[asterisk-users] Call not going through and failing because "never answered"
Andy Beak
andrewb at cellsmart.co.za
Tue Jul 20 22:51:22 CDT 2010
Hi Gareth,
Thanks for replying. Here is the SIP debug from the CLI. I assume that
the first two blocks are from having qualify=yes and the remaining are
from attempting to place a call.
Do you know what "SIP/2.0 480 No Routes Found" means? It looks like the
SIP provider cannot find my box.
Thanks,
Andy
CST4*CLI> sip set debug on
SIP Debugging enabled
Reliably Transmitting (no NAT) to 192.168.34.1:5060:
OPTIONS sip:192.168.34.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK7a19a314;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 192.168.0.14>;tag=as613ee548
To: <sip:192.168.34.1>
Contact: <sip:asterisk at 192.168.0.14>
Call-ID: 46609c1028646806347073e155e9107d at 192.168.0.14
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1
Date: Wed, 21 Jul 2010 03:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
CST4*CLI>
<--- SIP read from UDP:192.168.34.1:5060 --->
SIP/2.0 480 No Routes Found
Via: SIP/2.0/UDP
172.28.20.106:60017;received=172.28.20.106;branch=z9hG4bK7a19a314;rport=60017
From: "asterisk" <sip:asterisk at 192.168.0.14>;tag=as613ee548
To: <sip:192.168.34.1>;tag=aprqngfrt-6c6p2t00000c6
Call-ID: 46609c1028646806347073e155e9107d at 192.168.0.14
CSeq: 102 OPTIONS
<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog
'46609c1028646806347073e155e9107d at 192.168.0.14' Method: OPTIONS
-- Remote UNIX connection
-- Attempting call on SIP/MTN-NEW/084xxxyyyy for application
MP3Player(/var/www/andy/calls/foschini.mp3) (Retry 1)
== Using SIP RTP CoS mark 5
Audio is at 192.168.0.14 port 12350
Adding codec 0x2 (gsm) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.34.1:5060:
INVITE sip:084xxxyyyy at 192.168.34.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK305454ce;rport
Max-Forwards: 70
From: "Andy" <sip:084xxxyyyy at 192.168.0.14>;tag=as0140b1e4
To: <sip:084xxxyyyy at 192.168.34.1>
Contact: <sip:084xxxyyyy at 192.168.0.14>
Call-ID: 4df23ce8116c0229281fcdf5376afc51 at 192.168.0.14
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1
Date: Wed, 21 Jul 2010 03:47:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 386
v=0
o=root 1528469633 1528469633 IN IP4 192.168.0.14
s=Asterisk PBX 1.6.2.5-0ubuntu1
c=IN IP4 192.168.0.14
t=0 0
m=audio 12350 RTP/AVP 3 4 0 18 101
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
CST4*CLI>
<--- SIP read from UDP:192.168.34.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.28.20.106:60017;received=172.28.20.106;branch=z9hG4bK305454ce;rport=60017
From: "Andy" <sip:084xxxyyyy at 192.168.0.14>;tag=as0140b1e4
To: <sip:084xxxyyyy at 192.168.34.1>
Call-ID: 4df23ce8116c0229281fcdf5376afc51 at 192.168.0.14
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
CST4*CLI>
<--- SIP read from UDP:192.168.34.1:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP
192.168.0.14;received=172.28.20.106;branch=z9hG4bK305454ce;rport=60017
From: "Andy" <sip:084xxxyyyy at 192.168.0.14>;tag=as0140b1e4
To: <sip:084xxxyyyy at 192.168.34.1>;tag=SD7ojh898-4C466D62-548CB9C-0ADE2C09
Call-ID: 4df23ce8116c0229281fcdf5376afc51 at 192.168.0.14
CSeq: 102 INVITE
Reason: Q.850 ;cause=127 ;text="Interworking, unspecified"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.34.1:5060:
ACK sip:084xxxyyyy at 192.168.34.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK305454ce;rport
Max-Forwards: 70
From: "Andy" <sip:084xxxyyyy at 192.168.0.14>;tag=as0140b1e4
To: <sip:084xxxyyyy at 192.168.34.1>;tag=SD7ojh898-4C466D62-548CB9C-0ADE2C09
Contact: <sip:084xxxyyyy at 192.168.0.14>
Call-ID: 4df23ce8116c0229281fcdf5376afc51 at 192.168.0.14
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1
Content-Length: 0
---
> Channel SIP/MTN-NEW-00000003 was never answered.
[Jul 21 05:47:51] NOTICE[30442]: pbx_spool.c:339 attempt_thread: Call
failed to go through, reason (8) Congestion (circuits busy)
Really destroying SIP dialog
'4df23ce8116c0229281fcdf5376afc51 at 192.168.0.14' Method: INVITE
-- Remote UNIX connection disconnected
CST4*CLI>
On 20/07/2010 06:24 PM, Gareth Blades wrote:
> Posting a sip debug will probably be helpfull aswell as you can see
> exactly where the traffic is being sent and what the response was.
>
>
> Andy Beak wrote:
>
>> Hi,
>>
>> Thanks, I added that. I'll ask my network provider if they received
>> these message tomorrow morning. That will narrow things down to either
>> an Asterisk configuration or a network routing issue.
>>
>> There is not really a caller, I'm trying to use Asterisk as an Automated
>> Voice Message server to dial phone numbers and play an mp3.
>>
>> I'm using my mobile phone to test on and it doesn't ring. Asterisk
>> gives the following message immediately after reading the .call file
>> from the spool directory:
>>
>> -- Attempting call on SIP/MTN-NEW/mynumber for application
>> MP3Player(/myfile) (Retry 1)
>> == Using SIP RTP CoS mark 5
>> > Channel SIP/MTN-NEW-00000001 was never answered.
>> [Jul 20 18:07:37] NOTICE[22259]: pbx_spool.c:339 attempt_thread: Call
>> failed to go through, reason (8) Congestion (circuits busy)
>>
>> Because the phone doesn't ring and the error message appears immediately
>> I don't think it's a timeout issue.
>>
>> Will reading the source for pbx_spool.c at line 339 give any clues as to
>> what's happening or will that be a waste of time?
>>
>> Cheers,
>> Andy
>>
>>
>> On 20/07/2010 05:42 PM, Gareth Blades wrote:
>>
>>> If you add qualify=yes to the setting in sip.conf it will send a sip
>>> message to the peer every 60 seconds to check if it is alive.
>>> If you try to make a call while the peer is not alive it will fail
>>> immediatly rather than the caller hearing silence while your box waits
>>> for a reply timeout.
>>>
>>> Andy Beak wrote:
>>>
>>>
>>>> Hi,
>>>>
>>>> No that is the correct address. I know it is an internal IP.
>>>>
>>>> We have our machine hosted in racks at our SIP providers data center.
>>>>
>>>> They've patched a new port to our cabinet and linked that to a gateway
>>>> (172.28.20.105).
>>>>
>>>> As long as we use that gateway (and the IP address they assigned to us)
>>>> our traffic will reach their SBC.
>>>>
>>>> I've confirmed that traceroute follows the path it is supposed to:
>>>>
>>>> traceroute to 192.168.34.1 (192.168.34.1), 30 hops max, 60 byte packets
>>>> 1 192.168.0.1 (192.168.0.1) 0.656 ms 0.562 ms 0.501 ms
>>>> 2 172.28.20.105 (172.28.20.105) 1.211 ms 1.209 ms 1.196 ms
>>>> 3 192.168.34.5 (192.168.34.5) 23.270 ms 23.269 ms 23.328 ms
>>>> 4 * * *
>>>> 5 * * *
>>>> 6 * * *^C
>>>>
>>>> Is there a way to test in Asterisk if it is able to reach a particular
>>>> IP address? Do you think that there is a routing problem here?
>>>>
>>>> Thanks,
>>>> Andy
>>>>
>>>>
>>>>
>>>>
>>>> On 20/07/2010 04:58 PM, Zeeshan Zakaria wrote:
>>>>
>>>>
>>>>> This "host=192.168.34.1" is where you'll put your provider's IP
>>>>> address. Currently you are using some local address which is not your
>>>>> provider's IP address. Where did you get it from? Call your providrr
>>>>> and ask them the IP address of the server where you'll be sending your
>>>>> calls.
>>>>>
>>>>> Zeeshan A Zakaria
>>>>>
>>>>> --
>>>>> www.ilovetovoip.com<http://www.ilovetovoip.com>
>>>>>
>>>>>
>>>>>
>>>>>> On 2010-07-20 10:27 AM, "Andy Beak"<andrewb at cellsmart.co.za
>>>>>> <mailto:andrewb at cellsmart.co.za>> wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I set my list to subscribe to digest and I can't see how to reply to
>>>>>> your reply without starting a new thread.
>>>>>>
>>>>>> There is no need for SIP username and password because the provider
>>>>>> authenticates me on my IP address.
>>>>>>
>>>>>> I thought that "host=192.168.34.1" would be the sip provider IP
>>>>>> address.
>>>>>>
>>>>>> At this point I don't need to accept incoming calls or place
>>>>>> VOIP-to-VOIP. All I need to do is connect to their PBX to place a
>>>>>> call to a cellphone.
>>>>>>
>>>>>> I reread all the documentation I could find and couldn't see where
>>>>>> else in sip.conf I should set the provider IP.
>>>>>>
>>>>>> Thanks for your reply,
>>>>>> Andy
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>> In your sip.conf, there is no mention of your sip provider's IP
>>>>>>>
>>>>>>>
>>>>>> address, username and secret (pa...
>>>>>>
>>>>>> www.ilovetovoip.com<http://www.ilovetovoip.com>
>>>>>> <http://www.ilovetovoip.com>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>> On 2010-07-20 5:09 AM, "Andy Beak"<andrewb at xxxxxxxxxxxxxxx
>>>>>>>
>>>>>>>
>>>>>> <mailto:andrewb at xxxxxxxxxxxxxxx<mailto:andrewb at xxxxxxxxxxxxxxx>>>
>>>>>> wr...
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
>>>>>> -- Bandwidth and Colocation Provided byhttp://www.api-digital.com
>>>>>> <http://www.api-digital.com> --
>>>>>>
>>>>>>
>>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>>> http://www.aste...
>>>>>>
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>>> http://www.asterisk.org/hello
>>>>>>
>>>>>> asterisk-users mailing list
>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>>
>>>>>>
>>>>
>>>>
>>>
>>>
>>
>
>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/pkcs7-signature
Size: 5606 bytes
Desc: S/MIME Cryptographic Signature
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100721/df50828d/attachment.bin
More information about the asterisk-users
mailing list