Problem 2 Asterisk server

-I have 2 asterisk server is severPC and serverNB.
-I have 2 account : 24500 register to serverPC
24501 register to serverNB
-I want call between 24500 and 24501 but I have problem.
-I create follow link: … rs-via-sip
My Problem
sip : … sp=sharing
extension : … sp=sharing **severPC -> serverPC
log : … sp=sharing

Thank you and I’m sorry, my English isn’t very good.

Two many links to retrieve, but you may be having problems because you are using friend when you should use peer.

Please include the relevant configuration and logs as code or quote sections in your posting.

thx , david55
i use peer but same problem.

=================== Notebook ========================
include => internal
include => remote

exten => 24501,1,Dial(SIP/24501)

exten => 24500,1,Dial(SIP/serverPC/24500)
exten => 24500,2,Hangup()

include => internal

register => serverNB:pass_serverNB@


======================== PC ===========================
exten => 24500,1,Dial(SIP/24500)

include =>internal
include =>remote

exten => 24501,1,Dial(SIP/serverNB/24501)
exten => 24501,2,Hangup()

include => internal


register => serverPC:pass_serverPC@


I solve old problem ,bring to new problem.
I test call to 24500(PC) from 24501(NB).

==============cli on Notebook====================================================
== Using SIP RTP CoS mark 5
– Executing [24500@phones:1] Dial(“SIP/24501-00000014”, “SIP/serverPC/24500”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/serverPC/24500
[Feb 6 00:35:00] WARNING[2340]: chan_sip.c:20366 handle_response_invite: Received response: “Forbidden” from ‘“24501” sip:24501@;tag=as2d19741b’
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [24500@phones:2] Hangup(“SIP/24501-00000014”, “”) in new stack
== Spawn extension (phones, 24500, 2) exited non-zero on ‘SIP/24501-00000014’
[Feb 6 00:35:00] NOTICE[2340]: chan_sip.c:23107 handle_request_invite: Unable to create/find SIP channel for this INVITE

================cli on PC =======================================================
[Feb 6 14:35:00] NOTICE[2409]: chan_sip.c:23364 handle_request_invite: Failed to authenticate device “24501” sip:24501@;tag=as2d19741b

thank you.

Both servers must have the same secret, or you must specify both secret and remotesecret. remotesecret is the secret value for the other server. It defaults to secret. This is a relatively new feature. Previously you had to have both secrets the same or disable authentication on invites.

If you know the IP address, using host=dynamic is silly.

thx, david .
The same secret it work, but new problem agian is call drop in 30-40 second.

[Feb 6 20:47:09] WARNING[2340]: chan_sip.c:20457 handle_response_invite: just did sched_add waitid(405) for sip_reinvite_retry for dialog 6d6cedd96e5b9d100baa41016bf4c4b6@ in handle_response_invite

[Feb 6 20:58:02] WARNING[2332]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission 6002db9b6d66712311ebe116164b4983@ for seqno 104 (Critical Response) – See … nsmissions
Packet timed out after 31999ms with no response
[Feb 6 20:58:02] WARNING[2332]: chan_sip.c:4012 retrans_pkt: Hanging up call 6002db9b6d66712311ebe116164b4983@ - no reply to our critical packet (see … nsmissions).
== Spawn extension (serverNB_incoming, 24500, 1) exited non-zero on ‘SIP/serverNB-00000018’

If between two Asterisk boxes, look for firewall problems. If between Asterisk and something else, the something else is broken; disable directmedia and COLP (sendrpid=no).

thx ,david55 .
I solve my problem by reinstall asterisk .
I have one question, Can use softphone on notebook call to dahdi(GSM phone) on PC?? and explain please.

thank you.

Asterisk is a back to back user agent, so the channel technology doesn’t need to match between the A and B parties. For further details see