Unable to connect to VOIP provider

[andrew_art9]

Hi,
I Can ping Google, yahoo & everyone. however i am not able to connect to my Voip provider.
[root@vici ~]# asterisk -r
Asterisk 1.2.30.2, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.30.2 currently running on vici (pid = 2528)
Verbosity is at least 21
May 15 19:28:43 NOTICE[2624]: chan_sip.c:5529 sip_reg_timeout: -- Registration for 'andrew@207.182.133.26' timed out, trying again (Attempt #2)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/912039378928@default-0a5a,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/912039378928@default-0a5a,2", "/12039378928||To") in new stack
May 15 19:28:55 WARNING[11529]: channel.c:2621 ast_request: No channel type registered for ''
May 15 19:28:55 NOTICE[11529]: app_dial.c:1076 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Local/912039378928@default-0a5a,2", "") in new stack
== Spawn extension (default, 912039378928, 3) exited non-zero on 'Local/912039378928@default-0a5a,2'
-- Executing DeadAGI("Local/912039378928@default-0a5a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
May 15 19:29:03 NOTICE[2624]: chan_sip.c:5529 sip_reg_timeout: -- Registration for 'andrew@207.182.133.26' timed out, trying again (Attempt #3)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
May 15 19:29:23 NOTICE[2624]: chan_sip.c:5529 sip_reg_timeout: -- Registration for 'andrew@207.182.133.26' timed out, trying again (Attempt #4)

Is thr some Error from My End ? OR my voip provider has not configured MY ID?


Best Regards,
Andrew

[andrew_art9]

Below is the configuration i have done through web interface. I am using reliance internet connection.

register => andrew:XXXXX@207.182.133.26
[SIPtrunk]
disallow=all
allow=G729
type=friend
username=andrew
secret=XXXXXX
host=207.182.133.26
dtmfmode=inband
qualify=1000

SIPtrunk = SIP/SIPtrunk

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIPtrunk/${EXTEN:1}@SIPtrunk,55,o)
exten => _91NXXNXXXXXX,3,Hangup

[andrew_art9]

hello,
I need your help can't figure out whats wrong.




best regards

[codehaxor]

check your firewall, check if iptables and SElinux is active, i think 5060 udp is being blocked or the ip address is down, can you also try pinging the ip address

[andrew_art9]

Thanks for replying

[root@vici ~]# ping 207.182.133.26
PING 207.182.133.26 (207.182.133.26) 56(84) bytes of data.
64 bytes from 207.182.133.26: icmp_seq=1 ttl=111 time=374 ms
64 bytes from 207.182.133.26: icmp_seq=2 ttl=111 time=365 ms
64 bytes from 207.182.133.26: icmp_seq=3 ttl=111 time=372 ms
64 bytes from 207.182.133.26: icmp_seq=4 ttl=111 time=372 ms
64 bytes from 207.182.133.26: icmp_seq=5 ttl=111 time=367 ms
64 bytes from 207.182.133.26: icmp_seq=6 ttl=111 time=372 ms
64 bytes from 207.182.133.26: icmp_seq=7 ttl=111 time=372 ms
64 bytes from 207.182.133.26: icmp_seq=8 ttl=111 time=372 ms
64 bytes from 207.182.133.26: icmp_seq=9 ttl=111 time=371 ms
64 bytes from 207.182.133.26: icmp_seq=10 ttl=111 time=374 ms
64 bytes from 207.182.133.26: icmp_seq=11 ttl=111 time=366 ms
64 bytes from 207.182.133.26: icmp_seq=12 ttl=111 time=366 ms
64 bytes from 207.182.133.26: icmp_seq=13 ttl=111 time=365 ms

--- 207.182.133.26 ping statistics ---
13 packets transmitted, 13 received, 0% packet loss, time 12007ms
rtt min/avg/max/mdev = 365.664/370.338/374.533/3.408 ms


How do i check Iptables & other things

[andrew_art9]

Now i See all this while i am trying to dialout

[root@vici ~]# asterisk -r
Asterisk 1.2.30.2, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.30.2 currently running on vici (pid = 2469)
Verbosity is at least 21
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/cc100-085f4b48 was answered.
-- Executing MeetMe("SIP/cc100-085f4b48", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
May 18 19:07:44 WARNING[25890]: channel.c:2403 set_format: Unable to find a codec translation path from g729 to gsm
May 18 19:07:44 WARNING[25890]: file.c:828 ast_streamfile: Unable to open conf-onlyperson (format g729): No such file or directory
May 18 19:07:44 WARNING[25890]: channel.c:2403 set_format: Unable to find a codec translation path from g729 to slin
May 18 19:07:44 WARNING[25890]: app_meetme.c:1014 conf_run: Unable to set 'SIP/cc100-085f4b48' to write linear mode
-- Hungup 'Zap/pseudo-406827271'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-085f4b48'
-- Executing DeadAGI("SIP/cc100-085f4b48", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/cc100-085f4b48 was answered.
-- Executing MeetMe("SIP/cc100-085f4b48", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
May 18 19:08:04 WARNING[25986]: channel.c:2403 set_format: Unable to find a codec translation path from g729 to gsm
May 18 19:08:04 WARNING[25986]: file.c:828 ast_streamfile: Unable to open conf-onlyperson (format g729): No such file or directory
May 18 19:08:04 WARNING[25986]: channel.c:2403 set_format: Unable to find a codec translation path from g729 to slin
May 18 19:08:04 WARNING[25986]: app_meetme.c:1014 conf_run: Unable to set 'SIP/cc100-085f4b48' to write linear mode
-- Hungup 'Zap/pseudo-1253674958'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-085f4b48'
-- Executing DeadAGI("SIP/cc100-085f4b48", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== 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/8600051@default-5c01,2", "8600051|F") in new stack
> Channel Local/8600051@default-5c01,1 was answered.
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Executing AGI("Local/8600051@default-5c01,1", "agi://127.0.0.1:4577/call_log") in new stack
-- Playing 'conf-onlyperson' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-5c01,1", "/12032653545||To") in new stack
May 18 19:08:07 WARNING[26013]: channel.c:2621 ast_request: No channel type registered for ''
May 18 19:08:07 NOTICE[26013]: app_dial.c:1076 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Local/8600051@default-5c01,1", "") in new stack
== Spawn extension (default, 912032653545, 3) exited non-zero on 'Local/8600051@default-5c01,1'
-- Executing DeadAGI("Local/8600051@default-5c01,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Hungup 'Zap/pseudo-745516207'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-5c01,2'
-- Executing DeadAGI("Local/8600051@default-5c01,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1


Please help!!!!!!!!!!!!!

[andrew_art9]

This is how i have configured through Web interface

register => andrew:XXXXX@207.182.133.26:5060

[SIPtrunk]
disallow=all
allow=G729
type=friend
username=andrew
secret=XXXXXXX
host=207.182.133.26:5060
dtmfmode=inband
qualify=1000

SIPtrunk = SIP/SIPtrunk

exten => _9NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXXXX,2,Dial(SIPtrunk/${EXTEN:1}@SIPtrunk,55,o)
exten => _9NXXNXXXXXX,3,Hangup

[andrew_art9]

Help!!!!!!!!!!!!!!!!!!!!!!!!!

[andrew_art9]

I am Still loooking for Help Can't find any help elsewhere

[Op3r]

do you have g729 codec?

If not go get it at www.digium.com

[triplef]

make sure codec is ulaw, and you use fromuser

[williamconley]

allow=all will let asterisk negotiate to whatever codec you and your provider have in common. Asterisk will automatically compare yours and theirs and choose one you both have. Likely ulaw, but this will completely remove the restriction and you can work it out later (after you've made some calls).

[gzpxyj]

In most cases, g729 codec is not installed for the asterisk. You have to purchase the codec in order to use it. So the best way to start up is to use the ulaw or alaw. Your voip provider generally will provide you with the ulaw. So write some like that:
disallow=all
allow=ulaw
allow=gsm
...
That should get you working. After that, you can search for g729 codec and install it, then change it to allow=g729
Jim

[andrew_art9]

Thanks for repling, i will update if its connecting

[andrew_art9]

I have installed G729 Codec now i am able to connect with service provider however voice transmission is not happening. One side audio. i tried manually & it works (bypassed dialer).



Any recommendation would be helpfull

[williamconley]

start here: http://lmgtfy.com/?q=vicidial+"one%20way%20audio"

how exactly did you try "manually"? did you use asterisk on the same box but not vicidial? did you use a soft phone on another computer? did you use the same protocol and/or codec?

there are a lot of problems that can cause one-way audio, but most have to do with NAT. All have been discussed quite a bit.