[asterisk-bugs] [JIRA] (ASTERISK-24698) Asterisk 13.1.0 PJSIP over TCP gives PJSIP_ETPNOTSUITABLE
Richard Mudgett (JIRA)
noreply at issues.asterisk.org
Fri Jan 16 09:57:35 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24698?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Richard Mudgett updated ASTERISK-24698:
---------------------------------------
Description:
Hi - I raised this initially on http://forums.asterisk.org
but am getting no response there. I hope someone here can help.
I've just upgraded a local test system from Asterisk 13.0.1 to 13.1.0 and am having problems using pjsip over TCP (same config worked fine on 13.0.1).
I have 2 virtualbox VMs.
First VM runs OpenSIPS and rtpproxy.
Second VM runs Asterisk 13.1.0. chan_pjsip is configured to connect Asterisk to OpenSIPS (over TCP), with Linphone client connected to Asterisk using chan_sip (over UDP).
Asterisk registers ok with OpenSIPS over TCP on startup, but calls initated by the Linphone client (that should get forwarded on by Asterisk to OpenSIPS and then directed back to the same Asterisk) are currently failing. Call INVITE never gets forwarded to OpenSIPS by Asterisk.
I get:
{noformat}
*CLI> Dialing call 1002 <+449991002> to PJSIP/+44999123 at mytrunk....
[Jan 14 16:01:16] WARNING[20909]: pjsip:0 <?>: tsx0x7f4074009 ...Failed to send Request msg INVITE/cseq=13219 (tdta0x7f40740073e0)! err=171064 (Unsuitable transport selected (PJSIP_ETPNOTSUITABLE))
1002 <+449991002> to PJSIP/+44999123 at mytrunk DialStatus: CONGESTION HangupCause: 34
1002 <+449991002> to PJSIP/+44999123 at mytrunk DialStatus: CONGESTION HangupCause: 34 - using err_default
*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri...............................> <Status....> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.........................................................>
Match: <ip/cidr.........................>
Channel: <ChannelId......................................> <State.....> <Time(sec)>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
=========================================================================================
Endpoint: mytrunk Not in use 0 of inf
OutAuth: mytrunk/GW1
Aor: mytrunk 0
Contact: mytrunk/sip:10.10.20.12:5060 Unknown nan
Transport: simpletrans tcp 0 0 10.10.20.13:5060
Identify: mytrunk/mytrunk
Match: 10.10.20.12/32
{noformat}
pjsip.conf (for TCP test) contains...
{noformat}
;==============TRANSPORTS
[simpletrans]
type=transport
protocol=tcp
bind=10.10.20.13
;===============TRUNK
[mytrunk]
type=registration
transport=simpletrans
outbound_auth=mytrunk
outbound_proxy=sip:10.10.20.12\;transport=tcp
server_uri=sip:10.10.20.12
client_uri=sip:GW1 at 10.10.20.12
retry_interval=60
max_retries=1440
[mytrunk]
type=auth
auth_type=userpass
password=*******
username=GW1
[mytrunk]
type=aor
contact=sip:10.10.20.12:5060
[mytrunk]
type=endpoint
transport=simpletrans
context=externalIncoming
disallow=all
allow=gsm
outbound_auth=mytrunk
aors=mytrunk
force_rport=yes
direct_media=no
media_encryption_optimistic=no
[mytrunk]
type=identify
endpoint=mytrunk
match=10.10.20.12
{noformat}
If I change pjsip.conf (& OpenSIPS) to use UDP rather than TCP it all works as expected.
Prior to the upgrade I had the same config (excluding the new media_encryption_optimistic=no option) working fine with Asterisk 13.0.1. I don't think I've changed anything in my setup.
Is this a bug or do I need some additional (Asterisk 13.1.0 specific) pjsip.conf change to get this to work over TCP?
was:
Hi - I raised this initially on http://forums.asterisk.org
but am getting no response there. I hope someone here can help.
I've just upgraded a local test system from Asterisk 13.0.1 to 13.1.0 and am having problems using pjsip over TCP (same config worked fine on 13.0.1).
I have 2 virtualbox VMs.
First VM runs OpenSIPS and rtpproxy.
Second VM runs Asterisk 13.1.0. chan_pjsip is configured to connect Asterisk to OpenSIPS (over TCP), with Linphone client connected to Asterisk using chan_sip (over UDP).
Asterisk registers ok with OpenSIPS over TCP on startup, but calls initated by the Linphone client (that should get forwarded on by Asterisk to OpenSIPS and then directed back to the same Asterisk) are currently failing. Call INVITE never gets forwarded to OpenSIPS by Asterisk.
I get:
*CLI> Dialing call 1002 <+449991002> to PJSIP/+44999123 at mytrunk....
[Jan 14 16:01:16] WARNING[20909]: pjsip:0 <?>: tsx0x7f4074009 ...Failed to send Request msg INVITE/cseq=13219 (tdta0x7f40740073e0)! err=171064 (Unsuitable transport selected (PJSIP_ETPNOTSUITABLE))
1002 <+449991002> to PJSIP/+44999123 at mytrunk DialStatus: CONGESTION HangupCause: 34
1002 <+449991002> to PJSIP/+44999123 at mytrunk DialStatus: CONGESTION HangupCause: 34 - using err_default
*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri...............................> <Status....> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.........................................................>
Match: <ip/cidr.........................>
Channel: <ChannelId......................................> <State.....> <Time(sec)>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
=========================================================================================
Endpoint: mytrunk Not in use 0 of inf
OutAuth: mytrunk/GW1
Aor: mytrunk 0
Contact: mytrunk/sip:10.10.20.12:5060 Unknown nan
Transport: simpletrans tcp 0 0 10.10.20.13:5060
Identify: mytrunk/mytrunk
Match: 10.10.20.12/32
pjsip.conf (for TCP test) contains...
;==============TRANSPORTS
[simpletrans]
type=transport
protocol=tcp
bind=10.10.20.13
;===============TRUNK
[mytrunk]
type=registration
transport=simpletrans
outbound_auth=mytrunk
outbound_proxy=sip:10.10.20.12\;transport=tcp
server_uri=sip:10.10.20.12
client_uri=sip:GW1 at 10.10.20.12
retry_interval=60
max_retries=1440
[mytrunk]
type=auth
auth_type=userpass
password=*******
username=GW1
[mytrunk]
type=aor
contact=sip:10.10.20.12:5060
[mytrunk]
type=endpoint
transport=simpletrans
context=externalIncoming
disallow=all
allow=gsm
outbound_auth=mytrunk
aors=mytrunk
force_rport=yes
direct_media=no
media_encryption_optimistic=no
[mytrunk]
type=identify
endpoint=mytrunk
match=10.10.20.12
If I change pjsip.conf (& OpenSIPS) to use UDP rather than TCP it all works as expected.
Prior to the upgrade I had the same config (excluding the new media_encryption_optimistic=no option) working fine with Asterisk 13.0.1. I don't think I've changed anything in my setup.
Is this a bug or do I need some additional (Asterisk 13.1.0 specific) pjsip.conf change to get this to work over TCP?
> Asterisk 13.1.0 PJSIP over TCP gives PJSIP_ETPNOTSUITABLE
> ---------------------------------------------------------
>
> Key: ASTERISK-24698
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24698
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip
> Affects Versions: 13.1.0
> Environment: CentOS release 6.6 (Final)
> Linux localhost.localdomain 2.6.32-504.1.3.el6.x86_64 #1 SMP Tue Nov 11 17:57:25 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux
> Reporter: Ian Gilmour
> Assignee: Ian Gilmour
>
> Hi - I raised this initially on http://forums.asterisk.org
> but am getting no response there. I hope someone here can help.
> I've just upgraded a local test system from Asterisk 13.0.1 to 13.1.0 and am having problems using pjsip over TCP (same config worked fine on 13.0.1).
> I have 2 virtualbox VMs.
> First VM runs OpenSIPS and rtpproxy.
> Second VM runs Asterisk 13.1.0. chan_pjsip is configured to connect Asterisk to OpenSIPS (over TCP), with Linphone client connected to Asterisk using chan_sip (over UDP).
> Asterisk registers ok with OpenSIPS over TCP on startup, but calls initated by the Linphone client (that should get forwarded on by Asterisk to OpenSIPS and then directed back to the same Asterisk) are currently failing. Call INVITE never gets forwarded to OpenSIPS by Asterisk.
> I get:
> {noformat}
> *CLI> Dialing call 1002 <+449991002> to PJSIP/+44999123 at mytrunk....
> [Jan 14 16:01:16] WARNING[20909]: pjsip:0 <?>: tsx0x7f4074009 ...Failed to send Request msg INVITE/cseq=13219 (tdta0x7f40740073e0)! err=171064 (Unsuitable transport selected (PJSIP_ETPNOTSUITABLE))
> 1002 <+449991002> to PJSIP/+44999123 at mytrunk DialStatus: CONGESTION HangupCause: 34
> 1002 <+449991002> to PJSIP/+44999123 at mytrunk DialStatus: CONGESTION HangupCause: 34 - using err_default
> *CLI> pjsip show endpoints
> Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
> I/OAuth: <AuthId/UserName...........................................................>
> Aor: <Aor............................................> <MaxContact>
> Contact: <Aor/ContactUri...............................> <Status....> <RTT(ms)..>
> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
> Identify: <Identify/Endpoint.........................................................>
> Match: <ip/cidr.........................>
> Channel: <ChannelId......................................> <State.....> <Time(sec)>
> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
> =========================================================================================
> Endpoint: mytrunk Not in use 0 of inf
> OutAuth: mytrunk/GW1
> Aor: mytrunk 0
> Contact: mytrunk/sip:10.10.20.12:5060 Unknown nan
> Transport: simpletrans tcp 0 0 10.10.20.13:5060
> Identify: mytrunk/mytrunk
> Match: 10.10.20.12/32
> {noformat}
> pjsip.conf (for TCP test) contains...
> {noformat}
> ;==============TRANSPORTS
> [simpletrans]
> type=transport
> protocol=tcp
> bind=10.10.20.13
> ;===============TRUNK
> [mytrunk]
> type=registration
> transport=simpletrans
> outbound_auth=mytrunk
> outbound_proxy=sip:10.10.20.12\;transport=tcp
> server_uri=sip:10.10.20.12
> client_uri=sip:GW1 at 10.10.20.12
> retry_interval=60
> max_retries=1440
> [mytrunk]
> type=auth
> auth_type=userpass
> password=*******
> username=GW1
> [mytrunk]
> type=aor
> contact=sip:10.10.20.12:5060
> [mytrunk]
> type=endpoint
> transport=simpletrans
> context=externalIncoming
> disallow=all
> allow=gsm
> outbound_auth=mytrunk
> aors=mytrunk
> force_rport=yes
> direct_media=no
> media_encryption_optimistic=no
> [mytrunk]
> type=identify
> endpoint=mytrunk
> match=10.10.20.12
> {noformat}
> If I change pjsip.conf (& OpenSIPS) to use UDP rather than TCP it all works as expected.
> Prior to the upgrade I had the same config (excluding the new media_encryption_optimistic=no option) working fine with Asterisk 13.0.1. I don't think I've changed anything in my setup.
> Is this a bug or do I need some additional (Asterisk 13.1.0 specific) pjsip.conf change to get this to work over TCP?
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