TIPS XiVO - no audio when forwarding to cell

[wa4zlw]

@briankelly63:

Called in from cell 1 picked up on grandstream. audio. Transfered call to cell 2. Seeing rtp logging cell 2 never rings this time. CANCELLED transfer audio back to original; tried transfer again to alternate number on cell 2 same thing.

tried calling from my Incredible 13/12 using grandstream DP750 cordless into the grandstream desktop. as expected same as above. Hit transfer I heard the hold music coming out the handset originated on the grandstream and saw again lots of RTP logging but no dialout

I think we need to get upgraded first then try this again.

Thanks leon

EDit: CORRECTION: Didn't do transfer correctly from desk phone. it works with 2 way audio. did this multiple times with multiple source phones. SO right now it fails only when doing the auto call forward on initial ring in.

 

[wa4zlw]

@wardmundy:

Here's the snippet I posted from the upgrade. I couldnt find a way to attach a file here (probably under my nose)

root@xivo:~# xivo-upgrade
Upgrading xivo-upgrade
Reading package lists...
Building dependency tree...
Reading state information...
xivo-upgrade is already the newest version.
0 upgraded, 0 newly installed, 0 to remove and 74 not upgraded.
.
.
.
installed version : 16.12
proposed update : 16.13
Would you like to upgrade your system (all services will be restarted) [Y/n]? Y
root@xivo:~#

Leon

 

[wa4zlw]

@wardmundy: any updates on the update process?

thanks leon

 

[Josh North]

Did you ever resolve this issue? I'm running into what seems to be the exact same thing (no audio in either direction on forwards to external numbers).

 

[wa4zlw]

@Josh North: Nope. I even installed the latest codebase and it does the same thing. The no audio is ONLY on forwarded calls. I can receive and make calls from extensions.

Leon

 

[briankelly63]

I know you've mentioned you have a very tight firewall.... When 'answered' normally it's probably done in such a way that the firewall is able to handle it but when forwarded the packets are such that the firewall can't handle it. Try opening the RTP range to your server...

Take a look at this thread and then this section:


http://community.freepbx.org/t/solved-no-sound-when-do-followme-on-ext-call-to-ext-number/29367/7

"Okay, so I found the problem, but I'm not sure how to fix it.

Dicko, once again, you lead me down the right road. It was an network issue.

On our firewall, we allow all traffic from five IP's, the ones our VSP AnveoDirect says to allow SIP from. Well, I noticed that when a call isn't in some way "answered" by asterisk (this seems to be the case when you have a DID with an inbound route that goes to an outside number via Misc. Destinations), the IP that the RTP stream is coming from is not one of the five Anveo IP's, hence the traffic is blocked by the firewall.

I figured this out by looking at the CDR->SIP trace logs for each call. Toward the middle of the log, there's an IP that doesn't belong to Anveo or me. Here's what it looks like"

 

[wa4zlw]

@briankelly63: I looked at tat thread. I already have progressinband=yes BUT can not find prematuremedia on any config panels.

regarding the firewall, RTP outbound is 10000-20000, 5004 and 42872 UDP

And when the call comes in and forwarded I can see the SIP go out BUT no RTP ever hits the firewall at least in the monitor window. I have some packet captures from last week if anyone wants to look.

Thanks leon

 

[briankelly63]

http://serverfault.com/questions/38...risk-to-send-audio-before-call-is-established

noteworthy:
prematuremedia=no ;this does the exact opposite of what everyone assumes it does

 

[wa4zlw]

ok but I can not find where in the xivo gui to set this :-(

 

[briankelly63]

here's a few tidbits... looks like it's the ISDN setting but there may be a bug, maybe it's been fixed..

https://www.google.com/search?q=xivo+prematuremedia&ie=utf-8&oe=utf-8

 

[wa4zlw]

I just found it when you sent the reply

GENERAL SETTINGS --> SIP --> SIGNALLING

ISDN compatibility (early media):

I've changed it and it now works!. Of course google has the old site need to change it to wazo.community

thanks robert stack no longer needed!

Leon

 

[briankelly63]

One of the issues I have with Xivo is the renaming of certain Asterisk standard config item's. Every config item should have a 'hover' balloon that says what it really affects.

 

[wa4zlw]

my problem is the context sensitive help isn't - I think we're talking about the same thing.

a few other snits:

1. Where do we define outbound CNAME for a extension/user or trunk. SO easy in Freepbx
2. there is no ASterisk Info type display
3. there is no UCP type functionality so this is no good for my synagogue where the Rabbi uses a browser to go through VMs and manage them

 

[wardmundy]

Re: #3 Have you looked at the XiVO Client??

 

[wa4zlw]

looking at it now...i dont see anywhere that a list of VMs would be shown and what you could do with them like play, email, etc like in UCP. looks lilke you can call your VM but if you have a 100 VMs then you have to sit through them all. Also there is still french on the people object on the left side

 

[briankelly63]

It's easy to see why there is confusion about these two settings especially when used together....... Doesn't really explain why the RTP stream doesn't ultimately work after a delay.

progressinband=never

If we should generate in-band ringing
always use 'never' to never
use in-band signalling, even in cases
where some buggy devices might not render it
Valid values: yes, no, never Default: never

prematuremedia=no

Some ISDN links send empty media frames before
the call is in ringing or progress state. The SIP
channel will then send 183 indicating early media
which will be empty - thus users get no ring signal.
Setting this to "yes" will stop any media before we have
call progress (meaning the SIP channel will not send 183 Session
Progress for early media). Default is "no". Also make sure that
the SIP peer is configured with progressinband=never.


http://doxygen.asterisk.org/asterisk1.4/Config_sip.html

 

[wa4zlw]

GM @briankelly63: Yes confusing to say the least. I am using progress=yes, notifyringing checked and earlymedia checked. so far so good.

at work we have a siemens switch and its handled by unify. (multinational company) there's been issues where calls outside the LOCAL trunks ring yet calls going out the LONG DISTANCE trunk dont ring. They've been dealing with this for months. That's not my realm so I am out of the loop but way back I suggested somethings to them. Obviously a config issue it seems to me.

 

[wa4zlw]

here's another issue probably need to open a new thread ... an inbound call comes in and is forwarded out. the CALLERID displayed on the cell is that of the PBX NOT the actual callee yet the number displayed on the local extension is correct :-(

 

[briankelly63]

i think there are some threads on that already

 

[wa4zlw]

i think u are right. I did dial in and watched the xivo client and never saw the call come in so I guess that is not a realtime client?