Bug#433779: asterisk doesn't work correctly after boot
Hasse Hagen Johansen
hasse-debianbug at hagenjohansen.dk
Mon Aug 27 16:47:26 UTC 2007
>>>>> "Faidon" == Faidon Liambotis <paravoid at debian.org> writes:
Faidon> Hello, Thanks for all the information you provided, you've
Faidon> been very helpful.
Thank you too, I am grateful for your help and Tzafrir's help also
Faidon> On an unrelated matter, it saddens me that you decide to
Faidon> not use our packages. Can you pinpoint us on the fixes
Faidon> that Danish CID wants (a bug report on bugs.digium.com
Faidon> would be the best). We have patched Asterisk for such
Faidon> issues in the past (UK CID in particular), so there is
Faidon> precedent.
Yes. I would actually also very much like to use the existing
packages(It is a lot easier to manage packages, and I am running this
on a soekris box so compiling is _very_ painful)
The info I have seen is in bug 9 at
bugs.asterisk.org(http://bugs.digium.com/view.php?id=9) In that there
is also a link to person describing how danish CID works. I remember
something about being very speciel. It uses DTMF to send the number
but before the actual ring signal is send, so I wildcard FXO which I
am using should actually always listen to the phoneline,because there
is no signaling that would tell it to listen(like..hey here comes the
CID)
Maybe I am wrong and the 1.2 asterisk also have been patched, but I
haven't found something indicating this. Actually both zaptel &
asterisk has to be patched
Faidon> Hrm, that's weird. May be it is not registering properly?
>> The problem is that when the nameserver is not available:
>> 3. But I would still get a failure tone from my upstream
>> provider, and my local asterisk doesn't even "see anything"
>> from upstream like it does when all is working...I don't get
>> anything in the console at this point. I have restart or
>> restart asterisk for it to work
Faidon> Could you try doing the opposite at that step? i.e. try
Faidon> making an outbound call. It will probably work, even if
Faidon> the registration has failed, since most SIP providers
Faidon> don't require you to REGISTER before making calls (and
Faidon> INVITEs are authenticated).
Ok. I have just tried the calling out at that time. It doesn't work
either. I wouldn't expect it to if it not registered properly, because
musimi.dk is charging money when doing sip->pstn so they have to know
who to charge. I get this in the console with verbose set to 10
-- Executing Dial("SIP/1001-08188c80", "Sip/musimi/my-pstn-number|15") in new stack
Aug 27 18:00:57 WARNING[5347]: chan_sip.c:1991 create_addr: No such host: musimi
Aug 27 18:00:57 NOTICE[5347]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion("SIP/1001-08188c80", "") in new stack
== Spawn extension (internal, 35380070, 2) exited non-zero on 'SIP/1001-08188c80'
Aug 27 18:00:57 WARNING[2639]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8129e68', 10 retries!
Aug 27 18:00:57 ERROR[2657]: chan_sip.c:11408 sipsock_read: We could NOT get the channel lock for SIP/1001-08188c80!
Aug 27 18:00:57 ERROR[2657]: chan_sip.c:11409 sipsock_read: SIP MESSAGE JUST IGNORED: ACK
Aug 27 18:00:57 ERROR[2657]: chan_sip.c:11410 sipsock_read: BAD! BAD! BAD!
and sip show registry still looks fine:
deckard*CLI> sip show registry
Host Username Refresh State
musimi.dk:5060 my-username 285 Registered
Faidon> Could you perform something a bit harder for me, please?
Faidon> Shutdown your local DNS server and start Asterisk. Type
Faidon> "set verbose 10" and "sip set debug" on your Asterisk
Faidon> console. Wait a bit and then start your local DNS server.
Faidon> Then wait until the register timers hit and Asterisk
Faidon> registers to your SIP provider.
Ok. I did it, and a little more. I then rebooted my sip phone to
reregistrate with asterisk and tried place an outbound call. What it
seems like is that asterisk doesn't authenticate properly or
something?(and please ignore all the warnings from the modules I am
not using)
1001 is my local sip phone connecting to my local asterisk
<my real pstn number> is aphone hooked up to a "normal" pstn line
my-user is my username/phonenumber at my sip provider which also a
pstn number which my upstream musimi.dk is gateway'ing between
pstn/sip
Hope it helps
deckard:~# asterisk -c
Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster at 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.
=========================================================================
[ Booting...Aug 27 18:19:51 WARNING[7725]: manager.c:1714 init_manager: Invalid address '127.0.0.1,10.0.0.1' specified, using 0.0.0.0
Aug 27 18:19:51 NOTICE[7725]: cdr.c:1192 do_reload: CDR simple logging enabled.
.......Aug 27 18:19:52 NOTICE[7725]: config.c:863 ast_config_engine_register: Registered Config Engine odbc
.Aug 27 18:19:52 NOTICE[7725]: res_odbc.c:265 load_odbc_config: Adding ENV var: INFORMIXSERVER=my_special_database
Aug 27 18:19:52 NOTICE[7725]: res_odbc.c:265 load_odbc_config: Adding ENV var: INFORMIXDIR=/opt/informix
Aug 27 18:19:52 NOTICE[7725]: res_odbc.c:294 load_odbc_config: registered database handle 'asterisk' dsn->[asterisk]
Aug 27 18:19:52 NOTICE[7725]: res_odbc.c:554 odbc_obj_connect: Connecting asterisk
Aug 27 18:19:52 WARNING[7725]: res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified
Aug 27 18:19:52 NOTICE[7725]: res_odbc.c:599 load_module: res_odbc loaded.
.........Aug 27 18:19:52 WARNING[7725]: acl.c:244 ast_get_ip_or_srv: Unable to lookup 'musimi.dk'
......Aug 27 18:19:52 WARNING[7734]: chan_sip.c:1991 create_addr: No such host: musimi.dk
Aug 27 18:19:52 WARNING[7734]: chan_sip.c:5512 transmit_register: Probably a DNS error for registration to my-user at musimi.dk, trying REGISTER again (after 20 seconds)
..............................................................................................................Aug 27 18:19:56 WARNING[7725]: cdr_sqlite3_custom.c:124 load_config: cdr_sqlite3_custom: Failed to load configuration file. Module not activated.
Aug 27 18:19:56 WARNING[7725]: cdr_sqlite3_custom.c:216 load_module: cdr_sqlite3_custom: near "(": syntax error.
..... ]
Asterisk Ready.
*CLI> set verbose 10
Verbosity was 0 and is now 10
*CLI> sip debug
SIP Debugging enabled
*CLI> Aug 27 18:20:12 NOTICE[7734]: chan_sip.c:5429 sip_reg_timeout: -- Registration for 'my-user at musimi.dk' timed out, trying again (Attempt #1)
Aug 27 18:20:12 WARNING[7734]: chan_sip.c:1991 create_addr: No such host: musimi.dk
Destroying call '2903aa046831c3b422dcead831ff7b9f at 127.0.0.1'
Aug 27 18:20:12 WARNING[7734]: chan_sip.c:5512 transmit_register: Probably a DNS error for registration to my-user at musimi.dk, trying REGISTER again (after 20 seconds)
Aug 27 18:20:32 NOTICE[7734]: chan_sip.c:5429 sip_reg_timeout: -- Registration for 'my-user at musimi.dk' timed out, trying again (Attempt #2)
-- parse_srv: SRV mapped to host sip.musimi.dk, port 5060
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 87.54.25.114:5060:
REGISTER sip:musimi.dk SIP/2.0
Via: SIP/2.0/UDP 90.184.6.185:5060;branch=z9hG4bK34a9dbfb;rport
From: <sip:my-user at musimi.dk>;tag=as5be72fb8
To: <sip:my-user at musimi.dk>
Call-ID: 2903aa046831c3b422dcead831ff7b9f at 127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:my-user at my-internet-ip-address>
Event: registration
Content-Length: 0
---
<-- SIP read from 87.54.25.114:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 90.184.6.185:5060;branch=z9hG4bK34a9dbfb;rport=5060
From: <sip:my-user at musimi.dk>;tag=as5be72fb8
To: <sip:my-user at musimi.dk>;tag=71f7ae5f309317ddcbc68bbdd2fee19f.e47e
Call-ID: 2903aa046831c3b422dcead831ff7b9f at 127.0.0.1
CSeq: 102 REGISTER
WWW-Authenticate: Digest realm="musimi.dk", nonce="46d2fafda6bad0e39bf3cb9a5e6e29415d2d89e9"
Content-Length: 0
--- (8 headers 0 lines) ---
Responding to challenge, registration to domain/host name musimi.dk
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 87.54.25.114:5060:
REGISTER sip:musimi.dk SIP/2.0
Via: SIP/2.0/UDP my-internet-ip-address:5060;branch=z9hG4bK11b4540b;rport
From: <sip:my-user at musimi.dk>;tag=as5b5dcfed
To: <sip:my-user at musimi.dk>
Call-ID: 2903aa046831c3b422dcead831ff7b9f at 127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="my-user", realm="musimi.dk", algorithm=MD5, uri="sip:musimi.dk", nonce="46d2fafda6bad0e39bf3cb9a5e6e29415d2d89e9", response="8b11e2a350e15dff67eb5e7e41b00432", opaque=""
Expires: 120
Contact: <sip:my-user at my-internet-ip-address>
Event: registration
Content-Length: 0
---
<-- SIP read from 87.54.25.114:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP my-internet-ip-address:5060;branch=z9hG4bK11b4540b;rport=5060
From: <sip:my-user at musimi.dk>;tag=as5b5dcfed
To: <sip:my-user at musimi.dk>;tag=71f7ae5f309317ddcbc68bbdd2fee19f.b377
Call-ID: 2903aa046831c3b422dcead831ff7b9f at 127.0.0.1
CSeq: 103 REGISTER
Contact: <sip:my-user at my-internet-ip-address>;expires=300
Content-Length: 0
--- (8 headers 0 lines) ---
Scheduling destruction of call '2903aa046831c3b422dcead831ff7b9f at 127.0.0.1' in 32000 ms
Aug 27 18:20:33 NOTICE[7734]: chan_sip.c:9895 handle_response_register: Outbound Registration: Expiry for musimi.dk is 300 sec (Scheduling reregistration in 285 s)
<-- SIP read from 10.0.0.15:5060:
REGISTER sip:10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK3c1260a09058c5
From: <sip:1001 at 10.0.0.1;user=phone>;tag=BA5E2FDA40E55DB9DDF
To: <sip:1001 at 10.0.0.1;user=phone>
Call-ID: 20163-1B11-1120-3A62-1E52E575BE99 at 10.0.0.15
CSeq: 26 REGISTER
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Contact: <sip:1001 at 10.0.0.15:5060;transport=udp>
Expires: 0
Content-Length: 0
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.0.0.15 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK3c1260a09058c5;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=BA5E2FDA40E55DB9DDF
To: <sip:1001 at 10.0.0.1;user=phone>
Call-ID: 20163-1B11-1120-3A62-1E52E575BE99 at 10.0.0.15
CSeq: 26 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1001 at 10.0.0.1>
Content-Length: 0
---
Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK3c1260a09058c5;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=BA5E2FDA40E55DB9DDF
To: <sip:1001 at 10.0.0.1;user=phone>;tag=as7ed7f71b
Call-ID: 20163-1B11-1120-3A62-1E52E575BE99 at 10.0.0.15
CSeq: 26 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d28e4ea"
Content-Length: 0
---
Scheduling destruction of call '20163-1B11-1120-3A62-1E52E575BE99 at 10.0.0.15' in 15000 ms
Destroying call '20163-1B11-1120-3A62-1E52E575BE99 at 10.0.0.15'
Destroying call '2903aa046831c3b422dcead831ff7b9f at 127.0.0.1'
<-- SIP read from 10.0.0.15:5060:
REGISTER sip:10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK3815d44cb2685d
From: <sip:1001 at 10.0.0.1;user=phone>;tag=A233B927A3517FEAB1
To: <sip:1001 at 10.0.0.1;user=phone>
Call-ID: 19104-1B11-1120-4CF7-E0E25442529F at 10.0.0.15
CSeq: 1 REGISTER
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Contact: <sip:1001 at 10.0.0.15:5060;transport=udp>
Expires: 300
Content-Length: 0
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.0.0.15 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK3815d44cb2685d;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=A233B927A3517FEAB1
To: <sip:1001 at 10.0.0.1;user=phone>
Call-ID: 19104-1B11-1120-4CF7-E0E25442529F at 10.0.0.15
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1001 at 10.0.0.1>
Content-Length: 0
---
Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK3815d44cb2685d;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=A233B927A3517FEAB1
To: <sip:1001 at 10.0.0.1;user=phone>;tag=as61b1a764
Call-ID: 19104-1B11-1120-4CF7-E0E25442529F at 10.0.0.15
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6802ec8e"
Content-Length: 0
---
Scheduling destruction of call '19104-1B11-1120-4CF7-E0E25442529F at 10.0.0.15' in 15000 ms
<-- SIP read from 10.0.0.15:5060:
REGISTER sip:10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK905f5463325456
From: <sip:1001 at 10.0.0.1;user=phone>;tag=72EA942057922D52323
To: <sip:1001 at 10.0.0.1;user=phone>
Call-ID: 19104-1B11-1120-4CF7-E0E25442529F at 10.0.0.15
CSeq: 2 REGISTER
Authorization: DIGEST username="1001",realm="asterisk",nonce="6802ec8e",uri="sip:10.0.0.1",algorithm=MD5,response="52ffe99679063fac830e3431a68fe1a8"
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Contact: <sip:1001 at 10.0.0.15:5060;transport=udp>
Expires: 300
Content-Length: 0
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.0.0.15 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK905f5463325456;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=72EA942057922D52323
To: <sip:1001 at 10.0.0.1;user=phone>
Call-ID: 19104-1B11-1120-4CF7-E0E25442529F at 10.0.0.15
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1001 at 10.0.0.1>
Content-Length: 0
---
-- Saved useragent "ZyXEL P2000W VoIP Wi-Fi Phone" for peer 1001
Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK905f5463325456;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=72EA942057922D52323
To: <sip:1001 at 10.0.0.1;user=phone>;tag=as61b1a764
Call-ID: 19104-1B11-1120-4CF7-E0E25442529F at 10.0.0.15
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 300
Contact: <sip:1001 at 10.0.0.15:5060;transport=udp>;expires=300
Date: Mon, 27 Aug 2007 16:21:18 GMT
Content-Length: 0
---
Scheduling destruction of call '19104-1B11-1120-4CF7-E0E25442529F at 10.0.0.15' in 15000 ms
<-- SIP read from 10.0.0.15:5060:
INVITE sip:<my real pstn number>@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK464be59b7ddd49
From: <sip:1001 at 10.0.0.1;user=phone>;tag=87462097DB13672E83A9
To: <sip:<my real pstn number>@10.0.0.1;user=phone>
Call-ID: 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
CSeq: 1 INVITE
Supported: replaces
Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,SUBSCRIBE,NOTIFY,INFO,REFER
Contact: <sip:1001 at 10.0.0.15:5060;transport=udp>
Max-Forwards: 70
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Content-Type: application/sdp
Content-Length: 245
v=0
o=TelogyUnknown0000 9598 9598 IN IP4 10.0.0.15
s=RTP Audio
c=IN IP4 10.0.0.15
t=0 0
m=audio 2070 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (13 headers 11 lines) ---
Using INVITE request as basis request - 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
Sending to 10.0.0.15 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK464be59b7ddd49;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=87462097DB13672E83A9
To: <sip:<my real pstn number>@10.0.0.1;user=phone>;tag=as0f40b554
Call-ID: 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58726955"
Content-Length: 0
---
Scheduling destruction of call '43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15' in 15000 ms
Found user '1001'
<-- SIP read from 10.0.0.15:5060:
ACK sip:<my real pstn number>@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK464be59b7ddd49
From: <sip:1001 at 10.0.0.1;user=phone>;tag=87462097DB13672E83A9
To: <sip:<my real pstn number>@10.0.0.1;user=phone>;tag=as0f40b554
Call-ID: 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
CSeq: 1 ACK
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Content-Length: 0
--- (8 headers 0 lines) ---
<-- SIP read from 10.0.0.15:5060:
INVITE sip:<my real pstn number>@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK7f3d9c7288d857
From: <sip:1001 at 10.0.0.1;user=phone>;tag=87462097DB13672E83A9
To: <sip:<my real pstn number>@10.0.0.1;user=phone>
Call-ID: 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
CSeq: 2 INVITE
Proxy-Authorization: DIGEST
username="1001",realm="asterisk",nonce="58726955",uri="sip:<my real
pstn number>@10.0.0.1",algorithm=MD5,response="56544c8b706ad76578fc52eb31f85eb7"
Supported: replaces
Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,SUBSCRIBE,NOTIFY,INFO,REFER
Contact: <sip:1001 at 10.0.0.15:5060;transport=udp>
Max-Forwards: 70
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Content-Type: application/sdp
Content-Length: 245
v=0
o=TelogyUnknown0000 9598 9598 IN IP4 10.0.0.15
s=RTP Audio
c=IN IP4 10.0.0.15
t=0 0
m=audio 2070 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (14 headers 11 lines) ---
Using INVITE request as basis request - 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
Sending to 10.0.0.15 : 5060 (non-NAT)
Found user '1001'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.15:2070
Found description format PCMU
Found description format G729
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for <my real pstn number> in internal (domain 10.0.0.1)
list_route: hop: <sip:1001 at 10.0.0.15:5060;transport=udp>
Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK7f3d9c7288d857;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=87462097DB13672E83A9
To: <sip:<my real pstn number>@10.0.0.1;user=phone>
Call-ID: 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:<my real pstn number>@10.0.0.1>
Content-Length: 0
---
-- Executing Dial("SIP/1001-08188890", "Sip/musimi/<my real pstn number>|15") in new stack
Aug 27 18:21:27 WARNING[8077]: chan_sip.c:1991 create_addr: No such host: musimi
Destroying call '4af4190d48feae7d34d6b1505d982ab7 at 127.0.0.1'
Aug 27 18:21:27 NOTICE[8077]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion("SIP/1001-08188890", "") in new stack
Transmitting (no NAT) to 10.0.0.15:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK7f3d9c7288d857;received=10.0.0.15
From: <sip:1001 at 10.0.0.1;user=phone>;tag=87462097DB13672E83A9
To: <sip:<my real pstn number>@10.0.0.1;user=phone>;tag=as4caa0979
Call-ID: 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:<my real pstn number>@10.0.0.1>
Content-Length: 0
X-Asterisk-HangupCause: No route to destination
---
== Spawn extension (internal, <my real pstn number>, 2) exited non-zero on 'SIP/1001-08188890'
Aug 27 18:21:27 WARNING[7729]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x818ebd0', 10 retries!
<-- SIP read from 10.0.0.15:5060:
ACK sip:<my real pstn number>@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.15:5060;branch=z9hG4bK7f3d9c7288d857
From: <sip:1001 at 10.0.0.1;user=phone>;tag=87462097DB13672E83A9
To: <sip:<my real pstn number>@10.0.0.1;user=phone>;tag=as4caa0979
Call-ID: 43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15
CSeq: 2 ACK
User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone
Content-Length: 0
--- (8 headers 0 lines) ---
Destroying call '43871-1B11-1120-F660-833F5FA818E3 at 10.0.0.15'
Faidon> Then send us the debug output :) Don't forget to ommit
Faidon> sensitive information from the output. May be something
Faidon> is wrong with the way Asterisk registers; this will help
Faidon> us pinpoint it.
I hope I did :-)
Faidon> Thanks a lot, Faidon
Thanks a lot too :-)
/Hasse
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