Some errors

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Asterisk Users 3 Comments

On 03/15/2011 10:18 AM, Paul Belanger wrote:
> On 11-03-15 10:11 AM, Fellipe Paes wrote:
>> [Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
>> 0x794f840 (len 828) to (null) returned -1: Invalid argument
>> [Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3386 retrans_pkt:
>> Retransmission timeout reached on transmission
>> 521df80947598d560109d73f1493a76d@ for seqno 102
>> (Critical Request) — See doc/sip-retransmit.txt.
>> Packet timed out after 32000ms with no response
>> Well… everything works fine, but I don’t like this errors, any ideas?
> Theses are leftover issue with the IPv6 conversion for Asterisk 1.8.
> Collect a complete debug log[1] and open a new issue on the tracker.
> [1]

Umm… no, I don’t think they are. His log showed an attempt to use
Dial() to connect to ‘SIP/h’, and he had no peer named ‘h’ or is that an
IP address or DNS name. It should have failed a little more cleanly than
it did, but I’m sure that at least part of the problem is attempting to
dial a SIP endpoint that doesn’t exist (and dialing out from the ‘h’
extension as well).

3 thoughts on - Some errors

  • You can enable / disable IPv6 via the config files, but the core API /
    ABI have changed.

    But Kevin is correct, your issue does look to be a dialplan problem
    (SIP/h). I should have been more specific about handling the failure
    more gracefully.

  • Hello guys,

    I received the following answer from Digium, I understand and will change my dialplan, but I have one more question now, why I can’t use _. in my dialplan?

  • Since you are just learning Asterisk, I would *highly* recommend not
    using ‘exten => _X.’; this is a bad practice. Taking the time to
    understand how pattern matching works with Asterisk will help you in the
    long run.