TIPS XiVO - no audio when forwarding to cell

wa4zlw

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hi there:

Incredible PBX 13.2016-08-29 for xivo

Asterisk: UP NGINX: UP Consul: UP
Postfix: UP IPtables: UP SSH: UP
LAN port: UP Fail2Ban: UP Munin: UP
AGetty: UP PostgreSQL: UP NR VPN: UP

RAM:132MB Debian 8 (jessie) Disk:7.7GB

Asterisk 13.11.2 XiVO PBX 16.12

Private IP: 10.161.51.15

Public Info: 70.x.y.z

System Time: Wed Oct 26 20:04:55 EDT 2016


< OK >
I'm testing this under a HyperV guest. I got inbound working based on the info from another thread whereby the wrong context was set.

In and out calls work with my Grandstream endpoint with 2way audio. When an inbound comes in and forwards to my sideline # (or my cell primary) the call completes BUT there is no audio either way.

This is behind my Watchguard firewall and is a test. It will eventually end up replacing my DIgital Ocean instance of Incredible 13-12 which is stable.

anyone else experiencing anything similar? I tried putting my public IP in the entry box and that didnt change anything.

THanks leon
 
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As mentioned these are always double NAT, Firewall config issues. However.... I remember that at least in earlier Asterisk versions Asterisk bridges some calls differently then others. For example a forward might not provide two way audio but a simultaneous ring from a ring group to your cell phone may work. A transfer may or may not work too. I forget which was which but try a few options.

In pfsense (a firewall) the concept of a static port exists which prevents the firewall from rewriting ports....: By default pfSense rewrites the source port on all outbound traffic. This is necessary for proper NAT in some circumstances such as having multiple SIP phones behind a single public IP registering to a single external PBX. With a minority of providers, rewriting the source port of RTP can cause one way audio. In that case, setup manual outbound NAT and Static Port on all UDP traffic potentially with the exclusion of UDP 5060.

Of course check SIP ALG and then NAT setting on the extension although that shouldn't affect it.
 
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wa4zlw

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Hi there...thanks for the replies....

call comes in from voip.ms from cell phone to XiVO.
XiVO routes to grandstream AND to my other cell (using sideline alternate # or the primary #) ringing the other cell also goes back out over voip.ms. Both endpoints ring. If I pickup the grandstream 2 way audio. If I pickup the cell no audio either way. definitely seems related to only when it rings the 2nd cell.

I spent a great deal of time over a year ago, since I am a network engineer, on getting my watchguard firewall fine tuned; so I am confident it is NOT a firewall issue. THere is no SIP-ALG running since I found a bad bug in it over a year ago and waiting for a fix.

This seems to be something related to XiVO.

I'm not going to deploy this on D.O. until I finish my testing here locally.

THanks leon
 
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In Xivo under Sip Protocol Properties check the default settings tab. Check 'Inband RING event:' and 'NAT':. Try turning IRE on. experiment with NAT
 

wa4zlw

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Under SIP PROTOCOL PROP NAT is YES, IRE is ON
Tried putting my public in as well and that didnt change anything.

leon
 

wa4zlw

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ward...logging in with ssh doesnt change anything when it attempts to update things.

ldz
 

wa4zlw

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i found the procedure in one of the posts...doing it now
 

wa4zlw

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Code:
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:~#
 

wa4zlw

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here's tail end of RTP logging:

Code:
[Oct 27 11:02:14]     -- Called SIP/1NNNN_Test/484NNXXXXX
[Oct 27 11:02:15]     -- SIP/1NNNN_Test-00000008 is making progress passing it to Local/484NNXXXXX@default-00000002;2
[Oct 27 11:02:15]     -- Local/484NNXXXXX@default-00000002;1 is making progress passing it to SIP/1NNNN_Test-00000006
[Oct 27 11:02:25]     -- SIP/1NNNN_Test-00000008 answered Local/484NNXXXXX@default-00000002;2
[Oct 27 11:02:25]     -- Local/484NNXXXXX@default-00000002;1 answered SIP/1NNNN_Test-00000006
[Oct 27 11:02:25]     -- SIP/userid-00000007 Internal Gosub(hangup_handlers,userevent,1) start
[Oct 27 11:02:25]     -- Executing [userevent@hangup_handlers:1] NoOp("SIP/userid-00000007", "Sending Hangup userevent") in new stack
[Oct 27 11:02:25]     -- Executing [userevent@hangup_handlers:2] UserEvent("SIP/userid-00000007", "Hangup,XIVO_USERUUID: c3477860-fabb-4293-8e41-25a9da41227f") in new stack
[Oct 27 11:02:25]     -- Executing [userevent@hangup_handlers:3] Return("SIP/userid-00000007", "") in new stack
[Oct 27 11:02:25]   == Spawn extension (default, s, 1) exited non-zero on 'SIP/userid-00000007'
[Oct 27 11:02:25]     -- SIP/userid-00000007 Internal Gosub(hangup_handlers,userevent,1) complete GOSUB_RETVAL=
[Oct 27 11:02:25]     -- Channel Local/484NNXXXXX@default-00000002;1 joined 'simple_bridge' basic-bridge <4a5d90e2-1605-422f-a046-9abba9916b3f>
[Oct 27 11:02:25]     -- Channel SIP/1NNNN_Test-00000006 joined 'simple_bridge' basic-bridge <4a5d90e2-1605-422f-a046-9abba9916b3f>
[Oct 27 11:02:25]   == Extension Changed 701[default] new state Idle for Notify User iffuxrxf
[Oct 27 11:02:25]     -- Channel SIP/1NNNN_Test-00000008 joined 'simple_bridge' basic-bridge <35f7eb1a-5977-4041-9932-4856af252fcd>
[Oct 27 11:02:25]     -- Channel Local/484NNXXXXX@default-00000002;2 joined 'simple_bridge' basic-bridge <35f7eb1a-5977-4041-9932-4856af252fcd>
[Oct 27 11:02:42]     -- Channel SIP/1NNNN_Test-00000008 left 'simple_bridge' basic-bridge <35f7eb1a-5977-4041-9932-4856af252fcd>
[Oct 27 11:02:42]     -- Channel Local/484NNXXXXX@default-00000002;2 left 'simple_bridge' basic-bridge <35f7eb1a-5977-4041-9932-4856af252fcd>
[Oct 27 11:02:42]   == Spawn extension (outcall, dial, 7) exited non-zero on 'Local/484NNXXXXX@default-00000002;2'
[Oct 27 11:02:42]     -- Channel Local/484NNXXXXX@default-00000002;1 left 'simple_bridge' basic-bridge <4a5d90e2-1605-422f-a046-9abba9916b3f>
[Oct 27 11:02:42]     -- Channel SIP/1NNNN_Test-00000006 left 'simple_bridge' basic-bridge <4a5d90e2-1605-422f-a046-9abba9916b3f>
[Oct 27 11:02:42]   == Spawn extension (user, s, 37) exited non-zero on 'SIP/1NNNN_Test-00000006'
xivo*CLI>
 

wa4zlw

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Hi Sylvain...Not that I am aware of I believe it's all through the pbx

ldz
 

Sylvain Boily

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I think you don't have audio because you don't know where your rtp traffic going. You need to check your configuration. Check the SDP in SIP packet when you perform a transfer.
 

wa4zlw

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I'm not doing the transfer. the system is ringing both at the same time.
 
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While you are waiting for the next reply humor me and try what I suggested
"under Sip Protocol Properties check the default settings tab: Check 'Inband RING event:' ':. Try turning Inband Ring Event on.

It does more than it implies and forces the RTP
 
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Also if it's currently set up as a forward then set it up as a group ring and vice-versa. Additionally test a blind transfer. You'll probably see a difference in the traces as to how the bridging is done.
 

wa4zlw

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Hi there!

inband ringing IS setup as I mentioned yesterday, as well as I tried putting my public IP in and out.
I'm at work so I can't do that until I get home tonight

I only have one extension, one user and have my mobile setup in my user with pre mobility added

Again as long as I ignore the ringing on my cell it works. I will try and setup a few other items.

But, Ward said to upgrade and I reported last night that it tried to but then says it is on the current release even though the version are different

Leon
 

wa4zlw

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@wardmundy: yes, I posted a snippet of the results above last night. it does not upgrade. Please note awhile back when this first came out I was able to upgrade to the version I am on now.

Leon
 

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