Help getting gtalk channel running

I’ve seen some posts here about this, but haven’t found a way to get mine working. I’m trying to get native Google Voice calling working through Asterisk. I’m still pretty new to this whole thing. I’ve followed some configuration guides, and think I’ve got my .conf files correct.

My problem seems to be that the gtalk channel is loaded, yet not running. Observe:

[code]*CLI> core show channeltypes
Type Description Devicestate Indications Transfer


USTM UNISTIM Channel Driver no yes no
Phone Standard Linux Telephony API Driver no yes no
Console OSS Console Channel Driver no yes no
MGCP Media Gateway Control Protocol (MGCP) yes yes no
SIP Session Initiation Protocol (SIP) yes yes yes
Skinny Skinny Client Control Protocol (Skinny) yes yes no
Gtalk Gtalk Channel Driver no yes no
Agent Call Agent Proxy Channel yes yes no
IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes
MulticastR Multicast RTP Paging Channel Driver no no no
Jingle Jingle Channel Driver no yes no
Bridge Bridge Interaction Channel no no no
DAHDI DAHDI Telephony Driver yes yes no
Local Local Proxy Channel Driver yes yes no

14 channel drivers registered.
*CLI>[/code]

I’m running this on ~$ uname -a Linux gob 2.6.32-26-generic #48-Ubuntu SMP Wed Nov 24 09:00:03 UTC 2010 i686 GNU/Linux

And Asterisk: *CLI> core show version Asterisk SVN-trunk-r292826 built by root @ gob on a i686 running Linux on 2010-10-25 13:26:40 UTC *CLI>

Any help would be greatly appreciated!

That just indicates that devicestate isn’t supported for GTalk.

Did you follow the instructions here?

wiki.asterisk.org/wiki/display/ … ing+Google

Shoot. Perhaps not. I will look into getting these dependencies compiled. Then is it just a matter of re-make-ing Asterisk and installing?

No, if it shows up there, it’s running; you’re just looking at whether or not it supports devicestate, which it doesn’t.

That page on the Wiki should help you get it configured.

I can’t seem to compile iksemel. Grabbing the svn from svn.uludag.org.tr/projeler/iksemel, but when am getting errors compiling.

[code]root@gob:/usr/src/iksemel# ./autogen.sh
Running libtoolize…
Running aclocal…
configure.ac:48: warning: macro AM_PATH_LIBGNUTLS' not found in library Running autoheader... Running automake... configure.ac:8: installing./config.guess’
configure.ac:8: installing ./config.sub' configure.ac:5: installing./install-sh’
configure.ac:5: installing ./missing' doc/Makefile.am:5: installingdoc/texinfo.tex’
src/Makefile.am: installing `./depcomp’
Running autoconf…
configure.ac:48: error: possibly undefined macro: AM_PATH_LIBGNUTLS
If this token and others are legitimate, please use m4_pattern_allow.
See the Autoconf documentation.

Done!
Now run ./configure in order to create Makefiles.

root@gob:/usr/src/iksemel# ./configure --prefix=/usr
checking for a BSD-compatible install… /usr/bin/install -c
checking whether build environment is sane… yes
checking for a thread-safe mkdir -p… /bin/mkdir -p
checking for gawk… gawk
checking whether make sets $(MAKE)… yes
checking build system type… i686-pc-linux-gnu
checking host system type… i686-pc-linux-gnu
checking for gcc… gcc
checking whether the C compiler works… yes
checking for C compiler default output file name… a.out
checking for suffix of executables…
checking whether we are cross compiling… no
checking for suffix of object files… o
checking whether we are using the GNU C compiler… yes
checking whether gcc accepts -g… yes
checking for gcc option to accept ISO C89… none needed
checking for style of include used by make… GNU
checking dependency style of gcc… gcc3
checking for objdump… objdump
checking for a sed that does not truncate output… /bin/sed
checking for grep that handles long lines and -e… /bin/grep
checking for egrep… /bin/grep -E
checking for fgrep… /bin/grep -F
checking for ld used by gcc… /usr/bin/ld
checking if the linker (/usr/bin/ld) is GNU ld… yes
checking for BSD- or MS-compatible name lister (nm)… /usr/bin/nm -B
checking the name lister (/usr/bin/nm -B) interface… BSD nm
checking whether ln -s works… yes
checking the maximum length of command line arguments… 1572864
checking whether the shell understands some XSI constructs… yes
checking whether the shell understands “+=”… yes
checking for /usr/bin/ld option to reload object files… -r
checking for objdump… (cached) objdump
checking how to recognize dependent libraries… pass_all
checking for ar… ar
checking for strip… strip
checking for ranlib… ranlib
checking command to parse /usr/bin/nm -B output from gcc object… ok
checking how to run the C preprocessor… gcc -E
checking for ANSI C header files… yes
checking for sys/types.h… yes
checking for sys/stat.h… yes
checking for stdlib.h… yes
checking for string.h… yes
checking for memory.h… yes
checking for strings.h… yes
checking for inttypes.h… yes
checking for stdint.h… yes
checking for unistd.h… yes
checking for dlfcn.h… yes
checking for objdir… .libs
checking if gcc supports -fno-rtti -fno-exceptions… no
checking for gcc option to produce PIC… -fPIC -DPIC
checking if gcc PIC flag -fPIC -DPIC works… yes
checking if gcc static flag -static works… yes
checking if gcc supports -c -o file.o… yes
checking if gcc supports -c -o file.o… (cached) yes
checking whether the gcc linker (/usr/bin/ld) supports shared libraries… yes
checking whether -lc should be explicitly linked in… no
checking dynamic linker characteristics… GNU/Linux ld.so
checking how to hardcode library paths into programs… immediate
checking whether stripping libraries is possible… yes
checking if libtool supports shared libraries… yes
checking whether to build shared libraries… yes
checking whether to build static libraries… yes
checking for ANSI C header files… (cached) yes
checking for unistd.h… (cached) yes
checking for strings.h… (cached) yes
checking errno.h usability… yes
checking errno.h presence… yes
checking for errno.h… yes
checking for an ANSI C-conforming const… yes
checking for inline… inline
checking for size_t… yes
checking for struct stat.st_blksize… yes
checking for library containing recv… none required
checking for getopt_long… yes
checking for getaddrinfo… yes
./configure: line 11035: syntax error near unexpected token ,' ./configure: line 11035:AM_PATH_LIBGNUTLS(,'
root@gob:/usr/src/iksemel#[/code]

Current rev from svn is 574. I’ve gone all the way back to rev 500. Same result. There appears to be a bug logged for it at code.google.com/p/iksemel/issues … akechanges

I’d really like to get this working, but keep running into stumbling blocks!

Does anyone actually have Asterisk 1.8 up and running with chan_gtalk and res_jabber, all on Ubuntu 10.04, and have it working?

[quote=“malcolmd”]No, if it shows up there, it’s running; you’re just looking at whether or not it supports devicestate, which it doesn’t.

That page on the Wiki should help you get it configured.[/quote]

So it doesn’t have to show up in the list of channels? I think I’ve got my confs set, but maybe take a quick look?

gtalk.conf:

[code]root@gob:/etc/asterisk# cat gtalk.conf
[general]
context=local ; Context to dump call into
bindaddr=192.168.1.153 ; Address to bind to
externip=xxx.xxx.xxx.xxx ; Set your external ip if you are behind a NAT.
;stunaddr=mystunserver.com ; Get your external ip from a STUN server.
; Note, if the STUN query is successful, this will
; replace any value placed in externip;
allowguest=yes ; Allow calls from people not in list of peers

[guest] ; special account for options on guest account
disallow=all
allow=ulaw
context=gv-inbound
connection=jp_jabber

root@gob:/etc/asterisk#[/code]

jabber.conf:

[code]root@gob:/etc/asterisk# cat jabber.conf
[general]
debug=yes ; Enable debugging (disabled by default).
autoprune=no ; Auto remove users from buddy list. Depending on your
; setup (ie, using your personal Gtalk account for a test)
; you might lose your contacts list. Default is ‘no’.
autoregister=yes ; Auto register users from buddy list.

[jp_jabber]
type=client
serverhost=talk.google.com
username=xxxxxxxx@gmail.com/Talk
secret=xxxxxx
port=5222
usetls=yes
usesasl=yes
statusmessage="Connected via Asterisk"
timeout=100
root@gob:/etc/asterisk#[/code]

My Android phone in sip.conf:

[JP_Android] type=friend host=dynamic secret=xxxx context=gvjp callerid="JP_Android" <100> nat=yes

extensions.conf:

[code][local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => incominggv
include => gvjp
include => outboundgv
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

[users]
exten => 100,1,Dial(SIP/JP_Android,20)

[gvjp]
;append an area code if necessary
exten => _NXXXXXX,1,Set(CALLERID(dnid)=1508${CALLERID(dnid)})
exten => _NXXXXXX,n,Goto(1508${EXTEN},1)
;append a 1 if necessary
exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=1${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Goto(1${EXTEN},1)
;do our real dialing
exten => _1NXXNXXXXXX,1,Dial(gtalk/jabber_jp/${EXTEN}@voice.google.com)
exten => _+1NXXNXXXXXX,1,Dial(gtalk/jabber_jp/${EXTEN}@voice.google.com)
include => users

[gv-inbound]
exten => xxxxxxxxx@gmail.com,1,Answer()
exten => xxxxxxxxx@gmail.com,n,Wait(1)
exten => xxxxxxxxx@gmail.com,n,SendDTMF(1)
exten => xxxxxxxxx@gmail.com,n,Dial(SIP/JP_Android,20)

[/code]

And if I try to place a call with my Android, here is what sticks out on the Asterisk CLI:

[Dec 13 15:17:11] ERROR[14138]: chan_gtalk.c:1836 gtalk_request: No XMPP client to talk to, us (partial JID) : jabber_jp [Dec 13 15:17:11] WARNING[14138]: app_dial.c:2030 dial_exec_full: Unable to create channel of type 'gtalk' (cause 0 - Unknown)

Sorry. I’m really new to this, and hope that it’s just something stupid that I didn’t know about.

Ooo. And this can’t be good:

*CLI> gtalk show channels Channel Jabber ID Resource Read Write Segmentation fault root@gob:/etc/asterisk#

Also, I see this in /var/log/messages after a segfault:

Am I hosed?

I just updated my svn, and re-built Asterisk. Same issue, and still getting that segfault error 4 in chan_gtalk.so

[quote=“jp.briggs”]I can’t seem to compile iksemel. Grabbing the svn from svn.uludag.org.tr/projeler/iksemel, but when am getting errors compiling.

[code]root@gob:/usr/src/iksemel# ./autogen.sh
Running libtoolize…
Running aclocal…
configure.ac:48: warning: macro AM_PATH_LIBGNUTLS' not found in library Running autoheader... Running automake... configure.ac:8: installing./config.guess’
configure.ac:8: installing ./config.sub' configure.ac:5: installing./install-sh’
configure.ac:5: installing ./missing' doc/Makefile.am:5: installingdoc/texinfo.tex’
src/Makefile.am: installing `./depcomp’
Running autoconf…
configure.ac:48: error: possibly undefined macro: AM_PATH_LIBGNUTLS
If this token and others are legitimate, please use m4_pattern_allow.
See the Autoconf documentation.

Done!
Now run ./configure in order to create Makefiles.

root@gob:/usr/src/iksemel# ./configure --prefix=/usr
checking for a BSD-compatible install… /usr/bin/install -c
checking whether build environment is sane… yes
checking for a thread-safe mkdir -p… /bin/mkdir -p
checking for gawk… gawk
checking whether make sets $(MAKE)… yes
checking build system type… i686-pc-linux-gnu
checking host system type… i686-pc-linux-gnu
checking for gcc… gcc
checking whether the C compiler works… yes
checking for C compiler default output file name… a.out
checking for suffix of executables…
checking whether we are cross compiling… no
checking for suffix of object files… o
checking whether we are using the GNU C compiler… yes
checking whether gcc accepts -g… yes
checking for gcc option to accept ISO C89… none needed
checking for style of include used by make… GNU
checking dependency style of gcc… gcc3
checking for objdump… objdump
checking for a sed that does not truncate output… /bin/sed
checking for grep that handles long lines and -e… /bin/grep
checking for egrep… /bin/grep -E
checking for fgrep… /bin/grep -F
checking for ld used by gcc… /usr/bin/ld
checking if the linker (/usr/bin/ld) is GNU ld… yes
checking for BSD- or MS-compatible name lister (nm)… /usr/bin/nm -B
checking the name lister (/usr/bin/nm -B) interface… BSD nm
checking whether ln -s works… yes
checking the maximum length of command line arguments… 1572864
checking whether the shell understands some XSI constructs… yes
checking whether the shell understands “+=”… yes
checking for /usr/bin/ld option to reload object files… -r
checking for objdump… (cached) objdump
checking how to recognize dependent libraries… pass_all
checking for ar… ar
checking for strip… strip
checking for ranlib… ranlib
checking command to parse /usr/bin/nm -B output from gcc object… ok
checking how to run the C preprocessor… gcc -E
checking for ANSI C header files… yes
checking for sys/types.h… yes
checking for sys/stat.h… yes
checking for stdlib.h… yes
checking for string.h… yes
checking for memory.h… yes
checking for strings.h… yes
checking for inttypes.h… yes
checking for stdint.h… yes
checking for unistd.h… yes
checking for dlfcn.h… yes
checking for objdir… .libs
checking if gcc supports -fno-rtti -fno-exceptions… no
checking for gcc option to produce PIC… -fPIC -DPIC
checking if gcc PIC flag -fPIC -DPIC works… yes
checking if gcc static flag -static works… yes
checking if gcc supports -c -o file.o… yes
checking if gcc supports -c -o file.o… (cached) yes
checking whether the gcc linker (/usr/bin/ld) supports shared libraries… yes
checking whether -lc should be explicitly linked in… no
checking dynamic linker characteristics… GNU/Linux ld.so
checking how to hardcode library paths into programs… immediate
checking whether stripping libraries is possible… yes
checking if libtool supports shared libraries… yes
checking whether to build shared libraries… yes
checking whether to build static libraries… yes
checking for ANSI C header files… (cached) yes
checking for unistd.h… (cached) yes
checking for strings.h… (cached) yes
checking errno.h usability… yes
checking errno.h presence… yes
checking for errno.h… yes
checking for an ANSI C-conforming const… yes
checking for inline… inline
checking for size_t… yes
checking for struct stat.st_blksize… yes
checking for library containing recv… none required
checking for getopt_long… yes
checking for getaddrinfo… yes
./configure: line 11035: syntax error near unexpected token ,' ./configure: line 11035:AM_PATH_LIBGNUTLS(,'
root@gob:/usr/src/iksemel#[/code]

Current rev from svn is 574. I’ve gone all the way back to rev 500. Same result. There appears to be a bug logged for it at code.google.com/p/iksemel/issues … akechanges

I’d really like to get this working, but keep running into stumbling blocks!

Does anyone actually have Asterisk 1.8 up and running with chan_gtalk and res_jabber, all on Ubuntu 10.04, and have it working?[/quote]

Why are you trying to compile it from source? Have you tried the libiksemel-dev package?

apt-get install libiksemel-dev

[code]root@gob:/etc/asterisk# cat gtalk.conf
[general]
context=local ; Context to dump call into
bindaddr=192.168.1.153 ; Address to bind to
externip=xxx.xxx.xxx.xxx ; Set your external ip if you are behind a NAT.
;stunaddr=mystunserver.com ; Get your external ip from a STUN server.
; Note, if the STUN query is successful, this will
; replace any value placed in externip;
allowguest=yes ; Allow calls from people not in list of peers

[guest] ; special account for options on guest account
disallow=all
allow=ulaw
context=gv-inbound
connection=jp_jabber
[/code]

You’re using a real IP for externip, right? (just asking, not flaming).

[code]root@gob:/etc/asterisk# cat jabber.conf
[general]
debug=yes ; Enable debugging (disabled by default).
autoprune=no ; Auto remove users from buddy list. Depending on your
; setup (ie, using your personal Gtalk account for a test)
; you might lose your contacts list. Default is ‘no’.
autoregister=yes ; Auto register users from buddy list.

[jp_jabber]
type=client
serverhost=talk.google.com
username=xxxxxxxx@gmail.com/Talk
secret=xxxxxx
port=5222
usetls=yes
usesasl=yes
statusmessage="Connected via Asterisk"
timeout=100
root@gob:/etc/asterisk#[/code]

Looks fine.

[JP_Android] type=friend host=dynamic secret=xxxx context=gvjp callerid="JP_Android" <100> nat=yes

Looks fine.

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => incominggv
include => gvjp
include => outboundgv
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

[users]
exten => 100,1,Dial(SIP/JP_Android,20)

[gvjp]
;append an area code if necessary
exten => _NXXXXXX,1,Set(CALLERID(dnid)=1508${CALLERID(dnid)})
exten => _NXXXXXX,n,Goto(1508${EXTEN},1)
;append a 1 if necessary
exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=1${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Goto(1${EXTEN},1)
;do our real dialing
exten => _1NXXNXXXXXX,1,Dial(gtalk/jabber_jp/${EXTEN}@voice.google.com)
exten => _+1NXXNXXXXXX,1,Dial(gtalk/jabber_jp/${EXTEN}@voice.google.com)
include => users

[gv-inbound]
exten => xxxxxxxxx@gmail.com,1,Answer()
exten => xxxxxxxxx@gmail.com,n,Wait(1)
exten => xxxxxxxxx@gmail.com,n,SendDTMF(1)
exten => xxxxxxxxx@gmail.com,n,Dial(SIP/JP_Android,20)

Here’s a problem.

exten => _1NXXNXXXXXX,1,Dial(gtalk/jabber_jp/${EXTEN}@voice.google.com)
exten => _+1NXXNXXXXXX,1,Dial(gtalk/jabber_jp/${EXTEN}@voice.google.com)

Should be:

exten => _1NXXNXXXXXX,1,Dial(gtalk/jp_jabber/${EXTEN}@voice.google.com)
exten => _+1NXXNXXXXXX,1,Dial(gtalk/jp_jabber/${EXTEN}@voice.google.com)

You got the name of the connection backwards, from your definition in jabber.conf.

Oh man, you rock! Yeah, I do have valid IPs…just thought I’d blank them out for security? But totally valid question, given my experience, or lack thereof.

But yeah. I was stupid, and called the wrong jabber. Moron.

So now I can actually place a call to my work number, and it rings! I just don’t have any audio in either direction. Also, if I hang up (either end, tried both ways) it takes quite a while for the connected end to register that the call has dropped.

I assume this may be an issue with NAT and ports? On my router I have UDP 5060 forwarded to my Asterisk IP. I noticed rtp.conf has UDP 10000-20000 specified, so I put those in my router forwarded to Asterisk as well. Not sure why I’m not getting sound?

Thanks for your help! I was banging my head against the wall for quite some time. Just didn’t manage to see the mistake.

[quote=“jp.briggs”]Oh man, you rock! Yeah, I do have valid IPs…just thought I’d blank them out for security? But totally valid question, given my experience, or lack thereof.
[/quote]

Cool, common practice for sure, I just didn’t want to assume anything; thanks for not taking offense (“Is your computer plugged in?”). :wink:

[quote=“jp.briggs”]
So now I can actually place a call to my work number, and it rings! I just don’t have any audio in either direction. Also, if I hang up (either end, tried both ways) it takes quite a while for the connected end to register that the call has dropped.

I assume this may be an issue with NAT and ports? On my router I have UDP 5060 forwarded to my Asterisk IP. I noticed rtp.conf has UDP 10000-20000 specified, so I put those in my router forwarded to Asterisk as well. Not sure why I’m not getting sound?

Thanks for your help! I was banging my head against the wall for quite some time. Just didn’t manage to see the mistake.[/quote]

The connection dropping taking a bit seems to be something related to Google. What’s your network situation? Asterisk is behind NAT A, and you have a phone also behind NAT A that’s directly connected to Asterisk?

I’ve used Asterisk behind NAT A and a softphone behind NAT B and had success. What phone are you using?

[quote=“malcolmd”]

The connection dropping taking a bit seems to be something related to Google. What’s your network situation? Asterisk is behind NAT A, and you have a phone also behind NAT A that’s directly connected to Asterisk?

I’ve used Asterisk behind NAT A and a softphone behind NAT B and had success. What phone are you using?[/quote]

My Asterisk is behind a NAT from my FiOS connection. So far, I’ve been trying using SIPDroid on my Motorola Droid on VZW’s data network (which I believe is not NAT’d). I’ll give a try when I’m home today on my WiFi (behind the same NAT, actually on the same LAN as Asterisk).

Thanks,

Howdy,

Let’s see if that works, and then go from there.

The client I tested was the Blink SIP client for Mac here at work (Digium) behind the corporate firewall/NAT against Asterisk running behind NAT at home, on an AT&T line.

Alright. So here’s where I’m at.

Test devices: SIPDroid on Android and my wife’s laptop, logged into her gmail using her Google Voice chat account.

Both are on wifi on the same LAN as the Asterisk server.

If I call from SIPDroid to my wife’s GV number, her chat client rings. I can answer, and hear voice in both directions. Everything works great.

If I call from her gchat client to my GV number, it rings many times…sometimes her gchat client will ‘answer’ and I hear silence. Eventually SIPDroid will “pick up” but instantly say it’s been hung up.

If I switch SIPDroid off wifi and onto the VZW network and try to make the call, it’s similar to when I tried calling my work number. Rings, and I can answer, but no sound two way.

So I think it stands thus: When my SIP client is behind the NAT (so far have only been able to test this on the same LAN as asterisk) it works great. When not, I can place calls but no audio. Receiving calls seems to not really work at all.

All leads point to NAT traversal problems? Anything in the configs point to it? Other configs to look at?

I feel like I’m SO close!!!

Howdy,

Do you have anything set in the general section of your sip.conf file to indicate that Asterisk is behind NAT?
localnet
externip
?

[quote=“malcolmd”]Howdy,

Do you have anything set in the general section of your sip.conf file to indicate that Asterisk is behind NAT?
localnet
externip
?[/quote]

I hadn’t. But then looking at some forums here (voipuser.org/forum_topic_2130.html) I added:

externip=96.252.xxx.xxx localnet=192.168.1.0/255.255.255.0 srvlookup=yes canreinvite=no

No good? Sorry I suck at this so much. But I’m learning a lot anyway!

I get the call connected, still no sound either way. In the CLI I see:

[Dec 16 11:04:38] WARNING[23182]: chan_sip.c:3420 retrans_pkt: Retransmission timeout reached on transmission 465653436910@10.247.226.217-S for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt. Packet timed out after 31999ms with no response [Dec 16 11:04:38] WARNING[23182]: chan_sip.c:3449 retrans_pkt: Hanging up call 465653436910@10.247.226.217-S - no reply to our critical packet (see doc/sip-retransmit.txt).

Neither that number nor IP address is anything I recognize. I assume perhaps that’s the call that Asterisk places with Google’s servers?

Nope, Asterisk talks SIP to your handset’s SIP client and Jabber/XMPP to Google.

Whoa. Wait! I got sound! Both ways! Caloo-calay!

Mr. Hammond…the phones are working.

I think I’m golden here.

Thank you so much for your help! You absolutely rock on every level!

Cabbages and Kings?

What was the final fix?

:smile:

I hadn’t changed anything. It must’ve just needed one more initialization?

Can I ask one more question? More theory than everything else.

So as I understand SIP (which I don’t fully), normally, when a call is placed, SIP is used to connect the two endpoints. This traffic can pass through SIP servers on both ends. Once the session is initiated, the actual voice/video data is to/from the endpoints, no longer running through the SIP servers. Is this not the case when Asterisk is behind NAT? Is the voice traveling through Asterisk? I guess what makes me think that is this from the SIP debug:

[code]<— SIP read from UDP:174.252.xxx.xxx:61373 —>
SUBSCRIBE sip:JP_Android@xxxxxxxxx.com SIP/2.0
Via: SIP/2.0/UDP 174.252.xxx.xxx:61373;rport;branch=z9hG4bK+498d21fbd7875e983cc4bd6a2aec60e21+s138+1
From: sip:JP_Android@xxxxxxx.com;tag=s138+1+1950000+2a65246a
To: sip:JP_Android@xxxxxxxxx.com
Call-ID: 871168332553@10.247.226.217-S
Max-Forwards: 70
CSeq: 2 SUBSCRIBE
Contact: sip:JP_Android@174.252.xxx.xxx:61373;transport=udp
Expires: 184000
User-Agent: Sipdroid/2.0.1 beta/Droid
Event: message-summary
Accept: application/simple-message-summary
Authorization: Digest username=“JP_Android”, realm=“asterisk”, nonce=“69d1f8dd”, uri="sip:JP_Android@xxxxxxxxx.com", algorithm=MD5, response="9a479bf83a5f4c65268a7a7e65b921d6"
Content-Length: 0

<------------->[/code]

From my SIP device To my SIP device? Sounds fishy.

Thanks again!