[Asterisk-Users] Broadvoice + Reinvite = Not Working Anymore

[Dhennys Pestana]

Hello, people! I have the following scenario:

Analog Phone -> Sipura 2100 (internal ip) -> eth0-Asterisk1-ppp0 ->
Asterisk2 -> Broadvoice

The analog phone can dial "local" extensions on Asterisk1 server and
"remote" extensions on Asterisk2 server, no issues here.

My problem: when I try to dial from the analog phone through Broadvoice, I
can't get through when reinvite=yes between Asterisk1 and Asterisk2. But I
could a few weeks ago!!! That's what's really wierd. :/

Since the problem started to appear I tried a lot of changes in sip.conf (on
both servers) and even upgraded Asterisk2 (yesterday, from CVS) but the
problem is still there.

Calls get through, destination phone rings and when the party answers I can
hear them for less than a second, then no audio anymore.

Here's what I was able to do in the past:
- Sipura User registered on Asterisk1 with reinvite=no.
- Asterisk1 registered on Asterisk2 with reinvite=yes.
- Asterisk2 registered on Broadvoice with reinvite=yes.

So, any international call that should be routed to Broadvoice made from the
analog phone should be forward from Asterisk1 to Asterisk2 and then forward
to Broadvoice. Asterisk2 stays out of the media path so RTP packets should
be sent/received to/from Asterisk1 and Broadvoice.

Configuration files? Ok, See below.


sip.conf on Asterisk1:

[general]
context=default
recordhistory=yes
realm=myserver.domain.com
port=5060
bindaddr=0.0.0.0
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=900
videosupport=yes
disallow=all
allow=ulaw
musicclass=default
language=br
rtptimeout=60
rtpholdtimeout=600
useragent=Pestana PBX
nat=no
externip = 200.200.200.200
localnet = 172.20.100.0/24

[asterisk2]
type=peer
username=9999
secret=9999
host=asterisk2.domain.com
context=customers
canreinvite=yes
nat=no
insecure=very
rtptimeout=60
disallow=all
allow=ulaw

[1234]
type=friend
username=1234
secret=1234
host=dynamic
context=home
mailbox=1234 at home
canreinvite=no
rtptimeout=60
dtmfmode=rfc2833
callerid="1234" <1234>
nat=yes
disallow=all
allow=ulaw





sip.conf on Asterisk2:

[general]
context=default
recordhistory=yes
port=5060
bindaddr=0.0.0.0
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=900
videosupport=yes
compactheaders=yes
disallow=all
allow=ulaw
musicclass=default
language=en
rtptimeout=60
rtpholdtimeout=600
useragent=Tels PBX
nat=no

[broadvoice]
type=peer
username=myusername
secret=mysecret
fromuser=myusername
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
canreinvite=yes
dtmfmode=inband
nat=no
insecure=very
disallow=all
allow=ulaw

[9999]
type=friend
username=9999
secret=9999
host=dynamic
context=customers
mailbox=9999 at customers
rtptimeout=60
canreinvite=yes
dtmfmode=rfc2833
callerid="9999" <9999>
nat=no
disallow=all
allow=ulaw





extensions.conf on Asterisk1:

[home]
exten => _X.,1,Dial(SIP/${EXTEN}@asterisk2,60)





extensions.conf on Asterisk2:

[customers]
exten => _X.,1,Dial(SIP/${EXTEN}@broadvoice,60,m)





Here's what happens on Asterisk1 and Asterisk2 when the call is received
from Asterisk1 and forwarded to Broadvoice:

asterisk1*CLI>
    -- Executing Dial("SIP/1234-b9ca", "SIP/18778657669 at asterisk2|60")
in new stack
    -- Called 18778657669 at asterisk2
    -- SIP/asterisk2-466a is making progress passing it to SIP/1234-b9ca
    -- SIP/asterisk2-466a answered SIP/1234-b9ca
    -- Attempting native bridge of SIP/1234-b9ca and SIP/asterisk2-466a
    -- Attempting native bridge of SIP/1234-b9ca and SIP/asterisk2-466a
  == Spawn extension (home, 18778657669, 1) exited non-zero on
'SIP/1234-b9ca'
       > cdr_odbc: Query Successful!
asterisk1*CLI>


asterisk2*CLI>
    -- Executing Dial("SIP/200.200.200.200-081818b8",
"SIP/18778657669 at broadvoice|60|m") in new stack
    -- Called 18778657669 at broadvoice
    -- Started music on hold, class 'default', on
SIP/200.200.200.200-081818b8
    -- SIP/broadvoice-7727 is ringing
    -- SIP/broadvoice-7727 answered SIP/200.200.200.200-081818b8
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
Mar 12 01:03:01 WARNING[18046]: codec_gsm.c:164 gsmtolin_framein: Invalid
GSM data
    -- Stopped music on hold on SIP/200.200.200.200-081818b8
    -- Attempting native bridge of SIP/200.200.200.200-081818b8 and
SIP/broadvoice-7727
    -- Got SIP response 404 "Not Found" back from 147.135.0.128
    -- Got SIP response 404 "Not Found" back from 147.135.0.128
    -- Got SIP response 404 "Not Found" back from 147.135.0.128
    -- Got SIP response 404 "Not Found" back from 147.135.0.128
    -- Got SIP response 404 "Not Found" back from 147.135.0.128
    -- Got SIP response 404 "Not Found" back from 147.135.0.128
  == Spawn extension (default, 18778657669, 1) exited non-zero on
'SIP/200.200.200.200-081818b8'
asterisk2*CLI>




Notes:
- There's no GSM codec configured on any friend or peer on these servers, I
can't figure why the error message "Invalid GSM data" above.
- Error 404 sometimes becomes 400.
- The very first moment the call is completed, the audio is ok. I mean, it's
possible to hear the callee say "hello" (or any audio message from
Broadvoice) for less than a second, but nothing else beyond that.
- Asterisk1 is running Bri-Stuff 0.1.0-RC4 and can't be upgraded for the
moment.


I appologize for the long message, but I've been trying to fix this for
almost two weeks.

Thanks in advance for any input on this matter.

Best regards,

-Dhennys Pestana