Re: [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw)

2011-06-28 Thread Fellipe Paes 

Hello!
In your sip.conf use alaw as your first codec option and see what happens.Best 
regards,
Fellipe Paes

Date: Tue, 28 Jun 2011 15:29:11 +0530
From: theasterisk...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asked to transmit frame type slin,while native 
formats is 0x8 (alaw)


Asterisk 1.8.3.2

I have been getting this warning constantly on CLI in a call scenario where I 
use local channels to connect SIP with PSTN. 
I use callfile and local channel to first call a PSTN number and if answered, 
use local channel to call SIP phone with music on hold enabled in Dial string.

If I call PSTN from SIP  directly or vice versa I don't see this warning coming.
On SIP I have allowed only one codec(alaw).

[Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, 
while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw)



I also tried to yes/no option transcode_via_sln in asterisk.conf without any 
success.
Any idea?
Thanks,
--AM


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Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit

2011-04-04 Thread Fellipe Paes 

Hi Satish!

Few days ago I had the same problem, and was a problem in my dialplan.
Post your extensions.conf and let's see.
Best regards,

Fellipe

From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 4 Apr 2011 19:51:26 +
Subject: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit









Hey Guys,

Whenever i calling any extension i am getting following WARNING messages do you 
have any idea they coming from where?

-Satish



shirley*CLI
  == Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro(SIP/7527-0008, 
stdexten,7623,sip/7623sip/7624) in new stack
-- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, 
sip/7623sip/7624iax2/7623,20,t) in new stack
  == Using SIP RTP CoS mark 5
-- Called 7623
  == Using SIP RTP CoS mark 5
[Apr  4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect
[Apr  4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
-- Called 7624
-- Called 7623
-- SIP/7623-0009 is ringing
[Apr  4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: 
Auto-congesting call due to slow response
-- IAX2/0.0.29.199:4569-5537 is circuit-busy
-- Hungup 'IAX2/0.0.29.199:4569-5537'
[Apr  4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
-- SIP/7623-0009 connected line has changed. Saving it until answer for 
SIP/7527-0008
-- SIP/7623-0009 answered SIP/7527-0008
[Apr  4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 
'SIP/7527-0008' in macro 'stdexten'
  == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008'
[Apr  4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr  4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission 
timeout reached on transmission 
23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical 
Request) -- See doc/sip-retransmit.txt.
Packet timed out after 32000ms with no response

  

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Re: [asterisk-users] What is the most stable version of asterisk?

2011-03-25 Thread Fellipe Paes 

Hi Doug!

I use Asterisk 1.4 and 1.8, I can easily see that Asterisk 1.8 works better 
than 1.4.
Everything on Asterisk 1.8 seems better.
Best regards,

 From: d...@impalanetworks.com
 To: asterisk-users@lists.digium.com
 Date: Fri, 25 Mar 2011 08:32:04 -0600
 Subject: Re: [asterisk-users] What is the most stable version of asterisk?
 
 Do you have the same ratio of deployments using 1.4 as you do with 1.2? What 
 about 1.6 or 1.8? I simply question how accurate a comparison can be made 
 when one is comparing 50 1.2 deployments to 5 1.4 deployments? I'm sure it 
 says something, and I do appreciate the feedback.
 
 -
 Doug Mortensen
 Network Consultant
 Impala Networks
 P: 505.327.7300
 .
 
 -Original Message-
 From: C F [mailto:shma...@gmail.com] 
 Sent: Thursday, March 24, 2011 8:14 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] What is the most stable version of asterisk?
 
 I use mainly 1.2 with great success, mostly restarts are due to power outages.
 I recently started to upgrade to 1.4, so far so good. Too early to say, the 
 longest running 1.4 is only 234 days. While I have had 900+ days with 1.2
 
 
 
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[asterisk-users] Some errors

2011-03-15 Thread Fellipe Paes 

Hello folks,

since I started with asterisk 1.8.2 I got this messages in my console when 
finish a call.

 -- Executing [1610@from-e1:1] Dial(SIP/xxx-0027, SIP/1610,60) in new 
stack

  == Using SIP RTP CoS mark 5

-- Called 1610

-- SIP/1610-0028 is ringing

-- SIP/1610-0028 answered SIP/xxx-0027

-- Locally bridging SIP/xxx-0027 and SIP/1610-0028

  == Using SIP RTP CoS mark 5

-- Executing [h@from-e1:1] Dial(SIP/xxx-0029, SIP/h,60) in new stack

  == Using SIP RTP CoS mark 5

[Mar 15 11:06:47] ERROR[2173]: netsock2.c:245 ast_sockaddr_resolve: 
getaddrinfo(h, (null), ...): Name or service not known

[Mar 15 11:06:47] WARNING[2173]: chan_sip.c:5057 create_addr: No such host: h

[Mar 15 11:06:47] WARNING[2173]: acl.c:698 ast_ouraddrfor: Cannot connect

[Mar 15 11:06:47] WARNING[2173]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x79e0bc0 (len 826) to (null) returned -1: Invalid argument

-- Called h

[Mar 15 11:06:48] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x79e0bc0 (len 826) to (null) returned -1: Invalid argument

-- Executing [h@from-e1:1] Dial(SIP/xxx-0027, SIP/h,60) in new stack

  == Using SIP RTP CoS mark 5

[Mar 15 11:06:48] ERROR[2172]: netsock2.c:245 ast_sockaddr_resolve: 
getaddrinfo(h, (null), ...): Name or service not known

[Mar 15 11:06:48] WARNING[2172]: chan_sip.c:5057 create_addr: No such host: h

[Mar 15 11:06:48] WARNING[2172]: acl.c:698 ast_ouraddrfor: Cannot connect

[Mar 15 11:06:48] WARNING[2172]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x7a12140 (len 828) to (null) returned -1: Invalid argument

-- Called h

[Mar 15 11:06:48] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x7a12140 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:06:49] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x79e0bc0 (len 826) to (null) returned -1: Invalid argument

[Mar 15 11:06:49] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x7a12140 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:06:51] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x79e0bc0 (len 826) to (null) returned -1: Invalid argument

[Mar 15 11:06:51] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x7a12140 (len 828) to (null) returned -1: Invalid argument

  == Spawn extension (from-e1, h, 1) exited non-zero on 'SIP/xxx-0029'

-- Executing [h@from-e1:1] Dial(SIP/xxx SIP/h,60) in new stack

  == Using SIP RTP CoS mark 5

[Mar 15 11:06:54] ERROR[2173]: netsock2.c:245 ast_sockaddr_resolve: 
getaddrinfo(h, (null), ...): Name or service not known

[Mar 15 11:06:54] WARNING[2173]: chan_sip.c:5057 create_addr: No such host: h

[Mar 15 11:06:54] WARNING[2173]: acl.c:698 ast_ouraddrfor: Cannot connect

[Mar 15 11:06:54] WARNING[2173]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x794f840 (len 828) to (null) returned -1: Invalid argument

-- Called h

  == Spawn extension (from-e1, h, 1) exited non-zero on 'SIP/xxx-0029'

  == Spawn extension (from-e1, 1610, 1) exited non-zero on 'SIP/xxx-0027'

[Mar 15 11:06:55] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x79e0bc0 (len 826) to (null) returned -1: Invalid argument

[Mar 15 11:06:55] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x794f840 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:06:55] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x7a12140 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:06:56] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x794f840 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:06:58] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x794f840 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:07:02] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x794f840 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:07:03] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x79e0bc0 (len 826) to (null) returned -1: Invalid argument

[Mar 15 11:07:03] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x7a12140 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:07:10] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x794f840 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:07:19] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x79e0bc0 (len 826) to (null) returned -1: Invalid argument

[Mar 15 11:07:19] WARNING[1947]: chan_sip.c:3386 retrans_pkt: Retransmission 
timeout reached on transmission 
3bf2878e1658597810280835412e51b0@172.16.1.127:5060 for seqno 102 (Critical 
Request) -- See doc/sip-retransmit.txt.

Packet timed out after 32000ms with no response

[Mar 15 11:07:19] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
 0x7a12140 (len 828) to (null) returned -1: Invalid argument

[Mar 15 11:07:20] WARNING[1947]: chan_sip.c:3386 retrans_pkt: Retransmission 
timeout reached on transmission 
69848055147d5354302b461c34811d17@172.16.1.127:5060

Re: [asterisk-users] Some errors

2011-03-15 Thread Fellipe Paes 

Hi Paul,

thanks for your answer, I'll open this issue on the tracker, but now I have a 
new question, is there some way to deactivate IPv6 on Asterisk 1.8?
Best regards,

Fellipe

 Date: Tue, 15 Mar 2011 11:18:30 -0400
 From: pabelan...@digium.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Some errors
 
 On 11-03-15 10:11 AM, Fellipe Paes wrote:
  [Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x794f840 (len 828) to (null) returned -1: Invalid argument
 
  [Mar 15 11:07:26] WARNING[1947]: chan_sip.c:3386 retrans_pkt: 
  Retransmission timeout reached on transmission 
  521df80947598d560109d73f1493a76d@172.16.1.127:5060 for seqno 102 (Critical 
  Request) -- See doc/sip-retransmit.txt.
 
  Packet timed out after 32000ms with no response
  Well... everything works fine, but I don't like this errors, any ideas? 
 
 Theses are leftover issue with the IPv6 conversion for Asterisk 1.8. 
 Collect a complete debug log[1] and open a new issue on the tracker.
 
 [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
 
 -- 
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org
 
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Re: [asterisk-users] Some errors

2011-03-15 Thread Fellipe Paes 

Hello guys,

I received the following answer from Digium, I understand and will change my 
dialplan, but I have one more question now, why I can't use _. in my dialplan?

-

As suggested by Kevin Fleming on the asterisk-users list, this is not a

bug.



I believe the reason you're seeing this, is your usage of _. as an

extension.  Never do that.

-

Again, thanks for all.
Best regards,
Fellipe

 Date: Tue, 15 Mar 2011 13:19:44 -0400
 From: pabelan...@digium.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Some errors
 
 On 11-03-15 12:51 PM, Fellipe Paes wrote:
  thanks for your answer, I'll open this issue on the tracker, but now I have 
  a new question, is there some way to deactivate IPv6 on Asterisk 1.8?
 
 You can enable / disable IPv6 via the config files, but the core API / 
 ABI have changed.
 
 But Kevin is correct, your issue does look to be a dialplan problem 
 (SIP/h).  I should have been more specific about handling the failure 
 more gracefully.
 
 -- 
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org
 
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Re: [asterisk-users] Some errors

2011-03-15 Thread Fellipe Paes 

Hi Jason,

well, I was thinking something like this, but don't hurt to ask :D
Thank you for all guys.
Best regards,

Fellipe

 Date: Tue, 15 Mar 2011 12:37:05 -0500
 From: jpar...@digium.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Some errors
 
 On 03/15/2011 12:34 PM, Fellipe Paes wrote:
  why I can't use _. in my dialplan?
 
 
 Because it matches everything.  In this case, it's matching the 'h' exten.  
 So 
 when the call gets hung up, it goes to _. and does what you're seeing.
 
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Re: [asterisk-users] Some errors

2011-03-15 Thread Fellipe Paes 

Hello guys,

one more question, if I have the following dialplan and can't use _. how can I 
send everything that isn't 1620,1622,1610,9XXX to SIP/xxx?
Sorry I'm new with this * world.

root@*:/etc/asterisk# vim extensions.conf



[example]



exten = _.,1,Dial(SIP/xxx/${EXTEN},60)

exten = _.,n,Hangup()

exten = _1620,1,Dial(SIP/${EXTEN},60)

exten = _1620,n,Hangup()

exten = _1622,1,Dial(SIP/${EXTEN},60)

exten = _1622,n,Hangup()

exten = _9XXX,1,Dial(SIP/${EXTEN},60)

exten = _9XXX,n,Hangup()

exten = h,1,Hangup()

exten = _1610,1,Dial(SIP/${EXTEN})

exten = _1610,n,Hangup()   

Thanks for all.
Best regards,

Fellipe

From: fellipe...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 15 Mar 2011 18:14:45 +
Subject: Re: [asterisk-users] Some errors








Hi Jason,

well, I was thinking something like this, but don't hurt to ask :D
Thank you for all guys.
Best regards,

Fellipe

 Date: Tue, 15 Mar 2011 12:37:05 -0500
 From: jpar...@digium.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Some errors
 
 On 03/15/2011 12:34 PM, Fellipe Paes wrote:
  why I can't use _. in my dialplan?
 
 
 Because it matches everything.  In this case, it's matching the 'h' exten.  
 So 
 when the call gets hung up, it goes to _. and does what you're seeing.
 
 --
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Re: [asterisk-users] Some errors

2011-03-15 Thread Fellipe Paes 

Hello guys!

I've solved this question just adding X in dialplan and killing that h option, 
and of course with your help:

root@*:/etc/asterisk# vim extensions.conf



[example]



exten = _X.,1,Dial(SIP/xxx/${EXTEN},60)

exten = _X.,n,Hangup()

exten = _1620,1,Dial(SIP/${EXTEN},60)

exten = _1620,n,Hangup()

exten = _1622,1,Dial(SIP/${EXTEN},60)

exten = _1622,n,Hangup()

exten = _9XXX,1,Dial(SIP/${EXTEN},60)

exten = _9XXX,n,Hangup()

exten = _1610,1,Dial(SIP/${EXTEN})

exten = _1610,n,Hangup()   

Thanks for all.
Best regards,

Fellipe 

From: fellipe...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 15 Mar 2011 18:49:32 +
Subject: Re: [asterisk-users] Some errors








Hello guys,

one more question, if I have the following dialplan and can't use _. how can I 
send everything that isn't 1620,1622,1610,9XXX to SIP/xxx?
Sorry I'm new with this * world.

root@*:/etc/asterisk# vim extensions.conf



[example]



exten = _.,1,Dial(SIP/xxx/${EXTEN},60)

exten = _.,n,Hangup()

exten = _1620,1,Dial(SIP/${EXTEN},60)

exten = _1620,n,Hangup()

exten = _1622,1,Dial(SIP/${EXTEN},60)

exten = _1622,n,Hangup()

exten = _9XXX,1,Dial(SIP/${EXTEN},60)

exten = _9XXX,n,Hangup()

exten = h,1,Hangup()

exten = _1610,1,Dial(SIP/${EXTEN})

exten = _1610,n,Hangup()   

Thanks for all.
Best regards,

Fellipe

From: fellipe...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 15 Mar 2011 18:14:45 +
Subject: Re: [asterisk-users] Some errors








Hi Jason,

well, I was thinking something like this, but don't hurt to ask :D
Thank you for all guys.
Best regards,

Fellipe

 Date: Tue, 15 Mar 2011 12:37:05 -0500
 From: jpar...@digium.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Some errors
 
 On 03/15/2011 12:34 PM, Fellipe Paes wrote:
  why I can't use _. in my dialplan?
 
 
 Because it matches everything.  In this case, it's matching the 'h' exten.  
 So 
 when the call gets hung up, it goes to _. and does what you're seeing.
 
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