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Summary:ASTERISK-00482: [patch] app_sayunixtime.c needs to answer the channel if it hasn't already
Reporter:w0ss (w0ss)Labels:
Date Opened:2003-11-05 05:16:10.000-0600Date Closed:2008-01-15 14:37:59.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) app_sayunixtime_answer.diff.txt
Description:I just downloaded the newest version from CVS(Tuesday@~7pm) and I am getting an error whenever I call the asterisk box. I cannot here any audio on the budgtone. This works fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it rings but I get that same error when I pick up. When the skinny phone calls the Budgtone it works fine. I have 2 budgtone phones and it does this on both of them. This worked fine before I installed the newest version of asterisk.

   -- Executing Playback("SIP/budgtone-7ee9", "carried-away-by-monkeys") in new stack
   -- Playing 'carried-away-by-monkeys' (language 'en')
   -- Executing Playback("SIP/budgtone-7ee9", "lots-o-monkeys") in new stack
   -- Playing 'lots-o-monkeys' (language 'en')
WARNING[40966]: File chan_sip.c, Line 456 (retrans_pkt): Maximum retries exceeded on call d21f4608-1b1f-0a52-b657-2d9ca6239169@192.168.1.223 for seqno 1735 (Response)


With sip debug

Sip read:
INVITE sip:9998@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To: <sip:9998@192.168.1.2> Contact: <sip:budgtone@192.168.1.223> Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62159 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 263  v=0 o=budgtone 0 0 IN IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000

12 headers, 13 lines

Using latest request as basis request

Sending to 192.168.1.223 : 5060 (non-NAT)

Found audio format UNKN

Found audio format ALAW

Found audio format ULAW

Found audio format UNKN

Found audio format GSM

Found audio format UNKN

Found description format PCMU

Found description format PCMA

Found description format G723

Found description format G729

Found description format G726-32

Found description format G728

Capabilities: us - 524302, them - 285/0, combined - 12

Non-codec capabilities: us - 1, them - 0, combined - 0

Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To: <sip:9998@192.168.1.2>;tag=as67b6f854 Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62159 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact:  Proxy-Authenticate: Digest realm="asterisk", nonce="6c3e5732" Content-Length: 0  
to 192.168.1.223:5060

Sip read:
ACK sip:9998@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=ab86b88b-d30d-4b9a-8cfe-f143b09372bd To: <sip:9998@192.168.1.2>;tag=as67b6f854 Contact: <sip:budgtone@192.168.1.223> Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62159 ACK User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0  

11 headers, 0 lines

Sip read:
INVITE sip:9998@192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998@192.168.1.2> Contact: <sip:budgtone@192.168.1.223> Proxy-Authorization: DIGEST username="budgtone", realm="asterisk", algorithm=MD5, uri="sip:9998@192.168.1.2", nonce="6c3e5732", response="4e90c985822b15d83f297e8c4fe80372" Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 263  v=0 o=budgtone 0 0 IN IP4 192.168.1.223 s=- c=IN IP4 192.168.1.223 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000

13 headers, 13 lines

Using latest request as basis request

Sending to 192.168.1.223 : 5060 (non-NAT)

Found audio format UNKN

Found audio format ALAW

Found audio format ULAW

Found audio format UNKN

Found audio format GSM

Found audio format UNKN

Found description format PCMU

Found description format PCMA

Found description format G723

Found description format G729

Found description format G726-32

Found description format G728

Capabilities: us - 524302, them - 285/0, combined - 12

Non-codec capabilities: us - 1, them - 0, combined - 0

Looking for 9998 in default

list_route: hop: <sip:budgtone@192.168.1.223>

Transmitting (no NAT):
SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998@192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998@192.168.1.2> Content-Length: 0  
to 192.168.1.223:5060

   -- Executing Playback("SIP/budgtone-66e9", "carried-away-by-monkeys") in new stack

We're at 192.168.1.2 port 15592

Answering with capability 2

Answering with capability 4

Answering with capability 8

Reliably Transmitting (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998@192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4 192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000
to 192.168.1.223:5060

   -- Playing 'carried-away-by-monkeys' (language 'en')

Retransmitting #1 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998@192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4 192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000
to 192.168.1.223:5060

Retransmitting #2 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998@192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4 192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000
to 192.168.1.223:5060

   -- Executing Playback("SIP/budgtone-66e9", "lots-o-monkeys") in new stack

   -- Playing 'lots-o-monkeys' (language 'en')

   -- Registered 'blah' (AUTHENTICATED) at 192.168.1.214:5036

Retransmitting #3 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998@192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4 192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000
to 192.168.1.223:5060

Retransmitting #4 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998@192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4 192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000
to 192.168.1.223:5060

Retransmitting ASTERISK-1 (no NAT):
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.223 From: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 To: <sip:9998@192.168.1.2>;tag=as5481a27e Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 62160 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9998@192.168.1.2> Content-Type: application/sdp Content-Length: 176  v=0 o=root 7654 7654 IN IP4 192.168.1.2 s=session c=IN IP4 192.168.1.2 t=0 0 m=audio 15592 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000
to 192.168.1.223:5060
WARNING[40966]: File chan_sip.c, Line 456 (retrans_pkt): Maximum retries exceeded on call fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 for seqno 62160 (Response)

 == Spawn extension (default, 9998, 2) exited non-zero on 'SIP/budgtone-66e9'

set_destination: Parsing <sip:budgtone@192.168.1.223> for address/port to send to

set_destination: set destination to 192.168.1.223, port 5060

Reliably Transmitting:
BYE sip:budgtone@192.168.1.223 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4062184f From: <sip:9998@192.168.1.2>;tag=as5481a27e To: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 Contact: <sip:9998@192.168.1.2> Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0   (no NAT) to 192.168.1.223:5060

Sip read:
SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4062184f From: <sip:9998@192.168.1.2>;tag=as5481a27e To: "William Carlson" <sip:budgtone@192.168.1.2>;tag=caf3f868-bc63-d9e1-72bd-8cfe49e7b093 Call-ID: fd9e49e7-81fe-9a6d-7b39-69b0b88bce52@192.168.1.223 CSeq: 102 BYE User-Agent: Grandstream SIP UA 1.0.3.81 Contact: <sip:budgtone@192.168.1.223> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0  

10 headers, 0 lines

Message is BYE




Thanks for all your help
  Will
Comments:By: w0ss (w0ss) 2003-11-06 11:55:20.000-0600

special thanks to ironhelix He have figured out how to fix this. you have to

disallow=all
allow=ulaw

seems strange that it would give that error when it is a codec problem. oh well
By: w0ss (w0ss) 2003-11-06 14:39:55.000-0600

still having problems talking to asterisk. but at least now it seems to bridge calls.
By: John Todd (jtodd) 2003-11-08 13:30:07.000-0600

So can we close out this ticket?
By: w0ss (w0ss) 2003-11-08 16:31:02.000-0600

well I guess thats your decision... there is still a problem. I have seen 3 other people report the same thing or something very similir on the mailing list. It looks to be a problem with codec capibility exchange as the
disallow=all
allow=ulaw
somewhat fixes it. I am still unable to talk to asterisk on some applications. Voicemail2 works but sayunixtime and a basic AGI script does not.
 Thanks,
   Will
By: Brian West (bkw918) 2003-11-08 20:00:50.000-0600

My BT101 works fine with *.... I think you have network and/or config issues.. mine even works with nat.
By: w0ss (w0ss) 2003-11-08 20:22:44.000-0600

Are you useing the latest CVS? Also you say it's working fine did you try sayunixtime? I don't see how it could be a network problem it worked before I upgraded asterisk as for a config problem I guess it's possible but again all I did was upgrade asterisk.
By: swell (swell) 2003-11-12 01:31:15.000-0600

I just bought 2 new BT101's to test for my company.  I am having this same issue.  The firmware is the very latest.  I am on the very lattest CVS.  Currently all our extensions are analog on CAC FXS ports.  When I call from analog to BT everything works like a charm.  When I call from BT to analog the analog rings, I pick up the line and there is nothing.  In the BT receiver it continues to ring for like 2-3 seconds then both the BT and analog drop the line with a fast busy signal.  I have tried everything.  No luck.  Same problem on BT to BT.  I am going to try this disallow=all; allow=ulaw hack and see if things work tomorrow.  I am supposed to send an ethreal dump to grandstream tomorrow.
By: w0ss (w0ss) 2003-11-12 06:40:44.000-0600

another way I have found to fix this is even odder. Put an explisite answer.

exten => 123,1,answer
exten => 123,2,SayUnixTime


This does not work(for the budgetone works for all other channels/sip).
exten => 123,1,SayUnixTime
By: tekati (tekati) 2003-11-12 16:09:01.000-0600

Same problem here.  With the budgetone ATA-286 which is supposed to have the same engine as the BT100 series phones.  I can call from TDM40B > * > ATA286 no problem but from ATA286 > * > TDM40B or any other type of interface it rings then someone picks up the other extension and it continues to ring a couple more times then both parties hear the congestion busy signal at the same time.  Please let me know if there is any output that would help solve this issue.
By: Brian West (bkw918) 2003-11-13 15:58:09.000-0600

This sounds like an issue with the Grandstream products... jtodd would you agree?
By: zoa (zoa) 2003-11-20 13:45:09.000-0600

try decreasing the registration timeout on the grandstream, i think that helps...
By: Brian West (bkw918) 2003-11-20 13:49:02.000-0600

Works on grandstream and cisco.
By: Brian West (bkw918) 2003-11-20 23:06:31.000-0600

Fixed in CVS.
By: Digium Subversion (svnbot) 2008-01-15 14:37:59.000-0600

Repository: asterisk
Revision: 1771

U   trunk/apps/app_sayunixtime.c

------------------------------------------------------------------------
r1771 | markster | 2008-01-15 14:37:59 -0600 (Tue, 15 Jan 2008) | 2 lines

Answer if channel isn't up (bug ASTERISK-482)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=1771