Hi :)
I've had all this working before, but I'm revisiting it, and in short, I
currently have huge problems receiving incoming calls. I've been trying
with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel
and libpri as of yesterday afternoon.
Would someone mind helping? :)
My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set
as the 'DMZ Host' so all incoming IP traffic (even AH/ESP for IPSec
etc.) goes directly to that machine. I am not doing any firewalling, nor
is my ISP.
I've made my configuration as superficial as I can to ease diagnosis:
root@eddie:/etc/asterisk# ls -l
-rw-r--r-- 1 root root 104 Jun 23 21:21 extensions.conf
-rw-r--r-- 1 root root 164 Jun 23 19:25 iax.conf
-rw-r--r-- 1 root root 0 Jun 22 15:36 modem.conf
-rw-r--r-- 1 root root 387 Jun 23 21:22 modules.conf
-rw-r--r-- 1 root root 363 Jun 23 21:19 sip.conf
-rw-r--r-- 1 root root 0 Jun 22 15:36 voicemail.conf
root@eddie:/etc/asterisk# more extensions.conf
[general]
static=no
writeprotect=yes
[default]
exten => 3333,1,Dial(IAX2/janie|20|tr)
root@eddie:/etc/asterisk# more iax.conf
[general]
port=5036
[janie]
type=friend
username=janie
secret=mysecret
host=dynamic
context=default
auth=md5
notransfer=yes
root@eddie:/etc/asterisk# more modules.conf
[modules]
autoload=yes
noload => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => app_intercom.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_oss.so
noload => chan_skinny.so
noload => chan_mgcp.so
noload => chan_phone.so
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
noload => chan_zap.so
root@eddie:/etc/asterisk# more sip.conf
[general]
port = 5060
bindaddr = 10.0.0.1
context = default
disallow = all
allow = ulaw
allow = alaw
allow = gsm
externip = 213.232.83.29
localnet = 10.0.0.0
localmask = 255.255.255.0
register => 77830:MyPassword@fwd.pulver.com/3333
[fwd.pulver.com]
type=friend
secret=MyPassword
username=77830
host=fwd.pulver.com
==================================================
root@eddie:/etc/asterisk# asterisk -vvvvvvvvvvvc
[....]
Asterisk Ready.
*CLI> sip show registry
Host Username Refresh State
192.246.69.223:5060 77830 120 Registered
*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port
Status
fwd.pulver.com/ 192.246.69.223 255.255.255.255 5060
Unmonitored
*CLI> iax2 show registry
Host Username Perceived Refresh State
*CLI> iax2 show peers
Name/Username Host Mask Port Status
janie/janie 10.0.0.74 (D) 255.255.255.255 4569 Unmonitored
(janie is using iaxComm for Windows as the soft phone, and dialling
'3333' from iaxComm causes a call to come in on 'line 2' of
iaxComm)
If I now initiate an external call using FWD's "Call Me"
*CLI> ##### Testing 192.246.69.223 with 192.246.69.223
Target address 192.246.69.223 is not local, substituting externip
Setting NAT on RTP to -1
Stopping retransmission on '1710764988@alphacp' of Response 1: Found
[15 seconds pass]
Auto destroying call '1710764988@alphacp'
[20 seconds pass]
*CLI> ##### Testing 65.39.205.111 with 65.39.205.111
Target address 65.39.205.111 is not local, substituting externip
##### Testing 65.39.205.111 with 65.39.205.111
Target address 65.39.205.111 is not local, substituting externip
Check for res for
is not a local user
build_route: Contact hop: sip:65.39.205.111:5060
-- Executing Dial("SIP/fwd.pulver.com-0811c948",
"IAX2/janie|20|tr")
in new stack
SIMPLE DIAL (NO URL)
-- Called janie
-- Call accepted by 10.0.0.74 (format ULAW)
-- Format for call is ULAW
-- IAX2[janie]/4 is ringing
And so it is. I answer the softphone and:
Dropping incompatible voice frame on IAX2[janie]/4 of format GSM since
our native format has changed to ULAW
screams up the screen for each frame...
Does this make sense to anyone?
I have made a full SIP trace of the session available at
http://gdh.ca/siptrace.txt if it helps! :)
Final note, I have tried 'nat=yes' and 'nat=no' in the
[fwd.pulver.com] section of sip.conf but it makes no difference :(
Cheers,
Gavin.
Gavin Hamill wrote:> Hi :) > > I've had all this working before, but I'm revisiting it, and in > short, I currently have huge problems receiving incoming calls. I've > been trying with both FWD and voiptalk.org. I'm running CVS HEAD of > asterisk, zaptel and libpri as of yesterday afternoon. > > Would someone mind helping? :) > > My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set > as the 'DMZ Host' so all incoming IP traffic (even AH/ESP for IPSec > etc.) goes directly to that machine. I am not doing any firewalling, > nor > is my ISP. > > I've made my configuration as superficial as I can to ease diagnosis: > > root@eddie:/etc/asterisk# ls -l > -rw-r--r-- 1 root root 104 Jun 23 21:21 > extensions.conf > -rw-r--r-- 1 root root 164 Jun 23 19:25 iax.conf > -rw-r--r-- 1 root root 0 Jun 22 15:36 modem.conf > -rw-r--r-- 1 root root 387 Jun 23 21:22 modules.conf > -rw-r--r-- 1 root root 363 Jun 23 21:19 sip.conf > -rw-r--r-- 1 root root 0 Jun 22 15:36 voicemail.conf > > root@eddie:/etc/asterisk# more extensions.conf > [general] > static=no > writeprotect=yes > > [default] > exten => 3333,1,Dial(IAX2/janie|20|tr) > > root@eddie:/etc/asterisk# more iax.conf > [general] > port=5036 > > [janie] > type=friend > username=janie > secret=mysecret > host=dynamic > context=default > auth=md5 > notransfer=yes > > root@eddie:/etc/asterisk# more modules.conf > [modules] > autoload=yes > noload => pbx_gtkconsole.so > noload => pbx_kdeconsole.so > noload => app_intercom.so > load => res_musiconhold.so > noload => chan_alsa.so > noload => chan_oss.so > noload => chan_skinny.so > noload => chan_mgcp.so > noload => chan_phone.so > noload => chan_modem.so > noload => chan_modem_aopen.so > noload => chan_modem_bestdata.so > noload => chan_modem_i4l.so > noload => chan_zap.so > > root@eddie:/etc/asterisk# more sip.conf > [general] > port = 5060 > bindaddr = 10.0.0.1 > context = default > disallow = all > allow = ulaw > allow = alaw > allow = gsm > externip = 213.232.83.29 > localnet = 10.0.0.0 > localmask = 255.255.255.0 > > register => 77830:MyPassword@fwd.pulver.com/3333 > > [fwd.pulver.com] > type=friend > secret=MyPassword > username=77830 > host=fwd.pulver.com > > ==================================================> > root@eddie:/etc/asterisk# asterisk -vvvvvvvvvvvc > [....] > > Asterisk Ready. > *CLI> sip show registry > Host Username Refresh State > 192.246.69.223:5060 77830 120 Registered > *CLI> sip show peers > Name/username Host Dyn Nat ACL Mask Port > Status > fwd.pulver.com/ 192.246.69.223 255.255.255.255 5060 > Unmonitored > > *CLI> iax2 show registry > Host Username Perceived Refresh State > *CLI> iax2 show peers > Name/Username Host Mask Port Status > janie/janie 10.0.0.74 (D) 255.255.255.255 4569 Unmonitored > > (janie is using iaxComm for Windows as the soft phone, and dialling > '3333' from iaxComm causes a call to come in on 'line 2' of iaxComm) > > If I now initiate an external call using FWD's "Call Me" > > *CLI> ##### Testing 192.246.69.223 with 192.246.69.223 > Target address 192.246.69.223 is not local, substituting externip > Setting NAT on RTP to -1 Stopping retransmission on > '1710764988@alphacp' of Response 1: Found > > [15 seconds pass] > > Auto destroying call '1710764988@alphacp' > > > [20 seconds pass] > > > *CLI> ##### Testing 65.39.205.111 with 65.39.205.111 > Target address 65.39.205.111 is not local, substituting externip > ##### Testing 65.39.205.111 with 65.39.205.111 Target address > 65.39.205.111 is not local, substituting externip Check for res for > is not a local user > build_route: Contact hop: sip:65.39.205.111:5060 > -- Executing Dial("SIP/fwd.pulver.com-0811c948", > "IAX2/janie|20|tr") > in new stack > SIMPLE DIAL (NO URL) > -- Called janie > -- Call accepted by 10.0.0.74 (format ULAW) > -- Format for call is ULAW > -- IAX2[janie]/4 is ringing > > And so it is. I answer the softphone and: > > Dropping incompatible voice frame on IAX2[janie]/4 of format GSM since > our native format has changed to ULAW > > screams up the screen for each frame... > > Does this make sense to anyone? > > I have made a full SIP trace of the session available at > http://gdh.ca/siptrace.txt if it helps! :) > > Final note, I have tried 'nat=yes' and 'nat=no' in the > [fwd.pulver.com] section of sip.conf but it makes no difference :( > > Cheers, > Gavin.I remember once having same issue like this using a softfone. Playing with codecs on both sides did solve the problem. Sorry, that I can not be specific since I just can not remember the details. Ta Senad
Gavin Hamill wrote:> Hi :) > > I've had all this working before, but I'm revisiting it, and in short, I > currently have huge problems receiving incoming calls. I've been trying > with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel > and libpri as of yesterday afternoon. > > Would someone mind helping? :) > > My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set > as the 'DMZ Host' so all incoming IP traffic (even AH/ESP for IPSec > etc.) goes directly to that machine. I am not doing any firewalling, nor > is my ISP. > > I've made my configuration as superficial as I can to ease diagnosis: > > root@eddie:/etc/asterisk# ls -l > -rw-r--r-- 1 root root 104 Jun 23 21:21 extensions.conf > -rw-r--r-- 1 root root 164 Jun 23 19:25 iax.conf > -rw-r--r-- 1 root root 0 Jun 22 15:36 modem.conf > -rw-r--r-- 1 root root 387 Jun 23 21:22 modules.conf > -rw-r--r-- 1 root root 363 Jun 23 21:19 sip.conf > -rw-r--r-- 1 root root 0 Jun 22 15:36 voicemail.conf >I know that this is not related to your ultimate question, but I would not recommend giving read access to everyone. Even if you have guest disabled, this still leaves you vulnerable to snoops discovering your configuration. With that in hand, they can make phone calls on your dime. I would change the access rights on all of these files to 640. Stephen R. Besch
> > 65.39.205.111 is not local, substituting externip Check for res for > > is not a local user > > build_route: Contact hop: sip:65.39.205.111:5060 > > -- Executing Dial("SIP/fwd.pulver.com-0811c948", > > "IAX2/janie|20|tr") > > in new stack > > SIMPLE DIAL (NO URL) > > -- Called janie > > -- Call accepted by 10.0.0.74 (format ULAW) > > -- Format for call is ULAW > > -- IAX2[janie]/4 is ringing > > > > And so it is. I answer the softphone and: > > > > Dropping incompatible voice frame on IAX2[janie]/4 of format GSM since > > our native format has changed to ULAW > > > > screams up the screen for each frame... > > > > Does this make sense to anyone?FWD only supports ULAW comment out the line allow=GSM in the general section of the iax.conf Jason
Hi!> FWD only supports ULAW comment out the line allow=GSM in the general > section of the iax.confNonsense - FWD *does* permit the use of GSM. Cheers, Philipp
At 14:48 25/06/2004 +0200, you wrote:>Hi! > > > FWD only supports ULAW comment out the line allow=GSM in the general > > section of the iax.conf > >Nonsense - FWD *does* permit the use of GSM. > >Cheers, PhilippNot in iax only with sip Jason