-19 13:57:02.065-0500

Siva,

Please try setting "ignoresdpversion=yes" for the peer configuration "FreePBX-CUCM" and report back.

Also you are several versions behind on the 1.8 version of Asterisk.  I would recommend upgrading to a more recent version.
By: Siva (siva.percipia) 2014-05-19 17:15:08.854-0500

Michael,

Thank you for your quick response, i tried as per your suggestion now they are able to hear my voice very low but still MOH is playing at the background.The voice is low due to MOH is playing at the background.
{noformat}
 -- Started music on hold, class 'default', on SIP/digiumtrunk-0000074b
   -- Called SIP/siptrunkccm/4100
   -- SIP/siptrunkccm-00000754 is ringing
   -- SIP/siptrunkccm-00000754 is ringing
   -- SIP/siptrunkccm-00000754 is ringing
   -- Call on SIP/siptrunkccm-00000754 placed on hold
   -- Started music on hold, class 'default', on SIP/siptrunkccm-00000754
   -- SIP/2656-00000753 requested media update control 20, passing it to SIP/siptrunkccm-00000754
   -- Stopped music on hold on SIP/digiumtrunk-0000074b
   -- SIP/siptrunkccm-00000754 answered SIP/digiumtrunk-0000074b
   -- Locally bridging SIP/digiumtrunk-0000074b and SIP/siptrunkccm-00000754
[May 19 15:31:49] WARNING[14350]: channel.c:4934 ast_write: Codec mismatch on channel SIP/siptrunkccm-00000754 setting write format to slin from ulaw native formats 0x4 (ulaw)
{noformat}
By: Michael L. Young (elguero) 2014-05-20 09:20:45.574-0500

Siva,

Thanks for reporting back.  For an explanation on why "ignoresdpversion=yes" works when dealing with Cisco, please see ASTERISK-20642 and a more detailed explanation on ASTERISK-20633.

In regards to MOH still playing, something tells me that this may have already been fixed but I was unable to find anything with my searching.  Is it possible to try a newer version of 1.8?  That version was released in May of 2012.  There have been several security issues and bug issues fixed in the past 2 years.

When you do try a newer version of Asterisk 1.8, please attach a full debug log.  [Collecting Debug Information|https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information].

Thanks
By: Rusty Newton (rnewton) 2014-06-06 08:39:52.262-0500

[~siva.percipia] are you able to follow Michael's direction for testing with the latest 1.8?
By: Siva (siva.percipia) 2014-06-06 09:08:00.065-0500

Hello Rusty
I followed as per Michael suggestion, some times it's working and some times it's not working.I didn't upgraded the latest version in production environment.

I tried in my lab it's working.Sorry for the delay.

Michael thanks for your help on this.

Siva


By: Rusty Newton (rnewton) 2014-06-06 11:49:14.366-0500

Siva , you got it working in your lab, but not in production?

In both cases are you implementing ignoresdpversion=yes, but the difference is that the lab is using the latest version of 1.8?




By: Rusty Newton (rnewton) 2014-06-25 08:57:08.397-0500

No further response, and Siva's last commented sounded like he got things working by using the ignoresdpversion=yes option. Closing this out. Siva, please contact a bug marshal in #asterisk-bugs or #asterisk-dev on irc.freenode.net if you still feel there is a bug.