[asterisk-users] Maximum retries exceeded

[姚文超]

have searched this list and others, and see other pepole having this issue. 
However, I have not seen how to fix it.

 
Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Maximum retries 
exceeded on transmission 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 for 
seqno 1620 (Critical Response)
 
Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Hanging up call 
778f89593967725f0abe40eb1752504c (at) 10.10.206.53 no reply to our critical 
packet.

  
  
 

   
  
  -- Original --
  From:  "asterisk-users-request";
 Date:  Thu, Dec 23, 2010 10:52 AM
 To:  "229838677"<229838...@qq.com>; 
 
 Subject:  confirm 7544ab1cb7b90cb7d6583f225fe16c45193d0c45

  
Mailing list subscription confirmation notice for mailing list
asterisk-users

We have received a request from 125.70.77.55 for subscription of your
email address, "229838...@qq.com", to the
asterisk-users@lists.digium.com mailing list.  To confirm that you
want to be added to this mailing list, simply reply to this message,
keeping the Subject: header intact.  Or visit this web page:


http://lists.digium.com/mailman/confirm/asterisk-users/7544ab1cb7b90cb7d6583f225fe16c45193d0c45


Or include the following line -- and only the following line -- in a
message to asterisk-users-requ...@lists.digium.com:

confirm 7544ab1cb7b90cb7d6583f225fe16c45193d0c45

Note that simply sending a `reply' to this message should work from
most mail readers, since that usually leaves the Subject: line in the
right form (additional "Re:" text in the Subject: is okay).

If you do not wish to be subscribed to this list, please simply
disregard this message.  If you think you are being maliciously
subscribed to the list, or have any other questions, send them to
asterisk-users-ow...@lists.digium.com.--
_
-- 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

[asterisk-users] maximum retries exceeded on transmission Warnings

[Benjamin Jacob]

Hello All,
I've got the following warning messages a couple of days back:
/chan_sip.c: Maximum retries exceeded on transmission  for 
seqno 1 (Critical Response).

/Have got the warnings repeatedly for one Callid. If maximum retries 
have exceeded why should it give me those warnings again n again for the 
same callid, with a gap 4 seconds between each warning.
The callids mentioned in the warnings are of the inbound leg.

I've scoured the net, but haven't got anything conclusive. Have found 
responses ranging from firewall issues, no reception of ACKs, to bugs in 
some versions of Asterisk.

I am using Asterisk 1.4.4, all SIP calls, with PSTN termination provided 
by my service provider. Have no firewalls or iptables set on my server.
The calls did not seem to work even across a restart of asterisk.
Interestingly, the calls to and from the very same numbers worked later 
on the next day.

Anyone faced similar problems and was able to get the root of it? Or is 
it a bug?

cheerz
- Ben







EMAIL DISCLAIMER : This email and any files transmitted with it are 
confidential and intended solely for the use of the individual or entity to 
whom they are addressed. Any unauthorised distribution or copying is strictly 
prohibited. If you receive this transmission in error, please notify the sender 
by reply email and then destroy the message. Opinions, conclusions and other 
information in this message that do not relate to official business of Mascon 
shall be understood to be neither given nor endorsed by Mascon. Any information 
contained in this email, when addressed to Mascon clients is subject to the 
terms and conditions in governing client contract.

Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we 
can not guarantee that any email or attachment is free from computer viruses 
and you are strongly advised to undertake your own anti-virus precautions. 
Mascon grants no warranties regarding performance, use or quality of any e-mail 
or attachment and undertakes no liability for loss or damage, howsoever caused. 



___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum retries exceeded on transmission

[Apa Minerala]

Tom,
The device is voxbone from voxbone.com . I am using a DID as an access 
number...it worked with same config with asterisk 1.2.12 and a2billing 1.2.3, 
but doesn't work with asterisk 1.4.11 and a2billing 1.3 

Can you tell me what am I missing?

Apa

Tom Lynn <[EMAIL PROTECTED]> wrote: I suspect if you remove the callerid entry 
from this device's sip.conf definition things will work better.  

On 9/9/07, Apa Minerala < [EMAIL PROTECTED]> wrote:
 
 I have searched this list and others, and see other pepole having this 
 issue. However, I have not seen how to fix it.
 
 Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum
 retries exceeded on transmission
 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical 
 Response)
 
 Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up
 call 778f89593967725f0abe40eb1752504c no reply to our critical
 packet.
 
 What is the critical packet that is not being responded to? Please help. 
 
 

-
 Pinpoint customers who are looking for what you sell.   

___

Sign up now for AstriCon 2007!  September 25-28th.   http://www.astricon.net/

--Bandwidth and Colocation Provided by http://www.api-digital.com--
 
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users 


 ___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

   
-
Be a better Globetrotter. Get better travel answers from someone who knows.
Yahoo! Answers - Check it out.___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum retries exceeded on transmission

[Tom Lynn]

I suspect if you remove the callerid entry from this device's
sip.confdefinition things will work better.

On 9/9/07, Apa Minerala <[EMAIL PROTECTED]> wrote:
>
>
>
> I have searched this list and others, and see other pepole having this
> issue. However, I have not seen how to fix it.
>
> Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum
> retries exceeded on transmission
> 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical
> Response)
>
> Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging
> up
> call 778f89593967725f0abe40eb1752504c no reply to our critical
> packet.
>
> What is the critical packet that is not being responded to? Please help.
>
>  --
> Pinpoint customers
> who
> are looking for what you sell.
>
>
> ___
>
> Sign up now for AstriCon 2007!  September 25-28th.
> http://www.astricon.net/
>
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Maximum retries exceeded on transmission

[Apa Minerala]

 
 I have searched this list and others, and see other pepole having this
 issue. However, I have not seen how to fix it.
 
 Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum
 retries exceeded on transmission
 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical
 Response)
 
 Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up
 call 778f89593967725f0abe40eb1752504c no reply to our critical
 packet.
 
 What is the critical packet that is not being responded to? Please help.
 
 
   
-
Pinpoint customers who are looking for what you sell. ___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum retries exceeded on transmission

[Olle E Johansson]

13 apr 2007 kl. 16.45 skrev Brian Jones:

I've encountered a similar problem with Cisco equipment.  The Cisco  
proxy was not replying to Asterisk with an ACK after * sent an OK.


Since version 1.2.14, * was changed so that not receiving an ACK to  
an OK is considered a FATAL error.


The specific change that causes this problem is in sip_answer() in  
chan_sip.c:


res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2);

Changing the 2 to a 1 will probably fix it.  Note that this is NOT  
a bug in * but improper implementations--either caused by latency,  
or a software bug (not sending an ACK).  Perhaps it might be  
beneficial to have an option in sip.conf to change how * handles  
not receiving an ACK?  I know... it's someone else's problem, but  
might help those of us stuck with buggy implementations in  
production environments. :)


Answering late, but still answering to this mail from april that was  
highlighted to me by an Asterisk user.
This change is *not recommended* and will in worst case cause  
Asterisk to have channels hanging with open UDP ports

and eventually break your system.

Not sending an ACK on an INVITE-200 OK- ACK transaction is and should  
be a fatal error. If you change this, you really
need to know what you're up to. (And don't request help on the bug  
tracker :-) )


/Olle

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum retries exceeded on transmission

[Brian Jones]

I've encountered a similar problem with Cisco equipment.  The Cisco proxy
was not replying to Asterisk with an ACK after * sent an OK.

Since version 1.2.14, * was changed so that not receiving an ACK to an OK is
considered a FATAL error.

The specific change that causes this problem is in sip_answer() in
chan_sip.c:

res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2);

Changing the 2 to a 1 will probably fix it.  Note that this is NOT a bug in
* but improper implementations--either caused by latency, or a software bug
(not sending an ACK).  Perhaps it might be beneficial to have an option in
sip.conf to change how * handles not receiving an ACK?  I know... it's
someone else's problem, but might help those of us stuck with buggy
implementations in production environments. :)

Brian.


On 4/12/07, Joao Pereira <[EMAIL PROTECTED]> wrote:


Hello
Thanks a lot for your reply.
Im now using asterisk-1.2.10 and the problem disappeared.
Thanks
regards
Joao Pereira


Edoardo Serra wrote:
> Same to me !!
>
> Calls from OpenSER to Asterisk
>
> It happens only with Asterisk versions >= 1.2.14
>
> I'm going to capture some traffic
>
> Tnx for help
>
> Regards
>
> Alex Balashov ha scritto:
>>
>> Joao,
>>
>>   It sounds like the proxy is not acknowledging the Asterisk's
>> processing of the INVITE, but I could be wrong.  It would be helpful
>> to supply a packet capture between OpenSER and Asterisk so we could
>> see the setup flow.
>>
>> Thanks,
>>
>> -- Alex
>>
>> On Tue, 10 Apr 2007, Joao Pereira said something to this effect:
>>
>>> Hello
>>> My asterisk is receiving calls from OpenSER but all calls hangup in
>>> 20 seconds.
>>> This only happens because Im using Asterisk2Billing's AGI (without
>>> A2Billing it doesnt hang up).
>>> does someone knows whats the problem??
>>>
>>> Here is my Asterisk debug:
>>> (xxx.xxx.xxx.xxx  -> the phone's IP)
>>>
>>>
>>>
>>> Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread:
>>> Unable to spawn mp3player
>>> Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort
>>> noise support incomplete in Asterisk (RFC 3389). Please turn off on
>>> client if possible. Client IP: xxx.xxx.xxx.xxx
>>> Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum
>>> retries exceeded on transmission
>>> [EMAIL PROTECTED] for seqno 12282
>>> (Critical Response)
>>> Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging
>>> up call [EMAIL PROTECTED] - no
>>> reply to our critical packet.
>>> Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort
>>> noise support incomplete in Asterisk (RFC 3389). Please turn off on
>>> client if possible. Client IP: xxx.xxx.xxx.xxx
>>>
>>>
>>> Thanks for the help
>>> Regards
>>> Joao Pereira
>>>
>>>
>>> ___
>>> --Bandwidth and Colocation provided by Easynews.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> Alex Balashov <[EMAIL PROTECTED]>
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>
>
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum retries exceeded on transmission

[Joao Pereira]

Hello
Thanks a lot for your reply.
Im now using asterisk-1.2.10 and the problem disappeared.
Thanks
regards
Joao Pereira


Edoardo Serra wrote:

Same to me !!

Calls from OpenSER to Asterisk

It happens only with Asterisk versions >= 1.2.14

I'm going to capture some traffic

Tnx for help

Regards

Alex Balashov ha scritto:


Joao,

  It sounds like the proxy is not acknowledging the Asterisk's 
processing of the INVITE, but I could be wrong.  It would be helpful 
to supply a packet capture between OpenSER and Asterisk so we could 
see the setup flow.


Thanks,

-- Alex

On Tue, 10 Apr 2007, Joao Pereira said something to this effect:


Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 
20 seconds.
This only happens because Im using Asterisk2Billing's AGI (without 
A2Billing it doesnt hang up).

does someone knows whats the problem??

Here is my Asterisk debug:
(xxx.xxx.xxx.xxx  -> the phone's IP)



Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: 
Unable to spawn mp3player
Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort 
noise support incomplete in Asterisk (RFC 3389). Please turn off on 
client if possible. Client IP: xxx.xxx.xxx.xxx
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum 
retries exceeded on transmission 
[EMAIL PROTECTED] for seqno 12282 
(Critical Response)
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging 
up call [EMAIL PROTECTED] - no 
reply to our critical packet.
Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort 
noise support incomplete in Asterisk (RFC 3389). Please turn off on 
client if possible. Client IP: xxx.xxx.xxx.xxx



Thanks for the help
Regards
Joao Pereira


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users



--
Alex Balashov <[EMAIL PROTECTED]>
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users






___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum retries exceeded on transmission

[Andrey Solovjov]

I confirm the same behaviour. I use asterisk with Mera Softswitch (with 
SIP HIT).
After upgrading from 1.2.13 to 1.2.14 "Maximum retries exceeded..." 
messages began to appear in logs. About 10% of calls were lost. I've 
dumped such calls and don't see anything suspicous in Mera's packets. 
Asterisk doesn't reply for first several INVITEs from Mera but then it 
replies OK, Mera sends back ACK but it seems that asterisk ignores it 
and tries to send OK. After trying to send OK several times asterisk 
hangs up the call. I've attached the text file where this can be seen. 
Mera SS is 10.150.16.4. We see that asterisk replies to INVITEs after 4 
seconds. That's wierd. Server is not heavily loaded - about 10 
simultanious calls.
I've downgraded to 1.2.13 and problem has gone away. I guess there is 
something wrong with asterisk.

Regards.
Andrey Solovjov.

Edoardo Serra:

Same to me !!

Calls from OpenSER to Asterisk

It happens only with Asterisk versions >= 1.2.14

I'm going to capture some traffic

Tnx for help

Regards

Alex Balashov ha scritto:


Joao,

  It sounds like the proxy is not acknowledging the Asterisk's 
processing of the INVITE, but I could be wrong.  It would be helpful 
to supply a packet capture between OpenSER and Asterisk so we could 
see the setup flow.


Thanks,

-- Alex

On Tue, 10 Apr 2007, Joao Pereira said something to this effect:


Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 
20 seconds.
This only happens because Im using Asterisk2Billing's AGI (without 
A2Billing it doesnt hang up).

does someone knows whats the problem??

Here is my Asterisk debug:
(xxx.xxx.xxx.xxx  -> the phone's IP)



Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: 
Unable to spawn mp3player
Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort 
noise support incomplete in Asterisk (RFC 3389). Please turn off on 
client if possible. Client IP: xxx.xxx.xxx.xxx
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum 
retries exceeded on transmission 
[EMAIL PROTECTED] for seqno 12282 
(Critical Response)
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging 
up call [EMAIL PROTECTED] - no 
reply to our critical packet.
Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort 
noise support incomplete in Asterisk (RFC 3389). Please turn off on 
client if possible. Client IP: xxx.xxx.xxx.xxx



Thanks for the help
Regards
Joao Pereira


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users



--
Alex Balashov <[EMAIL PROTECTED]>
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users







|Time | 10.150.16.4   | 10.153.144.131|
|488,548  | INVITE SDP ( telephone-event)  |SIP From: 
sip:[EMAIL PROTECTED] To:sip:[EMAIL PROTECTED]:5060
| |(5060)   -->  (5060)   |
|489,047  | INVITE SDP ( telephone-event)  |SIP From: 
sip:[EMAIL PROTECTED] To:sip:[EMAIL PROTECTED]:5060
| |(5060)   -->  (5060)   |
|489,539  | INVITE SDP ( telephone-event)  |SIP From: 
sip:[EMAIL PROTECTED] To:sip:[EMAIL PROTECTED]:5060
| |(5060)   -->  (5060)   |
|490,544  | INVITE SDP ( telephone-event)  |SIP From: 
sip:[EMAIL PROTECTED] To:sip:[EMAIL PROTECTED]:5060
| |(5060)   -->  (5060)   |
|492,429  | 100 Trying|   |SIP Status
| |(5060)   <--  (5060)   |
|492,434  | 200 OK SDP ( telephone-event)  |SIP Status
| |(5060)   <--  (5060)   |
|492,435  | RTP (g711A)   |RTP Num packets:845  
Duration:19.980s ssrc:858984592
| |(21816)  <--  (16296)  |
|492,448  | ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|492,557  | 200 OK SDP ( telephone-event)  |SIP Status
| |(5060)   <--  (5060)   |
|492,560  | ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|492,683  | 200 OK SDP ( telephone-event)  |SIP Status
| |(5060)   <--  (5060)   |
|492,708  | ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|492,834  | RTP (g711A)   |RTP Num packets:11  
Duration:0.206s ssrc:1605848118
| |(21816)  -->  (16296)  |
|493,050  | 200 OK SDP ( telephone-event)  |SIP Status
| |(5060)   <--  (5060)   |
|493,051  

Re: [asterisk-users] Maximum retries exceeded on transmission

[Edoardo Serra]

Same to me !!

Calls from OpenSER to Asterisk

It happens only with Asterisk versions >= 1.2.14

I'm going to capture some traffic

Tnx for help

Regards

Alex Balashov ha scritto:


Joao,

  It sounds like the proxy is not acknowledging the Asterisk's 
processing of the INVITE, but I could be wrong.  It would be helpful 
to supply a packet capture between OpenSER and Asterisk so we could 
see the setup flow.


Thanks,

-- Alex

On Tue, 10 Apr 2007, Joao Pereira said something to this effect:


Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 
20 seconds.
This only happens because Im using Asterisk2Billing's AGI (without 
A2Billing it doesnt hang up).

does someone knows whats the problem??

Here is my Asterisk debug:
(xxx.xxx.xxx.xxx  -> the phone's IP)



Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: 
Unable to spawn mp3player
Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort 
noise support incomplete in Asterisk (RFC 3389). Please turn off on 
client if possible. Client IP: xxx.xxx.xxx.xxx
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum 
retries exceeded on transmission 
[EMAIL PROTECTED] for seqno 12282 
(Critical Response)
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging 
up call [EMAIL PROTECTED] - no 
reply to our critical packet.
Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort 
noise support incomplete in Asterisk (RFC 3389). Please turn off on 
client if possible. Client IP: xxx.xxx.xxx.xxx



Thanks for the help
Regards
Joao Pereira


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users



--
Alex Balashov <[EMAIL PROTECTED]>
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users





--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino

Tel: +39 011 678 100
Fax: +39 011 678 275

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Maximum retries exceeded on transmission

[Alex Balashov]

Joao,

  It sounds like the proxy is not acknowledging the Asterisk's 
processing of the INVITE, but I could be wrong.  It would be helpful to 
supply a packet capture between OpenSER and Asterisk so we could see the 
setup flow.


Thanks,

-- Alex

On Tue, 10 Apr 2007, Joao Pereira said something to this effect:


Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 20 
seconds.
This only happens because Im using Asterisk2Billing's AGI (without A2Billing 
it doesnt hang up).

does someone knows whats the problem??

Here is my Asterisk debug:
(xxx.xxx.xxx.xxx  -> the phone's IP)



Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to 
spawn mp3player
Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise 
support incomplete in Asterisk (RFC 3389). Please turn off on client if 
possible. Client IP: xxx.xxx.xxx.xxx
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries 
exceeded on transmission [EMAIL PROTECTED] 
for seqno 12282 (Critical Response)
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call 
[EMAIL PROTECTED] - no reply to our 
critical packet.
Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise 
support incomplete in Asterisk (RFC 3389). Please turn off on client if 
possible. Client IP: xxx.xxx.xxx.xxx



Thanks for the help
Regards
Joao Pereira


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users



--
Alex Balashov <[EMAIL PROTECTED]>
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Maximum retries exceeded on transmission

[Joao Pereira]

Hello
My asterisk is receiving calls from OpenSER but all calls hangup in 20 
seconds.
This only happens because Im using Asterisk2Billing's AGI (without 
A2Billing it doesnt hang up).

does someone knows whats the problem??

Here is my Asterisk debug:
(xxx.xxx.xxx.xxx  -> the phone's IP)



Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: 
Unable to spawn mp3player
Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise 
support incomplete in Asterisk (RFC 3389). Please turn off on client if 
possible. Client IP: xxx.xxx.xxx.xxx
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum 
retries exceeded on transmission 
[EMAIL PROTECTED] for seqno 12282 
(Critical Response)
Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up 
call [EMAIL PROTECTED] - no reply to 
our critical packet.
Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise 
support incomplete in Asterisk (RFC 3389). Please turn off on client if 
possible. Client IP: xxx.xxx.xxx.xxx



Thanks for the help
Regards
Joao Pereira


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Maximum retries exceeded on transmission

[AJ Grinnell]

I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it.
Sep 12 18:52:36 
WARNING
[4620]: chan_sip.c:1835 retrans_pkt: Maximum retries exceeded 
on transmission [EMAIL PROTECTED] for seqno 1620 
(Critical Response)
Sep 12 18:52:36 
WARNING
[4620]: chan_sip.c:1835 retrans_pkt: Hanging up call 
[EMAIL PROTECTED] no reply to our critical 
packet.What is the critical packet that is not being responded to? Please help.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on transmission

[Filipe Mordhorst]

Gentlemen, please help me with this one. When a try to
call this user, “drsouza”, I’m sent directly to the mailbox,
that is the 2º option in extension.conf but the user still gets the incoming
call messages in his softphone and he answers it, he just hear silence.

I looked for some help at google but it didn’t
help me very much.

 

asterisk01*CLI>

    -- Executing Dial("SIP/drsouza-79bb",
"ZAP/4/008007012700|60|tT") in new stack

    -- Called 4/008007012700

    -- Zap/4-1 answered SIP/drsouza-79bb

    -- Hungup 'Zap/4-1'

  == Spawn extension (ramais-digitro, 008007012700,
1) exited non-zero on 'SIP/drsouza-79bb'

    -- Executing Dial("SIP/drsouza-18ae",
"ZAP/4/034613744|60|tT") in new stack

    -- Called 4/034613744

    -- Zap/4-1 answered SIP/drsouza-18ae

    -- Hungup 'Zap/4-1'

  == Spawn extension (ramais-digitro, 034613744, 1)
exited non-zero on 'SIP/drsouza-18ae'

    -- Starting simple switch on 'Zap/4-1'

Mar 22 15:47:49 NOTICE[27300]: chan_zap.c:6063
ss_thread: Got event 18 (Ring Begin)...

Mar 22 15:47:49 NOTICE[27300]: chan_zap.c:6063
ss_thread: Got event 2 (Ring/Answered)...

Mar 22 15:47:53 NOTICE[27300]: chan_zap.c:6063
ss_thread: Got event 18 (Ring Begin)...

    -- Executing Goto("Zap/4-1",
"ramais-digitro|8603|1") in new stack

    -- Goto (ramais-digitro,8603,1)

    -- Executing Dial("Zap/4-1",
"SIP/drsouza|60|tTwW") in new stack

    -- Called drsouza

Mar 22 15:47:53 WARNING[1939]: chan_sip.c:1210
retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED]
for seqno 102 (Critical Request)

Mar 22 15:47:53 WARNING[1939]: chan_sip.c:1227
retrans_pkt: Hanging up call [EMAIL PROTECTED] - no
reply to our critical packet.

  == Everyone is busy/congested at this time
(1:0/0/1)

    -- Executing VoiceMail("Zap/4-1",
"8603") in new stack

    -- Playing 'vm-intro' (language 'br')

    -- Playing 'beep' (language 'br')

    -- Recording the message

    -- x=0, open writing: 
/var/spool/asterisk/voicemail/default/8603/INBOX/msg format: wav, 0x81cf9b8

    -- User hung up

    -- Recording was 1 seconds long but needs to be
at least 2 - abandoning

  == Spawn extension (ramais-digitro, 8603, 2) exited
non-zero on 'Zap/4-1'

    -- Hungup 'Zap/4-1'

    -- Starting simple switch on 'Zap/4-1'

Mar 22 15:48:20 NOTICE[27306]: chan_zap.c:6063
ss_thread: Got event 18 (Ring Begin)...

Mar 22 15:48:20 NOTICE[27306]: chan_zap.c:6063
ss_thread: Got event 2 (Ring/Answered)...

Mar 22 15:48:25 NOTICE[27306]: chan_zap.c:6063
ss_thread: Got event 18 (Ring Begin)...

    -- Executing Goto("Zap/4-1",
"ramais-digitro|8603|1") in new stack

    -- Goto (ramais-digitro,8603,1)

    -- Executing Dial("Zap/4-1",
"SIP/drsouza|60|tTwW") in new stack

    -- Called drsouza

Mar 22 15:48:25 WARNING[1939]: chan_sip.c:1210
retrans_pkt: Maximum retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102 (Critical Request)

Mar 22 15:48:25 WARNING[1939]: chan_sip.c:1227 retrans_pkt:
Hanging up call [EMAIL PROTECTED] - no reply to our
critical packet.

  == Everyone is busy/congested at this time
(1:0/0/1)

    -- Executing VoiceMail("Zap/4-1",
"8603") in new stack

    -- Playing 'vm-intro' (language 'br')

    -- Playing 'beep' (language 'br')

    -- Recording the message

    -- x=0, open writing: 
/var/spool/asterisk/voicemail/default/8603/INBOX/msg format: wav, 0x81cf9b8

    -- User hung up

  == Spawn extension (ramais-digitro, 8603, 2) exited
non-zero on 'Zap/4-1'

    -- Hungup 'Zap/4-1'

asterisk01*CLI>

 

 

Thanks for any help

Regards,



Filipe Mordhorst  
Joinville - SC - Brasil

 








smime.p7s
Description: S/MIME cryptographic signature
___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on call/phantom calls?

[Oscar Carriles]

I am confused due a side effect produced in my * installation.
It consists of 1 Sangoma A101 E1/lSDN PRI card connected to Telmex
service
16 analog phones thru SIP enabled SP5004 Micronet gateways & 4 SIP hard
phones.
Everything in a local network/no natting.
We are processing nearly 2000 calls/day outgoing/incoming
Everithing seems to be ok but after an hour or so I begin to see the
message “Maximum retries exceeded on call...”
On my logging console. This message continues to appear  with a climbing
frequency on different call ids till the entire system begin to
unregister my sip clients. Asterisk needs to be restarted as if it has
suffered  “a DOS attack”.

Prior to this situation arrives, I notice that “phantom calls” rings
phones but nobody there---
After a couple of weeks of debugging I notice that this situation could
be related to 3-way calling from the operator to other sip extensions.
This tranferred calls seems not to die after the normal operation of the
feature (flash/get tone/dial extension/speak with employee/hangup). I
have all my sip gateways set to support transfer, so SIP attended
transfer is done by the gateway and by zapata at the same time producing
the side effect?

Waiting some feedback
OCA 

 


-- 
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.375 / Virus Database: 267.15.2/251 - Release Date:
04/02/2006
 

___
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call.

[Peter Ankerstål]

Yes, using sip. The ports are forwarded. The calls going to the other asterisk
server works fine. The problem occurs only when people who are registred to my
server tries to call.
On Thu, 13 Oct 2005 08:30:17 +0100
"Steve Daniels" <[EMAIL PROTECTED]> wrote:

> Using SIP? IAX?
> 
> One way sound is usually a SIP and nat/firewall problem, make sure ports are 
> forwarded.
> 
> Steve
> - Original Message - 
> From: "Peter Ankerstål" <[EMAIL PROTECTED]>
> To: 
> Sent: Wednesday, October 12, 2005 10:39 PM
> Subject: [Asterisk-Users] Maximum retries exceeded on call.
> 
> 
> I have set up a asterisk-server behind NAT and peers to another asterisk
> and uses that one for outgoing calls. I have som clients on my asterisk
> and they could register to it well over internet. Not a problem. But when
> they try to call me the asterisk-server tells me this:
> 
> Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries 
> exceeded on call [EMAIL PROTECTED] for seqno 
> 32458501 (Non-critical Response)
> 
> Configs can be found at http://www.pulia.nu/~peter/asterisk/
> 
> When they call me they can hear me but I get no sound. Weird.
> Any Ideas?
> 
> 
> 
> -- 
> MVH
> Peter Ankerstål.
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
MVH
Peter Ankerstål.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call.

[Steve Daniels]

Using SIP? IAX?

One way sound is usually a SIP and nat/firewall problem, make sure ports are 
forwarded.


Steve
- Original Message - 
From: "Peter Ankerstål" <[EMAIL PROTECTED]>

To: 
Sent: Wednesday, October 12, 2005 10:39 PM
Subject: [Asterisk-Users] Maximum retries exceeded on call.


I have set up a asterisk-server behind NAT and peers to another asterisk
and uses that one for outgoing calls. I have som clients on my asterisk
and they could register to it well over internet. Not a problem. But when
they try to call me the asterisk-server tells me this:

Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries 
exceeded on call [EMAIL PROTECTED] for seqno 
32458501 (Non-critical Response)


Configs can be found at http://www.pulia.nu/~peter/asterisk/

When they call me they can hear me but I get no sound. Weird.
Any Ideas?



--
MVH
Peter Ankerstål.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on call.

[Peter Ankerstål]

I have set up a asterisk-server behind NAT and peers to another asterisk
and uses that one for outgoing calls. I have som clients on my asterisk
and they could register to it well over internet. Not a problem. But when
they try to call me the asterisk-server tells me this:

Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries 
exceeded on call [EMAIL PROTECTED] for seqno 32458501 (Non-critical Response)

Configs can be found at http://www.pulia.nu/~peter/asterisk/

When they call me they can hear me but I get no sound. Weird.
Any Ideas?



-- 
MVH
Peter Ankerstål.
___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] maximum retries exceeded on call

[Michael Häberle]

has somebody an advise.
Do I need to provide more information?

Regards
Michael

Michael Häberle wrote:

Hi,

I phone with phpagi and/or x-pro.
Sometimes I get this warning in the asterisk-console:
"maximum retries exceeded on call".
I noticed when this message shows up, asterisk hangs up the call (even 
when i'am in the middle of a call, according to our employess)


When they restart x-pro it seems to work properly again (at least some 
time).


Asterisk and the clients are in the same LAN.

I read the FAQ at voip-info.org but it didn't help.

Here is my sip.conf
--
[general]
context=telin
port=5060
bindaddr=0.0.0.0
srvlookup=yes
toos=lowdelay

allow=g726
allow=ulaw

rtptimeout=60
rtpholdtimeout=300

useragent=EASYCOM
nat=yes
-
after that comes the whole register-thing

here comes a sample user (all are the same)
-
[user]
context=telout
type=friend
secret=XXX
dtmfmode=rfc2833
host=dynamic
allow=all
canreinvite=no
-

in x-pro everything is standard (nothing changend but the 
network-settings and sip-proxy)


Since Iam neither a linux nor a asterisk-crack, I don't really have a 
clue what's going on.


Hope you can help me :)

Regards
Michael




--
Immosky AG

Service-Zentrale
Dufourstr. 5
CH-8702 Zollikon-Zürich

Tel+41 (0)43 344 52 52
Fax   +41 (0)43 344 52 58

www.immosky.ch
[EMAIL PROTECTED]

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] maximum retries exceeded on call

[Michael Häberle]

Hi,

I phone with phpagi and/or x-pro.
Sometimes I get this warning in the asterisk-console:
"maximum retries exceeded on call".
I noticed when this message shows up, asterisk hangs up the call (even 
when i'am in the middle of a call, according to our employess)


When they restart x-pro it seems to work properly again (at least some 
time).


Asterisk and the clients are in the same LAN.

I read the FAQ at voip-info.org but it didn't help.

Here is my sip.conf
--
[general]
context=telin
port=5060
bindaddr=0.0.0.0
srvlookup=yes
toos=lowdelay

allow=g726
allow=ulaw

rtptimeout=60
rtpholdtimeout=300

useragent=EASYCOM
nat=yes
-
after that comes the whole register-thing

here comes a sample user (all are the same)
-
[user]
context=telout
type=friend
secret=XXX
dtmfmode=rfc2833
host=dynamic
allow=all
canreinvite=no
-

in x-pro everything is standard (nothing changend but the 
network-settings and sip-proxy)


Since Iam neither a linux nor a asterisk-crack, I don't really have a 
clue what's going on.


Hope you can help me :)

Regards
Michael


--
Immosky AG

Service-Zentrale
Dufourstr. 5
CH-8702 Zollikon-Zürich

Tel+41 (0)43 344 52 52
Fax   +41 (0)43 344 52 58

www.immosky.ch
[EMAIL PROTECTED]

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded

[Joseph]

On Fri, 2005-07-15 at 06:37 -0400, Joseph wrote:
> Periodically I will get this type of message in the * log:
> 
> WARNING[18535]: Maximum retries exceeded on call 
> [EMAIL PROTECTED] for seqno 102 (Non-critical 
> Response)
> 
> The ip address listed sometimes is the * box itself and sometimes will 
> be a sip cisco sip phone.
> 
> When this happens often there will be lots of these messages, so 30 or 
> 40 in 2 minutes time.
> 
> I know it is a warning, but when this happens folks complain that calls 
> are dropped or they can not hear.
> 
> Asterisk suddenly lags in call completion and talk time.
> 
> How can I go about finding what causes this?
> 
> We have a normal call load of about 20 to 40 calls at a time, running 
> 1.0.7 stable.
> 
> We use the queue to handle incoming customer calls.
> And the AgentCallBackLogin.
> 
> Doing a google revealed little that seemed helpful or I did not google 
> right :)
> 
> I have thought of removing the mpg123 stuff and using raw files.
> But, I don't know that that is the real problem.
> 
> It seems especially strange that it complains about its on ip address 
> sometimes.
> 
> Any one else ever see this or have an idea where to trouble shoot it?

This setup has 2 nics. Would that make things any worse or is that
undesirable?

> 
-- 
respectfully, Joseph ===
-= **  =

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded

[Joseph]

Periodically I will get this type of message in the * log:

WARNING[18535]: Maximum retries exceeded on call 
[EMAIL PROTECTED] for seqno 102 (Non-critical 
Response)


The ip address listed sometimes is the * box itself and sometimes will 
be a sip cisco sip phone.


When this happens often there will be lots of these messages, so 30 or 
40 in 2 minutes time.


I know it is a warning, but when this happens folks complain that calls 
are dropped or they can not hear.


Asterisk suddenly lags in call completion and talk time.

How can I go about finding what causes this?

We have a normal call load of about 20 to 40 calls at a time, running 
1.0.7 stable.


We use the queue to handle incoming customer calls.
And the AgentCallBackLogin.

Doing a google revealed little that seemed helpful or I did not google 
right :)


I have thought of removing the mpg123 stuff and using raw files.
But, I don't know that that is the real problem.

It seems especially strange that it complains about its on ip address 
sometimes.


Any one else ever see this or have an idea where to trouble shoot it?

--

respectfully, Joseph

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded

[Rich Adamson]

> What does this mean?  I have a sipura 3000 with an analog line that I
> have created as a trunk.  Incoming calls make it to the sipura but not
> to the pbx.  However I can make outgoing calls but have no audio.  I
> thought it might be a codec issue so I set disallow= and
> commented out the "allow=".  I get the following in my logfile:
> build_route: Contact hop: satelliteout
> Jun 2 21:06:33 VERBOSE[2393]: -- SIP/satelliteout-af86 is ringing
> Jun 2 21:06:33 VERBOSE[2393]: -- SIP/satelliteout-af86 answered SIP/100-6eab
> Jun 2 21:06:33 VERBOSE[2393]: -- Attempting native bridge of
> SIP/100-6eab and SIP/satelliteout-af86
> Jun 2 21:06:39 WARNING[2393]: Maximum retries exceeded on call
> [EMAIL PROTECTED] for seqno 102 (Non-critical Response)
> 
> Satelliteout is my outbound trunk and the call is being made from extension 
> 100.
> Any idea what this means, I don't see anything that indicates an error
> when running asterisk -rv in the console, this is taken from the
> asterisk log files.
> 

Check to ensure canreinvite=no is defined in sip.conf entries for
both the sipura and your phone. Might also do disallow=all and allow=ulaw
in sip.conf and restart asterisk. See if either impacts the problem.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded

[Tim P]

What does this mean?  I have a sipura 3000 with an analog line that I
have created as a trunk.  Incoming calls make it to the sipura but not
to the pbx.  However I can make outgoing calls but have no audio.  I
thought it might be a codec issue so I set disallow= and
commented out the "allow=".  I get the following in my logfile:
build_route: Contact hop: satelliteout
Jun 2 21:06:33 VERBOSE[2393]: -- SIP/satelliteout-af86 is ringing
Jun 2 21:06:33 VERBOSE[2393]: -- SIP/satelliteout-af86 answered SIP/100-6eab
Jun 2 21:06:33 VERBOSE[2393]: -- Attempting native bridge of
SIP/100-6eab and SIP/satelliteout-af86
Jun 2 21:06:39 WARNING[2393]: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Non-critical Response)

Satelliteout is my outbound trunk and the call is being made from extension 100.
Any idea what this means, I don't see anything that indicates an error
when running asterisk -rv in the console, this is taken from the
asterisk log files.

Any help is greatly appreciated!
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call

[Andrew Furey]

> I see this message in my asterisk log sometimes.  Can someone explain to me
> what this means and how to correct the problem?
> 
> May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries
> exceeded on call 795fcf0c6
> [EMAIL PROTECTED] for seqno 18950 (Non-critical Response)
> May 17 13:39:12 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries
> exceeded on call 0ec0538d9
> [EMAIL PROTECTED] for seqno 40146 (Non-critical Response)
> May 17 13:39:19 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries
> exceeded on call 0ec0538d9
> [EMAIL PROTECTED] for seqno 40147 (Non-critical Response)

We tend to get this when asterisk tries to call a SIP extension which
has lost its connection for some reason (network troubles, power
outage, whatever). See if there are any calls being attempted at that
time...

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call

[Joel Jn-Francois]

None of our phones are being forwarded unless the phones are being 
forwarded unknowingly.

>We see this when person A forwards their phone to person B, who has
>forwarded their phone to Person A.
>so A->B->A
>or A->B->C->A and so on and so forth :)
Joel Jn-Francois wrote:
> I see this message in my asterisk log sometimes.  Can someone explain to
> me what this means and how to correct the problem?
>
> May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum
> retries exceeded on call 795fcf0c6
> [EMAIL PROTECTED] for seqno 18950 (Non-critical Response)
> May 17 13:39:12 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum
> retries exceeded on call 0ec0538d9
> [EMAIL PROTECTED] for seqno 40146 (Non-critical Response)
> May 17 13:39:19 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum
> retries exceeded on call 0ec0538d9
> [EMAIL PROTECTED] for seqno 40147 (Non-critical Response)
>
> Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call

[Asterisk]

We see this when person A forwards their phone to person B, who has 
forwarded their phone to Person A.

so A->B->A
or A->B->C->A and so on and so forth :)
No matter how many times I tell 'em ...
Julian
Joel Jn-Francois wrote:
I see this message in my asterisk log sometimes.  Can someone explain to 
me what this means and how to correct the problem?

May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum 
retries exceeded on call 795fcf0c6
[EMAIL PROTECTED] for seqno 18950 (Non-critical Response)
May 17 13:39:12 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum 
retries exceeded on call 0ec0538d9
[EMAIL PROTECTED] for seqno 40146 (Non-critical Response)
May 17 13:39:19 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum 
retries exceeded on call 0ec0538d9
[EMAIL PROTECTED] for seqno 40147 (Non-critical Response)

Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on call

[Joel Jn-Francois]

I see this message in my asterisk log sometimes.  Can someone explain to me 
what this means and how to correct the problem?

May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries 
exceeded on call 795fcf0c6
[EMAIL PROTECTED] for seqno 18950 (Non-critical Response)
May 17 13:39:12 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries 
exceeded on call 0ec0538d9
[EMAIL PROTECTED] for seqno 40146 (Non-critical Response)
May 17 13:39:19 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries 
exceeded on call 0ec0538d9
[EMAIL PROTECTED] for seqno 40147 (Non-critical Response)

Thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on call

[Walter Willis]

i am server with ser(iptel) and asterisk.

the prosess is:

the xlite the connect to ser, the ser redirect to asterisk and call for x100p

after the two minutes the call, the call is cut.

the error is:


Asterisk Ready.
*CLI> -- Executing Dial("SIP/sorcier.com.pe-41000490",
"Zap/1/499732") in new stack
-- Called 1/499732
-- Zap/1-1 answered SIP/sorcier.com.pe-41000490
-- Hungup 'Zap/1-1'
  == Spawn extension (incoming_ser_asterisk, 0499732, 1) exited
non-zero on 'SIP/sorcier.com.pe-41000490'
Apr 11 01:21:09 WARNING[1897]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 55926
(Critical Response)


wath happend?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on call

[Erick Weber V.]

Hi:
I have a asterisk server that shows the following Warning
Jan 24 10:23:37 WARNING[1116941120]: chan_sip.c:673 retrans_pkt: Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Non-critical Reque

If it show on the midel of the call rhe call will be droped or if it show at 
the begining of the call the call will show buisy ( No one is available to 
answer at this time)

Any help will be appreciate
Please help
Thanks
Erick W 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on call

[Me]

Any idea why I keep getting this on my Asterisk console? This is coming from
a Grandstream 101..

No NAT or anything, clean connection between phone and Asterisk on same
subnet via 100mbit.

***

Oct 26 15:19:52 WARNING[180235]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 102
(Non-critical Request)


--
Start Your Own ISP!
http://www.YourOwnISP.com



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call

[Eric C. Snowdeal III]

Eric C. Snowdeal III wrote: 

after registering the phones correctly and receiving a "200 o.k." 
message i can connect to other registered softphones and pstn 
endpoints [ via an voicepulse account ],  but after making the initial 
connection, i can't hear any sound and i get disconnected after 
getting the following error message:

chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call 
[EMAIL PROTECTED] for seqno 41270 

prompted by a recent email to the group [1] about setting the bindaddr, 
i took a closer look at the sip messages being sent back and forth and 
noticed that the contact header was incorrectly set to 127.0.0.1 in the 
200 o.k. message [2].  once i set the bindaddr to the * machine's public 
ip address everything worked fine and and contact header i.p.  address 
was set correctly.

what's odd, at least to me, is that unlike the recent email about a 
similar issue [1], my * box is on a non-natted, public ip address so i 
would have thought that keeping the default bindaddr  (0.0.0.0) would 
have worked, but obviously it didn't.

not sure how to interpret the dirth of responses, perhaps this was 
frighteningly obvious to everyone else.

[1] http://lists.digium.com/pipermail/asterisk-users/2004-June/051375.html
[2]
RECEIVE TIME: 7548279
RECEIVE << my.public.asterisk.ip:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
my.wan.ip:5060;rport;branch=z9hG4bKFE76D502C2BF11D8A741000D93C22AF4;received=my.wan.ip;rport=1029
From: snowdeal ;tag=1666554831
To: snowdeal ;tag=as7f7ed33f
Call-ID: [EMAIL PROTECTED]
CSeq: 43970 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 1800
Contact: ;expires=1800
Date: Sun, 20 Jun 2004 13:44:34 GMT
Content-Length: 0
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded w/SIP

[Andy Sackheim]

Brian:
 
Thanks!
 
I looked through the list and didn't see a 
correlation between what I was seeing and those parameters.  Must have 
missed it.
 
Thanks for your help.
 
Andy

  - Original Message - 
  From: 
  Brian K. West 
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, June 19, 2004 11:49 
  PM
  Subject: Re: [Asterisk-Users] Maximum 
  retries exceeded w/SIP 
  
  Usage of externip= and localnet= are what you are 
  looking for.
   
  These all have been covered more than once in the 
  mailing list...
   
  Remember GOOGLE IS YOUR FRIEND!! :P
   
  bkw
  
- Original Message - 
From: 
Andrew 
Sackheim 
To: [EMAIL PROTECTED] 

Sent: Saturday, June 19, 2004 9:29 
PM
Subject: [Asterisk-Users] Maximum 
retries exceeded w/SIP 

I struggled with this for several hours tonight.Turns out that if you have an * machine behind NAT, you must put the PUBLIC address in the bindaddr in sip.confIf you don't put it in, the Contact: header contains the NATted address and the sip phone can't get back to *.I don't know what happens if you mix and match sip phones on the local network -- it might not work unless the sipphone uses the public address as well. Hope this helps as I see this thread come up again and again... Andy ---Steve,

Sure, I could put all my machines on the public Internet, but that defeats the 
purpose of having a firewall in the first place.

As an alternative, I could only place the * server on the outside, but I'd 
rather not give the script-kiddies another box to pound.

Steve Totaro wrote:

> Can you disable your firewall?  i am about to start this phase of asterisk
> an would like help from one newbie to another.  otherwise this newbie will
> let you know how i did it.
> 
> 
> - Original Message -
> From: "Brad Waite" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Saturday, September 20, 2003 9:07 AM
> Subject: [Asterisk-Users] Maximum retries exceeded w/SIP
> 
> 
> 
>>First of all, I'd like to send a big "thank you" to all the folks who have
>>helped me get this far.
>>
>>Now on to the next problem.  Here's my current network setup:
>>
>>
>>The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
>>  |
>>  +--- Laptop (public IP)
>>
>>natd is set up with the following rules:
>>
>>redirect_port udp 10.0.0.253:1-2 1-2
>>redirect_port udp 10.0.0.253:5060 5060
>>
>>* is set up with the demo/sandbox config.
>>
>>I'm using XLite as my SIP client and have configured it on PC to work with
> 
> *.
> 
>>I'm able to do everything I've tried so far.  I should, though - I'm on
> 
> the inside.
> 
>>However, when trying to make a call from the outside (via Laptop),
> 
> something's
> 
>>breaking.  I've set up the SIP proxy in XLite to be the external interface
> 
> on
> 
>>the firewall, and am able to log into the proxy without difficulty.  And
> 
> while I
> 
>>can begin conversations, I can't keep them going for long.
>>
>>For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I
> 
> get most
> 
>>of the "demo-abouttotry" message - "I am about to attempt an IAX
> 
> connection to a
> 
>>demonstration server located at Di" - at which point it gets cut off.  The
>>console spits out the following error:
>>
>>File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call
>>[EMAIL PROTECTED] for seqno 12384
> 
> (Response)
> 
>>
>>Any ideas what could be going on?  My first guess is the firewall, but I
> 
> can't
> 
>>figure out why some of the packets would get through while others
> 
> apparently are
> 
>>not.  I'm at a loss.
>>
>>Brad Waite
>>aka HankPoacher
>>
>>___
>>Asterisk-Users mailing list
>>[EMAIL PROTECTED]
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

Re: [Asterisk-Users] Maximum retries exceeded w/SIP

[Brian K. West]

Usage of externip= and localnet= are what you are 
looking for.
 
These all have been covered more than once in the 
mailing list...
 
Remember GOOGLE IS YOUR FRIEND!! :P
 
bkw

  - Original Message - 
  From: 
  Andrew Sackheim 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, June 19, 2004 9:29 
  PM
  Subject: [Asterisk-Users] Maximum retries 
  exceeded w/SIP 
  
  I struggled with this for several hours tonight.Turns out that if you have an * machine behind NAT, you must put the PUBLIC address in the bindaddr in sip.confIf you don't put it in, the Contact: header contains the NATted address and the sip phone can't get back to *.I don't know what happens if you mix and match sip phones on the local network -- it might not work unless the sipphone uses the public address as well. Hope this helps as I see this thread come up again and again... Andy ---Steve,

Sure, I could put all my machines on the public Internet, but that defeats the 
purpose of having a firewall in the first place.

As an alternative, I could only place the * server on the outside, but I'd 
rather not give the script-kiddies another box to pound.

Steve Totaro wrote:

> Can you disable your firewall?  i am about to start this phase of asterisk
> an would like help from one newbie to another.  otherwise this newbie will
> let you know how i did it.
> 
> 
> - Original Message -
> From: "Brad Waite" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Saturday, September 20, 2003 9:07 AM
> Subject: [Asterisk-Users] Maximum retries exceeded w/SIP
> 
> 
> 
>>First of all, I'd like to send a big "thank you" to all the folks who have
>>helped me get this far.
>>
>>Now on to the next problem.  Here's my current network setup:
>>
>>
>>The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
>>  |
>>  +--- Laptop (public IP)
>>
>>natd is set up with the following rules:
>>
>>redirect_port udp 10.0.0.253:1-2 1-2
>>redirect_port udp 10.0.0.253:5060 5060
>>
>>* is set up with the demo/sandbox config.
>>
>>I'm using XLite as my SIP client and have configured it on PC to work with
> 
> *.
> 
>>I'm able to do everything I've tried so far.  I should, though - I'm on
> 
> the inside.
> 
>>However, when trying to make a call from the outside (via Laptop),
> 
> something's
> 
>>breaking.  I've set up the SIP proxy in XLite to be the external interface
> 
> on
> 
>>the firewall, and am able to log into the proxy without difficulty.  And
> 
> while I
> 
>>can begin conversations, I can't keep them going for long.
>>
>>For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I
> 
> get most
> 
>>of the "demo-abouttotry" message - "I am about to attempt an IAX
> 
> connection to a
> 
>>demonstration server located at Di" - at which point it gets cut off.  The
>>console spits out the following error:
>>
>>File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call
>>[EMAIL PROTECTED] for seqno 12384
> 
> (Response)
> 
>>
>>Any ideas what could be going on?  My first guess is the firewall, but I
> 
> can't
> 
>>figure out why some of the packets would get through while others
> 
> apparently are
> 
>>not.  I'm at a loss.
>>
>>Brad Waite
>>aka HankPoacher
>>
>>___
>>Asterisk-Users mailing list
>>[EMAIL PROTECTED]
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

[Asterisk-Users] Maximum retries exceeded w/SIP

[Andrew Sackheim]

I struggled with this for several hours tonight.Turns out that if you have an * machine behind NAT, you must put the PUBLIC address in the bindaddr in sip.confIf you don't put it in, the Contact: header contains the NATted address and the sip phone can't get back to *.I don't know what happens if you mix and match sip phones on the local network -- it might not work unless the sipphone uses the public address as well. Hope this helps as I see this thread come up again and again... Andy ---Steve,

Sure, I could put all my machines on the public Internet, but that defeats the 
purpose of having a firewall in the first place.

As an alternative, I could only place the * server on the outside, but I'd 
rather not give the script-kiddies another box to pound.

Steve Totaro wrote:

> Can you disable your firewall?  i am about to start this phase of asterisk
> an would like help from one newbie to another.  otherwise this newbie will
> let you know how i did it.
> 
> 
> - Original Message -
> From: "Brad Waite" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Saturday, September 20, 2003 9:07 AM
> Subject: [Asterisk-Users] Maximum retries exceeded w/SIP
> 
> 
> 
>>First of all, I'd like to send a big "thank you" to all the folks who have
>>helped me get this far.
>>
>>Now on to the next problem.  Here's my current network setup:
>>
>>
>>The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
>>  |
>>  +--- Laptop (public IP)
>>
>>natd is set up with the following rules:
>>
>>redirect_port udp 10.0.0.253:1-2 1-2
>>redirect_port udp 10.0.0.253:5060 5060
>>
>>* is set up with the demo/sandbox config.
>>
>>I'm using XLite as my SIP client and have configured it on PC to work with
> 
> *.
> 
>>I'm able to do everything I've tried so far.  I should, though - I'm on
> 
> the inside.
> 
>>However, when trying to make a call from the outside (via Laptop),
> 
> something's
> 
>>breaking.  I've set up the SIP proxy in XLite to be the external interface
> 
> on
> 
>>the firewall, and am able to log into the proxy without difficulty.  And
> 
> while I
> 
>>can begin conversations, I can't keep them going for long.
>>
>>For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I
> 
> get most
> 
>>of the "demo-abouttotry" message - "I am about to attempt an IAX
> 
> connection to a
> 
>>demonstration server located at Di" - at which point it gets cut off.  The
>>console spits out the following error:
>>
>>File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call
>>[EMAIL PROTECTED] for seqno 12384
> 
> (Response)
> 
>>
>>Any ideas what could be going on?  My first guess is the firewall, but I
> 
> can't
> 
>>figure out why some of the packets would get through while others
> 
> apparently are
> 
>>not.  I'm at a loss.
>>
>>Brad Waite
>>aka HankPoacher
>>
>>___
>>Asterisk-Users mailing list
>>[EMAIL PROTECTED]
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

Re: [Asterisk-Users] Maximum retries exceeded on call

[Eric C. Snowdeal III]

Holger Schurig wrote:
i'm new to asterisk and am having trouble placing outbound calls.  i
   

Bug Grandstream so that they finally fix their buggy software.
The GS phone sends occassional SIP packets to port 0, not to port 5060, as 
tcpdump or (better) ethereal will show you.

There's a page on this at voip-info.org.
 

thanks for the heads-up about grandstream, but as i stated in the 
original message, i'm using xten lite softphones.   hopefully this is 
the approproriate forum for this question; i believe this is not an xten 
configuration issue because i can connect to a ser/rtproxy/nathelper 
server without problems and i can connect directly to a voicepulse 
account, which leads me to believe that this is an * configuration 
problem on my part.  less likely, i suppose, is the chance that * isn't 
as robust in handling nat than ser or whatever voicepulse is running.

given the configuration files that i posted in the original message, are 
there any changes that i should make?  certainly the asterisk faq makes 
the solution seems straighforward [1]:

"Most likely you have a SIP client behind NAT that is trying to 
communicate with Asterisk without having the "nat=yes" setting in place 
in sip.conf. Another cause for this could be related to a user device 
that has an sip entry but has been physically removed (switched off or 
LAN-disconnected)."

but as my original message showed, i do have nat=yes in my sip.conf and 
i don't believe the latter scenario is true.

any help is greatly appreciated.
[1] http://www.voip-info.org/wiki-Asterisk+FAQ
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call

[Holger Schurig]

> i'm new to asterisk and am having trouble placing outbound calls.  i

Bug Grandstream so that they finally fix their buggy software.

The GS phone sends occassional SIP packets to port 0, not to port 5060, as 
tcpdump or (better) ethereal will show you.

There's a page on this at voip-info.org.


I'd love to see that we e-mail in MASSES to Grandstream, so that they fix 
their software. The problem is that it doesn't happen always. Try 
[EMAIL PROTECTED] :-)

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on call

[Eric C. Snowdeal III]

i'm new to asterisk and am having trouble placing outbound calls.  i 
know this topic has been discussed  ad nauseum in the past [1] , but i 
can't seem to find a workaround and i'm wondering if my newbie-ness is 
getting the best of me. 

after registering the phones correctly and receiving a "200 o.k." 
message i can connect to other registered softphones and pstn endpoints 
[ via an voicepulse account ],  but after making the initial connection, 
i can't hear any sound and i get disconnected after getting the 
following error message:

chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call 
[EMAIL PROTECTED] for seqno 41270  

i've compiled the stock asterisk tarball on a redhat 7.3 box with a 
public ip address.  the clients are xten lite softphone's running on 
ibooks with os 10.3.4.  the clients are natted behind a  linksys wrt54g 
wireless router running the sveasoft [2] firmware.  i'm perplexed, 
because i can get things to work fine if i use ser/rtpproxy instead of 
asterisk.  i can also connect directly to my voicepulse connect account 
with the xten softphone and things work great.  so i think i have the 
xten client configured properly and i know that the sveasoft firmware 
isn't throwing a monkey wrench into the picture.  i suppose i could 
configure ser to "front" asterisk since it appears to deal with the nat, 
but i'm wondering if i'm missing something basic.

my channel config files look like the  following:
sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind SIP channel to
context = default   ; Default context for incoming calls
disallow=all; Disallow all codecs
allow=ulaw  ; Allow codecs in order of preference
[2000]
type=friend   ; This device takes and makes calls
username=2000 ; Username on device
secret=supersecret ; Password for device
host=dynamic  ; This host is not on the same IP addr every time
context=from-sip-internal
nat=yes
canreinvite=no; Typically set to NO if behind NAT
qualify=500
[2001]
type=friend   ; This device takes and makes calls
username=2001 ; Username on device
secret=supersecret2 ; Password for device
host=dynamic  ; This host is not on the same IP addr every time
context=from-sip-internal
nat=yes
canreinvite=no; Typically set to NO if behind NAT
qualify=500
iax.conf
[general]
port=5036
bandwidth=low
disallow=all; Disallow all codecs
allow=ulaw  ; Allow codecs in order of preference
jitterbuffer=no
[voicepulse]
context = voicepulse-in
secret=topsecrect
auth=md5
type=friend
host=gw5.voicepulse.com
[1] 
http://www.google.com/search?q=retrans_pkt:+Maximum+retries+exceeded+on+call++site:http://lists.digium.com&hl=en&lr=&ie=UTF-8&start=10&sa=N

[2] http://www.sveasoft.com/modules/phpBB2/index.php
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Maximum retries exceeded problem...

[Michael Picher]

Also, working this a bit more...  if i do the echo 
test (extension 600) i get sorta the same thing...
 
== Spawn extension (default, asterisk, 1) exited non-zero 
on 'SIP/520-a25e'May  4 09:15:51 NOTICE[1125329600]: chan_sip.c:5655 
handle_request: Unknown SIPcommand 'PUBLISH' from 
'192.168.100.12'    -- Executing Playback("SIP/520-1a68", 
"demo-echotest") in new stack    -- Playing 'demo-echotest' 
(language 'en')May  4 09:15:58 WARNING[1125329600]: chan_sip.c:497 
retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 2 (Response)  == Spawn extension (default, 600, 1) exited 
non-zero on 'SIP/520-1a68'


From: Michael Picher [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, May 04, 2004 9:41 AMTo: 
[EMAIL PROTECTED]Subject: [Asterisk-Users] Maximum 
retries exceeded problem...

Searched the 
archives thoroughly... 
 
Can't find this 
specific problem...
 
Simple setup with 
Asterisk on RedHat.  No voice cards in the box, 2 SNOM 200 
phones...
 
Phones seem to work 
well, can leave VM, Message Waiting Indicator lights up but when I try to 
retrieve messages the call terminates and the following 
happens:
 
 
-- Executing 
VoiceMailMain("SIP/520-a25e", "Mike") in new stack    -- 
Playing 'vm-login' (language 'en')May  4 07:58:07 WARNING[1125329600]: 
chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 2 (Response)May  4 07:58:07 WARNING[1217602880]: 
app_voicemail.c:2748 vm_execmain: Couldn'tread username  == Spawn 
extension (default, asterisk, 1) exited non-zero on 
'SIP/520-a25e'asterisk*CLI>
 
Pertinent section of 
extensions.conf
 

  
exten => 504,1,Dial,sip/${EXTEN}|10
  
exten => 504,2,Voicemail(u504)
  
exten => 504,102,Voicemail(b504)
  
exten => 504,103,Hangup
  
exten => 520,1,Dial,sip/${EXTEN}|10
  
exten => 520,2,Voicemail(u520)
  
exten => 520,102,Voicemail(b520)
  
exten => 520,103,Hangup
  
exten => 
asterisk,1,VoicemailMain(${CALLERIDNUM})
 
Pertinent section of 
voicemail.conf
  
504 => 504,Tech Desk,[EMAIL PROTECTED]
  
520 => 520,Mike 
Picher,[EMAIL PROTECTED]

RE: [Asterisk-Users] Maximum retries exceeded problem...

[Michael Picher]

Sorry, forgot to include that...  Seems to be set right for the Snom phones
(from what I could gather).

[520]
type=friend
secret=blah
host=dynamic
callerid=Mike
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=192.168.0.12
mailbox=520 ; Mailbox for message waiting indicator
;restrictcid=yes; To have the callerid restriced -> sent as
ANI

[504]
type=friend
secret=blah
host=dynamic
callerid=TechDesk
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=192.168.0.13
mailbox=504 ; Mailbox for message waiting indicator
;restrictcid=yes; To have the callerid restriced -> sent as
ANI
 

-Original Message-
From: Justin Carlson [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, May 04, 2004 5:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Maximum retries exceeded problem...

I don't think your DTMF is set right look in sip.conf for the dtmf directive
for your phones.

cheers!


On Tue, 2004-05-04 at 13:41, Michael Picher wrote:
> Searched the archives thoroughly... 
>  
> Can't find this specific problem...
>  
> Simple setup with Asterisk on RedHat.  No voice cards in the box, 2 
> SNOM 200 phones...
>  
> Phones seem to work well, can leave VM, Message Waiting Indicator 
> lights up but when I try to retrieve messages the call terminates and 
> the following happens:
>  
>  
> -- Executing VoiceMailMain("SIP/520-a25e", "Mike") in new stack
> -- Playing 'vm-login' (language 'en') May  4 07:58:07 
> WARNING[1125329600]: chan_sip.c:497 retrans_pkt:
> Maximum retries
>  exceeded on call [EMAIL PROTECTED] for seqno 2 
> (Response
> )
> May  4 07:58:07 WARNING[1217602880]: app_voicemail.c:2748 vm_execmain:
> Couldn't
> read username
>   == Spawn extension (default, asterisk, 1) exited non-zero on 
> 'SIP/520-a25e'
> asterisk*CLI>
>  
> Pertinent section of extensions.conf
>  
> 
>   exten => 504,1,Dial,sip/${EXTEN}|10
> 
>   exten => 504,2,Voicemail(u504)
> 
>   exten => 504,102,Voicemail(b504)
> 
>   exten => 504,103,Hangup
> 
>   exten => 520,1,Dial,sip/${EXTEN}|10
> 
>   exten => 520,2,Voicemail(u520)
> 
>   exten => 520,102,Voicemail(b520)
> 
>   exten => 520,103,Hangup
> 
>   exten => asterisk,1,VoicemailMain(${CALLERIDNUM})
> 
>  
> Pertinent section of voicemail.conf
> 
>   504 => 504,Tech Desk,[EMAIL PROTECTED]
> 
>   520 => 520,Mike Picher,[EMAIL PROTECTED]
> 
> 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded problem...

[Justin Carlson]

I don't think your DTMF is set right look in sip.conf for the dtmf
directive for your phones.

cheers!


On Tue, 2004-05-04 at 13:41, Michael Picher wrote:
> Searched the archives thoroughly... 
>  
> Can't find this specific problem...
>  
> Simple setup with Asterisk on RedHat.  No voice cards in the box, 2
> SNOM 200 phones...
>  
> Phones seem to work well, can leave VM, Message Waiting Indicator
> lights up but when I try to retrieve messages the call terminates and
> the following happens:
>  
>  
> -- Executing VoiceMailMain("SIP/520-a25e", "Mike") in new stack
> -- Playing 'vm-login' (language 'en')
> May  4 07:58:07 WARNING[1125329600]: chan_sip.c:497 retrans_pkt:
> Maximum retries
>  exceeded on call [EMAIL PROTECTED] for seqno 2
> (Response
> )
> May  4 07:58:07 WARNING[1217602880]: app_voicemail.c:2748 vm_execmain:
> Couldn't
> read username
>   == Spawn extension (default, asterisk, 1) exited non-zero on
> 'SIP/520-a25e'
> asterisk*CLI>
>  
> Pertinent section of extensions.conf
>  
> 
>   exten => 504,1,Dial,sip/${EXTEN}|10
> 
>   exten => 504,2,Voicemail(u504)
> 
>   exten => 504,102,Voicemail(b504)
> 
>   exten => 504,103,Hangup
> 
>   exten => 520,1,Dial,sip/${EXTEN}|10
> 
>   exten => 520,2,Voicemail(u520)
> 
>   exten => 520,102,Voicemail(b520)
> 
>   exten => 520,103,Hangup
> 
>   exten => asterisk,1,VoicemailMain(${CALLERIDNUM})
> 
>  
> Pertinent section of voicemail.conf
> 
>   504 => 504,Tech Desk,[EMAIL PROTECTED]
> 
>   520 => 520,Mike Picher,[EMAIL PROTECTED]
> 
> 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded problem...

[Michael Picher]

Searched the 
archives thoroughly... 
 
Can't find this 
specific problem...
 
Simple setup with 
Asterisk on RedHat.  No voice cards in the box, 2 SNOM 200 
phones...
 
Phones seem to work 
well, can leave VM, Message Waiting Indicator lights up but when I try to 
retrieve messages the call terminates and the following 
happens:
 
 
-- Executing 
VoiceMailMain("SIP/520-a25e", "Mike") in new stack    -- 
Playing 'vm-login' (language 'en')May  4 07:58:07 WARNING[1125329600]: 
chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 2 (Response)May  4 07:58:07 WARNING[1217602880]: 
app_voicemail.c:2748 vm_execmain: Couldn'tread username  == Spawn 
extension (default, asterisk, 1) exited non-zero on 
'SIP/520-a25e'asterisk*CLI>
 
Pertinent section of 
extensions.conf
 

  
exten => 504,1,Dial,sip/${EXTEN}|10
  
exten => 504,2,Voicemail(u504)
  
exten => 504,102,Voicemail(b504)
  
exten => 504,103,Hangup
  
exten => 520,1,Dial,sip/${EXTEN}|10
  
exten => 520,2,Voicemail(u520)
  
exten => 520,102,Voicemail(b520)
  
exten => 520,103,Hangup
  
exten => 
asterisk,1,VoicemailMain(${CALLERIDNUM})
 
Pertinent section of 
voicemail.conf
  
504 => 504,Tech Desk,[EMAIL PROTECTED]
  
520 => 520,Mike 
Picher,[EMAIL PROTECTED]

Re: [Asterisk-Users] Maximum retries exceeded on call

[Barry Fawthrop]

check context perhaps try include in the extensions.conf




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call

[Edmund A. Hintz]

On Tue, Mar 16, 2004, Barry Fawthrop thus spake:

>check NAT setting  try taking it out of sip.conf, that worked for me

Nope. My bad-shoulda said up front that I've tried both with and
without nat=yes in sip.conf, no difference in symptoms.

Regards,

Ed Hintz
[EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded on call

[Barry Fawthrop]

check NAT setting  try taking it out of sip.conf, that worked for me
Barry

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded on call

[Edmund A. Hintz]

Running * with default config files except for sip.conf. Any call made is
dropped 5 seconds after connection, with the following messages:

Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call 6C94C1B1-77C4-11D8-91FB-
[EMAIL PROTECTED] for seqno 48221 (Response)
  == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7'
Mar 17 16:37:47 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call 6C94C1B1-77C4-11D8-91FB-
[EMAIL PROTECTED] for seqno 102 (Request)
Mar 17 16:43:43 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Request)

Setup is OBSD 3.4, asterisk 0.7.2, X-Lite softphone, and Cisco ATA186. PF
is disabled entirely, and all nodes are behind the same NAT, on the same
LAN. I see all kinds of posts in the archives with this problem, but no
clear solution. Suggestions?

Regards,

Ed Hintz
[EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] maximum retries exceeded

[jrhopper]

Hi robert,

I found that the disallow=all and then specify a codec with allow= was required in 
sip.conf.

[17471234567]
type=friend
username=17471234567
secret=censored
host=dynamic
nat=yes
disallow=all
allow=ulaw

Jon Hopper

robert ivanc <[EMAIL PROTECTED]> wrote ..
> Hi,
> 
> i've just got 2 grandstream phones and when I try to connect them with
> * 
> I get the following:
> 
>  -- Playing 'demo-abouttotry' (language 'en')
> WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
> exceeded on call [EMAIL PROTECTED] for 
> seqno 59134 (Response)
> 
> I've seen there was some discussion on this already but i couldn't find
> any resolution. Can anyone help?
> 
> Regards,
>   Robert Ivanc
> 
> 
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] maximum retries exceeded

[robert ivanc]

Hi,

i've just got 2 grandstream phones and when I try to connect them with * 
I get the following:

-- Playing 'demo-abouttotry' (language 'en')
WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries 
exceeded on call [EMAIL PROTECTED] for 
seqno 59134 (Response)

I've seen there was some discussion on this already but i couldn't find 
any resolution. Can anyone help?

Regards,
 Robert Ivanc


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded w/SIP

[Stephen Varga]

On Sat, 2003-09-20 at 23:00, Brad Waite wrote:
> Steve,
> 
> If that's the case, why is it that I could get the first 6 seconds of the 
> demo-abouttotry message?

RTP requires two one way UDP streams.

phone -> asterisk
phone <- asterisk

The RTP stream can be routed from the * box to the phone, but not the
other way (unless you did what you stated below). So essentially you
have a one-way conversation.

> As it turns out, if I set up a static route for my inside network on Laptop with 
> the external interface of the firewall as the gateway, everything works fine. 
> Of course, I had to turn off my anti-spoofing rules.

I am guessing you want to have a phone somewhere else on the Internet so
this solution does not meet your requirements.

> And what's the nat=yes option supposed to do in sip.conf?

I don't know the answer to that one. I am new the *, and have already
started down the path that you are going and wanted to help so you don't
have to repeat all troubles I had.

It sounds like you more than one real IP address to work with, if that
is the case there may be a way to make it work in your setup. Let me
know.

Steve

> Brad
> 
> 
> Stephen Varga wrote:
> 
> > Unfortunetly this setup does not work, when * sends SDP info in the
> > INVITE process on how to establish the audio session *'s real IP address
> > is in the packet and the outside phone tries to connect to this IP
> > address, which of course is unreachable because of the firewall. For
> > this to work you need to move * to the firewall and the firewall's ip
> > address in the SIP.CONF file.
> > 
> > HTH,
> > Steve
> > 
> > On Sat, 2003-09-20 at 12:07, Brad Waite wrote:
> > 
> >>First of all, I'd like to send a big "thank you" to all the folks who have 
> >>helped me get this far.
> >>
> >>Now on to the next problem.  Here's my current network setup:
> >>
> >>
> >>The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
> >>  |
> >>  +--- Laptop (public IP)
> >>
> >>natd is set up with the following rules:
> >>
> >>redirect_port udp 10.0.0.253:1-2 1-2
> >>redirect_port udp 10.0.0.253:5060 5060
> >>
> >>* is set up with the demo/sandbox config.
> >>
> >>I'm using XLite as my SIP client and have configured it on PC to work with *. 
> >>I'm able to do everything I've tried so far.  I should, though - I'm on the inside.
> >>
> >>However, when trying to make a call from the outside (via Laptop), something's 
> >>breaking.  I've set up the SIP proxy in XLite to be the external interface on 
> >>the firewall, and am able to log into the proxy without difficulty.  And while I 
> >>can begin conversations, I can't keep them going for long.
> >>
> >>For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I get 
> >>most 
> >>of the "demo-abouttotry" message - "I am about to attempt an IAX connection to a 
> >>demonstration server located at Di" - at which point it gets cut off.  The 
> >>console spits out the following error:
> >>
> >>File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call 
> >>[EMAIL PROTECTED] for seqno 12384 (Response)
> >>
> >>
> >>Any ideas what could be going on?  My first guess is the firewall, but I can't 
> >>figure out why some of the packets would get through while others apparently are 
> >>not.  I'm at a loss.
> >>
> >>Brad Waite
> >>aka HankPoacher
> >>
> >>___
> >>Asterisk-Users mailing list
> >>[EMAIL PROTECTED]
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> > 
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded w/SIP

[Brad Waite]

Steve,

If that's the case, why is it that I could get the first 6 seconds of the 
demo-abouttotry message?

As it turns out, if I set up a static route for my inside network on Laptop with 
the external interface of the firewall as the gateway, everything works fine. 
Of course, I had to turn off my anti-spoofing rules.

And what's the nat=yes option supposed to do in sip.conf?

Brad

Stephen Varga wrote:

Unfortunetly this setup does not work, when * sends SDP info in the
INVITE process on how to establish the audio session *'s real IP address
is in the packet and the outside phone tries to connect to this IP
address, which of course is unreachable because of the firewall. For
this to work you need to move * to the firewall and the firewall's ip
address in the SIP.CONF file.
HTH,
Steve
On Sat, 2003-09-20 at 12:07, Brad Waite wrote:

First of all, I'd like to send a big "thank you" to all the folks who have 
helped me get this far.

Now on to the next problem.  Here's my current network setup:

The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
 |
 +--- Laptop (public IP)
natd is set up with the following rules:

redirect_port udp 10.0.0.253:1-2 1-2
redirect_port udp 10.0.0.253:5060 5060
* is set up with the demo/sandbox config.

I'm using XLite as my SIP client and have configured it on PC to work with *. 
I'm able to do everything I've tried so far.  I should, though - I'm on the inside.

However, when trying to make a call from the outside (via Laptop), something's 
breaking.  I've set up the SIP proxy in XLite to be the external interface on 
the firewall, and am able to log into the proxy without difficulty.  And while I 
can begin conversations, I can't keep them going for long.

For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I get most 
of the "demo-abouttotry" message - "I am about to attempt an IAX connection to a 
demonstration server located at Di" - at which point it gets cut off.  The 
console spits out the following error:

File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call 
[EMAIL PROTECTED] for seqno 12384 (Response)

Any ideas what could be going on?  My first guess is the firewall, but I can't 
figure out why some of the packets would get through while others apparently are 
not.  I'm at a loss.

Brad Waite
aka HankPoacher
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded w/SIP

[Rich Adamson]

Brad,

I've played with XLite, but not with a firewall in this direction, so 
my comments might be off base.

> redirect_port udp 10.0.0.253:1-2 1-2
> redirect_port udp 10.0.0.253:5060 5060
> 
> * is set up with the demo/sandbox config.
> 
> I'm using XLite as my SIP client and have configured it on PC to work with *. 
> I'm able to do everything I've tried so far.  I should, though - I'm on the inside.
> 
> However, when trying to make a call from the outside (via Laptop), something's 
> breaking.  I've set up the SIP proxy in XLite to be the external interface on 
> the firewall, and am able to log into the proxy without difficulty.  And while I 
> can begin conversations, I can't keep them going for long.

I'd guess that udp/5060 is working fine, but the voice channel is being
dropped for a couple of possible reasons. The Xlite doc suggests the voice
channel will be using udp/8000-8006 where 8000 & 8001 are used for line #1,
etc. Based on the redirect_port statement above, I wonder if one-half of
the voice port is being blocked (and therefore times out), or, nat table
timeout might might be an issue.

> Any ideas what could be going on?  My first guess is the firewall, but I can't 
> figure out why some of the packets would get through while others apparently are 
> not.  I'm at a loss.

I'd download ethereal (or whatever other sniffer you'd like) and watch the
flow of packets. It should give you a pretty good clue what's happening
for real.

I'm not so sure you're going to want to live with direction that you're
heading (asterisk on the inside) as the nat function is going to limit
what can be done.  Example, even if you get this to work, trying to make
any other call through nat while the first one is happening will be a
problem; the first call nails up udp/5060, but the second call will have
the udp/5060 nat'ed to some other port which will fail.

Reversing the role of * and the laptop will work, and many others have that
very implementation working for a single instance of Xlite.

Depending upon what your real objectives are for *, I'd suggest either
moving * to the outside, or add another NIC to * and placing it on the
outside. You should be able to lock down that external interface in such
a way as to only allow selected tcp/udp ports to be used.



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded w/SIP

[Stephen Varga]

Unfortunetly this setup does not work, when * sends SDP info in the
INVITE process on how to establish the audio session *'s real IP address
is in the packet and the outside phone tries to connect to this IP
address, which of course is unreachable because of the firewall. For
this to work you need to move * to the firewall and the firewall's ip
address in the SIP.CONF file.

HTH,
Steve

On Sat, 2003-09-20 at 12:07, Brad Waite wrote:
> First of all, I'd like to send a big "thank you" to all the folks who have 
> helped me get this far.
> 
> Now on to the next problem.  Here's my current network setup:
> 
> 
> The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
>   |
>   +--- Laptop (public IP)
> 
> natd is set up with the following rules:
> 
> redirect_port udp 10.0.0.253:1-2 1-2
> redirect_port udp 10.0.0.253:5060 5060
> 
> * is set up with the demo/sandbox config.
> 
> I'm using XLite as my SIP client and have configured it on PC to work with *. 
> I'm able to do everything I've tried so far.  I should, though - I'm on the inside.
> 
> However, when trying to make a call from the outside (via Laptop), something's 
> breaking.  I've set up the SIP proxy in XLite to be the external interface on 
> the firewall, and am able to log into the proxy without difficulty.  And while I 
> can begin conversations, I can't keep them going for long.
> 
> For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I get 
> most 
> of the "demo-abouttotry" message - "I am about to attempt an IAX connection to a 
> demonstration server located at Di" - at which point it gets cut off.  The 
> console spits out the following error:
> 
> File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call 
> [EMAIL PROTECTED] for seqno 12384 (Response)
> 
> 
> Any ideas what could be going on?  My first guess is the firewall, but I can't 
> figure out why some of the packets would get through while others apparently are 
> not.  I'm at a loss.
> 
> Brad Waite
> aka HankPoacher
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded w/SIP

[Brad Waite]

Steve,

Sure, I could put all my machines on the public Internet, but that defeats the 
purpose of having a firewall in the first place.

As an alternative, I could only place the * server on the outside, but I'd 
rather not give the script-kiddies another box to pound.

Steve Totaro wrote:

Can you disable your firewall?  i am about to start this phase of asterisk
an would like help from one newbie to another.  otherwise this newbie will
let you know how i did it.
- Original Message -
From: "Brad Waite" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 20, 2003 9:07 AM
Subject: [Asterisk-Users] Maximum retries exceeded w/SIP


First of all, I'd like to send a big "thank you" to all the folks who have
helped me get this far.
Now on to the next problem.  Here's my current network setup:

The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
 |
 +--- Laptop (public IP)
natd is set up with the following rules:

redirect_port udp 10.0.0.253:1-2 1-2
redirect_port udp 10.0.0.253:5060 5060
* is set up with the demo/sandbox config.

I'm using XLite as my SIP client and have configured it on PC to work with
*.

I'm able to do everything I've tried so far.  I should, though - I'm on
the inside.

However, when trying to make a call from the outside (via Laptop),
something's

breaking.  I've set up the SIP proxy in XLite to be the external interface
on

the firewall, and am able to log into the proxy without difficulty.  And
while I

can begin conversations, I can't keep them going for long.

For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I
get most

of the "demo-abouttotry" message - "I am about to attempt an IAX
connection to a

demonstration server located at Di" - at which point it gets cut off.  The
console spits out the following error:
File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 12384
(Response)

Any ideas what could be going on?  My first guess is the firewall, but I
can't

figure out why some of the packets would get through while others
apparently are

not.  I'm at a loss.

Brad Waite
aka HankPoacher
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Maximum retries exceeded w/SIP

[Steve Totaro]

Can you disable your firewall?  i am about to start this phase of asterisk
an would like help from one newbie to another.  otherwise this newbie will
let you know how i did it.


- Original Message -
From: "Brad Waite" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 20, 2003 9:07 AM
Subject: [Asterisk-Users] Maximum retries exceeded w/SIP


> First of all, I'd like to send a big "thank you" to all the folks who have
> helped me get this far.
>
> Now on to the next problem.  Here's my current network setup:
>
>
> The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
>   |
>   +--- Laptop (public IP)
>
> natd is set up with the following rules:
>
> redirect_port udp 10.0.0.253:1-2 1-2
> redirect_port udp 10.0.0.253:5060 5060
>
> * is set up with the demo/sandbox config.
>
> I'm using XLite as my SIP client and have configured it on PC to work with
*.
> I'm able to do everything I've tried so far.  I should, though - I'm on
the inside.
>
> However, when trying to make a call from the outside (via Laptop),
something's
> breaking.  I've set up the SIP proxy in XLite to be the external interface
on
> the firewall, and am able to log into the proxy without difficulty.  And
while I
> can begin conversations, I can't keep them going for long.
>
> For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I
get most
> of the "demo-abouttotry" message - "I am about to attempt an IAX
connection to a
> demonstration server located at Di" - at which point it gets cut off.  The
> console spits out the following error:
>
> File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call
> [EMAIL PROTECTED] for seqno 12384
(Response)
>
>
> Any ideas what could be going on?  My first guess is the firewall, but I
can't
> figure out why some of the packets would get through while others
apparently are
> not.  I'm at a loss.
>
> Brad Waite
> aka HankPoacher
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Maximum retries exceeded w/SIP

[Brad Waite]

First of all, I'd like to send a big "thank you" to all the folks who have 
helped me get this far.

Now on to the next problem.  Here's my current network setup:

The Big I ---+--- FreeBSD FW --- * (10.0.0.253)  PC (10.0.0.1)
 |
 +--- Laptop (public IP)
natd is set up with the following rules:

redirect_port udp 10.0.0.253:1-2 1-2
redirect_port udp 10.0.0.253:5060 5060
* is set up with the demo/sandbox config.

I'm using XLite as my SIP client and have configured it on PC to work with *. 
I'm able to do everything I've tried so far.  I should, though - I'm on the inside.

However, when trying to make a call from the outside (via Laptop), something's 
breaking.  I've set up the SIP proxy in XLite to be the external interface on 
the firewall, and am able to log into the proxy without difficulty.  And while I 
can begin conversations, I can't keep them going for long.

For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I get most 
of the "demo-abouttotry" message - "I am about to attempt an IAX connection to a 
demonstration server located at Di" - at which point it gets cut off.  The 
console spits out the following error:

File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call 
[EMAIL PROTECTED] for seqno 12384 (Response)

Any ideas what could be going on?  My first guess is the firewall, but I can't 
figure out why some of the packets would get through while others apparently are 
not.  I'm at a loss.

Brad Waite
aka HankPoacher
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users