Hi there.
I have a strange issue i cant get my head around. If someone can enlighten me with what’s going on would be appreciated.
Sorry in advance… struggled to make this look pretty.
####################Summary####################
chan_sip config works fine on Asterisk 13 “everything is awesome…!”
We are doing our testing for migrating to Asterisk 16 as per the support life-cycle.
and on 16.5 using pjsip i have this issue with the same provider
- incoming calls from “Telstra cell phone” through VSP causes the error:
channel.c:5589 set_format: Unable to find a codec translation path: (g729) -> (alaw)
“outgoing calls to Telstra are fine”
NOTE: incoming calls from other providers are fine… just in from my Telstra Mobile is broken.
NOTE2: I dont have or want G729 codec anywhere
is this a configuration on my end?.
Why does it work fine on Asterisk 13 and chan_sip
Any tips ?
Thanks in advance.
#info
** running Asterisk 13 on ubuntu apt installed and everything is great
Asterisk 13.18.3~dfsg-1ubuntu4 built by nobody at buildd.debian. org on a unknown running Linux on 2018-02-28 06:44:47 UTC
** BUT pjsip config on 16.5 compiled and on same OS
-----SAME VSP and account details.
testpbx-2CLI> core show version
Asterisk 16.5.0 built by root @ testpbx-2 on a x86_64 running Linux on 2019-09-01 11:03:22 UTC
testpbx-2CLI>
CONFIG BELOW!!!
#################
#################
####################SIP HISTORY####################
**testpbx-2*CLI> pjsip show history where number = 00000**
<--- History Entry 0 Received from 203.2.134.1:5060 at 1567384890 --->
INVITE sip:PHONENUMBER @ PBX.ipaddr:5060;line=elacjeb SIP/2.0
Via: SIP/2.0/UDP 203.2.134.1:5060;received=203.2.134.1;branch=z9hG4bKdb18jb20dolp08dku100.1
From: <sip:MOBILENUMBER @ 203.2.134.129;user=phone>;tag=284947600-1567384890779-
To: "NodePhone" <sip:PHONENUMBER @ FQDN ;line=elacjeb>
Call-ID: BW004130779020919-1955374055 @ 203.2.134.129
CSeq: 934655694 INVITE
Contact: <sip:MOBILENUMBER @ 203.2.134.1:5060;transport=udp>
Supported: 100rel
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml, application/sdp, multipart/mixed
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 227
Content-Type: application/sdp
Content-Length: 227
v=0
o=BroadWorks 38774207 1 IN IP4 203.2.134.1
s=-
c=IN IP4 203.2.134.1
t=0 0
m=audio 20704 RTP/AVP 8 0 18 106 101
**a=rtpmap:106 G.729b/8000**
a=rtpmap:101 telephone-event/80
**testpbx-2*CLI> pjsip show history where number = 00002**
<--- History Entry 2 Sent to Endpoint_ipaddr:5060 at 1567384890 --->
INVITE sip:6001 @ Endpoint_ipaddr;transport=udp SIP/2.0
Via: SIP/2.0/UDP PBX.ipaddress:5060;rport;branch=z9hG4bKPjadea4019-002c-4d4a-b18c-e3139bba1cfe
From: "Main Ring Group" <sip:MOBILENUMBER @ PBX.ipaddress>;tag=690cb88b-0ba5-4ec9-beaa-3d509ad13960
To: <sip:6001 @ Endpoint_ipaddr>
Contact: <sip:asterisk @ PBX.ipaddress:5060>
Call-ID: 5b6b3433-5436-46a7-9864-cd33d18e7de5
CSeq: 17896 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: "Main Ring Group" <sip:MOBILENUMBER @ PBX.ipaddress>;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 16.5.0
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 1635044233 1635044233 IN IP4 PBX.ipaddress
s=Asterisk
c=IN IP4 PBX.ipaddress
t=0 0
m=audio 10176 RTP/AVP 8 101
**a=rtpmap:8 PCMA/8000**
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxpti
####################config####################
CLI> core show version
Asterisk 16.5.0 built by root @ testpbx-2 on a x86_64 running Linux on 2019-09-01 11:03:22 UTC
#########pjsip.conf #####################
[transport-udp]
type=transport
protocol=udp ;udp,tcp,tls,ws,wss
;bind=0.0.0.0
bind=0.0.0.0
[mytrunk]
type=registration
transport=transport-udp
outbound_auth=mytrunk_auth
server_uri=sip:sip.internode.on.net
client_uri=sip:username@FQDN
contact_user=username ; this is the DestNo.. otherwise becomes 's'.
retry_interval=60
forbidden_retry_interval=600
expiration=3600
line=yes
endpoint=mytrunk
[mytrunk_auth]
type=auth
auth_type=userpass
password=secress
username=username
;(DO NOT USE REALM)
;realm=BroadWorks ;Internode can be BroadWorks or leave this blank
[mytrunk]
type=endpoint
transport=transport-udp
;transport=transport-udp-nat
context=inbound_handler
disallow=all
allow=alaw
allow=ulaw
outbound_auth=mytrunk_auth
aors=mytrunk
direct_media=no
force_rport=yes
[mytrunk]
type=aor
contact=sip:username@FQDN
[mytrunk]
type=identify
endpoint=mytrunk
match=203.2.134.1
;match=203.2.134.129
[anonymous]
type=endpoint
context=inbound_handler
disallow=all
allow=alaw
allow=ulaw
;============= Extension 6001 =============;
[6001]
type=endpoint
transport=transport-udp
context=national
send_rpid=yes
disallow=all
allow=alaw
allow=ulaw
aors=6001
[6001]
type=auth
auth_type=userpass
password=6001
username=secret
[6001]
type=aor
max_contacts=2
##############################
,
debug flow of call
>-- Executing [2244668811 @ inbound_handler:3] NoOp("PJSIP/mytrunk-00000032", "EXTEN is 2244668811") in new stack
-- Executing [2244668811 @ inbound_handler:4] Set("PJSIP/mytrunk-00000032", "DestNo=2244668811") in new stack
-- Executing [2244668811 @ inbound_handler:5] NoOp("PJSIP/mytrunk-00000032", "DestNo is 2244668811") in new stack
-- Executing [2244668811 @ inbound_handler:6] Goto("PJSIP/mytrunk-00000032", "sw_11_2244668811,10") in new stack
-- Goto (inbound_handler,sw_11_2244668811,10)
-- Executing [sw_11_2244668811 @ inbound_handler:10] Goto("PJSIP/mytrunk-00000032", "_.,main_no") in new stack
-- Goto (inbound_handler,_.,8)
-- Executing [_. @ inbound_handler:8] GotoIf("PJSIP/mytrunk-00000032", "1?9:13") in new stack
-- Goto (inbound_handler,_.,9)
-- Executing [_. @ inbound_handler:9] Set("PJSIP/mytrunk-00000032", "CALLERID(name)=Main Ring Group") in new stack
-- Executing [_. @ inbound_handler:10] Dial("PJSIP/mytrunk-00000032", "PJSIP/6001,30") in new stack
-- Called PJSIP/6001
-- PJSIP/6001-00000033 is ringing
-- PJSIP/6001-00000033 is ringing
-- PJSIP/6001-00000033 answered PJSIP/mytrunk-00000032
> 0x7f299002ef10 -- Strict RTP learning after remote address set to: endpoint.ipaddr:7078
> 0x7f299004ed80 -- Strict RTP learning after remote address set to: 203.2.134.1:23436
-- Channel PJSIP/6001-00000033 joined 'simple_bridge' basic-bridge <be88a861-020d-4f78-b15a-f4ec574258a4>
-- Channel PJSIP/mytrunk-00000032 joined 'simple_bridge' basic-bridge <be88a861-020d-4f78-b15a-f4ec574258a4>
> Bridge be88a861-020d-4f78-b15a-f4ec574258a4: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'PJSIP/mytrunk-00000032' and 'PJSIP/6001-00000033' in stack
> 0x7f299002ef10 -- Strict RTP switching to RTP target address endpoint.ipaddr:7078 as source
> 0x7f299004ed80 -- Strict RTP switching to RTP target address 203.2.134.1:23436 as source
[Sep 6 10:39:35] WARNING[15748][C-0000001b]: translate.c:488 ast_translator_build_path: No translator path: (ending codec is not valid)
[Sep 6 10:39:35] WARNING[15748][C-0000001b]: translate.c:488 ast_translator_build_path: No translator path: (ending codec is not valid)
> Bridge be88a861-020d-4f78-b15a-f4ec574258a4: switching from native_rtp technology to simple_bridge
[Sep 6 10:39:35] WARNING[15750][C-0000001b]: channel.c:5589 set_format: Unable to find a codec translation path: (g729) -> (alaw)
[Sep 6 10:39:35] WARNING[15748][C-0000001b]: channel.c:6549 ast_channel_make_compatible_helper: No path to translate from PJSIP/6001-00000033 to PJSIP/mytrunk-00000032
[Sep 6 10:39:35] WARNING[15748][C-0000001b]: channel.c:6549 ast_channel_make_compatible_helper: No path to translate from PJSIP/6001-00000033 to PJSIP/mytrunk-00000032
[Sep 6 10:39:35] WARNING[15748][C-0000001b]: channel.c:6549 ast_channel_make_compatible_helper: No path to translate from PJSIP/6001-00000033 to PJSIP/mytrunk-00000032
[Sep 6 10:39:35] WARNING[15748][C-0000001b]: channel.c:6549 ast_channel_make_compatible_helper: No path to translate from PJSIP/6001-00000033 to PJSIP/mytrunk-00000032
-- Channel PJSIP/mytrunk-00000032 left 'simple_bridge' basic-bridge <be88a861-020d-4f78-b15a-f4ec574258a4>
-- Channel PJSIP/6001-00000033 left 'simple_bridge' basic-bridge <be88a861-020d-4f78-b15a-f4ec574258a4>
[Sep 6 10:39:35] WARNING[15748][C-0000001b]: channel.c:5589 set_format: Unable to find a codec translation path: (g729) -> (alaw)
== Spawn extension (inbound_handler, _., 10) exited non-zero on 'PJSIP/mytrunk-00000032'
-- Executing [h @ inbound_handler:1] Wait("PJSIP/mytrunk-00000032", "1") in new stack