Deka, Rajib IN MAA SL
2011-Apr-07 09:24 UTC
[asterisk-users] asterisk SIP MESSAGE method support
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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/8ec3b210/attachment.htm>
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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/848f86bf/attachment.htm>
Deka, Rajib IN MAA SL
2011-Apr-07 13:32 UTC
[asterisk-users] asterisk SIP MESSAGE method support
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 -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Thursday, April 07, 2011 6:20 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 81, Issue 19 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-request at lists.digium.com You can reach the person managing the list at asterisk-users-owner at lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/8ec3b210/attachment-0001.htm> ------------------------------ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/b00b4082/attachment-0001.htm> ------------------------------ 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/848f86bf/attachment-0001.htm> ------------------------------ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/09036c23/attachment-0001.htm> ------------------------------ 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/0993ebe7/attachment-0001.htm> ------------------------------ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/85066224/attachment.htm> ------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Register Now: http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 81, Issue 19 ********************************************** 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.
On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote:> Is the following is the link for getting the source, > http://svn.asterisk.org/svn/asterisk/trunk/Please try not to reply to the entire digest.. S
Deka, Rajib IN MAA SL
2011-Apr-07 13:59 UTC
[asterisk-users] asterisk SIP MESSAGE method support
Is the following trunk has development version of out-of-call messaging capability, also what is the version of asterisk, http://svn.asterisk.org/svn/asterisk/trunk/ Regards, Rajib ------------------------------ Message: 10 Date: Thu, 7 Apr 2011 14:42:35 +0100 From: Steven Howes <steve-lists at geekinter.net> 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: <D5D50321-4B5B-41BD-B8A3-8BCCEAFC28C8 at geekinter.net> Content-Type: text/plain; charset=us-ascii On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote:> Is the following is the link for getting the source, > http://svn.asterisk.org/svn/asterisk/trunk/Please try not to reply to the entire digest.. S 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.