Thanks for the quick response James. Indeed was the provider but due to my
fault. I messed up the VoIP portal configuration and I was getting this weird
behavior in my end. Is all running nice and clean now
Thanks again
Ionel Chila
________________________________
From: James Babiak <ja...@routed.com>
To: astlinux-users@lists.sourceforge.net
Sent: Friday, May 10, 2013 11:41 PM
Subject: Re: [Astlinux-users] Weird Behavior after upgrade
Ionel,
Based on your asterisk log, it looks to me like your call is
resulting in two invites coming in at the exact same time to
asterisk. There are two different SIP channels (00000008 and
00000009). The first one (8) is processed first and starts ringing
DAHDI/3, but the second one fails since DAHDI/3 is now busy. I'm not
sure why enabling/disabling voicemail would result in any
difference, at least as far as your actual call getting through. You
may want to do a SIP trace and see what's going on. It looks like a
possible issue with your provider.
--James
On 05/10/2013 11:19 PM, Ionel Chila wrote:
>
>I get some really weird behavior on my system. This is my extensions.conf for
>the incoming VoIp provider which comes into the "bypass"
>
>
>When I call my home number (2812080888) from my cell, the phone rings in my
>house but my cell goes straight to the voicemail (101). I have to remove the
>Voicemail (101) so it can work. See the errors bellow. I don;t know if this
>is Astlinux 1.1.1 related but it started to happen after the upgrade. Anyone
>can see what's going on?
>
>
>[bypass]
>exten =>
2812080888,1,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?blacklisted,s,1)
>exten => 2812080888,2,DIAL(DAHDI/3)
>;exten => 2812080888,3,Voicemail(101)
>exten => 2812080888,102,Hangup()
>
>
>
>
>
>
>HOME-PBX*CLI>
>HOME-PBX*CLI>
> == Using SIP RTP CoS mark 5
> == Using SIP RTP CoS mark 5
> -- Executing [2812080888@chila:1] GotoIf("SIP/vitel-inbound-00000008",
>"0?blacklisted,2812080888,1") in new stack
> -- Executing [2812080888@chila:2] Playback("SIP/vitel-inbound-00000008",
>"wait-moment") in new stack
> -- Executing [2812080888@chila:1] GotoIf("SIP/vitel-inbound-00000009",
>"0?blacklisted,2812080888,1") in new stack
> -- Executing [2812080888@chila:2] Playback("SIP/vitel-inbound-00000009",
>"wait-moment") in new stack
> -- <SIP/vitel-inbound-00000008> Playing 'wait-moment.ulaw' (language 'en')
> -- <SIP/vitel-inbound-00000009> Playing 'wait-moment.ulaw' (language 'en')
> -- Executing [2812080888@chila:3] WaitExten("SIP/vitel-inbound-00000008",
>"1") in new stack
> -- Executing [2812080888@chila:3] WaitExten("SIP/vitel-inbound-00000009",
>"1") in new stack
> -- Timeout on SIP/vitel-inbound-00000008, continuing...
> -- Executing [2812080888@chila:4] Goto("SIP/vitel-inbound-00000008",
>"bypass,2812080888,1") in new stack
> -- Goto (bypass,2812080888,1)
> -- Executing [2812080888@bypass:1] GotoIf("SIP/vitel-inbound-00000008",
>"0?blacklisted,s,1") in new stack
> -- Executing [2812080888@bypass:2] Dial("SIP/vitel-inbound-00000008",
>"DAHDI/3") in new stack
> -- Called DAHDI/3
> -- DAHDI/3-1 is ringing
> -- Timeout on SIP/vitel-inbound-00000009, continuing...
> -- Executing [2812080888@chila:4] Goto("SIP/vitel-inbound-00000009",
>"bypass,2812080888,1") in new stack
> -- Goto (bypass,2812080888,1)
> -- Executing [2812080888@bypass:1] GotoIf("SIP/vitel-inbound-00000009",
>"0?blacklisted,s,1") in new stack
> -- Executing [2812080888@bypass:2] Dial("SIP/vitel-inbound-00000009",
>"DAHDI/3") in new stack
>
>
>[May 10 22:08:10] WARNING[564][C-0000000b]: app_dial.c:2437 dial_exec_full:
>Unable to create channel of type 'DAHDI' (cause 17 - User busy)
> == Everyone is busy/congested at this time (1:1/0/0)
> -- Auto fallthrough, channel 'SIP/vitel-inbound-00000009' status is 'BUSY'
>
>
> -- DAHDI/3-1 is ringing
>
>
>[May 10 22:08:14] WARNING[497]: chan_sip.c:4169 retrans_pkt: Retransmission
>timeout reached on transmission 1225fc9e4d26ce2b258e71c5541cae06@66.241.96.221
>for seqno 102 (Critical Response) -- See
>https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>Packet timed out after 6400ms with no response
>[May 10 22:08:14] WARNING[497]: chan_sip.c:4198 retrans_pkt: Hanging up call
>1225fc9e4d26ce2b258e71c5541cae06@66.241.96.221 - no reply to our critical
>packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>
>
> -- DAHDI/3-1 is ringing
> -- DAHDI/3-1 is ringing
> -- DAHDI/3-1 is ringing
> -- DAHDI/3-1 answered SIP/vitel-inbound-00000008
> -- Hanging up on 'DAHDI/3-1'
> -- Hungup 'DAHDI/3-1'
> == Spawn extension (bypass, 2812080888, 2) exited non-zero on
>'SIP/vitel-inbound-00000008'
>
>
>Regards
>Ionel Chila
>
>
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