[Asterisk-Users] problems with asterisk and the IAX protocol

[Pamela Weis]

Hello group,

I wanted to try out the asterisk iax protocol between two asterisk
machines but have several problems with it.
My scenario looks like follows. I am using asterisk 0.9.0 on both machines.

SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2

Both SER and asterisk run on a machine with a public IP address. When
the telephone on one side makes a call the telephone on the other side
rings. But whenever I pick up the call, asterisk2 hangs up without much
warning and then the telephone rings unexpectedly again and again.

Here is the output of the two asterisk machines:
asterisk 1:
*CLI>
    -- Accepting AUTHENTICATED call from 62.116.33.72, requested format
= 256, actual format = 256
    -- Executing Dial("IAX2[asterisk2 at asterisk2]/1",
"SIP/123 at 62.116.54.194") in new stack
    -- Called 123 at 62.116.54.194
    -- SIP/62.116.54.194-b71d is ringing
    -- SIP/62.116.54.194-b71d answered IAX2[asterisk2 at asterisk2]/1
  == Spawn extension (local, 123, 1) exited non-zero on
'IAX2[asterisk2 at asterisk2]/1'
    -- Hungup 'IAX2[asterisk2 at asterisk2]/1'
    -- Accepting AUTHENTICATED call from 62.116.33.72, requested format
= 256, actual format = 256
    -- Executing Dial("IAX2[asterisk2 at asterisk2]/2",
"SIP/123 at 62.116.54.194") in new stack
    -- Called 123 at 62.116.54.194
    -- SIP/62.116.54.194-6749 is ringing

---
asterisk2:
*CLI>     -- Executing Dial("SIP/-0811bef8",
"IAX2/asterisk2:19 at 62.116.54.194/123 at local") in new stack
    -- Called asterisk2:19 at 62.116.54.194/123 at local
    -- Call accepted by 62.116.54.194 (format G729A)
    -- Format for call is G729A
    -- IAX2[asterisk]/1 stopped sounds
    -- IAX2[asterisk]/1 stopped sounds
    -- IAX2[asterisk]/1 answered SIP/-0811bef8
Aug  5 17:57:00 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call 3c26d0834834-znq2uf92hxij at 10-33-10-103 for
seqno 1 (Response)
    -- Hungup 'IAX2[asterisk]/1'
  == Spawn extension (sip, 123, 1) exited non-zero on 'SIP/-0811bef8'
Aug  5 17:57:05 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call 3c26d0834834-znq2uf92hxij at 10-33-10-103 for
seqno 1 (Response)
Aug  5 17:57:06 WARNING[65541]: chan_sip.c:497 retrans_pkt: Maximum
retries exceeded on call 3c26d0834834-znq2uf92hxij at 10-33-10-103 for
seqno 102 (Request)
    -- Executing Dial("SIP/-0811bef8",
"IAX2/asterisk2:19 at 62.116.54.194/123 at local") in new stack
    -- Called asterisk2:19 at 62.116.54.194/123 at local
    -- Call accepted by 62.116.54.194 (format G729A)
    -- Format for call is G729A
    -- IAX2[asterisk]/2 stopped sounds
    -- Hungup 'IAX2[asterisk]/2'
  == No one is available to answer at this time

----

I also have another question to asterisk and NAT:
o) If one asterisk machine and the telephones are behind NAT, do I need
a proxy to get the speech through, or should asterisk work this out on
its own?

Any help with my problem will be greatly appreciated. Thanks in advance.

Pamela Weis