chan_sip.c:3370 __sip_xmit

[system]

Hallo,

i’m here to find help with my AVM Fritz.Box and Asterisk.

Sometimes it connect to the asterisk, but most there’s no connection.

My firewall allows Input connection to TCP / UDP 5060.

@asterisk -vvvvvr

[Apr 11 15:10:51] WARNING[24090]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xee5840 (len 953) to my-ip:61306 returned -1: Operation not permitted
[Apr 11 15:10:52] WARNING[24090]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xee5840 (len 953) to my-ip:61306 returned -1: Operation not permitted
– Registered SIP ‘4601’ at MY-IP:61350
> Saved useragent “Twinkle/1.4.2” for peer 4601
[Apr 11 15:10:54] WARNING[24090]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xee5840 (len 953) to my-ip:61306 returned -1: Operation not permitted
[Apr 11 15:10:56] WARNING[24090]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xeda2f0 (len 955) to my-ip:61306 returned -1: Operation not permitted
[Apr 11 15:10:58] WARNING[24090]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xee5840 (len 953) to my-ip:61306 returned -1: Operation not permitted
[Apr 11 15:11:06] WARNING[24090]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xee5840 (len 953) to my-ip:61306 returned -1: Operation not permitted
[Apr 11 15:11:12] WARNING[24090]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xeda2f0 (len 955) to my-ip:61306 returned -1: Operation not permitted

From the same pc in network twinkle can connect, as you can see.

Any Ideas?

Thanks and best regards
Frank

[david55]

You need to open the port ranges configured at each end for RTP.

[system]

@cat_/etc/asterisk/rtp.conf

…
rtpstart=10000
rtpend=20000
…

I think that are the port range.

Must firewall rule output and/or input and tcp or udp?

can i change the rtpstart / rtpend?

Thanks and best regards.

[david55]

Both ends.

You need to change the ranges on the devices that are connecting Asterisk, or fully open all outbound UDP.

[system]

@cat_sip.conf

[1000]
type=friend
secret=secret
host=dynamic

[1001]
type=friend
secret=secret
host=dynamic

@cat_extensions.conf

[default]
exten => 3300,1,Answer()
exten => 3300,2,Playback(hello-world)
exten => 3300,3,Hangup()

exten => 1000,1,Dial(SIP/1000)
exten => 1001,1,Dial(SIP/1001)

i can phone from 1000 to 3300 (but i don’t have sound, i don’t hear hello world)
I can phone from 1001 to 3300 (but i don’t have sound, i don’t hear hello world)
I can’t phone from 1000 to 1001
I can’t phone from 1001 to 1000

@cat_asterisk -vvvvvr

[Apr 11 16:08:47] WARNING[22607]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 6b3esssssddffdfsdfsdf760ecc0298@MY-IP:5060 for seqno 102 (Critical Request) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response
[Apr 11 16:08:47] WARNING[22607]: chan_sip.c:3670 retrans_pkt: Hanging up call 6b3esssssddffdfsdfsdf760ecc0298@MY-IP:5060 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).

i don’t know what to do.

Thanks.

[david55]

You will need to provide more details (a sip trace, possibly using sip set debug on) to work out which packet wasn’t getting through, however you might want to note that some soft phones have broken imlementations of sip re-invites.