[asterisk-users] INVITE retransmission by 1.8... (Steve Murphy)

Stefan Schmidt sst at sil.at
Mon Mar 19 03:55:41 CDT 2012

Am 18.03.12 19:53, schrieb Freddi Hansen:
>> I have a site that moved to the latest 1.8 revision, and began to
>> have problems with phones in "far away places" (South America,
>> and the MidEast).
>> What I see is that when a Dial() is issued, the sip channel driver
>> sends out an INVITE to the phone.  Very soon thereafter,  Asterisk
>> retransmits the INVITE. The phone sends back a 100 Trying, and
>> then, usually, a 400 response. I may be misinterpreting what I see,
>> but I *think* that the response from the phone is delayed just enough
>> to invoke the retransmission. The phone responds to the second invite
>> with a 400 code, which Asterisk interprets as an error, and the call
>> immediately
>> goes into hangup.
>> Has anyone else seen this? It doesn't happen all the time, and only
>> with certain
>> phones. It comes and goes, but when it comes, phones become
>> unreachable. It
>> seems to be in this state the majority of the time for certain phones.
>> While most
>> phones seem far away, some are in Florida.
>> We replaced the 1.8 version of Asterisk with a 1.6.2 version, and the
>> issue has
>> gone away.  I know, I know, I could give more detail, fill out a bug
>> report, but
>> this is the early stages. I may be misinterpreting what I'm seeing.
>>  Anyone else seen this sort of thing? Any words of wisdom?

There is a option in sip.conf you should use for this and its called T1

the normal retransmit timeout is either T1*2 counting up for each
retransmission (T1*4,*8,...T1*64) or it depends on the maxms values
which a qualify gives you.
So if you enable qualify and you have values like 300 ms to the phone
the retransmission should honor this, or as i said just change the t1
value from 500 ms (default) to 1000 or 2000 ms for example.

then asterisk should not try to retransmit the next invite so early.

best regards


> hi,
> one of our gateways is used for SIP over satelite links and we se the
> same thing on default installation.
> The fix is to change chan_sip.c
> #define DEFAULT_RETRANS  to a higher value, we use 3000.
> The retransmit timer at the far end (pap2t) is increased to 3 times its
> standard values.
> It probably breaks some sip specs but its needed to keep it working when
> roundtrip gets to big.
> Freddi
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Stefan Schmidt
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