Topic: Sail 3.2 Install on SME error during install

[jameswilson]

Following this page

http://www.sailpbx.com/twiki/bin/view/Main/DocChapter03h

All ok untill this command

yum install asterisk14 asterisk14-configs asterisk14-voicemail dahdi-linux dahdi-tools asterisk-sounds-extra-en-ulaw asterisk-sounds-moh-opsound-ulaw --enablerepo=*

Then it stalls at installing asterisk sounds-en. Gets about 30% then just sits there.

Any ideas

[SARK devs]

HI

I have also seen this error.  It seems to be a problem with the asterisk-sounds-extra-en-ulaw rpm.   Top get around it I think I ran yum without it and then installed it oln its own as a separate step.

Kind Regards

S

[jameswilson]

yep that worked for me too.

[fpausp]

Hi,

I tried to install the latest sail and it worked for me, is this also possible with asterisk16 or asterisk18 ?

Why do you use asterisk14 ?


Best
fpausp

[fpausp]

Sorry, that was a bit to fast, I can not reach http://serverip/sail

How can I fix this ?


regards
fpausp

[jameswilson]

try
https://<server_ip>:8443

[fpausp]

Yes but the normal login (admin - adminpwd) is not working ?

[fpausp]

OK, I think thats the right place:

http://www.sailpbx.com/mediawiki/index.php/Beta_V3R2


The first url:

http://www.sailpbx.com/twiki/bin/view/Main/DocChapter03h


mislead me ... anyway thanks for your help.



Best
fpausp

[jameswilson]

i think the default password is asterisk

[fpausp]

Yes ...


Now I tried the "new" howto and got a new problem:

Code: [Select]

Transaction Check Error:
  file /etc/e-smith/templates/etc/hosts.allow/tftpd from install of smesailenv-3.2.0-17.noarch conflicts with file from package smeserver-tftp-server-1.0-2.el4.sme.noarch


[jameswilson]

dont know mate sorry, i installed on virgin sme

[fpausp]

I was able to install it after removing smeserver-tftp-server-1.0-2.el4.sme.noarch ...

I was not able to run dahdi under a PAE-Kernel, is there a way to use PAE ?

[fpausp]

I get this error when I try to use a SIP-Trunk:

Code: [Select]

  == Using SIP RTP CoS mark 5
    -- Executing [02626xxxxxxx@internal:1] AGI("SIP/401-0000001d", "sarkhpe,OutCos,02626xxxxxxx,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- <SIP/401-0000001d>AGI Script sarkhpe completed, returning 0
    -- Executing [02626xxxxxxx@401opencos:1] AGI("SIP/401-0000001d", "sarkhpe,OutCluster,02626xxxxxxx,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
    -- <SIP/401-0000001d>AGI Script sarkhpe completed, returning 0
    -- Executing [02626xxxxxxx@qrxvtmny:1] AGI("SIP/401-0000001d", "sarkhpe,OutRoute,DEFAULT,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=14400)
Channel will hangup at 2012-07-15 02:25:43.311 CEST.
    -- AGI Script Executing Application: (Dial) Options: (SIP/02626xxxxxxx@peer590||Tr)
[Jul 14 22:25:43] WARNING[28541]: pbx.c:1417 pbx_exec: The application delimiter is now the comma, not the pipe.  Did you forget to convert your dialplan?  (Dial(SIP/02626xxxxxxx@peer590||Tr))
  == Using SIP RTP CoS mark 5
[Jul 14 22:25:43] ERROR[28541]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("peer590||Tr", "(null)", ...): Name or service not known
[Jul 14 22:25:43] WARNING[28541]: chan_sip.c:5330 create_addr: No such host: peer590||Tr
[Jul 14 22:25:43] WARNING[28541]: acl.c:708 ast_ouraddrfor: Cannot connect
[Jul 14 22:25:43] WARNING[28541]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x93f6158 (len 855) to (null) returned -1: Invalid argument
    -- Called SIP/02626xxxxxxx@peer590||Tr
[Jul 14 22:25:43] WARNING[2981]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x93f6158 (len 855) to (null) returned -1: Invalid argument
[Jul 14 22:25:44] WARNING[2981]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x93f6158 (len 855) to (null) returned -1: Invalid argument
[Jul 14 22:25:46] WARNING[2981]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x93f6158 (len 855) to (null) returned -1: Invalid argument
[Jul 14 22:25:50] WARNING[2981]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x93f6158 (len 855) to (null) returned -1: Invalid argument
[Jul 14 22:25:58] WARNING[2981]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x93f6158 (len 855) to (null) returned -1: Invalid argument
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
[Jul 14 22:26:14] WARNING[2981]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x93f6158 (len 855) to (null) returned -1: Invalid argument
[Jul 14 22:26:15] WARNING[2981]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 1a9a3cca3fa274087127d5ef28ece7f8@192.168.xxx.xxx:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[Jul 14 22:26:15] WARNING[2981]: chan_sip.c:3651 retrans_pkt: Hanging up call 1a9a3cca3fa274087127d5ef28ece7f8@192.168.xxx.xxx:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- SIP/peer590||Tr-0000001e is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Playback) Options: (beep)
    -- <SIP/401-0000001d> Playing 'beep.gsm' (language 'en-gb')
    -- AGI Script Executing Application: (Playtones) Options: (congestion)
    -- AGI Script Executing Application: (Congestion) Options: ()

[SARK devs]

OK -guys - we're going to have to impose a little discipline here...

Please don't use a thread for multiple issues.. start a new one.  Otherwise it all gets horribly confusing to everyone.

James/fpausp please re-post your new questions to new threads and I will answer them as best I can.  Please don't post anything further to this thread now.

thanks

S