Originate SIP trunk + SIP trunk

Asterisk Asterisk Support
1

Hi All,

I have been banging my head against a wall for hours over this and cannot see the answer so any help is gratefully appreciated.

I have a SIP trunk to voipcheap.com and with it i am trying to do an Originate from my Cell Phone to another Cell Phone. Both calls go out via SIP and there is no audio at either end. When I try the same but Originating the call from one of my SIP Phones internally it works just fine.

[AGIDial]
exten => _90.,1,Set(CALLFILENAME=${ContactID}_${TIMESTAMP}) 
exten => _90.,n,NoOP(${ContactID})
exten => _90.,n,Monitor(wav,${CALLFILENAME},m) 
exten => _90.,n,Dial(SIP/0044${EXTEN:2}@VoipCheap)
exten => _90.,n,Macro(fastbusy)

The box is fairly powerful so that should not be a side that would affect this.

Any Ideas?

2

make sure your rtp.conf ports are not firewalled. That can cause this sort of issue… very frustrating :smile:

3

Yep ports are not firewalled as I can already make calls, the system is behind a NAT though and the ports are forwarded.

It is really strange that i can make two calls out although I cannot do this through an originate.

HELP Please!!!

4

log fragment for failed call ?

5 
[Nov  9 11:40:59] WARNING[6789]: translate.c:86 powerof: No bits set? 0
[Nov  9 11:40:59] WARNING[5173]: translate.c:86 powerof: No bits set? 0
--- set_address_from_contact host '194.221.62.206'
[Nov  9 11:41:08] WARNING[6790]: translate.c:86 powerof: No bits set? 0
--- set_address_from_contact host '194.221.62.206'
[Nov  9 11:43:29] WARNING[6818]: translate.c:86 powerof: No bits set? 0
6

More Debug

  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'mark' logged on from 192.168.1.200
[Nov  9 11:55:34] WARNING[7707]: translate.c:86 powerof: No bits set? 0
--- set_address_from_contact host '194.221.62.206'
       > Channel SIP/VoipCheap-09568e90 was answered.
  == Manager 'mark' logged off from 192.168.1.200
    -- Executing [90XXXXXXXXXX@AGIDial:1] Set("SIP/VoipCheap-09568e90", "CALLFILENAME=PERSONALCALL_") in new stack
    -- Executing [90XXXXXXXXXX@AGIDial:2] NoOp("SIP/VoipCheap-09568e90", "PERSONALCALL") in new stack
    -- Executing [90XXXXXXXXXX@AGIDial:3] Monitor("SIP/VoipCheap-09568e90", "wav|PERSONALCALL_|m") in new stack
    -- Executing [90XXXXXXXXXX@AGIDial:4] Dial("SIP/VoipCheap-09568e90", "SIP/0044XXXXXXXXXX@VoipCheap") in new stack
[Nov  9 11:55:42] WARNING[7708]: translate.c:86 powerof: No bits set? 0
    -- Called 0044XXXXXXXXXX@VoipCheap
    -- SIP/VoipCheap-095736e0 is making progress passing it to SIP/VoipCheap-09568e90
--- set_address_from_contact host '194.221.62.206'
    -- SIP/VoipCheap-095736e0 answered SIP/VoipCheap-09568e90
  == Spawn extension (AGIDial, 90XXXXXXXXXX, 4) exited non-zero on 'SIP/VoipCheap-09568e90'
7

Have just checked the latest version from svn and still no change!!! Must be config although no ideas…Anyone?

8

Even if the calls go thru there can still be a nat issue. The call goes over one port while the rtp stream goes over many others. Please post your sip.conf. Also I have seen $1000.00 firewalls drop voip packets for abslutely no reason even though there was rule to let the packets thru.