Strange problem with Telasip DID


#1

I have configured telasip DID with following entried in sip_custom.conf

[telasip]
username=xxxx (fake)
type=peer
secret=xxxxx
quality=yes
nat=yes
insecure=very
fromuser=xxxx
host=gw4.telasip.com
#disallow=all
#allow=ulaw
#allow=alaw
fromdomain=gw4.telasip.com
context=from-telasip

and Register string in sip.conf under general and extensions.conf has following entries

[from-telasip]
exten => 1134817097,1,Answer
exten => 1134817097,2,Wait(1)
exten => 1134817097,3,Background(pls-hold-while-try)
exten => 1134817097,4,NoOp(Incoming call for Suzie on TelaSIP #8431234567)
exten => 1134817097,5,Dial(SIP/71469,20,m)
exten => 1134817097,6,VoiceMail(71469@default)
exten => 1134817097,7,Hangup

The problem is I can receive one incoming call to this DID successfully. Then I tried to call this DID, it say it is not avaiable. SO in Asterisk CLI I type reload to reload Asterisk. Then incoming call works again, then next one is not, then reload, it works, so and so. What could be the problem? Please help.


#2

quality=yes

First, this should read
qualiFy=yes


#3

Thanks for your reply. I changed it, but the problem is still the same.


#4

I have configured telasip DID with following entried in sip_custom.conf

and the sip_custom.inf is included in sip.conf via
#include=sip_custom.conf

,with the leading pound ?


#5

Thanks again for your Reply. Yes, the sip_custom.inf is included in sip.conf via #include=sip_custom.conf.


#6

Can you reload asterisk, then do a “SIP SHOW PEERS” at the CLI to check, if the peer is online.

Then do a call.

After that, do another SIP SHOW PEERS (without reloading before).
What does it read now ?


#7

Thanks for your reply. Below is the output. The first call is sucessfull. After we hung up and make second call to this DID # , it say not available.

*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
telasip/xli 4.79.19.59 N 5060 OK (41 ms)
1 sip peers [1 online , 0 offline]
*CLI> – SIP Seeding peer from astdb: ‘71467’ at 71467@192.168.1.100:5060 for 60

*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
71467/71467 192.168.1.100 D N 5060 Unmonitored
telasip/xli 4.79.19.59 N 5060 OK (28 ms)
2 sip peers [2 online , 0 offline]
– Executing Answer(“SIP/xli-2da8”, “”) in new stack
– Executing Wait(“SIP/xli-2da8”, “1”) in new stack
– Executing BackGround(“SIP/xli-2da8”, “pls-hold-while-try”) in new stack
– Playing ‘pls-hold-while-try’ (language ‘en’)
– Executing NoOp(“SIP/xli-2da8”, “Incoming call for Suzie on TelaSIP #8431234567”) in new stack
– Executing Dial(“SIP/xli-2da8”, “SIP/71469|20|m”) in new stack
– SIP Seeding peer from astdb: ‘71469’ at 71469@83.221.214.90:5060 for 1800
– Called 71469
– Started music on hold, class ‘default’, on channel ‘SIP/xli-2da8’
– SIP/71469-e48d is ringing
– Nobody picked up in 20000 ms
– Stopped music on hold on SIP/xli-2da8
– Executing VoiceMail(“SIP/xli-2da8”, “71469@default”) in new stack
uniqueid => 71469
customer_id => 71469
context => default
mailbox => 71469
password => 2882
fullname => roman
email => romanvaluev@gmail.com
attach => yes
saycid => yes
hidefromdir => no
– Playing ‘vm-intro’ (language ‘en’)
– Playing ‘beep’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/71469/INBOX/msg0004 format: wav49, 0x9029600
– x=1, open writing: /var/spool/asterisk/voicemail/default/71469/INBOX/msg0004 format: gsm, 0x9029900
– x=2, open writing: /var/spool/asterisk/voicemail/default/71469/INBOX/msg0004 format: wav, 0x9022180
– User hung up
== Spawn extension (from-telasip, 7134817097, 6) exited non-zero on ‘SIP/xli-2da8’

*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
71469/71469 83.221.214.90 D N 5060 Unmonitored
xli/xli 4.79.19.59 5060 Unmonitored
71467/71467 192.168.1.100 D N 5060 Unmonitored
telasip/xli 4.79.19.59 N 5060 OK (28 ms)
4 sip peers [4 online , 0 offline]
*CLI>


#8

the #include will not do anything - in Asterisk config processing, the ‘#’ denotes a comment, so you’re telling it to ignore that line.


#9

No, you need # in front sip_coustom.conf so that Asterisk can include sip_custom.conf in sip.conf. I tested it without #, it won’t work either.


#10

sorry, i was thinking of something else :frowning:


#11

Thanks a lot anyway.


#12

is the most recent log session from a good call or a bad one? regardless, can you post examples from both?


#13

When the second is made, i still see it ringing and timing out, assuming your log is from the second call (cuz the voicemail was fired up).

You can see that …67 (or …69) is ringing but running into timeout.
So i guess its “ghostringing”: The phone itself isnt ringing after the first successful call, correct ?

To me, it looks like a problem with the astdb.
The seeding (CLI output) of these extension where AFTER the first call took place, or is the seeding line also there, when you made the first call ?

Thats why we need both CLI outputs, from both calls.

Also, please try to reach the “unreachable” extension (when it happens) iternally from the other extension. Is THIS working at that moment ?

Then, raise the timoutvalue to 60s please and let it ring till judgementday…is the call ringing then, but very very delayed ?

Also, change passwords please. Your PBX is reachable from the outside and your passwords where posted. So everyone could log into your PBX as …67/69 now.