I have setup asterisk server but facing issue on call routing
Outside calls not landling on my softphone through provider (sip provider)
Error
[Dec 5 19:04:49] NOTICE[3756][C-00000001]: chan_sip.c:25894 handle_request_invite: Call from ‘provider’ (203.161.164.69:5060) to extension ‘s’ rejected because extension not found in context ‘from-pstn’.
That parse for To headers won’t work with phrase < address spec > format headers. It also won’t work if there is a space after the :.
If the ITSP is providing a proper direct in dialling service, the request URI should contain dialled digits. If not, To should be irrelevant and all numbers traffic from that peer should be treated alike.
Potentially and probably only when registration is not in use.
If registration is in place it will contain whatever customer put the Contact header during registration, i.e. what was in callbackextension= or in register=
Hi Now there’s little improvement but getting this error
– Executing [100@phones:1] NoOp(“SIP/provider-00000000”, “”) in new stack
– Executing [100@phones:2] MixMonitor(“SIP/provider-00000000”, “1480999380.0.wav,ab”) in new stack
– Executing [100@phones:3] Dial(“SIP/provider-00000000”, “SIP/sandeepk”) in new stack
== Begin MixMonitor Recording SIP/provider-00000000 [Dec 6 15:43:00] WARNING[23627][C-00000000]: app_dial.c:2455 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
** == Everyone is busy/congested at this time (1:0/0/1)**
– Executing [100@phones:4] StopMixMonitor(“SIP/provider-00000000”, “”) in new stack
== MixMonitor close filestream (mixed)
– Executing [100@phones:5] Hangup(“SIP/provider-00000000”, “”) in new stack
== Spawn extension (phones, 100, 5) exited non-zero on ‘SIP/provider-00000000’
== End MixMonitor Recording SIP/provider-00000000
[Dec 6 16:05:57] WARNING[23586]: chan_sip.c:4037 retrans_pkt: Retransmission timeout reached on transmission xxxx@xxxx for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
I hope we are very close to fix it, please lets do it, thanks
[quote=“sandeepk85, post:16, topic:68982”]
[Dec 6 16:05:57] WARNING[23586]: chan_sip.c:4037 retrans_pkt: Retransmission timeout reached on transmission xxxx@xxxx for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
I hope we are very close to fix it, please lets do it, thanks
The Retrans packet one is normally a firewall or NAT issues, i.e. either the request is never reaching the ITSP or the reponse is not getting back. In the latter case will be because it is being blocked, or because it is being sent to the wrong address (e.g. to your LAN private address,
Note, if your current configuration, above, is complete, it is not suitable for use from inside NAT (the nat options there are mainly for when the phones are inside, but Asterisk is outside). You need to tell Asterisk how to find its public address.
It seems the initial issue is solved and now you’re facing absolutely different problem.
You need to enable sip debug for ‘sandeepk’ and check what’s going on during the call attempt towards that peer.
Make sure your softphone has all the NAT traversal techniques switched off.
Add your VPN subnet as ‘localnet=’ in Asterisk.
I’m assuming that your phone has a direct VPN connection to your server without external VPN provider.
Further investigation will require clear readable sip trace, otherwise we will be only able to guess and provide general recommendations which might have no direct relation to the particular issue.
Now i needs to forward incoming sip call to another sip number (other country)
please guide how i can achieve this. Don’t i needs to register destination sip number first in SIP file /asterisk so that destination phone can receive the calls?
Is it possible to terminate coming sip call to any other mobile/landline even when its not registered in our asterisk server
please shed some light on this
what other ports needs open, other things i needs to take care of
I’m newbie to asterisk hence don’t have full clarity on the call flow