Maximum Retries

Asterisk Asterisk Support
1

Hi,

I can’t join other softphone (X-Lite), I have always the same error :

WARNING[32084]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 68d5680928423a585ae1ae6b28cbcb1e@192.168.200.2 for seqno 102 (Critical Request)

I don’t know where the problem comes ? is it a sip configuration mistake ? a bug in asterisk ? … where should I see to solve and understand the problem ?

Thx for your help …

2

Please send the entire verbose CLI for what you are trying to do. From the Dial string through to where the error occurs, or if it is an inbound call with the client first connects.

3

I just want to call from 5169 to 6951 or any other phone …

This is mi piece of CLI log.

 [chan_sip.so] => (Session Initiation Protocol (SIP))
  == Parsing '/etc/asterisk/sip.conf': Found
    -- SIP Seeding '5169' at 5169@192.168.200.102:5060 for 1800
    -- SIP Seeding '6951' at 6951@192.168.200.108:5060 for 1800
    -- SIP Seeding '0176' at 0176@192.168.200.106:5060 for 1800
    -- SIP Seeding '7777' at 7777@192.168.200.109:5060 for 1800
  == SIP Listening on 0.0.0.0:5060
  == Using TOS bits 0
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
  == Registered application 'SIPDtmfMode'
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
  == Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger restarted
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/enum.conf': Found
  == Parsing '/etc/asterisk/rtp.conf': Found
  == RTP Allocating from port range 10000 -> 20000
Asterisk Ready.
*CLI> Oct 19 12:07:04 WARNING[2780]: chan_oss.c:257 sound_thread: Read error on sound device: Resource temporarily unavailable
    -- Executing Macro("SIP/5169-2e58", "comportement_default|SIP/6951") in new stack
    -- Executing Dial("SIP/5169-2e58", "SIP/6951|20|t") in new stack
    -- Called 6951
Oct 19 12:07:20 WARNING[2780]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 6e58e74b13dc417a3b3288d169224842@192.168.200.2 for seqno 102 (Critical Request)
  == No one is available to answer at this time
    -- Executing VoiceMail("SIP/5169-2e58", "u6951") in new stack
    -- Playing 'vm-theperson' (language 'en')
    -- Playing 'digits/6' (language 'en')
    -- Playing 'digits/9' (language 'en')
    -- Playing 'digits/5' (language 'en')
    -- Playing 'digits/1' (language 'en')
    -- Playing 'vm-isunavail' (language 'en')
    -- Playing 'vm-intro' (language 'en')
    -- Playing 'beep' (language 'en')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/6951/INBOX/msg0002 format: wav49, 0x1769f798
    -- x=1, open writing:  /var/spool/asterisk/voicemail/default/6951/INBOX/msg0002 format: gsm, 0x1769f920
    -- x=2, open writing:  /var/spool/asterisk/voicemail/default/6951/INBOX/msg0002 format: wav, 0x176a3be0
    -- User hung up
  == Spawn extension (macro-comportement_default, s, 2) exited non-zero on 'SIP/5169-2e58' in macro 'comportement_default'
  == Spawn extension (mytest, 6951, 1) exited non-zero on 'SIP/5169-2e58'

extensions.conf

[general]
static=yes
writeprotect=yes


[globals]
PHONE1=SIP/5169
PHONE2=SIP/6951
PHONE3=SIP/0176
PHONE4=SIP/7777
PHONE5=SIP/3351



[macro-comportement_default]
; ${ARG1} - Phoneline

exten => s,1,Dial(${ARG1},20,t)
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup

[mytest]

exten => 0,1,Ringing(4)
exten => 0,2,Answer
exten => 0,3,Playback(demo-congrats)
exten => 0,4,Wait(1)
exten => 0,5,Hangup

;Internal Phoneline
exten => 5169,1,Macro(comportement_default,${PHONE1})
exten => 6951,1,Macro(comportement_default,${PHONE2})
exten => 0176,1,Macro(comportement_default,${PHONE3})
exten => 7777,1,Macro(comportement_default,${PHONE4})

sip.conf


[general]
;context=incoming                ; Default context for incoming calls
recordhistory=yes               ; Record SIP history by default
                                ; (see sip history / sip no history
;realm=192.168.200.2            ; Realm for digest authentication
                                ; defaults to "asterisk"
                                ; Realms MUST be globally unique according to RFC 3261
                                ; Set this to your host name or domain name
port=5060                       ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=no                    ; Enable DNS SRV lookups on outbound calls

disallow=all                    ; First disallow all codecs
allow=ilbc                      ; Allow codecs in order of preference
allow=ulaw
allow=gsm                       ; Note: codec order is respected only in 

[general]
                                ; This may also be set for individual users/peers
[5169]
type=friend
username=5169
secret=5169
callerid="Nicolas <5169>"
context=mytest
host=dynamic

[6951]
type=friend
username=6951
secret=6951
callerid="Florent <6951>"
context=mytest
host=dynamic

[0176]
type=friend
username=0176
secret=0176
callerid="Fabien <0176>"
context=mytest
host=dynamic

[7777]
type=friend
username=7777
secret=7777
callerid="Mathieu <7777>"
context=mytest
host=dynamic
4

it seems the the you are having problems with chan_oss…

WARNING[2780]: chan_oss.c:257 sound_thread: Read error on sound device: Resource temporarily unavailable
5

I’ve just solved the oss pb with a new compilation of kernel inclunding the sound driver of my card.

But the Maximum packet retries still appears …

I don’t know where the problem comes …

thx for your help …

6

I’ve changed the xlite by an other softphone and the behaviour is the same. F***

whatever softphone I’m using there is the same message and I can’t join other sipphone.


Oct 20 11:45:25 WARNING[5035]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 05702a4c4699b40a4b2862a02c9c703a@192.168.200.2 for seqno 102 (Critical Request)
7

I have been getting the same error, but it is only when I use a soft phone like Xlite or SJPhone. The “hard” IP phones like Polycom does not generate that error. So far I have been unable to track it down - keep digging :smile:

8

[quote=“gourou69”] -- Executing Dial("SIP/5169-2e58", "SIP/6951|20|t") in new stack -- Called 6951 Oct 19 12:07:20 WARNING[2780]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 6e58e74b13dc417a3b3288d169224842@192.168.200.2 for seqno 102 (Critical Request)[/quote]
Firstly, is your softphone registering with the Asterisk server? To check this, do:

sip show registry

If it isn’t you’ll have to make sure the phone’s configured properly. However, if the IP address of the softphone is always the same, it’s probably better not to use

host=dynamic

Instead, use

host=192.168.0.123

or whatever. If you do that, the softphone doesn’t have to register with the Asterisk server.