I can't call external phone, ouside lan but he calls in fine


#1

Hi,

I have a GXP2000 and a Budge Tone 100,

I can only connect them both if I set ulaw or G729 on the sip.conf file,

If I set GSM as I woulded wanted, it doesn’t work.

If I set allow=gsm on the GXP2000 I can play prompts but if I use G729 I can’t.

If I set allow=g729 on sip.conf on sections of both of the phones they can call each other but can’t call to asterisk to reproduce a menu driven prompt.

my first question is:

How can I have both worlds to work: when I dial to a menu on asterisk to use GSM and when I dial to other exension to use another codec.
(I need an efficient one since 7701 will be away from my office and ulaw couln’t make it, sound was very poor)
(Budge tone seems not to support GSM in my configuration, is this true?)

my second question is:
since 7701 away on a dynamic non public IP, how can I reach it, does it need to get registered to my asterisk? How do I do that, I tried but doesnt register for some reason

this is my sip.conf section

[7701]
type=friend
secret=1234
host=dynamic
defaultip=10.0.0.101

context=softphone
canreinvite=yes
dtmfmode=info
disallow=all
;allow=ulaw

allow=g729

;allow=gsm

[7702]
type=friend
secret=prueba
;host=10.0.0.102
host=dynamic
defaultip=10.0.0.102
context=softphone
canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
dtmfmode=info ; either RFC2833 or INFO for the BudgeTone

disallow=all ; need to disallow=all before we can use allow=
;allow=gsm
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!

allow=g729
allow=gsm

thanks


#2

In first Question,
If you use GSM codec, your ip hard phone must support this codec too. All most of IP hard phone don’t support this codec, so two phone can not connect and talk together. Try to use soft phone like Xlite (support gsm codec). But when you call to asterisk ivr you can hear voice from asterisk because asterisk convert to gsm codec automatic and codec G729 is licence codec if you don’t have this licence codec you can use only pass through.

In secord Question,
[7701]
type=friend
secret=1234
host=dynamic
;defaultip=10.0.0.101 ;comment this line
context=softphone
canreinvite=yes
;dtmfmode=info
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=gsm
qualify=yes ;asterisk will ping this client 7701
;allow=g729


#3

Hi, thanks for your help.

you are right, I think the Grandstream Budg phone 100 probably wont support GSM, as I woulded wanted.

and the G729 is only passthrough.

what other protocol would you recomend for a remote location with ADSL 128down/64up. Mulaw sounds very poor.

thanks


#4

Ok, I’ve tested with the phone on my same lan and I can call both ways, if I send the phone to an external lan, even with qualify = yes the phone can call any phone on the inside (where asterisk also resides) but I can’t call the external phone.

I still have the problem of the protocol (Budge Tone does not allow GSM, and 729 is passthrough, and muLaw sounds good one way but coming back from the remote site sounds poor (remote site only has 64 Kbps UP and 128 Dwn)

Any clues on both issues

this is the current sip.conf (7701 is the remote one)

[7701]
nat=yes
type=friend
secret=1234
host=dynamic
context=softphone
canreinvite=yes
dtmfmode=rfc2883
qualify=yes

disallow=all
allow=ulaw
;allow=gsm
;allow=g729

[7702]
type=friend
secret=prueba
host=dynamic
defaultip=10.0.0.102
context=softphone
canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
dtmfmode=info ; either RFC2833 or INFO for the BudgeTone

disallow=all ; need to disallow=all before we can use allow=

;allow=gsm
allow=ulaw

this is the trace:

-- Executing Dial("SIP/7702-0f1a", "SIP/7701|60|tr") in new stack

Jan 13 10:21:15 NOTICE[19606]: app_dial.c:759 dial_exec: Unable to create channel of type ‘SIP’
== Everyone is busy/congested at this time
– Executing Congestion(“SIP/7702-0f1a”, “”) in new stack

sip show peers:

7701 (Unspecified) D N 255.255.255.255 0 UNKNOWN

I’m definately missing something here :blush: