Koozali.org: home of the SME Server

Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: kb-ohnemus on June 10, 2011, 10:29:02 AM

Title: Updated to 2.6.1-11 / asterisk 1.4, some problems
Post by: kb-ohnemus on June 10, 2011, 10:29:02 AM

Hi

I've been using sail 2.1.14 / asterisk 1.2 on SME7 for quite a while and recently updated 8.0 beta / sail 2.6.1-11 / asterisk 1.4.13 keeping the old configuration. Well, partly, I had to edit some parts.

So running this for a while I noticed some minor problems:

1. I have a HFC-ISDN-Card using MISDN (two channels) and one sipgate account. In the route the two MISDN channels are first and second, third ist sipgate. So calling out MISDN is used as long as one channel is free. This works for all phones (PAP2, Qutecom, Twinkle...) except for this one: http://www.nettalk.ws/ke1020.htm
Calling out it tries to call MISDN twice an then sipgate ist used. I get this in asterisk:

Code: [Select]

   -- Executing [01708037314@internal:1] AGI("SIP/5001-000000b7", "selintra|OutCluster|01708037314") in new stack                                           
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
    -- AGI Script selintra completed, returning 0
    -- Executing [01708037314@qrxvtmny:1] AGI("SIP/5001-000000b7", "selintra|OutRoute|Out") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra                                                                                               
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=3600)                                                                             
    -- Channel will hangup at 2011-06-10 08:11:00 UTC.                                                                                                       
    -- AGI Script Executing Application: (Dial) Options: (MISDN/g:TEPP/01708037314||T)                                                                       
    -- Called g:TEPP/01708037314                                                                                                                             
[Jun 10 09:11:00] WARNING[25097]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf                                                           
[Jun 10 09:11:00] WARNING[25097]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf                                                           
    -- No one is available to answer at this time (1:0/0/0)                                                                                                 
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=3600)                                                                             
    -- Channel will hangup at 2011-06-10 08:11:00 UTC.                                                                                                       
    -- AGI Script Executing Application: (Dial) Options: (MISDN/g:TEPP/01708037314||T)                                                                       
    -- Called g:TEPP/01708037314                                                                                                                             
[Jun 10 09:11:01] WARNING[25097]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf                                                           
[Jun 10 09:11:01] WARNING[25097]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf                                                           
[Jun 10 09:11:01] WARNING[25097]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf                                                           
    -- No one is available to answer at this time (1:0/0/0)                                                                                                 
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=3600)                                                                             
    -- Channel will hangup at 2011-06-10 08:11:01 UTC.                                                                                                       
    -- AGI Script Executing Application: (Dial) Options: (SIP/01708037314@2540158||T)                                                                       
    -- Called 01708037314@2540158                                                                                                                           
2. Calling out using sipgate I sometimes (maybe 50%) I get a "please hang up and try again". Sometimes it works. Asterisk says: Got SIP response 484 "Address incomplete" back from 217.10.79.9 

Code: [Select]

    -- AGI Script Executing Application: (Dial) Options: (SIP/01708037314@2540158||T)                                                                       
    -- Called 01708037314@2540158                                                                                                                           
    -- Got SIP response 484 "Address incomplete" back from 217.10.79.9                                                                                       
  == Everyone is busy/congested at this time (1:0/0/1)                                                                                                       
    -- AGI Script Executing Application: (Background) Options: (were-sorry)                                                                                 
    -- <SIP/5001-000000b7> Playing 'were-sorry' (language 'en')                                                                                             
    -- AGI Script Executing Application: (Background) Options: (call-cannot-complete)                                                                       
[Jun 10 09:11:09] WARNING[25097]: file.c:665 ast_openstream_full: File call-cannot-complete does not exist in any format                                     
[Jun 10 09:11:09] WARNING[25097]: file.c:995 ast_streamfile: Unable to open call-cannot-complete (format 0x8 (alaw)): No such file or directory             
[Jun 10 09:11:09] WARNING[25097]: pbx.c:5846 pbx_builtin_background: ast_streamfile failed on SIP/5001-000000b7 for call-cannot-complete                     
    -- AGI Script Executing Application: (Background) Options: (please-hang-up-and-try-again)                                                               
    -- <SIP/5001-000000b7> Playing 'please-hang-up-and-try-again' (language 'en')                                                                           
    -- AGI Script selintra completed, returning 0                                                                                                           
  == Auto fallthrough, channel 'SIP/5001-000000b7' status is 'CHANUNAVAIL'                                                                                   
    -- Executing [h@qrxvtmny:1] Hangup("SIP/5001-000000b7", "") in new stack                                                                                 
  == Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/5001-000000b7'                                                                                 
3. Hash transfer won't do anything. I hit the hash key and nothing happens. Allow hash transfer is enabled.

Regards
Manuel

Title: Re: Updated to 2.6.1-11 / asterisk 1.4, some problems
Post by: SARK devs on June 11, 2011, 03:48:02 PM

Hello Manuel

1. I guess if it's just that phone then the phone is doing something that the others aren't.  This isn't a known problem with other phones (Snom, Cisco, Polycom, Aastra etc).
2. The dial is being sent according to your wishes but for some reason SIPgate doesn't like what you've sent. 
3. In SAIL, hash dial is set as ## by default.  This is because # is used a lot by corporate IVR systems (often to denote the end of an account number or some such). Anyway, you can set it to whatever you like by changing it in sark_features_featuremap.conf and restarting asterisk.

Kind Regards

S