Re: [asterisk-users] No sound with internal calls depending on which phones

2015-11-12 Thread Pete Mundy 
Hi Denis

That advice is correct for disabling RTP support in the phone and if you have 
achieved this then your quoted error about SRTP in the Asterisk console (when 
the call is failing) should no longer be appearing.

This will help show that it was a red herring all along.

The next step (IMO) is to use the Snom's built-in packet capture capabilities 
to grab a packet capture of a failed conversation from each phone then post it 
somewhere with a link to the list so that others can inspect the SIP signalling 
to discover where the issue lies.

You may also need to provide some information about your network configuration, 
IP ranges, firewall etc (a little diagram goes a long way).

For information on how to use the packet capture capabilities on the phone 
refer the Snom user's guide. I'm pretty sure it's well documented.

Hope this helps and look forward to investigating the packet captures for you :)

Pete


On 13/11/2015, at 5:46 AM, (lists) Denis BUCHER  wrote:

> Dear Sam, dear jg, dear Mitul, dear all,
> 
> Thanks a lot for your advices! I had the same idea, but it still doesn't work!
> 
> Maybe I changed the wrong option on the GUI configuration ?
> I went to menu "Setup" > "Identity 1" > "RTP" > "RTP Encryption:" > "off" on 
> both phones.
> And in the configuration I see "user_srtp1!: off"
> 
> Is this right ?



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Re: [asterisk-users] No sound with internal calls depending on which phones

2015-11-12 Thread Ishfaq Malik 
That is correct for turning SRTP off on a Snom phone.

On 12 November 2015 at 16:46, (lists) Denis BUCHER 
wrote:

> Dear Sam, dear jg, dear Mitul, dear all,
>
> Thanks a lot for your advices! I had the same idea, but it still doesn't
> work!
>
> Maybe I changed the wrong option on the GUI configuration ?
> I went to menu "Setup" > "Identity 1" > "RTP" > "RTP Encryption:" > "off"
> on both phones.
> And in the configuration I see "user_srtp1!: off"
>
> Is this right ?
>
> Denis
>
>
> Le 12.11.2015 17:05, Sam Basan a écrit :
>
> Snom default configuration is SRTP enabled.
>
> You should disable the SRTP from the phone web GUI configuration
>
>
>
>
>
>
>
> *Sincerely,*
>
>  [image: cid:image001.jpg@01D0D5C4.27A0CBA0]
>
> *Sam Basan*
>
> [image: cid:image003.png@01C918DA.6B3E4530]
>
>
>
> *From:* Mitul Limbani [mailto:mi...@enterux.in ]
> *Sent:* Thursday, November 12, 2015 5:25 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>  
> *Subject:* Re: [asterisk-users] No sound with internal calls depending on
> which phones
>
>
>
> You might have to disable srtp negotiations inside the phone web ui
> options.
>
> Mitul
>
> On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" 
> wrote:
>
> Dear all,
>
> I have a very strange problem :
>
>- external calls work perfectly,
>- internal calls between some phones too,
>- but internal call between two similar phones don't work !!! (Snom
>710)
>
> When we have sound, there are no errors in asterisk. When we do not have
> sound, there is the following error :
>
>- [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
>module loaded, can't setup SRTP session.
>
> This is a working internal call :
>
>   == Using SIP RTP CoS mark 5
> -- Executing [301@local:1] Dial("SIP/dbucher-", "SIP/phone1")
> in new stack
>   == Using SIP RTP CoS mark 5
> -- Called phone1
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 answered SIP/dbucher-
> -- Remotely bridging SIP/dbucher- and SIP/phone1-0001
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Got  RTP packet from192.168.128.99:49646 (type 126, seq 031575, ts
> 01, len 00)
> [Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read:
> Unknown RTP codec 126 received from '192.168.128.99:49646'
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
>   == Spawn extension (local, 301, 1) exited non-zero on
> 'SIP/dbucher-'
>
> This is a non-working call :
>
>   == Using SIP RTP CoS mark 5
> [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
> module loaded, can't setup SRTP session.
> -- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002",
> "SIP/phone1") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called phone1
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
> -- Remotely bridging SIP/hsolutionspf5-0002 and SIP/phone1-0003
> Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (typ

Re: [asterisk-users] No sound with internal calls depending on which phones

2015-11-12 Thread (lists) Denis BUCHER 

Dear Sam, dear jg, dear Mitul, dear all,

Thanks a lot for your advices! I had the same idea, but it still doesn't 
work!


Maybe I changed the wrong option on the GUI configuration ?
I went to menu "Setup" > "Identity 1" > "RTP" > "RTP Encryption:" > 
"off" on both phones.

And in the configuration I see "user_srtp1!: off"

Is this right ?

Denis

Le 12.11.2015 17:05, Sam Basan a écrit :


Snom default configuration is SRTP enabled.

You should disable the SRTP from the phone web GUI configuration

**

**

*Sincerely,*

cid:image001.jpg@01D0D5C4.27A0CBA0

*Sam Basan*

cid:image003.png@01C918DA.6B3E4530

*From:*Mitul Limbani [mailto:mi...@enterux.in]
*Sent:* Thursday, November 12, 2015 5:25 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion 

*Subject:* Re: [asterisk-users] No sound with internal calls depending 
on which phones


You might have to disable srtp negotiations inside the phone web ui 
options.


Mitul

On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" 
mailto:dbuche...@hsolutions.ch>> wrote:


Dear all,

I have a very strange problem :

  * external calls work perfectly,
  * internal calls between some phones too,
  * but internal call between two similar phones don't work !!!
(Snom 710)

When we have sound, there are no errors in asterisk. When we do
not have sound, there is the following error :

  * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp:
No SRTP module loaded, can't setup SRTP session.

This is a working internal call :

  == Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-",
"SIP/phone1") in new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 answered SIP/dbucher-
-- Remotely bridging SIP/dbucher- and
SIP/phone1-0001
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Got  RTP packet from 192.168.128.99:49646
<http://192.168.128.99:49646> (type 126, seq 031575, ts
01, len 00)
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190
ast_rtp_read: Unknown RTP codec 126 received from
'192.168.128.99:49646 <http://192.168.128.99:49646>'
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
  == Spawn extension (local, 301, 1) exited non-zero on
'SIP/dbucher-'

This is a non-working call :

  == Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp:
No SRTP module loaded, can't setup SRTP session.
-- Executing [301@local:1]
Dial("SIP/hsolutionspf5-0002", "SIP/phone1") in new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
-- Remotely bridging SIP/hsolutionspf5-0002 and
SIP/phone1-0003
Sent RTP P2P packet to 192.168.128.228:65494
<http://192.168.128.228:65494> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128

Re: [asterisk-users] No sound with internal calls depending on which phones

2015-11-12 Thread Sam Basan 
Snom default configuration is SRTP enabled.

You should disable the SRTP from the phone web GUI configuration

 

 

 

Sincerely,

 

Sam Basan



 

From: Mitul Limbani [mailto:mi...@enterux.in] 
Sent: Thursday, November 12, 2015 5:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] No sound with internal calls depending on which 
phones

 

You might have to disable srtp negotiations inside the phone web ui options. 

Mitul 

On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" mailto:dbuche...@hsolutions.ch> > wrote:

Dear all,

I have a very strange problem :

*   external calls work perfectly,
*   internal calls between some phones too,
*   but internal call between two similar phones don't work !!! (Snom 710)

When we have sound, there are no errors in asterisk. When we do not have sound, 
there is the following error :

*   [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP 
module loaded, can't setup SRTP session.

This is a working internal call :



  == Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-", "SIP/phone1") in 
new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 answered SIP/dbucher-
-- Remotely bridging SIP/dbucher- and SIP/phone1-0001
Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646>  
(type 00, len 000160)
Got  RTP packet from192.168.128.99:49646 <http://192.168.128.99:49646>  
(type 126, seq 031575, ts 01, len 00)
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown 
RTP codec 126 received from '192.168.128.99:49646 <http://192.168.128.99:49646> 
'
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 <http://192.168.128.231:57818>  
(type 00, len 000160)
  == Spawn extension (local, 301, 1) exited non-zero on 'SIP/dbucher-'

This is a non-working call :



  == Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module 
loaded, can't setup SRTP session.
-- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002", "SIP/phone1") 
in new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
-- Remotely bridging SIP/hsolutionspf5-0002 and SIP/phone1-0003
Sent RTP P2P packet to 192.168.128.228:65494 <http://192.168.128.228:65494>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.228:65494 <http://192.168.128.228:65494>  
(type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 <http://192.168.128.231:51350>  
(type 03, len 33)
  == Spawn extension (local, 301, 1) exited non-zero on 
'SIP/hsolutionspf5-0002'

I tried many options to disable SRTP but without success :

*   canreinvite = no
*   canreinvite = nonat
*   srtpcapable=no
*   encryption=no
*   directmedia=nonat
*   ...or noload => res_srtp.so in modules.conf

Re: [asterisk-users] No sound with internal calls depending on which phones

2015-11-12 Thread jg 

Am 12.11.2015 um 16:22 schrieb (lists) Denis BUCHER:

Dear all,

I have a very strange problem :

  * external calls work perfectly,
  * internal calls between some phones too,
  * but internal call between two similar phones don't work !!! (Snom 710)

When we have sound, there are no errors in asterisk. When we do not have sound, there is the 
following error :


  * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module 
loaded, can't
setup SRTP session.

This is a working internal call :

  == Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-", "SIP/phone1") in 
new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 answered SIP/dbucher-
-- Remotely bridging SIP/dbucher- and SIP/phone1-0001
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Got  RTP packet from192.168.128.99:49646 (type 126, seq 031575, ts 01, 
len 00)
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 
received from '192.168.128.99:49646'

Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
  == Spawn extension (local, 301, 1) exited non-zero on 'SIP/dbucher-'

This is a non-working call :

  == Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module loaded, can't 
setup SRTP session.

-- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002", "SIP/phone1") 
in new stack
  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
-- Remotely bridging SIP/hsolutionspf5-0002 and SIP/phone1-0003
Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
  == Spawn extension (local, 301, 1) exited non-zero on 
'SIP/hsolutionspf5-0002'

I tried many options to disable SRTP but without success :

  * canreinvite = no
  * canreinvite = nonat
  * srtpcapable=no
  * encryption=no
  * directmedia=nonat
  * ...or noload => res_srtp.so in modules.conf


Any help would be GREATLY appreciated !

Denis

P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)




Please check
http://wiki.snom.com/wiki/index.php/Settings/user_srtp
and make sure the flag is off.

If you install Asterisk with the srtp module, then you need to set the auth-tag to AES-80, but I 
haven't played with this option for quite some time.


jg
-- 
_
-- 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

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

[asterisk-users] No sound with internal calls depending on which phones

2015-11-12 Thread (lists) Denis BUCHER 

Dear all,

I have a very strange problem :

 * external calls work perfectly,
 * internal calls between some phones too,
 * but internal call between two similar phones don't work !!! (Snom 710)

When we have sound, there are no errors in asterisk. When we do not have 
sound, there is the following error :


 * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
   module loaded, can't setup SRTP session.

This is a working internal call :

  == Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-", 
"SIP/phone1") in new stack

  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 answered SIP/dbucher-
-- Remotely bridging SIP/dbucher- and SIP/phone1-0001
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Got  RTP packet from192.168.128.99:49646 (type 126, seq 031575, ts 
01, len 00)
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read: 
Unknown RTP codec 126 received from '192.168.128.99:49646'

Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
  == Spawn extension (local, 301, 1) exited non-zero on 
'SIP/dbucher-'

This is a non-working call :

  == Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP 
module loaded, can't setup SRTP session.
-- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002", 
"SIP/phone1") in new stack

  == Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
-- Remotely bridging SIP/hsolutionspf5-0002 and 
SIP/phone1-0003

Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
  == Spawn extension (local, 301, 1) exited non-zero on 
'SIP/hsolutionspf5-0002'

I tried many options to disable SRTP but without success :

 * canreinvite = no
 * canreinvite = nonat
 * srtpcapable=no
 * encryption=no
 * directmedia=nonat
 * ...or noload => res_srtp.so in modules.conf


Any help would be GREATLY appreciated !

Denis

P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)

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Re: [asterisk-users] No sound with internal calls depending on which phones

2015-11-12 Thread Mitul Limbani 
You might have to disable srtp negotiations inside the phone web ui
options.

Mitul
On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" 
wrote:

> Dear all,
>
> I have a very strange problem :
>
>- external calls work perfectly,
>- internal calls between some phones too,
>- but internal call between two similar phones don't work !!! (Snom
>710)
>
> When we have sound, there are no errors in asterisk. When we do not have
> sound, there is the following error :
>
>- [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
>module loaded, can't setup SRTP session.
>
> This is a working internal call :
>
>   == Using SIP RTP CoS mark 5
> -- Executing [301@local:1] Dial("SIP/dbucher-", "SIP/phone1")
> in new stack
>   == Using SIP RTP CoS mark 5
> -- Called phone1
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 is ringing
> -- SIP/phone1-0001 answered SIP/dbucher-
> -- Remotely bridging SIP/dbucher- and SIP/phone1-0001
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
> Got  RTP packet from192.168.128.99:49646 (type 126, seq 031575, ts
> 01, len 00)
> [Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read:
> Unknown RTP codec 126 received from '192.168.128.99:49646'
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
>   == Spawn extension (local, 301, 1) exited non-zero on
> 'SIP/dbucher-'
>
> This is a non-working call :
>
>   == Using SIP RTP CoS mark 5
> [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP
> module loaded, can't setup SRTP session.
> -- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002",
> "SIP/phone1") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called phone1
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 is ringing
> -- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
> -- Remotely bridging SIP/hsolutionspf5-0002 and SIP/phone1-0003
> Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
> Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
>   == Spawn extension (local, 301, 1) exited non-zero on
> 'SIP/hsolutionspf5-0002'
>
> I tried many options to disable SRTP but without success :
>
>- canreinvite = no
>- canreinvite = nonat
>- srtpcapable=no
>- encryption=no
>- directmedia=nonat
>- ...or noload => res_srtp.so in modules.conf
>
>
> Any help would be GREATLY appreciated !
>
> Denis
>
> P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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>
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