Customer has hung up: SIP/SIPOUT-xxxx

[jarubio]

Hi,
I've a problem with vicidial, with all agents at this moment.

when the agent log in (with idefisk IAX), the conference is connected and when the first call is made to the customer, the agent receive a pop-up that say: Customer has hung up: SIP/SIPOUT-d666 , but the call on idesfisk is not hunged up. The agent hung up the call on the idefist interface and then need to re-login to vicidial.

This happen after a power problems, when all the servers were powered off, so maybe this cause some corruption on the DB?

Any idea to debug the vicidial and found the problem?

Thank you in advance

[mflorell]

You would need to make sure your AST_update.pl script is running.

[jarubio]

It is running.

root 22534 1 0 13:05 ? 00:00:00 /usr/bin/SCREEN -d -m -S ASTupdate /home/cron/AST_update.pl
root 22535 22534 2 13:05 pts/8 00:00:01 /usr/bin/perl /home/cron/AST_update.pl

thank you.

[mflorell]

Do you have the same Dial string as your register string in your sip.conf?

what appears in the CALLS IN THIS SESSION link at the bottom of the vicidial.php page when this happens?

[jarubio]

register => asterisk1:1020@10.0.0.92

[SIPOUT]
type=peer
secret=1020
username=asterisk1
host=10.0.0.92
fromuser=asterisk1
disallow=all
allow=g729
context=default

exten => _81XXXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _81XXXXXXXX,2,Dial(SIP/SIPOUT/${EXTEN},55,o)
exten => _81XXXXXXXX,3,hangup

The agents are calling without using vicidial without any problem, so , I think is not an asterisk issue.

There is nothing on the "Show conference call channel information" link at the bottom of the vicidial.php page. I mean there is no showing the IAX or SIP channels.

[mflorell]

For VICIDIAL you need to use the actual register information for dialing not the SIP account name or the Local/ channels will not resolve their names properly.

[jarubio]

do you mean that I need to use asterisk1 instead SIPOUT on extensions.conf ?

regards,

[jarubio]

I used asterisk1 on sip.conf and extensions.conf with the same result. "customer has hung up".

[mflorell]

exten => _81XXXXXXXX,2,Dial(SIP/asterisk1:1020@10.0.0.92/${EXTEN},55,o)

If this does not work please post Real Asterisk CLI output(not asterisk -r) of a call being placed and the transfer agi script running.

[jarubio]

did'nt work.

What do you mean as Real Astersik CLI output and the transfer agi script running? could you please detail me this?

thank you

[mflorell]

run asterisk as "asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvgc"

log in to vicidial.php.

click RESUME

after the call hangs up on the asterisk CLI, post the output from that screen.

[jarubio]

I tried my best, because the servers are on production, it is very difficult to debug one call. This is for a call on the asterisk where the call is originated, I mean a child asterisk server where the sofphone is logged in.

+++++ CALL LOG START : |1155640098.2|Local/8600195@default-49d7,1|8183744423|Local|M0814165828000224827|2006-08-15 6:08:18
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/8600195@default-49d7,1", "SIP/SIPOUT/8183744423|55|o") in new stack
-- Called SIPOUT/8183744423
-- SIP/SIPOUT-0ee0 is ringing
-- SIP/SIPOUT-0ee0 is making progress passing it to Local/8600195@default-49d7,1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/SIPOUT-0ee0 answered Local/8600195@default-49d7,1
-- Registered IAX2 '4248' (AUTHENTICATED) at 10.0.0.45:4569
== Spawn extension (default, 8600195, 1) exited non-zero on 'IAX2/4245-24'
-- Executing DeadAGI("IAX2/4245-24", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+++++ CALL LOG HUNGUP: |1155640092.0|IAX2/4245-24|h|2006-08-15 6:08:37|min: |
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("IAX2/4245-24", "VD_hangup.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
AGI Environment Dump:
-- accountcode =
-- callerid = unknown
-- calleridname = S0608141658228600195
-- callingani2 = 0
-- callingpres = 0
-- callingtns = 0
-- callington = 0
-- channel = IAX2/4245-24
-- context = default
-- dnid = unknown
-- enhanced = 0.0
-- extension = h
-- language = en
-- priority = 2
-- rdnis = unknown
-- request = VD_hangup.agi
-- type = IAX2
-- uniqueid = 1155640092.0
AGI Environment Dump: |1155640092.0|IAX2/4245-24|h|IAX2|S0608141658228600195|S0608141658228600195|2|

DEBUG:

VD_hangup : S0608141658228600195 IAX2/4245-24 2
+++++ VD hangup START : |1155640092.0|IAX2/4245-24|h|IAX2|S0608141658228600195|228600195|2006-08-15 6:08:38||2|S0608141658228600195|


|SELECT lead_id,callerid FROM vicidial_auto_calls where uniqueid = '1155640092.0' limit 1;|

VD hangup: no VDAC record found: 1155640092.0 S0608141658228600195
-- AGI Script VD_hangup.agi completed, returning 0
-- Hungup 'IAX2/4245-24'
== Spawn extension (default, 8183744423, 2) exited non-zero on 'Local/8600195@default-49d7,1'
-- Executing DeadAGI("Local/8600195@default-49d7,1", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+++++ CALL LOG HUNGUP: |1155640098.2|Local/8600195@default-49d7,1|h|2006-08-15 6:08:41|min: 0.38|
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/8600195@default-49d7,1", "VD_hangup.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
AGI Environment Dump:

[jarubio]

I need some help here... do you have any idea to trace the problem? maybe any method to debug vicidial? I can not see anything on the show channels on the vicidial page, so there is any problem with live_channels or another table that is not uptaded or something like that?



Thank you.

[mflorell]

from this output I really don't know how you are trying to dial. Could you give an in-depth explanation of how you are dialing?

It looks like you are still using the account name to dial out, not the registration string.

What dial level?

Can you send more of the Asterisk CLI output?

[jarubio]

Ok, I've this escenario (couriusly working fine the last week).

I've 5 servers with asterisk/vicidial with 20 idefisk softphones. (child)
I've 1 server with Webserver/MySQL/VIcidial/asterisk, this server have the sip trunks with the carrier (I can't have more than one server). (main)

The child servers are connected through a SIP channel to the main server, doing transcoding gsm -> g729 before the call is sent. This channel is named SIPOUT and is registered with a user named asterisk"x". When a call is generated by vicidial (all phone numbers begins with "81") the extensions.conf on the child server send the call to the main server using the SIPOUT channel, and then the main server send the call to the carrier using the SIP Trunk.

This configuration were working without problems the last week. But today after the power problems, I've the hung up customer error on all agents.

I changed one child server to use the registration name on the dialplan, as you suggest me (with not results) , but, I think the call generated on the CLI output that I was sent was on other server with the SIPOUT string.
---------------

I've autodialer off
--------------

I will send more CLI output on the next post.

[jarubio]

CLI output on child server:

-- Call accepted by 10.0.0.153 (format gsm)
-- Format for call is gsm
== Manager 'sendcron' logged off from 127.0.0.1
> Channel IAX2/4245-19 was answered.
-- Executing MeetMe("IAX2/4245-19", "8600191") in new st
ack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600191'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600191@default-aa22,2", "8600191
") in new stack
> Channel Local/8600191@default-aa22,1 was answered.
-- Executing AGI("Local/8600191@default-aa22,1", "call_log.agi|81
83425916") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+++++ CALL LOG START : |1155654913.2|Local/8600191@default-aa22,1|8183425916|Local|M0814210524000224838|2006-08-15 10:15:15
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("Local/8600191@default-aa22,1", "SIP/SIPOUT/818
3425916|55|o") in new stack
-- Called SIPOUT/8183425916
-- SIP/SIPOUT-1deb is ringing
-- SIP/SIPOUT-1deb is making progress passing it to Local/8600191@default-aa22,1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/SIPOUT-1deb answered Local/8600191@default-aa22,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600191, 1) exited non-zero on 'IAX2/4245-19'
-- Executing DeadAGI("IAX2/4245-19", "call_log.agi|h") i
n new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+++++ CALL LOG HUNGUP: |1155654906.0|IAX2/4245-19|h|2006-08-15 10:15:29|min: |
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("IAX2/4245-19", "VD_hangup.agi|h")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
AGI Environment Dump:
-- accountcode =
-- callerid = unknown
-- calleridname = S0608142105168600191
-- callingani2 = 0
-- callingpres = 0
-- callingtns = 0
-- callington = 0
-- channel = IAX2/4245-19
-- context = default
-- dnid = unknown
-- enhanced = 0.0
-- extension = h
-- language = en
-- priority = 2
-- rdnis = unknown
-- request = VD_hangup.agi
-- type = IAX2
-- uniqueid = 1155654906.0
AGI Environment Dump: |1155654906.0|IAX2/4245-19|h|IAX2|S0608142105168600191|S0608142105168600191|2|

DEBUG:

VD_hangup : S0608142105168600191 IAX2/4245-19 2
+++++ VD hangup START : |1155654906.0|IAX2/4245-19|h|IAX2|S0608142105168600191|168600191|2006-08-15 10:15:29||2|S0608142105168600191
|


|SELECT lead_id,callerid FROM vicidial_auto_calls where uniqueid = '1155654906.0' limit 1;|

VD hangup: no VDAC record found: 1155654906.0 S0608142105168600191
-- AGI Script VD_hangup.agi completed, returning 0
-- Hungup 'IAX2/4245-19'
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 8183425916, 2) exited non-zero on 'Local/8600191@default-aa22,1'
-- Executing DeadAGI("Local/8600191@default-aa22,1", "call_log.ag
i|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+++++ CALL LOG HUNGUP: |1155654913.2|Local/8600191@default-aa22,1|h|2006-08-15 10:15:48|min: 0.55|
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/8600191@default-aa22,1", "VD_hangup.a
gi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
AGI Environment Dump:
-- accountcode =
-- callerid = SANTANDER
-- calleridname = M0814210524000224838
-- callingani2 = 0
-- callingpres = 0
-- callingtns = 0
-- callington = 0
-- channel = Local/8600191@default-aa22,1
-- context = default
-- dnid = unknown
-- enhanced = 0.0
-- extension = h
-- language = en
-- priority = 2
-- rdnis = unknown
-- request = VD_hangup.agi
-- type = Local
-- uniqueid = 1155654913.2
AGI Environment Dump: |1155654913.2|Local/8600191@default-aa22,1|h|Local|M0814210524000224838|M0814210524000224838|2|

DEBUG:

VD_hangup : M0814210524000224838 Local/8600191@default-aa22,1 2
+++++ VD hangup START : |1155654913.2|Local/8600191@default-aa22,1|h|Local|M0814210524000224838|224838|2006-08-15 10:15:48||2|M08142
10524000224838|
-- VDhangup Local DEBUG: ||M0814210524000224838|||
+++++ VDAD START LOCAL CHANNEL: EXITING- 2
-- AGI Script VD_hangup.agi completed, returning 0
-- Hungup 'Zap/pseudo-1831249821'
== Spawn extension (default, 8600191, 1) exited non-zero on 'Local/8600191@default-aa22,2'
-- Executing DeadAGI("Local/8600191@default-aa22,2", "call_log.ag
i|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+++++ CALL LOG HUNGUP: |1155654913.3|Local/8600191@default-aa22,2|h|2006-08-15 10:15:49|min: |
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("Local/8600191@default-aa22,2", "VD_hangup.a
gi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
AGI Environment Dump:
-- accountcode =
-- callerid = SANTANDER
-- calleridname = M0814210524000224838
-- callingani2 = 0
-- callingpres = 0
-- callingtns = 0
-- callington = 0
-- channel = Local/8600191@default-aa22,2
-- context = default
-- dnid = unknown
-- enhanced = 0.0
-- extension = h
-- language = en
-- priority = 2
-- rdnis = unknown
-- request = VD_hangup.agi
-- type = Local
-- uniqueid = 1155654913.3
AGI Environment Dump: |1155654913.3|Local/8600191@default-aa22,2|h|Local|M0814210524000224838|M0814210524000224838|2|

DEBUG:

VD_hangup : M0814210524000224838 Local/8600191@default-aa22,2 2
+++++ VD hangup START : |1155654913.3|Local/8600191@default-aa22,2|h|Local|M0814210524000224838|224838|2006-08-15 10:15:49||2|M08142
10524000224838|
-- VDhangup Local DEBUG: ||M0814210524000224838|||
+++++ VDAD START LOCAL CHANNEL: EXITING- 2
-- AGI Script VD_hangup.agi completed, returning 0


---------------------------------------------------------------------------


CLI Output on main server:

-- Executing AGI("SIP/asterisk3-e8c9", "call_log.agi|8183425916
") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+++++ CALL LOG START : |1155607526.0|SIP/asterisk3-e8c9|8183425916|SIP|M0814210524000224838|2006-08-14 21:05:26
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("SIP/asterisk3-e8c9", "SIP/TRKO_MCT_01/81834259
16|55|o") in new stack
-- Called TRKO_MCT_01/8183425916
-- SIP/TRKO_MCT_01-b283 is ringing
-- SIP/TRKO_MCT_01-b283 is making progress passing it to SIP/asterisk3-e8c9
-- SIP/TRKO_MCT_01-b283 answered SIP/asterisk3-e8c9
-- Attempting native bridge of SIP/asterisk3-e8c9 and SIP/TRKO_MCT_01-b283

[mflorell]

What is the reason you are using SIP between servers instead of IAX?

[jarubio]

I using SIP in order to decrease load on the main server, all signalling and transcoding to g729 are on the child server.

Well, I've a good news, I've restarted de AST_update.pl script on the child servers, and now the live_channels are good updated and the calls and vicidial are working toghether without problems.

BUT, , I've other problem on one of the child servers, when I log into vicidial, and the conference rings on the softphones, I received the "no conference available, please try again".

This is the log on the server:

> Channel IAX2/4238-7 was answered.
-- Executing MeetMe("IAX2/4238-7", "8600161") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Playing 'conf-invalid' (language 'en')


I've the 8600161 on the vicidial_conferences with the correct server_ip, either on the extensions.conf and meetme.conf.

thank you for all your help.

[jarubio]

I've not loaded the ztdummy module, I loaded and now everything is working fine. Thank you for your support