Hi, I am encountering problem making concurrent calls using A sangoma card, It seems that the 2nd call get a congested or buzy,I connect via sip-->asterisk-->dahdi attached is the PRI debug messages -- Making new call for cref 32771> DL-DATA request> Protocol Discriminator: Q.931 (8) len=42> TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator)> Message Type: SETUP (5)TEI=0 Transmitting N(S)=5, window is open V(A)=5 K=7> Protocol Discriminator: Q.931 (8) len=42> TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator)> Message Type: SETUP (5)> [04 03 80 90 a3]> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)> Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)> User information layer 1: A-Law (35)> [18 03 a1 83 81]> Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0> ChanSel: As indicated in following octets> Ext: 1 Coding: 0 Number Specified Channel Type: 3> Ext: 1 Channel: 1 Type: CPE]> [6c 0c 21 81 30 31 36 33 36 37 37 30 36 32]> Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)> Presentation: Presentation permitted, user number passed network screening (1) '016367XXXX' ]> [70 0b 80 30 31 39 36 35 30 31 30 32 34]> Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '019650XXXX' ]q931.c:5039 q931_setup: Call 32771 enters state 1 (Call Initiated). Hold state: Idle-- Making new call for cref 32772 > DL-DATA request> Protocol Discriminator: Q.931 (8) len=42> TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent from originator)> Message Type: SETUP (5)TEI=0 Transmitting N(S)=6, window is open V(A)=5 K=7 > Protocol Discriminator: Q.931 (8) len=42> TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent from originator)> Message Type: SETUP (5)> [04 03 80 90 a3]> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)> Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)> User information layer 1: A-Law (35)> [18 03 a1 83 82]> Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0> ChanSel: As indicated in following octets> Ext: 1 Coding: 0 Number Specified Channel Type: 3> Ext: 1 Channel: 2 Type: CPE]> [6c 0c 21 81 30 31 36 33 36 37 37 30 36 32]> Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)> Presentation: Presentation permitted, user number passed network screening (1) '016367XXXX' ]> [70 0b 80 30 31 39 33 36 37 31 30 32 34]> Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '019367XXXX' ]q931.c:5039 q931_setup: Call 32772 enters state 1 (Call Initiated). Hold state: Idle< Protocol Discriminator: Q.931 (8) len=10< TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent to originator)< Message Type: SETUP ACKNOWLEDGE (13)< [18 03 a9 83 81]< Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0< ChanSel: As indicated in following octets< Ext: 1 Coding: 0 Number Specified Channel Type: 3< Ext: 1 Channel: 1 Type: CPE]Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 24 (cs0, Channel Identification)q931.c:7390 post_handle_q931_message: Call 32771 enters state 2 (Overlap Sending). Hold state: Idle < Protocol Discriminator: Q.931 (8) len=10< TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent to originator)< Message Type: CALL PROCEEDING (2)< [18 03 a9 83 81]< Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0< ChanSel: As indicated in following octets< Ext: 1 Coding: 0 Number Specified Channel Type: 3< Ext: 1 Channel: 1 Type: CPE]Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 24 (cs0, Channel Identification)q931.c:7104 post_handle_q931_message: Call 32771 enters state 3 (Outgoing Call Proceeding). Hold state: Idle < Protocol Discriminator: Q.931 (8) len=9< TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent to originator)< Message Type: RELEASE COMPLETE (90)< [08 02 82 a2]< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2)< Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion (resource unavailable) (2) ]Received message for call 0x8cc6f80 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 8 (cs0, Cause)q931.c:7197 post_handle_q931_message: Call 32772 enters state 0 (Null). Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state IdleNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Idle < Protocol Discriminator: Q.931 (8) len=9< TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent to originator)< Message Type: ALERTING (1)< [1e 02 82 88]< Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2)< Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ]Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 30 (cs0, Progress Indicator)q931.c:6983 post_handle_q931_message: Call 32771 enters state 4 (Call Delivered). Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Delivered, peerstate Call Received, hold-state Idleq931.c:4845 q931_disconnect: Call 32771 enters state 11 (Disconnect Request). Hold state: Idle> DL-DATA request> Protocol Discriminator: Q.931 (8) len=9> TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator)> Message Type: DISCONNECT (69)TEI=0 Transmitting N(S)=7, window is open V(A)=7 K=7 > Protocol Discriminator: Q.931 (8) len=9> TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator)> Message Type: DISCONNECT (69)> [08 02 81 90]> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)> Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]< Protocol Discriminator: Q.931 (8) len=5< TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent to originator)< Message Type: RELEASE (77)Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0 TEI/SAPI 0/0q931.c:7237 post_handle_q931_message: Call 32771 enters state 0 (Null). Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request, hold-state Idle> DL-DATA request> Protocol Discriminator: Q.931 (8) len=9> TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator)> Message Type: RELEASE COMPLETE (90)TEI=0 Transmitting N(S)=8, window is open V(A)=8 K=7 > Protocol Discriminator: Q.931 (8) len=9> TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator)> Message Type: RELEASE COMPLETE (90)> [08 02 81 90]> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)> Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]q931_hangup: other hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state IdleNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Idle-------------- next part -------------- An HTML attachment was scrubbed... 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Mc GRATH Ricardo
2012-Apr-20 14:59 UTC
[asterisk-users] dahdi cannot make simaltaneous calls
Hi rosli sukri
Well what I see release it becomes from outbound PRI server (PSTN or whatever).
By the way it seems according to the PRI trace first call label 32771 have an
uncompleted process, voice communication isn?t established (Answer connect
acknowledge etc message), called party ringer, and answer?
In case of 32772 outbound PRI server side it no response to Setup request,
moreover after send Setup message it response with a release message cause
Network Congestion (resource unavailable).
So it can conclude outbound system ???-->dahdi -->asterisk,-->Sip can?t
process simultaneous call.
Best regards
Mc GRATH Ricardo
E-Mail mcgrathr at mail2web.com
________________________________________
From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at
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[asterisk-users-request at lists.digium.com]
Sent: 20 April 2012 10:22
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 93, Issue 28
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."
Today's Topics:
1. Re: g729 freezes 1.8 (A J Stiles)
2. Re: g729 freezes 1.8 (Kevin P. Fleming)
3. dahdi cannot make simaltaneous calls (rosli sukri)
4. Company info (Josu? Conti)
5. Re: Company info (Josu? Conti)
6. Auto answer Asterisk ; Unable to create channel of type
(motty.cruz)
7. Re: g729 freezes 1.8 (Jeff Brower)
8. Re: asterisk 1.4.39 and dahdi 2.6 on Ubuntu (bilal ghayyad)
9. Re: asterisk 1.4.39 and dahdi 2.6 (Chad Wallace)
10. Re: Company info (Steven Howes)
11. Re: Company info (Doug Lytle)
12. Re: Company info (Josu? Conti)
13. Experience with virtual servers? (Brynjolfur Thorvardsson)
14. Re: Experience with virtual servers? (Arthur Stanfield)
15. Re: Experience with virtual servers? (Danny Nicholas)
16. Re: Experience with virtual servers?
(Stuart Elvish - IP Exchange Systems)
----------------------------------------------------------------------
Message: 1
Date: Thu, 19 Apr 2012 17:52:17 +0100
From: A J Stiles <asterisk_list at earthshod.co.uk>
Subject: Re: [asterisk-users] g729 freezes 1.8
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <201204191752.17519.asterisk_list at earthshod.co.uk>
Content-Type: Text/Plain; charset="iso-8859-6"
On Thursday 19 April 2012, samuel wrote:> Just in case it helps:
>
> It turned out that from asterisk version 1.8.4 on, the g729 binaries are
> different from the previous versions so it was a version mismatch between
> the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
>
> Thanks to the Digium support department that found out the issue.
Someone really needs to get the mPlayer folks (based on the Continent, where
mathematics is not patentable) to create an Open Source g729 codec
implementation .....
--
AJS
Answers come *after* questions.
------------------------------
Message: 2
Date: Thu, 19 Apr 2012 12:59:56 -0500
From: "Kevin P. Fleming" <kpfleming at digium.com>
Subject: Re: [asterisk-users] g729 freezes 1.8
To: asterisk-users at lists.digium.com
Message-ID: <4F90529C.8070107 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 04/19/2012 11:52 AM, A J Stiles wrote:> On Thursday 19 April 2012, samuel wrote:
>> Just in case it helps:
>>
>> It turned out that from asterisk version 1.8.4 on, the g729 binaries
are
>> different from the previous versions so it was a version mismatch
between
>> the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
>>
>> Thanks to the Digium support department that found out the issue.
>
> Someone really needs to get the mPlayer folks (based on the Continent,
where
> mathematics is not patentable) to create an Open Source g729 codec
> implementation .....
Source code availability is not the issue; the reference source code is
easily obtained from the ITU-T. Many of the G.729 patent holders are
companies based in Europe, so I suspect they would have a different
opinion than you do about the legitimacy of their patent claims on G.729 :-)
In any case (and of course IANAL), it is my understanding that the
patents that cover the base G.729 recommendation, along with Appendices
A and B, will all expire in the next year or so. We'll have to see what
that means for the market, especially with new, more freely licensed,
codecs coming out that provide substantially better performance.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
------------------------------
Message: 3
Date: Fri, 20 Apr 2012 02:19:56 +0800
From: rosli sukri <roslisukri at live.com>
Subject: [asterisk-users] dahdi cannot make simaltaneous calls
To: <asterisk-users at lists.digium.com>
Message-ID: <COL116-W45ED9719D810E9CD102FD1BE3D0 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"
Hi, I am encountering problem making concurrent calls using A sangoma card, It
seems that the 2nd call get a congested or buzy,I connect via
sip-->asterisk-->dahdi attached is the PRI debug messages
-- Making new call for cref 32771> DL-DATA request> Protocol
Discriminator: Q.931 (8) len=42> TEI=0 Call Ref: len= 2 (reference 3/0x3)
(Sent from originator)> Message Type: SETUP (5)TEI=0 Transmitting N(S)=5,
window is open V(A)=5 K=7> Protocol Discriminator: Q.931 (8) len=42> TEI=0 Call Ref: len= 2
(reference 3/0x3) (Sent from originator)> Message Type: SETUP (5)> [04 03
80 90 a3]> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)> Ext: 1 Trans mode/rate:
64kbps, circuit-mode (16)> User information
layer 1: A-Law (35)> [18 03 a1 83 81]> Channel ID (len= 5) [ Ext: 1
IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0>
ChanSel: As indicated in following octets> Ext: 1
Coding: 0 Number Specified Channel Type: 3> Ext: 1
Channel: 1 Type: CPE]> [6c 0c 21 81 30 31 36 33 36 37 37 30 36 32]>
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)> Presentation:
Presentation permitted, user number passed network screening (1)
'016367XXXX' ]> [70 0b 80 30 31 39 36 35 30 31 30 32 34]> Called
Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan
(0) '019650XXXX' ]q931.c:5039 q931_setup: Call 32771 enters state 1
(Call Initiated). Hold state: Idle-- Making new call for cref 32772
> DL-DATA request> Protocol Discriminator: Q.931 (8) len=42> TEI=0
Call Ref: len= 2 (reference 4/0x4) (Sent from originator)> Message Type:
SETUP (5)TEI=0 Transmitting N(S)=6, window is open V(A)=5 K=7
> Protocol Discriminator: Q.931 (8) len=42> TEI=0 Call Ref: len= 2
(reference 4/0x4) (Sent from originator)> Message Type: SETUP (5)> [04 03
80 90 a3]> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)> Ext: 1 Trans mode/rate:
64kbps, circuit-mode (16)> User information
layer 1: A-Law (35)> [18 03 a1 83 82]> Channel ID (len= 5) [ Ext: 1
IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0>
ChanSel: As indicated in following octets> Ext: 1
Coding: 0 Number Specified Channel Type: 3> Ext: 1
Channel: 2 Type: CPE]> [6c 0c 21 81 30 31 36 33 36 37 37 30 36 32]>
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)> Presentation:
Presentation permitted, user number passed network screening (1)
'016367XXXX' ]> [70 0b 80 30 31 39 33 36 37 31 30 32 34]> Called
Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan
(0) '019367XXXX' ]q931.c:5039 q931_setup: Call 32772 enters state 1
(Call Initiated). Hold state: Idle
< Protocol Discriminator: Q.931 (8) len=10< TEI=0 Call Ref: len= 2
(reference 3/0x3) (Sent to originator)< Message Type: SETUP ACKNOWLEDGE
(13)< [18 03 a9 83 81]< Channel ID (len= 5) [ Ext: 1 IntID: Implicit
Other(PRI) Spare: 0 Exclusive Dchan: 0< ChanSel: As
indicated in following octets< Ext: 1 Coding: 0
Number Specified Channel Type: 3< Ext: 1 Channel: 1
Type: CPE]Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0,
call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 24 (cs0, Channel
Identification)q931.c:7390 post_handle_q931_message: Call 32771 enters state 2
(Overlap Sending). Hold state: Idle
< Protocol Discriminator: Q.931 (8) len=10< TEI=0 Call Ref: len= 2
(reference 3/0x3) (Sent to originator)< Message Type: CALL PROCEEDING (2)<
[18 03 a9 83 81]< Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI)
Spare: 0 Exclusive Dchan: 0< ChanSel: As indicated in
following octets< Ext: 1 Coding: 0 Number Specified
Channel Type: 3< Ext: 1 Channel: 1 Type: CPE]Received
message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0
TEI/SAPI 0/0-- Processing IE 24 (cs0, Channel Identification)q931.c:7104
post_handle_q931_message: Call 32771 enters state 3 (Outgoing Call Proceeding).
Hold state: Idle
< Protocol Discriminator: Q.931 (8) len=9< TEI=0 Call Ref: len= 2
(reference 4/0x4) (Sent to originator)< Message Type: RELEASE COMPLETE
(90)< [08 02 82 a2]< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) Spare: 0 Location: Public network serving the local user (2)<
Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion
(resource unavailable) (2) ]Received message for call 0x8cc6f80 on 0x8ca59a0
TEI/SAPI 0/0, call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 8 (cs0,
Cause)q931.c:7197 post_handle_q931_message: Call 32772 enters state 0 (Null).
Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling q931_hangup,
ourstate Null, peerstate Null, hold-state IdleNEW_HANGUP DEBUG: Destroying the
call, ourstate Null, peerstate Null, hold-state Idle
< Protocol Discriminator: Q.931 (8) len=9< TEI=0 Call Ref: len= 2
(reference 3/0x3) (Sent to originator)< Message Type: ALERTING (1)< [1e 02
82 88]< Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Public network serving the local user (2)<
Ext: 1 Progress Description: Inband information or appropriate pattern now
available. (8) ]Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0,
call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 30 (cs0, Progress
Indicator)q931.c:6983 post_handle_q931_message: Call 32771 enters state 4 (Call
Delivered). Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling
q931_hangup, ourstate Call Delivered, peerstate Call Received, hold-state
Idleq931.c:4845 q931_disconnect: Call 32771 enters state 11 (Disconnect
Request). Hold state: Idle> DL-DATA request> Protocol Discriminator: Q.931 (8) len=9> TEI=0 Call
Ref: len= 2 (reference 3/0x3) (Sent from originator)> Message Type:
DISCONNECT (69)TEI=0 Transmitting N(S)=7, window is open V(A)=7 K=7
> Protocol Discriminator: Q.931 (8) len=9> TEI=0 Call Ref: len= 2
(reference 3/0x3) (Sent from originator)> Message Type: DISCONNECT (69)>
[08 02 81 90]> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0)
Spare: 0 Location: Private network serving the local user (1)>
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]
< Protocol Discriminator: Q.931 (8) len=5< TEI=0 Call Ref: len= 2
(reference 3/0x3) (Sent to originator)< Message Type: RELEASE (77)Received
message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0
TEI/SAPI 0/0q931.c:7237 post_handle_q931_message: Call 32771 enters state 0
(Null). Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling
q931_hangup, ourstate Null, peerstate Release Request, hold-state
Idle> DL-DATA request> Protocol Discriminator: Q.931 (8) len=9> TEI=0 Call
Ref: len= 2 (reference 3/0x3) (Sent from originator)> Message Type: RELEASE
COMPLETE (90)TEI=0 Transmitting N(S)=8, window is open V(A)=8 K=7
> Protocol Discriminator: Q.931 (8) len=9> TEI=0 Call Ref: len= 2
(reference 3/0x3) (Sent from originator)> Message Type: RELEASE COMPLETE
(90)> [08 02 81 90]> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) Spare: 0 Location: Private network serving the local user (1)>
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]q931_hangup:
other hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate
Null, hold-state IdleNEW_HANGUP DEBUG: Destroying the call, ourstate Null,
peerstate Null, hold-state Idle
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------------------------------
Message: 4
Date: Thu, 19 Apr 2012 17:11:57 -0300
From: Josu? Conti <josueconti at gmail.com>
Subject: [asterisk-users] Company info
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CANVtW_DRZqD+bguEZezY5jqA5Xw1kWRjodCy4Gwaxb+wzrXtLw at
mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Dear all,
Please let me know if anybody have informations about a company called
Convergia, like your products, ASR/ACD or more details.
With Best Regards
Josue
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------------------------------
Message: 5
Date: Thu, 19 Apr 2012 18:09:39 -0300
From: Josu? Conti <josueconti at gmail.com>
Subject: Re: [asterisk-users] Company info
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CANVtW_DCoG+5guXOMu31yLSG9YLhJE1M6WfhsrT_QyESTkAy-w at
mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
This is your website:
http://www.convergia.com/
Thanks in advanced for any informations.
Best Regards
Josue
Em 19 de abril de 2012 17:11, Josu? Conti <josueconti at gmail.com>
escreveu:
> Dear all,
> Please let me know if anybody have informations about a company called
> Convergia, like your products, ASR/ACD or more details.
>
> With Best Regards
>
> Josue
>
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------------------------------
Message: 6
Date: Thu, 19 Apr 2012 14:29:45 -0700
From: "motty.cruz" <motty.cruz at gmail.com>
Subject: [asterisk-users] Auto answer Asterisk ; Unable to create
channel of type
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <C9DD3D4CF11F4EAE97472100826F2F82 at loaner1>
Content-Type: text/plain; charset="us-ascii"
Hello, I'm trying to get s extensions to autoanswer to Centos computer
speakers, the computer is a Dell Optiplex 170L embeded sound card. I'm
running Centos 6.2 i386 with Asterisk 1.8.10
Does anybody know how to fix error below?
-- Executing [s at default:1] Dial("SIP/publicip-00000001",
"console/sda1,20,A(trek)") in new stack
[Apr 19 14:25:25] WARNING[2966]: chan_oss.c:377 find_desc: could not find
<sda1>
[Apr 19 14:25:25] WARNING[2966]: chan_oss.c:850 oss_request: oss_request ty
<console> data 0x0xb6a448e8 <sda1>
[Apr 19 14:25:25] NOTICE[2966]: chan_oss.c:852 oss_request: Device sda1 not
found
[Apr 19 14:25:25] WARNING[2966]: app_dial.c:2218 dial_exec_full: Unable to
create channel of type 'console' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s at default:2] Hangup("SIP/publicip-00000001",
"") in new
stack
== Spawn extension (default, s, 2) exited non-zero on
'SIP/publicip-00000001'
#lspci -v
00:1f.5 Multimedia audio controller: Intel Corporation 82801EB/ER
(ICH5/ICH5R) AC'97 Audio Controller (rev 02)
Subsystem: Dell Device 017a
Flags: bus master, medium devsel, latency 0, IRQ 17
I/O ports at ee00 [size=256]
I/O ports at edc0 [size=64]
Memory at feb7fa00 (32-bit, non-prefetchable) [size=512]
Memory at feb7f900 (32-bit, non-prefetchable) [size=256]
Capabilities: [50] Power Management version 2
Kernel driver in use: Intel ICH
Kernel modules: snd-intel8x0
Thanks,
------------------------------
Message: 7
Date: Thu, 19 Apr 2012 16:31:05 -0500 (CDT)
From: "Jeff Brower" <jbrower at signalogic.com>
Subject: Re: [asterisk-users] g729 freezes 1.8
To: "A J Stiles" <asterisk_list at earthshod.co.uk>
Cc: asterisk-users at lists.digium.com
Message-ID:
<4739.64.219.188.225.1334871065.squirrel at office.signalogic.com>
Content-Type: text/plain;charset=iso-8859-1
AJ-
> On Thursday 19 April 2012, samuel wrote:
>> Just in case it helps:
>>
>> It turned out that from asterisk version 1.8.4 on, the g729 binaries
are
>> different from the previous versions so it was a version mismatch
between
>> the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
>>
>> Thanks to the Digium support department that found out the issue.
>
> Someone really needs to get the mPlayer folks (based on the Continent,
where
> mathematics is not patentable) to create an Open Source g729 codec
> implementation .....
<IMO>
Transformations are patented, not mathematics. This is true in the US also, if
you want to create solid patents.
Algorithms are certainly patented in both US and Europe -- Fraunhofer and MP3 is
a good example. Although many people
disagree whether this should be allowed, there has yet to be high level court
cases to decide the issue.
As for G729 and other codecs, they can also be implemented in hardware, in which
case there would be novel circuit
apparatus that does the job. And if someone used software to get the same
results -- violation. People too often
think that just because something can conveniently be done in software,
traditional patent law no longer matters...
that doesn't mean hardware approaches somehow disappeared. People who write
algorithm patents know this and make them
more solid using hardware techniques as "additional methods".
You're dreaming if you think you can use G729 in ways other than what the
patent holders grant. The only reason you
don't get bothered (yet) is if you're not making money.
</IMO>
-Jeff
------------------------------
Message: 8
Date: Thu, 19 Apr 2012 15:59:19 -0700 (PDT)
From: bilal ghayyad <bilmar_gh at yahoo.com>
Subject: Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6 on Ubuntu
To: asterisk-users at lists.digium.com
Message-ID:
<1334876359.30755.YahooMailClassic at
web162004.mail.bf1.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
Dears;
I see this at the /var/log/asterisk/messages:
[Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open
/dev/dahdi/transcode: No such file or directory
Again, I am installing asterisk and dahdi at Ubuntu (uname -a
Linux House 3.0.0-17-server #30-Ubuntu SMP Thu Mar 8 22:15:30 UTC 2012 x86_64
x86_64 x86_64 GNU/Linux).
I do not know if you were talking about the messages logs or about someting
else?
Anyway, these are the logs that I see at the messages after running
/etc/init.d/asterisk restart:
[Apr 20 01:49:48] NOTICE[1657] cdr.c: CDR simple logging enabled.
[Apr 20 01:49:48] NOTICE[1657] loader.c: 142 modules will be loaded.
[Apr 20 01:49:48] WARNING[1657] res_smdi.c: No SMDI interfaces are available to
listen on, not starting SMDI listener.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: Starting AEL load process.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: calculated config
file name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: parsed config file
name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: checked config file
name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: compiled config file
name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: merged config file
name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: verified config file
name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open
/dev/dahdi/transcode: No such file or directory
Regards
Bilal
---------------
> > Dear Warren;
> >
> > Yes I am compiling and installing dahdi first and then
> I start by asterisk 1.4.39 but I do not find chan_dahdi
> under /usr/lib/asterisk/modules, but if I used asterisk 1.8,
> it is working fine.
> >
> >? From the other side: I tried asterisk 1.4.44 and
> same thing (I am not able to see the chan_dahdi) !!
> >
> > By the way, I am using ubuntu.
> >
> > Which asterisk 1.4 version that you tried it with dahdi
> and you were able to find the chan_dahdi?
> >
> > Really I tried too many attempts and until now I am not
> able to find a solution ! What I am missing?
>
> You are asking people to help you guess what is wrong,
> instead of
> looking at the output of the Asterisk configure script. When
> the
> configure script checks for DAHDI, if that process fails for
> some
> reason, it will tell you why. That information is required
> for anyone to
> be able to help you.
>
> --
> Kevin P. Fleming
------------------------------
Message: 9
Date: Thu, 19 Apr 2012 18:49:51 -0700
From: Chad Wallace <cwallace at lodgingcompany.com>
Subject: Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6
To: asterisk-users at lists.digium.com
Message-ID: <20120419184951.76ef3759 at ws80.int.tlc>
Content-Type: text/plain; charset=US-ASCII
On Wed, 18 Apr 2012 06:12:45 -0700 (PDT)
bilal ghayyad <bilmar_gh at yahoo.com> wrote:
> Yes, first thing I do is the make all and make install for dahdi,
> then I do ./configure and make and make install for asterisk. But I
> do not find the chan_dahdi under the /usr/lib/asterisk/modules. WHY?
You probably need to run make menuselect after ./configure and before
make to select dahdi for building & installation.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
------------------------------
Message: 10
Date: Fri, 20 Apr 2012 09:40:18 +0100
From: Steven Howes <steve-lists at geekinter.net>
Subject: Re: [asterisk-users] Company info
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <07B57BEB-FB4B-4AFD-BEF9-077E35C1D242 at geekinter.net>
Content-Type: text/plain; charset="iso-8859-1"
Can't tell if this is a transparent attempt at advertising, or...?
S
On 19 Apr 2012, at 22:09, Josu? Conti wrote:
> This is your website:
>
> http://www.convergia.com/
>
> Thanks in advanced for any informations.
>
> Best Regards
>
> Josue
>
> Em 19 de abril de 2012 17:11, Josu? Conti <josueconti at gmail.com>
escreveu:
> Dear all,
> Please let me know if anybody have informations about a company called
Convergia, like your products, ASR/ACD or more details.
>
> With Best Regards
>
> Josue
>
> --
> _____________________________________________________________________
> -- 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
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------------------------------
Message: 11
Date: Fri, 20 Apr 2012 05:34:49 -0400
From: Doug Lytle <support at drdos.info>
Subject: Re: [asterisk-users] Company info
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <sig.04573a6929.4F912DB9.7050108 at drdos.info>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Steven Howes wrote:> Can't tell if this is a transparent attempt at advertising, or...?
>
If he doesn't asking any further questions, we'll know.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
------------------------------
Message: 12
Date: Fri, 20 Apr 2012 09:45:33 -0300
From: Josu? Conti <josueconti at gmail.com>
Subject: Re: [asterisk-users] Company info
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CANVtW_DkqXzphhLjDsEzA-G1PphB=YA5EAfW3s6iFHLSuND9JQ at
mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Dear Steven, no is not.
I?m looking for sip connections and this company is a opportunity, but I
would like to know if anybody have informations or use your products, just
it.
Like, this company is confident?
My apologies if seemed this.
With Best Regards
Josue
Em 20 de abril de 2012 05:40, Steven Howes <steve-lists at
geekinter.net>escreveu:
> Can't tell if this is a transparent attempt at advertising, or...?
>
> S
>
> On 19 Apr 2012, at 22:09, Josu? Conti wrote:
>
> This is your website:
>
> http://www.convergia.com/
>
> Thanks in advanced for any informations.
>
> Best Regards
>
> Josue
>
> Em 19 de abril de 2012 17:11, Josu? Conti <josueconti at gmail.com>
escreveu:
>
>> Dear all,
>> Please let me know if anybody have informations about a company called
>> Convergia, like your products, ASR/ACD or more details.
>>
>> With Best Regards
>>
>> Josue
>>
>
> --
> _____________________________________________________________________
> -- 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
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
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------------------------------
Message: 13
Date: Fri, 20 Apr 2012 14:51:03 +0200
From: Brynjolfur Thorvardsson <binni at itanet.nu>
Subject: [asterisk-users] Experience with virtual servers?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<FD89C36847B5104A9530494F3FB66E40ECC31D6802 at host.itanet.nu>
Content-Type: text/plain; charset="us-ascii"
Hi All
Does anybody have experience with running Asterisk on virtual servers? I have
been experimenting with two suppliers and I am not altogether happy with sound
quality etc.
Is it perhaps foolish to try and install a "production" Asterisk
server on a virtual machine? With dedicated servers being comparatively cheap
(although still several times more expensive than virtual servers), perhaps that
is the way I should be going? I have heard someone mention "Asterisk
friendly" VPS providers, how can you tell if they are or aren't
friendly?
We currently have our Asterisk server running on a five year old single AMD CPU
32 bit machine with 512Mb and that works fine. Even the cheapest virtual server
vendors offer servers that seem much more powerful but after testing I am not so
sure any more!
Any info would be very welcome!
Regards
Binni
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Message: 14
Date: Fri, 20 Apr 2012 14:05:14 +0100 (BST)
From: Arthur Stanfield <aj at dmcip.com>
Subject: Re: [asterisk-users] Experience with virtual servers?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <663195107.180017.1334927114649.JavaMail.root at pomona>
Content-Type: text/plain; charset=utf-8
Hi Binni,
We run a number of Asterisk servers on virtual machines. I'm not heavily
involved in the virtualisation side of the business so i'm afraid i
can't give you much advice on it, Past saying it is possible to have an
Asterisk System up and running reliably on virtual machines.
Our virtualisation platform is KVM based.
Hopefully someone with more knowledge than me will be able to help!.
Cheers,
AJ.
----- Original Message -----
From: "Brynjolfur Thorvardsson" <binni at itanet.nu>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Friday, 20 April, 2012 1:51:03 PM
Subject: [asterisk-users] Experience with virtual servers?
Hi All
Does anybody have experience with running Asterisk on virtual servers? I have
been experimenting with two suppliers and I am not altogether happy with sound
quality etc.
Is it perhaps foolish to try and install a ?production? Asterisk server on a
virtual machine? With dedicated servers being comparatively cheap (although
still several times more expensive than virtual servers), perhaps that is the
way I should be going? I have heard someone mention ?Asterisk friendly? VPS
providers, how can you tell if they are or aren?t friendly?
We currently have our Asterisk server running on a five year old single AMD CPU
32 bit machine with 512Mb and that works fine. Even the cheapest virtual server
vendors offer servers that seem much more powerful but after testing I am not so
sure any more!
Any info would be very welcome!
Regards
Binni
--
_____________________________________________________________________
-- 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
------------------------------
Message: 15
Date: Fri, 20 Apr 2012 08:11:42 -0500
From: "Danny Nicholas" <danny at debsinc.com>
Subject: Re: [asterisk-users] Experience with virtual servers?
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <002501cd1ef7$211a4ea0$634eebe0$@debsinc.com>
Content-Type: text/plain; charset="utf-8"
As long as you are using SIP trunking, Asterisk will perform nicely. If you
want PRI or DAHDI trunks, that's a different bridge to cross.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Arthur Stanfield
Sent: Friday, April 20, 2012 8:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Experience with virtual servers?
Hi Binni,
We run a number of Asterisk servers on virtual machines. I'm not heavily
involved in the virtualisation side of the business so i'm afraid i
can't give you much advice on it, Past saying it is possible to have an
Asterisk System up and running reliably on virtual machines.
Our virtualisation platform is KVM based.
Hopefully someone with more knowledge than me will be able to help!.
Cheers,
AJ.
----- Original Message -----
From: "Brynjolfur Thorvardsson" <binni at itanet.nu>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Friday, 20 April, 2012 1:51:03 PM
Subject: [asterisk-users] Experience with virtual servers?
Hi All
Does anybody have experience with running Asterisk on virtual servers? I have
been experimenting with two suppliers and I am not altogether happy with sound
quality etc.
Is it perhaps foolish to try and install a ?production? Asterisk server on a
virtual machine? With dedicated servers being comparatively cheap (although
still several times more expensive than virtual servers), perhaps that is the
way I should be going? I have heard someone mention ?Asterisk friendly? VPS
providers, how can you tell if they are or aren?t friendly?
We currently have our Asterisk server running on a five year old single AMD CPU
32 bit machine with 512Mb and that works fine. Even the cheapest virtual server
vendors offer servers that seem much more powerful but after testing I am not so
sure any more!
Any info would be very welcome!
Regards
Binni
--
_____________________________________________________________________
-- 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
--
_____________________________________________________________________
-- 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
------------------------------
Message: 16
Date: Fri, 20 Apr 2012 20:15:24 +0700
From: Stuart Elvish - IP Exchange Systems <asterisk.lists at ipesys.com>
Subject: Re: [asterisk-users] Experience with virtual servers?
To: asterisk-users at lists.digium.com
Message-ID: <4F91616C.30100 at ipesys.com>
Content-Type: text/plain; charset=windows-1252
Hi Binni,
It often depends on how over-subscribed / over-sold the server is as
well as CPU scheduling. People often suggest KVM VPS' over OpenVZ etc.
There are a few companies that have VPS products specifically designed
for voice hosting (presumably a lower ratio of VM's per server and
upstream bandwidth better suited for voice) and this may be worth
investigating. Some even provide templates for provisioning your
container / VM with TrixBox or Elastix.
If you are looking for a particular geographical location or a have a
specific solution (conferencing / trunk lines etc) your options won't be
as many.
Kind Regards
Stuart
On 04/20/2012 07:51 PM, Brynjolfur Thorvardsson wrote:> Hi All
>
>
>
> Does anybody have experience with running Asterisk on virtual servers? I
> have been experimenting with two suppliers and I am not altogether happy
> with sound quality etc.
>
>
>
> Is it perhaps foolish to try and install a ?production? Asterisk server
> on a virtual machine? With dedicated servers being comparatively cheap
> (although still several times more expensive than virtual servers),
> perhaps that is the way I should be going? I have heard someone mention
> ?Asterisk friendly? VPS providers, how can you tell if they are or
> aren?t friendly?
>
>
>
> We currently have our Asterisk server running on a five year old single
> AMD CPU 32 bit machine with 512Mb and that works fine. Even the cheapest
> virtual server vendors offer servers that seem much more powerful but
> after testing I am not so sure any more!
>
>
>
> Any info would be very welcome!
>
>
>
> Regards
>
>
>
> Binni
>
>
>
> --
> _____________________________________________________________________
> -- 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
------------------------------
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Register Now: http://www.astricon.net/
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End of asterisk-users Digest, Vol 93, Issue 28
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