Greške #13693
Zatvoren
evo šta asterisk kaže:
-- Executing [1@demo:1] Dial("SIP/officesa_hernad-08203dc0", ""sip/stan1"| 200") in new stack
-- Called stan1
-- SIP/stan1-0820d988 is ringing
-- SIP/stan1-0820d988 answered SIP/officesa_hernad-08203dc0
-- Native bridging SIP/officesa_hernad-08203dc0 and SIP/stan1-0820d988
== Spawn extension (demo, 1, 1) exited non-zero on 'SIP/officesa_hernad-08203dc0'
ako kažem u sip.conf canreinvite=no
[stan1]
type=friend
username=stan1
password=xxx
host=dynamic
callerid="stan-1"
allow=all
qualify=yes
nat=no
canreinvite=no
[officesa_hernad]
type=friend
username=officesa_hernad
password=xx
host=dynamic
callerid="officesa-hernad"
allow=all
qualify=yes
nat=no
canreinvite=no
dobijem isti rezultat
-- Executing [1@demo:1] Dial("SIP/officesa_hernad-0820b978", ""sip/stan1"| 200") in new stack
-- Called stan1
-- SIP/stan1-0820f8e0 is ringing
-- SIP/stan1-0820f8e0 answered SIP/officesa_hernad-0820b978
-- Packet2Packet bridging SIP/officesa_hernad-0820b978 and SIP/stan1-0820f8e0
jedino što prijavljuje Packet2Packet bridging a ne Native bridging
kvaka je što dođe do spajanja bez da druga strana digne slušalicu ?!
IAX -> SIP pak radi bez problema
ovo bi moglo biti interesantno
http://www.mail-archive.com/wengophone-devel@lists.openwengo.com/msg05477.html
[Wengophone-devel] Wengophone 2.2 and Asterisk problem
Alejandro Facultad
Thu, 11 Oct 2007 17:33:06 -0700
I'm using wengophone 2.2 for Linux and I log into an Asterisk SIP server. I have configured the Asterisk to permit "canreinvite=yes" so SIP control packets go among clients and Asterisk and RTP data packets go among clients directly (bypassing Asterisk).
But when I establish a call between two Wengophones, I sniffer the communication and I realize that SIP+RTP packets go among clients and Asterisk always (RTP does not bypass Asterisk).
Does Wengophone have an option that set up "canreinvite=no" always or something like this to avoid RTP data packets travel directly among Wengos and not via Asterisk ???
Thanking in advance
greetings
A.F.
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