[asterisk-users] asterisk SIP MESSAGE method support
Deka, Rajib IN MAA SL
rajib.deka at siemens.com
Thu Apr 7 08:32:24 CDT 2011
Is the following is the link for getting the source,
http://svn.asterisk.org/svn/asterisk/trunk/
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com
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Today's Topics:
1. asterisk SIP MESSAGE method support (Deka, Rajib IN MAA SL)
2. Re: Iptables configuration to handle brute force
registrations? (Gilles)
3. Re: BRI Configuration help me (mahesh katta)
4. Re: Iptables configuration to handle brute, force
registrations? (Gilles)
5. Compiling asterisk using NDK build (Nikhil)
6. Re: asterisk SIP MESSAGE method support (Olivier)
7. Re: BRI Configuration help me (Tzafrir Cohen)
8. Re: Compiling asterisk using NDK build (Tzafrir Cohen)
9. Re: BRI Configuration help me (mahesh katta)
10. Re: Trunk form asterisk1 to asterisk2 fails (GiGi)
11. Re: Asterisk 1.8.3 (Satish Patel)
12. Re: Asterisk 1.8.3 (Bryant Zimmerman)
13. Re: BRI Configuration help me (mahesh katta)
----------------------------------------------------------------------
Message: 1
Date: Thu, 7 Apr 2011 14:54:23 +0530
From: "Deka, Rajib IN MAA SL" <rajib.deka at siemens.com>
Subject: [asterisk-users] asterisk SIP MESSAGE method support
To: "asterisk-users at lists.digium.com"
<asterisk-users at lists.digium.com>
Message-ID:
<2658E54B540D284981EA57E6A549EA70A592F02E96 at INBLRK77M1MSX.in002.siemens.net>
Content-Type: text/plain; charset="us-ascii"
Hello List,
I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call.
But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks?
Regards,
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com<http://www.siemens.com>
Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com
________________________________
Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system.
Thank You.
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Message: 2
Date: Thu, 07 Apr 2011 12:51:48 +0200
From: Gilles <codecomplete at free.fr>
Subject: Re: [asterisk-users] Iptables configuration to handle brute
force registrations?
To: asterisk-users at lists.digium.com
Message-ID: <ko5rp6huuoqu2suivok9f0p0nccb4n987r at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson
<gordon+asterisk at drogon.net> wrote:
>Have a look at these:
Thanks much Gordon. I'll study the scripts you mentionned. It looks
like iptables is good enough and I won't have to install a second tool
to watch the logs and reconfigure iptables on the fly.
------------------------------
Message: 3
Date: Thu, 7 Apr 2011 16:48:13 +0530
From: mahesh katta <maheshkatta at flexydial.com>
Subject: Re: [asterisk-users] BRI Configuration help me
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTikP-CfWjOGw5--D48EuHT=Afr_nsQ at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Sir,
my files are in fistmail that is my configuration.
and till its disconnecting the line
On Thu, Apr 7, 2011 at 2:35 PM, Tzafrir Cohen <tzafrir.cohen at xorcom.com>wrote:
> Hi,
>
> Un-top-posting
>
> On Thu, Apr 07, 2011 at 12:53:18PM +0530, mahesh katta wrote:
> >
> > On Wed, Apr 6, 2011 at 9:04 PM, Tzafrir Cohen <tzafrir.cohen at xorcom.com
> >wrote:
> >
> > > On Wed, Apr 06, 2011 at 07:15:00PM +0530, mahesh katta wrote:
> > > > Sir,
> > > >
> > > > i am using goautodial server , bri card is showing ok but when i try
> to
> > > call
> > > > that showing below ,
> > > > This configuration is in doing in dubai , so kindly help me how can
> > > connet
> > > > the call from this ,
> > > > what is my mistake is in this
> > > >
> > > >
> > >
> > > > :::chan-dahdi.conf
> > > > [channels]
> > > >
> > > > #include
> > > > dahdi-channels.conf
> > >
> > > Is this line originally broken?
>
> I believe this line belongs here:
>
> >
> > This was comming and even i enterd that file last.
>
> Though I'm still not sure what you mean. If it is broken, it shouldn't
> be. It should be on the same line.
>
> >
> > >
> > > Anyway, you should have it in the end of chan_dahdi.conf .
> > >
> > > What do you have in /etc/asterisk/dahdi-channels.conf ?
> > >
> > > What's the output of lsdahdi ? dahdi_hardware ?
>
> > [root at go ~]#
> > dahdi_hardware
> >
> > pci:0000:04:02.0 wcb4xxp+ d161:b410 Digium Wildcard B410P
>
> >
> > then also its not connecting
>
> Fine. How about my other questions?
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.cohen at xorcom.com
> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>
> --
> _____________________________________________________________________
> -- 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
>
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Message: 4
Date: Thu, 07 Apr 2011 13:27:25 +0200
From: Gilles <codecomplete at free.fr>
Subject: Re: [asterisk-users] Iptables configuration to handle brute,
force registrations?
To: asterisk-users at lists.digium.com
Message-ID: <6r5rp6lru9eihujp3v0hrq0sr17rq47hbr at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Tue, 5 Apr 2011 17:38:15 -0400, Paul Dugas
<paul at dugasenterprises.com> wrote:
>First, this appears to be working for me though I'm not 100% sure of
>that and cannot guarantee it will for you in any way, shape or form.
>With the lawyering out of the way...
Thanks a lot, Paul.
------------------------------
Message: 5
Date: Thu, 07 Apr 2011 16:58:33 +0530
From: Nikhil <d.nikhil at cem-solutions.net>
Subject: [asterisk-users] Compiling asterisk using NDK build
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4D9D9FE1.2000101 at cem-solutions.net>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi all,
Does anyone compiled asterisk using NKD build in android. Please
give some suggestions.
Thanks
Nikhil
------------------------------
Message: 6
Date: Thu, 7 Apr 2011 13:38:20 +0200
From: Olivier <oza_4h07 at yahoo.fr>
Subject: Re: [asterisk-users] asterisk SIP MESSAGE method support
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTik0eYZXgmwN5=PGKSMx2OTUg02MDA at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
2011/4/7 Deka, Rajib IN MAA SL <rajib.deka at siemens.com>
> Hello List,
>
>
>
> I have found that asterisk supports only forwards in-dialog MESSAGE method.
> That is, if the MESSAGE method is sent within an active call.
>
>
>
> But according our requirement we need to send MESSAGE method to the other
> leg without being in a call (general stateless proxy forward).
>
There is ongoing development to enhance Text support in Asterisk's trunk.
Out-of-call messaging is one those features.
Regards
> Is it possible to do this in asterisk using some tricks?
>
>
>
> Regards,
>
>
>
> *Rajib Deka*
>
> SIEMENS Ltd.
>
> Robert V Chandran Tower, First Floor, West Wing,
>
> #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
>
> www.siemens.com
>
>
>
> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com
>
>
>
> ------------------------------
> Important notice: This e-mail and any attachment there to contains
> corporate proprietary information. If you have received it by mistake,
> please notify us immediately by reply e-mail and delete this e-mail and its
> attachments from your system.
> Thank You.
>
> --
> _____________________________________________________________________
> -- 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: 7
Date: Thu, 7 Apr 2011 14:51:07 +0300
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] BRI Configuration help me
To: asterisk-users at lists.digium.com
Message-ID: <20110407115107.GI6408 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
On Thu, Apr 07, 2011 at 04:48:13PM +0530, mahesh katta wrote:
> Sir,
>
> my files are in fistmail that is my configuration.
>
> and till its disconnecting the line
/me gives up. Anybody else wants to take a shot here?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 8
Date: Thu, 7 Apr 2011 14:51:43 +0300
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] Compiling asterisk using NDK build
To: asterisk-users at lists.digium.com
Message-ID: <20110407115143.GJ6408 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
On Thu, Apr 07, 2011 at 04:58:33PM +0530, Nikhil wrote:
> Hi all,
> Does anyone compiled asterisk using NKD build in android. Please
> give some suggestions.
Have you tried? What errors do you get?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 9
Date: Thu, 7 Apr 2011 17:29:48 +0530
From: mahesh katta <maheshkatta at flexydial.com>
Subject: Re: [asterisk-users] BRI Configuration help me
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTikuj5dga_t15qgcHabwxvsg+w5rrw at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
any buddy is there for this solution.
On Thu, Apr 7, 2011 at 5:21 PM, Tzafrir Cohen <tzafrir.cohen at xorcom.com>wrote:
> On Thu, Apr 07, 2011 at 04:48:13PM +0530, mahesh katta wrote:
> > Sir,
> >
> > my files are in fistmail that is my configuration.
> >
> > and till its disconnecting the line
>
> /me gives up. Anybody else wants to take a shot here?
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.cohen at xorcom.com
> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>
> --
> _____________________________________________________________________
> -- 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
>
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Message: 10
Date: Thu, 7 Apr 2011 12:05:59 +0000 (UTC)
From: GiGi <kalss21 at gmail.com>
Subject: Re: [asterisk-users] Trunk form asterisk1 to asterisk2 fails
To: asterisk-users at lists.digium.com
Message-ID: <loom.20110407T140357-310 at post.gmane.org>
Content-Type: text/plain; charset=us-ascii
Jonas Kellens <jonas.kellens <at> telenet.be> writes:
>
>
> On 03/16/2011 08:39 PM, Jonas Kellens wrote:
>
>
> Found the answer to my own question : fromuser in the peer definition
> Kind regards,
> Jonas.
>
>
> --
> _____________________________________________________________________
Can you extend a little bit this fix? I have a similar problem forwarding a call
to another Asterisk. Thank you.
------------------------------
Message: 11
Date: Thu, 7 Apr 2011 08:20:56 -0400
From: Satish Patel <satish_lx at hotmail.com>
Subject: Re: [asterisk-users] Asterisk 1.8.3
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BLU0-SMTP34DAB2A0D1D6A3C6D096D190A40 at phx.gbl>
Content-Type: text/plain; charset="us-ascii"; format=flowed; delsp=yes
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning to put this in
production. Please suggest me what should I do?
--
Sent from my iPhone
On Apr 6, 2011, at 8:54 PM, Edwin Lam <edwin.lam at officegeneral.com>
wrote:
> On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
>>
>> Thanks for your response. I have added the patch for 18818 per
>> Michel Verbrask's
>> recomendation. It appers that it has made quite a difference. I
>> don't have an PRI
>> connections as all of our PRI's are connected via SIP gateways. I
>> did run into
>> serveral instances wher I had to kill -9 the process as well but
>> post patch I have
>> been in good shape know on wood. I hope there will be a new release
>> that will
>> address the stability issues very soon if they release 1.8.4
>> without cleaning this
>> up I won't move unitl it is addressed.
>
> looking back at the messages file for the past 2 days. it
> just hanged on totally different events none of which related
> to Local channels.
>
> as far as the PRI not hearing early media issue. here's the
> excerpt from the messages file after "pri debug on" command:
>
> *********************
>
> -- Executing [18008291011 at out_going_x:1] Dial("SIP/
> 4988-6-00000b45", "DAHDI/r1/18008291011,,f") in new stack
> -- Making new call for cref 32974
> -- Requested transfer capability: 0x00 - SPEECH
>
> > DL-DATA request
> > Protocol Discriminator: Q.931 (8) len=51
> > TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent from originator)
> > Message Type: SETUP (5)
> TEI=0 Transmitting N(S)=87, window is open V(A)=87 K=7
>
> > Protocol Discriminator: Q.931 (8) len=51
> > TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent from originator)
> > Message Type: SETUP (5)
> > [04 03 80 90 a2]
> > 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: u-Law (34)
> > [18 03 a1 83 8a]
> > 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: 10 Type: CPE]
> > [28 06 b1 45 64 77 69 6e]
> > Display (len= 6) Charset: 31 [ Edwin ]
> > [6c 0c 21 83 34 31 35 34 33 39 34 39 38 38]
> > Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
> ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> > Presentation: Presentation allowed of
> network provided number (3) '4154394988' ]
> > [70 0c 80 31 38 30 30 38 32 39 31 30 31 31]
> > Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0)
> NPI: Unknown Number Plan (0) '18008291011' ]
> q931.c:5039 q931_setup: Call 32974 enters state 1 (Call Initiated).
> Hold state: Idle
> -- Called r1/18008291011
>
> < Protocol Discriminator: Q.931 (8) len=13
> < TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent to originator)
> < Message Type: STATUS (125)
> < [08 03 80 ab 28]
> < Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare:
> 0 Location: User (0)
> < Ext: 1 Cause: Access information discarded (43),
> class = Network Congestion (resource unavailable) (2) ]
> < Cause data 1: 28 (40)
> < [14 01 01]
> < Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0)
> Call state: Call Initiated (1)
> Received message for call 0x8fd2298 on 0x90d9cf0 TEI/SAPI 0/0, call-
> >pri is 0x90d9cf0 TEI/SAPI 0/0
> -- Processing IE 8 (cs0, Cause)
> -- Processing IE 20 (cs0, Call State)
>
> < Protocol Discriminator: Q.931 (8) len=10
> < TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent to originator)
> < Message Type: CALL PROCEEDING (2)
> < [18 03 a9 83 8a]
> < 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: 10 Type: CPE]
> Received message for call 0x8fd2298 on 0x90d9cf0 TEI/SAPI 0/0, call-
> >pri is 0x90d9cf0 TEI/SAPI 0/0
> -- Processing IE 24 (cs0, Channel Identification)
> q931.c:7104 post_handle_q931_message: Call 32974 enters state 3
> (Outgoing Call Proceeding). Hold state: Idle
> -- DAHDI/34-1 is proceeding passing it to SIP/4988-6-00000b45
>
> < Protocol Discriminator: Q.931 (8) len=13
> < TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent to originator)
> < Message Type: PROGRESS (3)
> < [08 02 82 ff]
> < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare:
> 0 Location: Public network serving the local user (2)
> < Ext: 1 Cause: Interworking, unspecified (127),
> class = Interworking (7) ]
> < [1e 02 82 81]
> < 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: Call
> is not end-to-end ISDN; further call progress information may be
> available inband. (1) ]
> Received message for call 0x8fd2298 on 0x90d9cf0 TEI/SAPI 0/0, call-
> >pri is 0x90d9cf0 TEI/SAPI 0/0
> -- Processing IE 8 (cs0, Cause)
> -- Processing IE 30 (cs0, Progress Indicator)
> -- PROGRESS with cause code 127 received
> -- DAHDI/34-1 is making progress passing it to SIP/4988-6-00000b45
>
> ***********************************
>
> i used the same SIP station to dial the same 800 number
> on both versions (1.8.3.2 & 1.6.2.17). the output are
> pretty much identical except on 1.8.3.2, after the
> "PROGRESS with cause code 127..." message. i would hear
> nothing until the other side timed out & hang up, whereas on
> 1.6.2.17. i got the "DAHDI/... is making progress passing it to
> SIP..."
> message and can hear the early media from the other side.
>
>
>> For Now 1.8.3..2 is very bad.
>
> agreed...
>
>
>
> --
> Edwin Lam <edwin.lam at officegeneral.com>
> Systems Engineer, OfficeWyze, Inc.
> Ph: +1 415 439 4988 Fax: +1 415 283 3370
> http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
>
>
> --
> _____________________________________________________________________
> -- 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: 12
Date: Thu, 7 Apr 2011 08:37:58 -0400
From: "Bryant Zimmerman" <BryantZ at zktech.com>
Subject: Re: [asterisk-users] Asterisk 1.8.3
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <7d15dea3$727b4c79$5739a33d$@com>
Content-Type: text/plain; charset="us-ascii"
On Apr 6, 2011, at 8:54 PM, Edwin Lam <edwin.lam at officegeneral.com>
wrote:
> On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
>>
>> Thanks for your response. I have added the patch for 18818 per
>> Michel Verbrask's
>> recomendation. It appers that it has made quite a difference. I
>> don't have an PRI
>> connections as all of our PRI's are connected via SIP gateways. I
>> did run into
>> serveral instances wher I had to kill -9 the process as well but
>> post patch I have
>> been in good shape know on wood. I hope there will be a new release
>> that will
>> address the stability issues very soon if they release 1.8.4
>> without cleaning this
>> up I won't move unitl it is addressed.
>
> looking back at the messages file for the past 2 days. it
> just hanged on totally different events none of which related
> to Local channels.
>
> as far as the PRI not hearing early media issue. here's the
> excerpt from the messages file after "pri debug on" command:
>
> *********************
>
> -- Executing [18008291011 at out_going_x:1] Dial("SIP/
... Parts Removed see origional response
> -- Processing IE 30 (cs0, Progress Indicator)
> -- PROGRESS with cause code 127 received
> -- DAHDI/34-1 is making progress passing it to SIP/4988-6-00000b45
>
> ***********************************
>
> i used the same SIP station to dial the same 800 number
> on both versions (1.8.3.2 & 1.6.2.17). the output are
> pretty much identical except on 1.8.3.2, after the
> "PROGRESS with cause code 127..." message. i would hear
> nothing until the other side timed out & hang up, whereas on
> 1.6.2.17. i got the "DAHDI/... is making progress passing it to
> SIP..."
> message and can hear the early media from the other side.
>
>
>> For Now 1.8.3..2 is very bad.
>
> agreed...
From: "Satish Patel" <satish_lx at hotmail.com>
Sent: Thursday, April 07, 2011 8:22 AM
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning to put this in
production. Please suggest me what should I do?
Satish
For me 1.8.3.2 has been the worst build that I have tried to use as far a
stability in a very long time. We are having issues with deadlocks and
voicemail.
I don't have a good option for you if you want to run 1.8 currently the
most stable release version I have found is 1.8.2.3 but I am having the
Voicemail issues there as well.
Things like messages not deleting propperly and hanging up the mail box so
users can't check them.
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Message: 13
Date: Thu, 7 Apr 2011 18:15:59 +0530
From: mahesh katta <maheshkatta at flexydial.com>
Subject: Re: [asterisk-users] BRI Configuration help me
To: asterisk-users at lists.digium.com
Message-ID: <BANLkTimBiL4Pe4MYhAwEkoDf84DOVa6rfw at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Sir,
I am using B410p card which BRI. and Mediatrix4400 is bri line provider in
dubai.
below configuration is my bri card configuration. and when try to connect
the call its going disconnect on cli getting
[Apr 6 09:36:37] WARNING[6433]: channel.c:3443 ast_request: No channel type
registered for 'Dahdi'
[Apr 6 09:36:37] WARNING[6433]: app_dial.c:1296 dial_exec_full: Unable to
create channel of type 'Dahdi' (cause 66 - Channel not
implemented)
i more times i changed my configuration. it was comming same
please help me
:::/etc/asterisk/chan-dahdi.conf
[channels]
language=en
context=default
usecallerid=yes
hidecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
relaxdtmf=yes
rxgain=0.0
txgain=0.0
;group=1
;callgroup=1
;pickupgroup=1
busydetect=yes
busycount=6
immediate=no
resetinterval=never
switchtype=euroisdn
signalling=bri_cpe
pridialplan=unknown
prilocaldialplan=unknown
group=0
channel => 1,2,4,5,7,8,10,11
#include dahdi-cahnnes.conf
;;;/etc/asterisk/dahdi-channels.conf
group=0,11
context=from-pstn
switchtype =
euroisdn
signalling =
bri_cpe
channel =>
1-2
context =
default
group =
63
; Span 2: B4/0/2 "B4XXP (PCI) Card 0 Span 2" AMI/CCS
RED
group=0,12
context=from-pstn
switchtype =
euroisdn
signalling =
bri_cpe
channel =>
4-5
context =
default
group =
63
; Span 3: B4/0/3 "B4XXP (PCI) Card 0 Span 3" AMI/CCS
RED
group=0,13
context=from-pstn
switchtype =
euroisdn
signalling =
bri_cpe
channel =>
7-8
context =
default
group =
63
; Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" (MASTER) AMI/CCS
RED
group=0,14
context=from-pstn
switchtype =
euroisdn
signalling =
bri_cpe
channel =>
10-11
context =
default
group = 63
;;;/etc/dahdi/system.conf
# your manual changes will be LOST.
# Dahdi Configuration
File
#
# This file is parsed by the Dahdi Configurator,
dahdi_cfg
#
# Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" AMI/CCS
RED
span=1,1,0,ccs,ami
# termtype:
te
bchan=1-2
dchan=3
echocanceller=mg2,1-2
# Span 2: B4/0/2 "B4XXP (PCI) Card 0 Span 2" AMI/CCS
RED
span=2,2,0,ccs,ami
# termtype:
te
bchan=4-5
dchan=6
echocanceller=mg2,4-5
# Span 3: B4/0/3 "B4XXP (PCI) Card 0 Span 3" AMI/CCS
RED
span=3,3,0,ccs,ami
# termtype:
te
bchan=7-8
dchan=9
echocanceller=mg2,7-8
# Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" (MASTER) AMI/CCS
RED
span=4,4,0,ccs,ami
# termtype:
te
bchan=10-11
dchan=12
echocanceller=mg2,10-11
# Global
data
loadzone =
us
defaultzone = us
On Thu, Apr 7, 2011 at 5:34 PM, Tzafrir Cohen <tzafrir.cohen at xorcom.com>wrote:
> Sent in private mail. I suggest that you don't follow up this message
> directly to the list.
>
> Also:
>
> On Thu, Apr 07, 2011 at 05:29:48PM +0530, mahesh katta wrote:
> > any buddy is there for this solution.
>
> Hint: look up the thread. I asked you some questions. Answer them (to
> the list). Then we can move on.
>
> --
> Tzafrir Cohen
> icq#16849755 jabber:tzafrir.cohen at xorcom.com
> +972-50-7952406 mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
>
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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