Iphone no audio soft/sip phones external only

Asterisk Asterisk Support
1

can not get audio on sip/soft phone apps for iPhone, Bria, media 5, 3cxPhone. external only. Everything works if I am on the local network.

I can make and send calls but there no audio.

Everything works internal and externa for X-Lite on my mac.

sip.conf

[101]
type=friend
username=100
secret=1234
host=dynamic
context=internal
callerid=Iphone
canreinvite=no
reinvite=no
qualify=1000
nat=yes
mailbox=100@default
; allow codec
allow=all
;allow=ulaw
;allow=alaw
;allow=g723.1
;allow=g729
;allow=ilbc
;allow=gsm

extension.conf

[general]
static=yes
writeprotect=yes
priorityjumping=no
autofallthrough=no

[globals]
OFFICE_OPEN_OVERRIDE=
;include trunks.include
DIALOUT=9
INTERNATIONAL-PREFIX=011
RINGTIME=30
TL_DASH=-
TL_MULTI=1
OPERATOR=0
RECORDING_FORMAT=WAV
TL_ENABLE_MAXCALLS_CHECK=0
autofallthrough=yes

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[incoming_calls]

[internal]
exten => s,1,Verbose(1|Echo test application)
exten => s,n,Echo()
exten => s,n,Hangup()

exten => 101,1,Log(NOTICE,“101 ACCOUNT”)
exten => 101,n,Dial(SIP/101,120,Tt)

exten => 102,1,Log(NOTICE,“102 ACCOUNT”)
exten => 102,n,Dial(SIP/102,120,Tt)

[phones]
include => internal
include => default

[callme]
exten => s,1,Verbose(1|Echo test application)
exten => s,n,Echo()
exten => s,n,Hangup()

exten => 101,1,Log(NOTICE,“101 ACCOUNT”)
exten => 101,n,Dial(SIP/101,120,Tt)

exten => 102,1,Log(NOTICE,“102 ACCOUNT”)
exten => 102,n,Dial(SIP/102,120,Tt)

2

Very few people with NAT need nat=; most need externip or stun options.

You haven’t adequately described your network topology, but I cannot see any options that would allow it to operate across a NAT router.

canreinvite is obsolete, but you haven’t said which version of Asterisk you are using. At least on versions that support canreinvite, there is no “reinvite” option.

3

Asterisk: 1.8.8.0-1digium1~oneiric / ubuntu
network topology: Star (home network)

I set canreinvite to yes still no audio.
On the advanced settings on Bria app I stet the Current Strategy to Server Manage

The phone is connecting from external and calling other extensions internal and external, there is no audio on the iPhone. Internally there is audio

X-Lite soft-phone running on an internal and external machine and it connects and has audio.

4

Why did you put nat=yes if your network has no external connectivity? (NB most people use this inappropriately even when they do have NAT.)

I didn’t say use “canreinvite”. I said it was meaningless to current versions of Asterisk!

One way audio is normally associated with network topology issues. As I assume star means a single nBaseT hub and no external connectivity, and a single sub-network, I’m not sure how that could happen.

5

The server is on my home network and I have the DMZ pointing to it.

When the iPhone is connected to the home network everything works.

If I connect to the server through 3g and my office (not home office), the phone will connect to the server at home, call and receive call from the server and other extensions. But, there is no audio.

When I connect from my office using X-lite on my computer, it has audio and everything works.

6

So is the topology:

I-phone (public IP) - 3G (no VoIP blocking) - the cloud - xDSL (public address) - NAT router (private address) - office LAN - loopback to same NAT router (I-phone public address passed through) - cloud - xDSL (public address) - Home NAT router (private address) - Asterisk (private address) - destination phone (private address)?

Note there is no VLAN in the above.

7

topology is more like this…

This works (home)
connects, send, receive and has audio:

iphone to xlite
iPhone (private address) - NAT router (private address) - Astrisk (private address) - NAT router (private address) - X-Lite (private address)

x-lite to iphone
iPhone (private address) - NAT router (private address) - Astrisk (private address) - NAT router (private address) - X-Lite (private address)

asterisk to x-lite (recorded outgoing file)
Astrisk (private address) - NAT router (private address) - X-Lite (private address)

asterisk to iPhone (recorded outgoing file)
Astrisk (private address) - NAT router (private address) - iphone (private address)

This works (office)
connects, send, receive and has audio

asterisk to x-lite (recorded outgoing file)
Astrisk (private address) - NAT router (private address)- xDSL (public address) - cloud - xDSL (public address) - NAT router (private address) - X-Lite (private address)

Does not work (office)
connects, send, receive and no audio

iphone to xlite
iPhone (private address) - NAT router (private address) - xDSL (public address) - cloud - xDSL (public address) - NAT router (private address) - Astrisk (private address) - NAT router (private address)- xDSL (public address) - cloud - xDSL (public address) - NAT router (private address) - X-Lite (private address)

xlite to iphone
X-Lite (private address) - NAT router (private address) - xDSL (public address) - cloud - xDSL (public address) - NAT router (private address) - Astrisk (private address) - NAT router (private address)- xDSL (public address) - cloud - xDSL (public address) - NAT router (private address) - iPhone(private address)

asterisk to iphone (recorded outgoing file)
Astrisk (private address) - NAT router (private address)- xDSL (public address) - cloud - xDSL (public address) - NAT router (private address) - iPhone(private address)

Does not work
connects, send, receive and no audio (3g):

iphone to x-lite
iPhone (ip from Verison) - 3G (Verison) - cloud - xDSL (public address) - NAT router (private address) - Astrisk (private address) - NAT router (private address)- xDSL (public address) - cloud - xDSL (public address) - NAT router (private address) - X-Lite (private address)

x-lite to iphone
X-Lite (private address) - NAT router (private address) - xDSL (public address) - cloud - xDSL (public address) - NAT router (private address) - Astrisk (private address) - NAT router (private address)- xDSL (public address) - cloud - 3G (Verison) iPhone (ip from Verison)

Asterisk to iphone (recorded outgoing file)
Astrisk (private address) - NAT router (private address)- xDSL (public address) - cloud - 3G (Verison) iPhone (ip from Verison)

8

Same issue i was, and I set the externip with my static ip of my internet provider then voice is coming. but still i m facing the issue in xlite to xlite call there is no voice.

9

I am able to get the iphone soft phone to connect from off site to the asterisk server and then to the x-lite softphone which is off site with audio on both the iphone and x-lite. I used both Brai and Medit5-fone on the iphone with calling and receiving success.

I did this by adding
localnet=xxx.xxx.xxx.xxx /255.255.255.0 ; local router
externip=xxx.xxx.xxx.xxx ; public ip of router

the new problem now is that the x-lite has no sound when it receives a call from the asterisk box via a call file. (auto dailer). but if i remove the localnet and externip it there is audio from the test message recording

and asterisk will not call the iphone via call file and will not delete the file

call file: -------------------------------------------------------------

Channel: sip/101
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Callerid: asterisk
AlwaysDelete: Yes

Application: Playback
Data: /var/lib/asterisk/sounds/custom/testMsg

error when call call file calls iphone-------------------------------

[Dec 30 11:16:50] WARNING[25310]: pbx_spool.c:278 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test3.call: Operation not permitted
– Attempting call on sip/101 for application Playback(/var/lib/asterisk/sounds/custom/testMsg) (Retry 1)
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
[Dec 30 11:17:02] WARNING[25310]: pbx_spool.c:278 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test3.call: Operation not permitted
– Attempting call on sip/101 for application Playback(/var/lib/asterisk/sounds/custom/testMsg) (Retry 1)
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
[Dec 30 11:17:21] NOTICE[9854]: pbx_spool.c:353 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
[Dec 30 11:17:21] WARNING[9854]: pbx_spool.c:278 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test3.call: Operation not permitted
[Dec 30 11:17:23] WARNING[25279]: chan_sip.c:3629 retrans_pkt: Retransmission timeout reached on transmission wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response

10

Some X-Lite versions, at least, have broken re-invite handling. If that is not the case, you need to provide the SIP trace. (They ignore re-invites, when they should either honour or reject them.)

You appear to be running non-root with unsuitable file ownership. In my view, you should run Asterisk as root unless you have a very good understanding of Linux file and system call permissions.

11

got the x-lite problem fixed. added the following

nat=yes
canreinvite=update,nonat

Now to see why it will not send the automated message to the iphone

12

I don’t think nat is doing anything for you. canreinvite is a deprecated name for directmedia. I suspect that directmedia=no would be safer with X-Lite. Yours is probably working because you also have a NAT case.

13

actually it is now:

nat=no
canreinvite=update,nonat

still wondering why it will not send the run the call file for the recorded message to the iphone…

14

Hi every one

I have installed Asterisk 1.8.7 in Ubuntu oneiric. And I have a LAN with Asterisk server and two phones:
A hard phone SPA 942 (extension 2000) and a 3CXPhone (extension 2004) both are in the same LAN and have the IP 192.168.0.3 and 192.168.0.2 respectively. With a subnet 192.168.0.0/255.255.255.0
The two phones above work perfectly I can make a call from and to each of witch and I have two way audio. It’s Ok

And I have another 3CXPhone in an external LAN (extension 2001) and an IP 192.168.1.36
This phone can ring the two local phones But with no sound (no audio at all).
And no one of the local phones (2000-2004) can ring the external phone (2001)

The external phone (2001) can register to my asterisk server but it’s status is UNREACHABLE.
I can recognize that all the phones are registered with there local IP address.
when I execute the command :
sip show peers I get this

Name/username Host Dyn Forcerport ACL Port Status
2000/hard phone 192.168.0.3 D 5060 OK
2001/soft phone 192.168.0.2 D 3697 OK
2004/Az soft Phone 192.168.1.36 D 3215 UNREACHABLE

The sip.conf file

localnet =192.168.0.0./255.255.225.0
externip=
nat=comedia
qualify=yes

all the peers have:
type = friend
host = dynamic

and my iptables is as follow:

iptables --flush           		
iptables --table nat --flush
iptables --delete-chain     
iptables --table nat --delete-chain
iptables --table nat --append POSTROUTING --out-interface eth0 -j MASQUERADE
iptables --append FORWARD --in-interface eth1 -j ACCEPT

echo 1 > /proc/sys/net/ipv4/ip_forward

I’m struggling here simply couldn’t make it work. And I believe that it is a network issu.
Any help will be appreciated.
Best regards