Asterisk to forward to different numbers

Asterisk Asterisk Support
1

I’m not sure if this is a problem with my setup, or if my problem is specific to BroadVoice.

We have two lines (sip1 & sip2) with BroadVoice. We have 10 numbers each assigned to each line. Forwarding seems to be working, but there is NO audio flowing either way once the call is bridged. I’m 99.9% sure this is not a firewall issue. Can someone tell me what I’ve done wrong?

Here is my asterisk config:
[sip.conf]

register => 919XXXXXXX:XXXXXXXXXX@sip.broadvoice.com
register => 919XXXXXXY:XXXXXXXXXX@sip.broadvoice.com

[sip2.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=919XXXXXXX
authuser=919XXXXXXX
secret=XXXXXXXXX
username=919XXXXXXX
insecure=invite,port
context=default
authname=919XXXXXXX
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
disallow=all
allow=ulaw

[sip1.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=919XXXXXXX
authuser=919XXXXXXX
secret=XXXXXXXXXXX
username=919XXXXXXX
insecure=invite,port
context=default
authname=919XXXXXXX
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
disallow=all
allow=ulaw

Important Part of exten.conf

[default] exten => INCOMINGNUMBER,1,dial(SIP/FORWARDNUMBER@sip1.broadvoice.com,30)

Log Output

Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [919XXXXXXX@default:1] Dial("SIP/sip2.broadvoice.com-50006b68", "SIP/919XXXXXXX@sip1.broadvoice.com,30") in new stack == Using SIP RTP CoS mark 5 -- Called 919XXXXXXX@sip1.broadvoice.com << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/sip1.broadvoice.com-5400b4e8] -- SIP/sip1.broadvoice.com-5400b4e8 is ringing << [ TYPE: Control (4) SUBCLASS: Unknown control '14' (14) ] [SIP/sip1.broadvoice.com-5400b4e8] -- SIP/sip1.broadvoice.com-5400b4e8 is making progress passing it to SIP/sip2.broadvoice.com-50006b68 << [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/sip1.broadvoice.com-5400b4e8] -- SIP/sip1.broadvoice.com-5400b4e8 answered SIP/sip2.broadvoice.com-50006b68 -- Packet2Packet bridging SIP/sip2.broadvoice.com-50006b68 and SIP/sip1.broadvoice.com-5400b4e8 == Spawn extension (default, 919XXXXXXX, 1) exited non-zero on 'SIP/sip2.broadvoice.com-50006b68'

2

Hello

I’d Check

  1. Is incoming RTP (UDP) ports natted/allowed to get into asterisk?
  2. Is your firewall rewriting sip headers? (sip gateway/proxy in some firewalls)
  3. Is your nat steup betwen asterisk and firewall OK? --> users/externip…
  4. Connect from other sip phone outside the office anf wireshark traffic