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

[Pamela Weis]

Hi Kevin,

no you didn't miss the reply and I've not resolved it yet.
Have you got similar problems?

Pamela

Kevin Fjelsted wrote:

>Pamela,
>Did you resolve the problems you described?
>I didn't see a reply on the list but I may have missed it.
>
>-Kevin
>
>-----Original Message-----
>From: Pamela Weis [mailto:peawy at gmx.at]
>Sent: Thursday, August 05, 2004 10:22 AM
>To: asterisk-users at lists.digium.com
>Subject: [Asterisk-Users] problems with asterisk and the IAX protocol
>
>
>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
>
>
>
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