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Chris Higgins

2004-Mar-16 20:03 UTC

[Asterisk-Users] Sipura line 1 outgoing voice problem?

Back in January I started having a problem with my Sipura (and there was 
at least one other on the list with the same problem) that if I answer 
an incoming call (via X100P) on line 1 of my Sipura, the caller cannot 
hear any voice from the internal extension.  If the internal user puts 
the external user on hold (via flash hook) and returns, both directions 
of audio are fine.

Line 2 never has had this problem.  For the meantime, I switched the 
internal phones so that my wife's favorite phone is line 2 and I told 
her to not pick up with line 1.  Not a very permanent solution :)

NAT is not an issue as the Sipura and * are on the same network.  Is 
anyone else having this problem?  It looks like other people are using 
Sipura (I saw one user with 30 of them ?!) and am surprised that nobody 
else is complaining about this problem.  I am willing to step through 
some sip debug if anyone is interested in the output.

* version: Asterisk CVS-02/08/04-22:22:57
Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem 
would go away)

Relevent config sections:

--8<--  sip.conf  --8<--

[cordless1]
type=friend
username=cordless1
secret=xxx
host=dynamic
context=cordless1
dtmfmode=info
mailbox=1234
canreinvite=no
disallow=all
allow=alaw

[cordless2]
type=friend
username=cordless2
secret=xxx
host=dynamic
context=cordless2
dtmfmode=info
mailbox=1234
canreinvite=no
disallow=all
allow=alaw


-- Chris

Senad Jordanovic

2004-Mar-17 03:58 UTC

head link

[Asterisk-Users] Sipura line 1 outgoing voice problem?

Chris Higgins wrote:> Back in January I started having a problem with my Sipura (and there
> was 
> at least one other on the list with the same problem) that if I answer
> an incoming call (via X100P) on line 1 of my Sipura, the caller cannot
> hear any voice from the internal extension.  If the internal user puts
> the external user on hold (via flash hook) and returns, both
> directions 
> of audio are fine.
I have not had this problem... And I use X100P as well in same setup.
BUT... There are other problems I have or had.
> 
> [cordless1]
> type=friend
> username=cordless1
> secret=xxx
> host=dynamic
> context=cordless1
> dtmfmode=info
> mailbox=1234
> canreinvite=no
> disallow=all
> allow=alaw
>
If you are using your SPA 2000 directly with * maybe it is better to
have "canreinvite" set to yes. ??? 

Also.. I think that auto default dtmfmode for SPA is AVT (which is
RFC2833)... So check that.!!!

cveazey@blackhillsfiber.com

2004-Mar-17 09:18 UTC

head link

[Asterisk-Users] Sipura line 1 outgoing voice problem?

__________________
Back in January I started having a problem with my Sipura (and there was
at least one other on the list with the same problem) that if I answer
an incoming call (via X100P) on line 1 of my Sipura, the caller cannot
hear any voice from the internal extension.  If the internal user puts
the external user on hold (via flash hook) and returns, both directions
of audio are fine.

Line 2 never has had this problem.  For the meantime, I switched the
internal phones so that my wife's favorite phone is line 2 and I told
her to not pick up with line 1.  Not a very permanent solution :)

NAT is not an issue as the Sipura and * are on the same network.  Is
anyone else having this problem?  It looks like other people are using
Sipura (I saw one user with 30 of them ?!) and am surprised that nobody
else is complaining about this problem.  I am willing to step through
some sip debug if anyone is interested in the output.

* version: Asterisk CVS-02/08/04-22:22:57
Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem
would go away)

Relevent config sections:

--8<--  sip.conf  --8<--

[cordless1]
type=friend
username=cordless1
secret=xxx
host=dynamic
context=cordless1
dtmfmode=info
mailbox=1234
canreinvite=no
disallow=all
allow=alaw

[cordless2]
type=friend
username=cordless2
secret=xxx
host=dynamic
context=cordless2
dtmfmode=info
mailbox=1234
canreinvite=no
disallow=all
allow=alaw


-- Chris
_______________________________________________





I had the exact same problem with a Mediatrix 1102....doing a flash hook 
brought both sides of the conversation together.  I found out that my 
sip.conf file had GSM as the first priority codec and the 1102 doesn't 
support GSM.  I kept that the same but put a "disallow = gsm"
statement in
my sip entry for the 1102 so g.711ulaw would be the first negotiated 
codec.  That fixed the problem.

VZ
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Mark Musone

2004-Apr-18 08:33 UTC

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[Asterisk-Users] Sipura line 1 outgoing voice problem?

I'm not sure if anybody has determined the cause/fix for this problem,
but I am getting the same problem.

I turned on syslog debugging and there were some interesting results:

1. phone call answered on line2:

Apr 18 11:14:13 192.168.1.21 [0:0]AUD ALLOC CALL (port=16428)
Apr 18 11:14:14 192.168.1.21 [0:0]RTP Rx Up
Apr 18 11:14:14 192.168.1.21 [1:0]AUD ALLOC CALL (port=16430)
Apr 18 11:14:14 192.168.1.21 [1:0]RTP Rx Up
Apr 18 11:14:17 192.168.1.21 [1]Off Hook
Apr 18 11:14:17 192.168.1.21 [1]CID interrupted
Apr 18 11:14:17 192.168.1.21 [1:0]RTP Rx 1st PKT @16430(2)
Apr 18 11:14:17 192.168.1.21 [0:0]AUD Rel Call
Apr 18 11:14:17 192.168.1.21 CC:Ended
Apr 18 11:14:17 192.168.1.21 CC:Connected
Apr 18 11:14:18 192.168.1.21 [1:0]ENC INIT 0
Apr 18 11:14:18 192.168.1.21 [1:0]RTP Tx Up (pt=0->c0a80114:15840)
Apr 18 11:14:18 192.168.1.21 [1:0]RTCP Tx Up
Apr 18 11:14:18 192.168.1.21 [1:0]DEC INIT 0
Apr 18 11:14:22 192.168.1.21 [1]On Hook
Apr 18 11:14:22 192.168.1.21 [1:0]AUD Rel Call
Apr 18 11:14:22 192.168.1.21 DLG Terminated
Apr 18 11:14:22 192.168.1.21 Sess Terminated
Apr 18 11:14:49 192.168.1.21 DLG Terminated
Apr 18 11:14:49 192.168.1.21 Sess Terminated
Apr 18 11:14:54 192.168.1.21 CC:Clean Up


2. Phone call answered on line1 (no outgoing voice)

Apr 18 11:01:16 192.168.1.21 [0:0]AUD ALLOC CALL (port=16424)
Apr 18 11:01:16 192.168.1.21 [0:0]RTP Rx Up
Apr 18 11:01:16 192.168.1.21 [1:0]AUD ALLOC CALL (port=16426)
Apr 18 11:01:16 192.168.1.21 [1:0]RTP Rx Up
Apr 18 11:01:18 192.168.1.21 [0]Off Hook
Apr 18 11:01:18 192.168.1.21 [0]CID interrupted
Apr 18 11:01:18 192.168.1.21 Codec 135 not defined in DPT
Apr 18 11:01:18 192.168.1.21 Codec 135 not defined in DPT
Apr 18 11:01:18 192.168.1.21 CC:Connected
Apr 18 11:01:18 192.168.1.21 No Common TxCodec
Apr 18 11:01:18 192.168.1.21 [0:0]RTP Rx 1st PKT @16424(2)
Apr 18 11:01:19 192.168.1.21 [1:0]AUD Rel Call
Apr 18 11:01:19 192.168.1.21 CC:Ended
Apr 18 11:01:19 192.168.1.21 [0:0]DEC INIT 0
Apr 18 11:01:21 192.168.1.21 [0]On Hook
Apr 18 11:01:21 192.168.1.21 [0:0]AUD Rel Call
Apr 18 11:01:21 192.168.1.21 DLG Terminated
Apr 18 11:01:21 192.168.1.21 Sess Terminated
Apr 18 11:01:50 192.168.1.21 DLG Terminated
Apr 18 11:01:50 192.168.1.21 Sess Terminated
Apr 18 11:01:53 192.168.1.21 CC:Clean Up

Seems like for some reason, line1 is not agreeing on the Tx codec..i
tried playing around with the codecs configured in both the sipura
configs and asterisk, but could not find anything that worked..


Below is the interesting flash-hook fix:

Apr 18 11:17:29 192.168.1.21 [0:0]AUD ALLOC CALL (port=16432)
Apr 18 11:17:29 192.168.1.21 [0:0]RTP Rx Up
Apr 18 11:17:29 192.168.1.21 [1:0]AUD ALLOC CALL (port=16434)
Apr 18 11:17:29 192.168.1.21 [1:0]RTP Rx Up
Apr 18 11:17:32 192.168.1.21 [0]Off Hook
Apr 18 11:17:32 192.168.1.21 [0]CID interrupted
Apr 18 11:17:32 192.168.1.21 Codec 135 not defined in DPT
Apr 18 11:17:32 192.168.1.21 Codec 135 not defined in DPT
Apr 18 11:17:32 192.168.1.21 [0:0]RTP Rx 1st PKT @16432(2)
Apr 18 11:17:32 192.168.1.21 CC:Connected
Apr 18 11:17:32 192.168.1.21 No Common TxCodec
Apr 18 11:17:32 192.168.1.21 [1:0]AUD Rel Call
Apr 18 11:17:32 192.168.1.21 CC:Ended
Apr 18 11:17:32 192.168.1.21 [0:0]DEC INIT 0
Apr 18 11:17:35 192.168.1.21 [0]Hook Flash
Apr 18 11:17:35 192.168.1.21 [0:0]RTP Tx Dn
Apr 18 11:17:35 192.168.1.21 [0:0]RTP Rx Dn
Apr 18 11:17:36 192.168.1.21 CC:Hold
Apr 18 11:17:36 192.168.1.21 [0:0]RTP Tx Dn
Apr 18 11:17:36 192.168.1.21 CC:Remote Resume
Apr 18 11:17:36 192.168.1.21 [0:0]RTCP Tx Up
Apr 18 11:17:38 192.168.1.21 CC:Connected
Apr 18 11:17:38 192.168.1.21 [0:0]RTP Tx Dn
Apr 18 11:17:39 192.168.1.21 [0:0]ENC INIT 0
Apr 18 11:17:39 192.168.1.21 [0:0]RTP Tx Up (pt=0->c0a80114:10040)
Apr 18 11:17:39 192.168.1.21 [0:0]RTCP Tx Up
Apr 18 11:17:39 192.168.1.21 CC:Remote Resume
Apr 18 11:17:39 192.168.1.21 [0:0]RTCP Tx Up
Apr 18 11:17:39 192.168.1.21 [0:0]RTP Rx Up
Apr 18 11:17:39 192.168.1.21 [0:0]RTP Rx 1st PKT @16432(2)
Apr 18 11:17:39 192.168.1.21 [0:0]DEC INIT 0
Apr 18 11:17:38 192.168.1.21 [0]Hook Flash
Apr 18 11:17:46 192.168.1.21 [0]On Hook
Apr 18 11:17:47 192.168.1.21 [0:0]AUD Rel Call
Apr 18 11:17:56 192.168.1.21 [1]RegOK. NextReg in 180
Apr 18 11:17:56 192.168.1.21 [0]RegOK. NextReg in 180
Apr 18 11:18:04 192.168.1.21 DLG Terminated
Apr 18 11:18:04 192.168.1.21 Sess Terminated
Apr 18 11:18:10 192.168.1.21 DLG Terminated
Apr 18 11:18:10 192.168.1.21 Sess Terminated
Apr 18 11:18:18 192.168.1.21 CC:Clean Up


I tried doing some sniffs with ethereal and sip debug, but neither
seemed to sow any info like this (of course, I don't know the sip debug
stuff that well)

w
My feeling is that this is a Sipura problem. I've upgraded to the
firmare 2.02, but still no difference.

Besides that, my configs for line1 and line2 in the sipura configuration
are exactly the same, except for SIP port (5060 and 5061 respectively)
and the extention numbers (2201 and 2202 respectively)

My asterisk config for the sip lines are:

[2202]
type=friend
host=dynamic
context=home
secret=XXXx
callerid="SPA2" <2202>
mailbox=2202
dtmfmode=rfc2833
canreinvite=no
nat=0

[2201]
type=friend
host=dynamic
context=home
secret=Xxxx
callerid="SPA1" <2201>
mailbox=2202
dtmfmode=rfc2833
canreinvite=no
nat=0



I'm also sending a copy of this to the sipura tech support..


Hope this either helps others to possibly find a fix, or if anyone
_does_ have a fix, please let me know!

I'm pretty close to just returning it, as to me, this simply does not
work.

Thanks!

-Mark



----- Original Message ----- 
From: Matt McIntyre
To: asterisk-users@lists.digium.com
Sent: Tuesday, March 23, 2004 6:59 PM
Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?


I am experiencing this same problem and was wondering if anyone has come
to
a resolution.  I have contacted Sipura but have not heard any response
yet
and am having trouble determining for sure whether the problem resides
with
Asterisk or the Sipura.  As I have noticed that there are many users on
the
list who use the Sipura unit without this problem (and even a fellow
with
one unit that worked and one that did) I think the Sipura must be
suspect.

__________________
Back in January I started having a problem with my Sipura (and there was
at least one other on the list with the same problem) that if I answer
an incoming call (via X100P) on line 1 of my Sipura, the caller cannot
hear any voice from the internal extension.  If the internal user puts
the external user on hold (via flash hook) and returns, both directions
of audio are fine.

Line 2 never has had this problem.  For the meantime, I switched the
internal phones so that my wife's favorite phone is line 2 and I told
her to not pick up with line 1.  Not a very permanent solution :)

Victor Perez

2004-Apr-18 09:32 UTC

head link

[Asterisk-Users] Sipura line 1 outgoing voice problem?

I don't know if this helps, but I started having this problem after I sent
out a fax. My fax machine was connected to line 1 at that time. I tried changing
the FAX detection settings but no luck.


Regards,
Victor Perez



-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Mark Musone
Sent: Sunday, April 18, 2004 10:33 AM
To: asterisk-users@lists.digium.com
Cc: support@sipura.com
Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?


I'm not sure if anybody has determined the cause/fix for this problem,
but I am getting the same problem.

I turned on syslog debugging and there were some interesting results:

...

My feeling is that this is a Sipura problem. I've upgraded to the
firmare 2.02, but still no difference.

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