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Call won't go though --- (Ripping my hair out)

Offline Teviot

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Call won't go though --- (Ripping my hair out)
« on: July 05, 2010, 07:31:35 AM »
Hi

I have a problem.  Call that used to go though with no problem don't any more.  Have looked at it changed a codec an settings to match but still nothing

Code: [Select]
pbx*CLI>
    -- Executing [131332@internal:1] AGI("SIP/201-09dd28a0", "selintra|OutCluster|131332") 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 [131332@qrxvtmny:1] AGI("SIP/201-09dd28a0", "selintra|OutRoute|13 Numbers") 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 2010-07-05 06:28:32 UTC.
    -- AGI Script Executing Application: (Dial) Options: (IAX2/pbxsail@sailpbx/131332||T)
    -- Called pbxsail@sailpbx/131332
[Jul  5 15:28:32] WARNING[20414]: chan_iax2.c:7750 socket_process: Call rejected by 220.157.76.100: <Unknown>
    -- Hungup 'IAX2/sailpbx-16388'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- AGI Script Executing Application: (Background) Options: (were-sorry)
[Jul  5 15:28:32] WARNING[20785]: file.c:602 ast_openstream_full: File were-sorry does not exist in any format
[Jul  5 15:28:32] WARNING[20785]: file.c:912 ast_streamfile: Unable to open were-sorry (format 0x4 (ulaw)): No such file or directory
[Jul  5 15:28:32] WARNING[20785]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/201-09dd28a0 for were-sorry
    -- AGI Script Executing Application: (Background) Options: (call-cannot-complete)
[Jul  5 15:28:32] WARNING[20785]: file.c:602 ast_openstream_full: File call-cannot-complete does not exist in any format
[Jul  5 15:28:32] WARNING[20785]: file.c:912 ast_streamfile: Unable to open call-cannot-complete (format 0x4 (ulaw)): No such file or directory
[Jul  5 15:28:32] WARNING[20785]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/201-09dd28a0 for call-cannot-complete
    -- AGI Script Executing Application: (Background) Options: (please-hang-up-and-try-again)
[Jul  5 15:28:32] WARNING[20785]: file.c:602 ast_openstream_full: File please-hang-up-and-try-again does not exist in any format
[Jul  5 15:28:32] WARNING[20785]: file.c:912 ast_streamfile: Unable to open please-hang-up-and-try-again (format 0x4 (ulaw)): No such file or directory
[Jul  5 15:28:32] WARNING[20785]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/201-09dd28a0 for please-hang-up-and-try-again
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/201-09dd28a0' status is 'CHANUNAVAIL'
    -- Executing [h@qrxvtmny:1] Hangup("SIP/201-09dd28a0", "") in new stack
  == Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/201-09dd28a0'


All other call go though with out an problems

Please help
Regards
M0GLJ
......................................................
I am new to SAIL SME Server v8b6 and have been using SME for many years.
I have already done some research and only ask questions if I still can't work it out.

Offline chris burnat

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Re: Call won't go though --- (Ripping my hair out)
« Reply #1 on: July 05, 2010, 08:08:30 AM »
Are you with Exetel?  They have changed their server last Saturday, not working for me anylonger...

I have also tried your number with Pennytel (I have used 52 prefix to force the call to Pennytel) , no joy:

 -- Executing [52131332@internal:1] AGI("SIP/5000-00000143", "selintra|OutCluster|xxxxxxx") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
    -- AGI Script selintra completed, returning 0
    -- Executing [52131332@qrxvtmny:1] AGI("SIP/5000-00000143", "selintra|OutTrunk|xxxxxxxxx") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/131332@PENYTEL||)
    -- Called 131332@PENYTEL
[Jul  5 16:00:22] WARNING[5536]: chan_sip.c:2013 retrans_pkt: Maximum retries exceeded on transmission 7ad6218237c67af711da9c750733af52@xxxxxxxxx for seqno 103 (Critical Request) -- See doc/sip-retransmit.txt.
[Jul  5 16:00:22] WARNING[5536]: chan_sip.c:2035 retrans_pkt: Hanging up call 7ad6218237c67af711da9c750733af52@xxxxxxxxx - no reply to our critical packet (see doc/sip-retransmit.txt).
  == Everyone is busy/congested at this time (1:0/0/1)
    -- AGI Script Executing Application: (Background) Options: (were-sorry)
    -- <SIP/5000-00000143> Playing 'were-sorry' (language 'en-gb')
    -- AGI Script Executing Application: (Background) Options: (call-cannot-complete)
    -- <SIP/5000-00000143> Playing 'call-cannot-complete' (language 'en-gb')
    -- AGI Script Executing Application: (Background) Options: (please-hang-up-and-try-again)
    -- <SIP/5000-00000143> Playing 'please-hang-up-and-try-again' (language 'en-gb')
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/5000-00000143' status is 'CHANUNAVAIL'
    -- Executing [h@qrxvtmny:1] Hangup("SIP/5000-00000143", "") in new stack
  == Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/5000-00000143'

PS: lucky you to still have hairs to rip off.....
- chris
If it does not work out of the box, please fill in a Bug Report @ Bugzilla (http://bugs.contribs.org)  - check: http://wiki.contribs.org/Bugzilla_Help .  Thanks.

Offline SARK devs

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Re: Call won't go though --- (Ripping my hair out)
« Reply #2 on: July 05, 2010, 01:10:39 PM »
The far end is bouncing the call.  You need to look at that machine - not this one.

You also need to install the Asterisk additional sounds pack or set the "play tones" settings to YES in Globals.  Right now you're getting errors in Asterisk because you have it set to play the autoattendent messages but the recordings aren't there.

Quote
    -- AGI Script Executing Application: (Background) Options: (were-sorry)
[Jul  5 15:28:32] WARNING[20785]: file.c:602 ast_openstream_full: File were-sorry does not exist in any format
[Jul  5 15:28:32] WARNING[20785]: file.c:912 ast_streamfile: Unable to open were-sorry (format 0x4 (ulaw)): No such file or directory
[Jul  5 15:28:32] WARNING[20785]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/201-09dd28a0 for were-sorry

Kind Regards

S

Offline Teviot

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Re: Call won't go though --- (Ripping my hair out)
« Reply #3 on: July 06, 2010, 04:28:44 AM »
The far end is bouncing the call.  You need to look at that machine - not this one.

You also need to install the Asterisk additional sounds pack or set the "play tones" settings to YES in Globals.  Right now you're getting errors in Asterisk because you have it set to play the autoattendent messages but the recordings aren't there.

Kind Regards

S

Which asterisk additional sound do I need to install and where would I get them from?

I'm using Asterisk 1.4.21.1 with sail-2.2.4-47

« Last Edit: July 06, 2010, 05:05:42 AM by teviot »
Regards
M0GLJ
......................................................
I am new to SAIL SME Server v8b6 and have been using SME for many years.
I have already done some research and only ask questions if I still can't work it out.

Offline chris burnat

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Re: Call won't go though --- (Ripping my hair out)
« Reply #4 on: July 06, 2010, 05:47:36 AM »
Which asterisk additional sound do I need to install and where would I get them from?

I'm using Asterisk 1.4.21.1 with sail-2.2.4-47

if you want the UK sounds (James is best!) check:
http://forums.contribs.org/index.php/topic,46173.0.html
- chris
If it does not work out of the box, please fill in a Bug Report @ Bugzilla (http://bugs.contribs.org)  - check: http://wiki.contribs.org/Bugzilla_Help .  Thanks.

Offline Teviot

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Re: Call won't go though --- (Ripping my hair out)
« Reply #5 on: July 06, 2010, 06:12:38 AM »
Tried to install with the following result

Code: [Select]
[root@sail ~]# wget http://sarkpbx.com/sail/languagepack/Asterisk-1.4.28+/sme-ast-en-uk-gpl-sounds-1.2.0-4.noarch.rpm
--13:51:32--  http://sarkpbx.com/sail/languagepack/Asterisk-1.4.28+/sme-ast-en-uk-gpl-sounds-1.2.0-4.noarch.rpm
           => `sme-ast-en-uk-gpl-sounds-1.2.0-4.noarch.rpm'
Resolving sarkpbx.com... 82.195.146.160
Connecting to sarkpbx.com|82.195.146.160|:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: 9,758,165 (9.3M) [text/plain]

100%[=================================================================================>] 9,758,165     84.86K/s    ETA 00:00

13:52:43 (137.34 KB/s) - `sme-ast-en-uk-gpl-sounds-1.2.0-4.noarch.rpm' saved [9758165/9758165]

[root@sail ~]# yum localinstall sme-ast-en-uk-gpl-sounds-1.2.0-4.noarch.rpm
Loading "protect-packages" plugin
Loading "installonlyn" plugin
Loading "fastestmirror" plugin
Loading "smeserver" plugin
Setting up Local Package Process
Examining sme-ast-en-uk-gpl-sounds-1.2.0-4.noarch.rpm: sme-ast-en-uk-gpl-sounds - 1.2.0-4.noarch
Marking sme-ast-en-uk-gpl-sounds-1.2.0-4.noarch.rpm as an update to sme-ast-en-uk-gpl-sounds - 1.0.0-3.noarch
Resolving Dependencies
--> Populating transaction set with selected packages. Please wait.
---> Package sme-ast-en-uk-gpl-sounds.noarch 0:1.2.0-4 set to be updated
--> Running transaction check
Setting up repositories
smecontribs               100% |=========================| 1.9 kB    00:00     
smeaddons                 100% |=========================| 1.9 kB    00:00     
smeextras                 100% |=========================| 1.9 kB    00:00     
base                      100% |=========================| 1.1 kB    00:00     
updates                   100% |=========================|  951 B    00:00     
smeos                     100% |=========================|  951 B    00:00     
smeupdates                100% |=========================| 1.9 kB    00:00     
Loading mirror speeds from cached hostfile
Reading repository metadata in from local files
Excluding Packages from CentOS - os
Finished
Excluding Packages from CentOS - updates
Finished
Excluding Packages from CentOS - os
Finished
Excluding Packages from CentOS - updates
Finished
--> Processing Dependency: asterisk14 >= 1.4.28 for package: sme-ast-en-uk-gpl-sounds
--> Restarting Dependency Resolution with new changes.
--> Populating transaction set with selected packages. Please wait.
warning: only V3 signatures can be verified, skipping V4 signature
---> Package asterisk14.i386 1:1.4.31-91.el4 set to be updated
--> Running transaction check
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
warning: only V3 signatures can be verified, skipping V4 signature
--> Processing Dependency: libtonezone.so.2.0 for package: asterisk14
--> Processing Dependency: asterisk-moh-opsound-wav for package: asterisk14
--> Processing Dependency: asterisk >= 1.4.7 for package: asterisk-addons
--> Processing Dependency: asterisk-core-sounds-en-gsm for package: asterisk14
--> Processing Dependency: libradiusclient-ng.so.2 for package: asterisk14
--> Processing Dependency: libgsm.so.1 for package: asterisk14
--> Processing Dependency: libiksemel.so.3 for package: asterisk14
--> Restarting Dependency Resolution with new changes.
--> Populating transaction set with selected packages. Please wait.
---> Package asterisk-moh-opsound-wav.noarch 0:2.03-56 set to be updated
---> Package libradiusclient-ng2.i386 0:0.5.6-0.el4 set to be updated
---> Package libtonezone2.i386 1:2.3.0-66.el4 set to be updated
---> Package asterisk-core-sounds-en-gsm.noarch 0:1.4.19-61 set to be updated
---> Package iksemel.i386 0:1.4-1.el4.rf set to be updated
---> Package libgsm1.i386 0:1.0.13-2.el4 set to be updated
--> Running transaction check
--> Processing Dependency: asterisk >= 1.4.7 for package: asterisk-addons
--> Processing Dependency: libgnutls.so.11 for package: iksemel
--> Processing Dependency: libgnutls.so.11(GNUTLS_REL_1_0_9) for package: iksemel
--> Restarting Dependency Resolution with new changes.
--> Populating transaction set with selected packages. Please wait.
---> Package gnutls.i386 0:1.0.20-4.el4_8.7 set to be updated
--> Running transaction check
--> Processing Dependency: asterisk >= 1.4.7 for package: asterisk-addons
--> Finished Dependency Resolution
Error: Missing Dependency: asterisk >= 1.4.7 is needed by package asterisk-addons
[root@sail ~]#

Then I relised that I need the other version which is already installed

Could my problem be because I upgraded to the latest version of SAIL?
Regards
M0GLJ
......................................................
I am new to SAIL SME Server v8b6 and have been using SME for many years.
I have already done some research and only ask questions if I still can't work it out.