Hi
And also when i dial from my extension[SJphone]. It displays the string operational but i could not hear any voice.
Regards
Karthik.
dial where/who/what ? do you have a debug log file fragment for one of these silent calls ?
Hi
Sorry. I misquoted in my last but one post.
Please use this phrase - "But i could not hear the other party voice to whom i am dialling?.
"
After 20 seconds,
The console shows the nobody picked up in 20 seconds. Please refer the below messages.
The message which i receive when i dial from the console
karthik*CLI> dial 0xxxxxxxxx@net4indiaout-inbound
– Executing Dial(“OSS/dsp”, “SIP/xxxxxxxxx@sipprovider-out|30|Ttr”) in new stack
– Called xxxxxxxxx@sipprovider-out
== Console is full duplex
– Nobody picked up in 30000 ms
– Executing Answer(“OSS/dsp”, “”) in new stack
<< Console call has been answered >>
Apr 5 17:16:23 WARNING[8469]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule ‘t’ in context ‘net4indiaout-inbound’
<< Hangup on console >>
Regards
Karthik.A
Hi
I have pasted the log
Apr 5 17:31:42 NOTICE[8559] cdr.c: CDR simple logging enabled.
Apr 5 17:31:52 WARNING[8559] pbx_dundi.c: Unable to look up host 'karthik.a’
Apr 5 17:32:02 WARNING[8559] chan_skinny.c: Unable to get our IP address, Skinny disabled
Apr 5 17:32:44 WARNING[8559] chan_mgcp.c: Unable to get our IP address, MGCP disabled
Apr 5 17:33:16 WARNING[8603] app_setcidname.c: SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead.
Apr 5 17:33:59 WARNING[8603] pbx.c: Timeout, but no rule ‘t’ in context ‘net4indiaout-inbound’
Regards
Karthik
Hi
Please help me out!
Regards
Karthik
Hi
I was able to make call from asterisk console to a pstn provider.
I removed all the options from the dial application
I increased my verbose level
I have a strange problem ,I was able to reach the destination but i the other party could not hear me.
And When i dial from my extension the console replies as " call leg transaction does not exist
The console output which i got when i dialed from extension
– Executing Dial(“SIP/1234-08cb2f20”, “SIP/123456789@sipprovider-out”) in new stack
– Called 123456789@sipprovider-out
– SIP/sipprovider-out-08cb8460 answered SIP/1234-08cb2f20
– Attempting native bridge of SIP/1234-08cb2f20 and SIP/sipprovider-out-08cb8460
– Got SIP response 481 “Call/Transaction Does Not Exist” back from sipprovider
Please help
Regards
Karthik
Hello,
I think it’s a codec problem, try to disallow all codecs and just allow one to test. You can do that in sip.conf in every SIP peer.
Regards!
Daniel 
Hi Dinesh,
It might be the problem with NAT. Try to set the nat option in your general settings NAT=yes.
Thanks,
Suresh