Re: [asterisk-users] BRI Configuration help me
[root@go ~]# dahdi_hardware pci::04:02.0 wcb4xxp+ d161:b410 Digium Wildcard B410P This was comming and even i enterd that file last. then also its not connecting On Wed, Apr 6, 2011 at 9:04 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: 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? 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 ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@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 -- _ -- 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
[asterisk-users] Asterisk Avaya SIP Trunking One Way Audio
I am facing one way audio problem in sip trunking between asterisk and avaya. +-+ ++ | avaya sip |---| P1 | +-+ ++ | | | +-+ | Asterisk | WAN - | | LAN +-+ | / ++ / | P2 |--+ ++ When P1 dial P2, P2 hears voice clear but P1 could not hear any voice. My sip.conf is [avaya] type=peer fromdomain=xx.xx.xx.xx host=xx.xx.xx.xx disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=yes -- Regards, Shariq Khan 0333-3501125 -- _ -- 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
Re: [asterisk-users] BRI Configuration help me
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.co...@xorcom.comwrote: 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@go ~]# dahdi_hardware pci::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.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@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
[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.comhttp://www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@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
Re: [asterisk-users] Iptables configuration to handle brute force registrations?
On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson gordon+aster...@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. -- _ -- 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
Re: [asterisk-users] BRI Configuration help me
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.co...@xorcom.comwrote: 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.co...@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@go ~]# dahdi_hardware pci::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.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@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 -- _ -- 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
Re: [asterisk-users] Iptables configuration to handle brute, force registrations?
On Tue, 5 Apr 2011 17:38:15 -0400, Paul Dugas p...@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. -- _ -- 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
[asterisk-users] Compiling asterisk using NDK build
Hi all, Does anyone compiled asterisk using NKD build in android. Please give some suggestions. Thanks Nikhil -- _ -- 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
Re: [asterisk-users] asterisk SIP MESSAGE method support
2011/4/7 Deka, Rajib IN MAA SL rajib.d...@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.d...@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 -- _ -- 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
Re: [asterisk-users] BRI Configuration help me
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.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@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
Re: [asterisk-users] Compiling asterisk using NDK build
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.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@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
Re: [asterisk-users] BRI Configuration help me
any buddy is there for this solution. On Thu, Apr 7, 2011 at 5:21 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: 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.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@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 -- _ -- 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
Re: [asterisk-users] Trunk form asterisk1 to asterisk2 fails
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. -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
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@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@out_going_x:1] Dial(SIP/ 4988-6-0b45, 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/4988-6-0b45 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
Re: [asterisk-users] Asterisk 1.8.3
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@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@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/4988-6-0b45 *** 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...@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. -- _ -- 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
Re: [asterisk-users] BRI Configuration help me
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.co...@xorcom.comwrote: 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.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@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 -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
Holy cow!! Can I just build 1.8.2 over existing 1.8.3 ? When we are going to fix all this thing??? -- Sent from my iPhone On Apr 7, 2011, at 8:37 AM, Bryant Zimmerman brya...@zktech.com wrote: On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@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@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/4988-6-0b45 *** 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...@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. -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote: On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@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@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/4988-6-0b45 *** 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...@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. 1.8.2 is unusable if you use RealTime without the patch in this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
We don't have realtime configuration everything is in plain text file. Is 1.8.3 has realtime issue or general issue? -- Sent from my iPhone On Apr 7, 2011, at 8:51 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote: On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@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@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/4988-6-0b45 *** 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...@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. 1.8.2 is unusable if you use RealTime without the patch in this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
On Apr 7, 2011, at 8:51 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote: On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@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@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/4988-6-0b45 *** 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...@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. 1.8.2 is unusable if you use RealTime without the patch in this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 From: Satish Patel satish...@hotmail.com Sent: Thursday, April 07, 2011 9:06 AM We don't have realtime configuration everything is in plain text file. Is 1.8.3 has realtime issue or general issue? Satish I have seen my issues with the realtime disabled and using just plain text. The issues get worse for me when we move to our realtime confgs. So from my perspective I would say you might get farther with realtime off but I would not bank on it. -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
On 11-04-07 08:20 AM, Satish Patel wrote: Is it ture 1.8.3 is unstable? We are planning to put this in production. Please suggest me what should I do? This is a loaded question, since it really depends on what you plan to do. What does your migration plan look like? What sort of testing have you done with Asterisk? Blindly moving into production with _anything_ is a recipe for trouble. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
Right now I'm testing 1.8.3 in devlopment and respose it pretty good without realtime. (I didn't set realtime). I ran stress test with sipp and pass 5000 call with RTP and no issue at all. I got hogging at system resource but no issue at asterisk. Look like I might go with 1.8.3 and later upgrade with 1.8.4 asap. -- Sent from my iPhone On Apr 7, 2011, at 9:12 AM, Bryant Zimmerman brya...@zktech.com wrote: On Apr 7, 2011, at 8:51 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote: On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@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@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/4988-6-0b45 *** 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...@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. 1.8.2 is unusable if you use RealTime without the patch in this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 From: Satish Patel satish...@hotmail.com Sent: Thursday, April 07, 2011 9:06 AM We don't have realtime configuration everything is in plain text file. Is 1.8.3 has realtime issue or general issue? Satish I have seen my issues with the realtime disabled and using just plain text. The issues get worse for me when we move to our realtime confgs. So from my perspective I would say you might get farther with realtime off but I would not bank on it. -- _ -- 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
Re: [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.d...@siemens.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com Sent: Thursday, April 07, 2011 6:20 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 81, Issue 19 Send asterisk-users mailing list submissions to asterisk-users@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-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@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.d...@siemens.com Subject: [asterisk-users] asterisk SIP MESSAGE method support To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Message-ID: 2658e54b540d284981ea57e6a549ea70a592f02...@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.comhttp://www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.d...@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 codecompl...@free.fr Subject: Re: [asterisk-users] Iptables configuration to handle brute force registrations? To: asterisk-users@lists.digium.com Message-ID: ko5rp6huuoqu2suivok9f0p0nccb4n9...@4ax.com Content-Type: text/plain; charset=us-ascii On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson gordon+aster...@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 maheshka...@flexydial.com Subject: Re: [asterisk-users] BRI Configuration help me To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: BANLkTikP-CfWjOGw5--D48EuHT=afr_...@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.co...@xorcom.comwrote: 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.co...@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
[asterisk-users] AgentCallbackLogin slow in Asterisk 1.4
Good morning ... I'm using Asterisk 1.4.40 AgentCallbackLogin in a Call Center. What is happening isthat when the Call Center has more than 15 simultaneous calls the login application isextremely slow to fall into the low priority, ie, the agent can log in, but takes about 1minute to drop in priority below. .. I've tried to recompile the asterisk, I installed other version of 1.4, but nothing helped ...Detail that the server is new, very good and even making the conversion to MP3recordings, rarely surpassed 10% for processing. Does anyone have any idea what might be causing this slowdown? I thought of usingthe AddQueueMember, but would have to change much in design, so is my second choice for solution. att Eduardo-- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
Best I can tell, multi-tenant parking also hasn't worked in any of the 1.8.x releases. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- 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
Re: [asterisk-users] asterisk SIP MESSAGE method support
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 -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
- Original Message - On 11-04-07 08:20 AM, Satish Patel wrote: Is it ture 1.8.3 is unstable? We are planning to put this in production. Please suggest me what should I do? This is a loaded question, since it really depends on what you plan to do. What does your migration plan look like? What sort of testing have you done with Asterisk? Blindly moving into production with _anything_ is a recipe for trouble. And don't forget that call pickup crashes Asterisk from what would appear release 1.8.1 upwards! We have had to back level to that latest 1.6 branch. https://issues.asterisk.org/view.php?id=18654 -- Thanks, Phil -- _ -- 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
Re: [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-li...@geekinter.net Subject: Re: [asterisk-users] asterisk SIP MESSAGE method support To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: d5d50321-4b5b-41bd-b8a3-8bcceafc2...@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. -- _ -- 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
Re: [asterisk-users] asterisk SIP MESSAGE method support
On 11-04-07 09:59 AM, Deka, Rajib IN MAA SL wrote: 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/ I don't believe the branches has been merged into trunk, you can use russellb's branch [1]. [1] http://svn.digium.com/svn/asterisk/team/russell/messaging/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] Asterisk 1.8.x Skips DTMF Digits on a First DAHDI Initiated Call
Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner.As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1) redial attempt is sufficient. Replicated from different analog phones. Troubleshooting and observations: 1. Provided external power to the TDM400P with FXS daughter cards. It did not help. 2. Using RXGAIN / TXGAIN in /etc/asterisk/dahdi-channels.conf makes situation worse. 3. Played with echo cancellation parameters with no luck. Any ideas will be appreciated. Thank you, Vladimir *DTMF log samples for two consecutive test calls each. * 1. Called ext. 400. Dialed 400 -- call failed, redialed by the Redial button -- call went through. 1.1. 08:46:57 -- passes 40 to the channel and fails 1.2. 08:48:17 -- passes 400 to the channel and goes through. [Apr 7 08:46:57] DTMF[9076] channel.c: DTMF begin '4' received on DAHDI/5-1 [Apr 7 08:46:57] DTMF[9076] channel.c: DTMF begin ignored '4' on DAHDI/5-1 [Apr 7 08:46:57] DTMF[9076] channel.c: DTMF end '4' received on DAHDI/5-1, duration 89 ms [Apr 7 08:46:57] DTMF[9076] channel.c: DTMF end passthrough '4' on DAHDI/5-1 [Apr 7 08:46:57] DTMF[9076] channel.c: DTMF begin '0' received on DAHDI/5-1 [Apr 7 08:46:57] DTMF[9076] channel.c: DTMF begin ignored '0' on DAHDI/5-1 [Apr 7 08:46:58] DTMF[9076] channel.c: DTMF end '0' received on DAHDI/5-1, duration 76 ms [Apr 7 08:46:58] DTMF[9076] channel.c: DTMF end passthrough '0' on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF begin '4' received on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF begin ignored '4' on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF end '4' received on DAHDI/5-1, duration 89 ms [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF end passthrough '4' on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF begin '0' received on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF begin ignored '0' on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF end '0' received on DAHDI/5-1, duration 89 ms [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF end passthrough '0' on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF begin '0' received on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF begin ignored '0' on DAHDI/5-1 [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF end '0' received on DAHDI/5-1, duration 76 ms [Apr 7 08:48:17] DTMF[9115] channel.c: DTMF end passthrough '0' on DAHDI/5-1 2. Called ext. 330. Dialed 330 -- call failed, redialed by the Redial button -- call went through. 2.1. 09:48:15 -- passes 3 to the channel and fails 2.2. 09:48:30 -- passes 330 to the channel and goes through. [Apr 7 09:48:15] DTMF[9536] channel.c: DTMF begin '3' received on DAHDI/5-1 [Apr 7 09:48:15] DTMF[9536] channel.c: DTMF begin ignored '3' on DAHDI/5-1 [Apr 7 09:48:15] DTMF[9536] channel.c: DTMF end '3' received on DAHDI/5-1, duration 89 ms [Apr 7 09:48:15] DTMF[9536] channel.c: DTMF end passthrough '3' on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF begin '3' received on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF begin ignored '3' on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF end '3' received on DAHDI/5-1, duration 89 ms [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF end passthrough '3' on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF begin '3' received on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF begin ignored '3' on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF end '3' received on DAHDI/5-1, duration 89 ms [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF end passthrough '3' on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF begin '0' received on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF begin ignored '0' on DAHDI/5-1 [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF end '0' received on DAHDI/5-1, duration 76 ms [Apr 7 09:48:30] DTMF[9539] channel.c: DTMF end passthrough '0' on DAHDI/5-1 *Configuration: * Asterisk 1.8.3.2 DAHDI Version: 2.4.1 Echo Canceller: MG2, HPEC FreePBX http://www.freepbx.org 2.9.0rc1.1 /*pbx*CLI dahdi show status*/ Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM410P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) /*/etc/dahdi/system.conf*/ # Autogenerated by /usr/sbin/dahdi_genconf on Sun Sep 26 00:01:18 2010 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCTDM/0 Wildcard TDM410P Board 1
Re: [asterisk-users] Asterisk 1.8.3
2011/4/7 Bryant Zimmerman brya...@zktech.com 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. Hi, If my memory serves me right, first usable 1.4 version was 1.4.21 or something. Time will tell if things are improving and hopefully next 1.10 would be usable from the very start (from 1.10.0). Is the asterisk testing framework easy enough to work with so that we could feed new tests into it and help devs to identify such regressions before GA release ? Cheers -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, April 07, 2011 10:27 AM To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8.3 2011/4/7 Bryant Zimmerman brya...@zktech.com 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. Hi, If my memory serves me right, first usable 1.4 version was 1.4.21 or something. Time will tell if things are improving and hopefully next 1.10 would be usable from the very start (from 1.10.0). Is the asterisk testing framework easy enough to work with so that we could feed new tests into it and help devs to identify such regressions before GA release ? Cheers [Danny Nicholas] 1.4.21 was the last ZAPTEL version. All versions from 1.4.22 forward have been DAHDI. Stability and usability depend on what variables you throw at it and your relative skill set. -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
2011/4/7 Danny Nicholas da...@debsinc.com -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Thursday, April 07, 2011 10:27 AM *To:* brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk 1.8.3 2011/4/7 Bryant Zimmerman brya...@zktech.com 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. Hi, If my memory serves me right, first usable 1.4 version was 1.4.21 or something. Time will tell if things are improving and hopefully next 1.10 would be usable from the very start (from 1.10.0). Is the asterisk testing framework easy enough to work with so that we could feed new tests into it and help devs to identify such regressions before GA release ? Cheers *[Danny Nicholas] * *1.4.21 was the last ZAPTEL version. All versions from 1.4.22 forward have been DAHDI. * True. *Stability and usability depend on what variables you throw at it and your relative skill set.* Of course -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
On 11-04-07 11:26 AM, Olivier wrote: Is the asterisk testing framework easy enough to work with so that we could feed new tests into it and help devs to identify such regressions before GA release ? +1 There is a learning curve to creating tests for the testsuite[1], but nothing too drastic. I'd suggest installing in on a local system and run it to see it in action. We already have a few tests in place, but always looking for more. To anybody that takes the time to write and submit a test to the issue tracker / reviewboard, I would help triage it and help get it merged ASAP. :) [1] http://svn.digium.com/svn/testsuite/asterisk/trunk/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] No ringback even though progressinband=yes is set
Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- 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
Re: [asterisk-users] Asterisk 1.8.3
On 11-04-07 09:45 AM, --[ UxBoD ]-- wrote: And don't forget that call pickup crashes Asterisk from what would appear release 1.8.1 upwards! We have had to back level to that latest 1.6 branch. https://issues.asterisk.org/view.php?id=18654 I ran into this issue as well on 1.8.3.2, but I didn't try a newer version, and someone else reported on the issue they don't have that problem with 1.8.4-rc2. Could someone who has this issue on 1.8.3.2 or earlier re-test with the latest 1.8 branch to determine if this is still an issue? Leif. -- _ -- 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
[asterisk-users] MOH on DAHDI PRI Channels
Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1 connected with it. When the called party press hold on his phone then asterisk start MOH?? -- Regards, Shariq Khan 0333-3501125 -- _ -- 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
Re: [asterisk-users] MOH on DAHDI PRI Channels
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shariq Khan Sent: Thursday, April 07, 2011 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] MOH on DAHDI PRI Channels Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1 connected with it. When the called party press hold on his phone then asterisk start MOH?? -- Regards, Shariq Khan 0333-3501125 [Danny Nicholas] Question #1 Dial(DAHDI/1/5551212,20,m) will play moh until the other end answers Question #2 Don't think so since you're asking Asterisk to detect on hold from outside (this might be do-able in a SIP environment, but DAHDI tends to be copper). Hope this is correct/helps. -- _ -- 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
Re: [asterisk-users] MOH on DAHDI PRI Channels
_ [Shariq Khan] Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1 connected with it. When the called party press hold on his phone then asterisk start MOH?? [Danny Nicholas] Question #1 Dial(DAHDI/1/5551212,20,m) will play moh until the other end answers Question #2 Don't think so since you're asking Asterisk to detect on hold from outside (this might be do-able in a SIP environment, but DAHDI tends to be copper). Hope this is correct/helps. [Don Kelly] Looks like the call is to the DAHDI Channel from an outside caller, so the called party is inside. This is simple MOH. -- _ -- 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
Re: [asterisk-users] MOH on DAHDI PRI Channels
Danny, Thanks for the support, but i need to hold the customer and play MOH after answering the call. As you know that the signalling codes of SIP and ISDN are almost same, that's why i was thinking that MOH can work on DAHDI as well. -- Regards, Shariq Khan 0333-3501125 On Thu, Apr 7, 2011 at 9:25 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Shariq Khan *Sent:* Thursday, April 07, 2011 11:19 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] MOH on DAHDI PRI Channels Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1 connected with it. When the called party press hold on his phone then asterisk start MOH?? -- Regards, Shariq Khan 0333-3501125 *[Danny Nicholas] * *Question #1* *Dial(DAHDI/1/5551212,20,m) will play moh until the other end answers* *Question #2* *Don’t think so since you’re asking Asterisk to detect “on hold” from outside (this might be do-able in a SIP environment, but DAHDI tends to be copper).* *Hope this is correct/helps.* -- _ -- 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
Re: [asterisk-users] No ringback even though progressinband=yes is set
On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, Steve -- _ -- 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
Re: [asterisk-users] chan_dahdi unknown dependency problem
asterisk-users-boun...@lists.digium.com wrote: On 03/30/2011 01:32 PM, SebA wrote: So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a make menuselect in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) res_smdi gets built fine, dahdi is installed and working, tonezone is installed, pri is installed, ss7 is not installed, openr2 is not installed. Surely one does not need ss7 and openr2 if one has pri! So what else could be the problem? --- # ls -l /usr/lib/asterisk/modules/chan_dahdi.so ls: /usr/lib/asterisk/modules/chan_dahdi.so: No such file or directory # ls /usr/lib/asterisk/modules/res_smdi.so -l -rwxr-xr-x 1 root root 227620 Mar 30 18:35 /usr/lib/asterisk/modules/res_smdi.so # ls -l /usr/lib/libtonezone* -rwxr-xr-x 1 root root 216276 Mar 30 17:45 /usr/lib/libtonezone.a lrwxrwxrwx 1 root root 18 Mar 30 17:45 /usr/lib/libtonezone.so - libtonezone.so.2.0 lrwxrwxrwx 1 root root 18 Mar 30 17:45 /usr/lib/libtonezone.so.1 - libtonezone.so.2.0 lrwxrwxrwx 1 root root 18 Mar 30 17:45 /usr/lib/libtonezone.so.1.0 - libtonezone.so.2.0 lrwxrwxrwx 1 root root 18 Mar 30 17:45 /usr/lib/libtonezone.so.2 - libtonezone.so.2.0 -rwxr-xr-x 1 root root 214066 Mar 30 17:45 /usr/lib/libtonezone.so.2.0 # ls -l /usr/lib/libpri* -rw-r--r-- 1 root root 1224116 Mar 30 16:49 /usr/lib/libpri.a lrwxrwxrwx 1 root root 13 Mar 30 16:49 /usr/lib/libpri.so - libpri.so.1.4 -rwxr-xr-x 1 root root 790374 Mar 30 16:49 /usr/lib/libpri.so.1.4 snip Have you re-run the configure script after installing these other libraries/packages? If so, look in the config.log file for the tests that check for them, and see what is failing. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org Hi Kevin, Yes, I even tried it on another (similar) machine, but with the same result. (The same machines used to run chan_zap with no issues on Asterisk 1.4.x.) config.log is 670KB, 23,000+ lines. What should I search for in here? dahdi has 212 occurrences, but chan_dahdi has 0. Kind regards, Sebastian A -- _ -- 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
Re: [asterisk-users] chan_dahdi unknown dependency problem
asterisk-users-boun...@lists.digium.com wrote: On 03/30/2011 01:32 PM, SebA wrote: So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a make menuselect in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) When I run 'make menuselect', this is what I see for chan_dahdi: DAHDI Telephony Depends on: res_smdi(M), dahdi(E), tonezone(E) Can use: pri(E), ss7(E), openr2(E) Yours says 'depends on' for all of these items, which means you *must* have them installed. Have you made any changes to the Asterisk source code? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Nope, I can tell you that ss7 and openr2 are certainly not installed! I have made no changes to the Asterisk source code. Kind regards, Sebastian A -- _ -- 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
Re: [asterisk-users] chan_dahdi unknown dependency problem
I presume you mean contrib/scripts/install_prereq but I'm not sure how to use it or whether it is applicable to this situation. I had a look over the source code and it seems to be heavily dependent on what distribution you are running. For Debian, quite a lot are listed, but for Redhat it is only the essentials: PACKAGES_RH=gcc gcc-c++ ncurses-devel openssl-devel This distribution is AsteriskNOW 1.0 or so (the rPath one), so I doubt that it would be recognized by that script. It looks like it only recognized Debian, Redhat and OpenBSD. Kind regards, Sebastian A -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: 31 March 2011 23:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] chan_dahdi unknown dependency problem Run pre requirement check script I don't know the name but it's located inside asterisk source dir inside contrib I had same issue and has been fixed by that. -- Sent from my iPhone On Mar 31, 2011, at 5:47 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 03/30/2011 01:32 PM, SebA wrote: So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a make menuselect in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) When I run 'make menuselect', this is what I see for chan_dahdi: DAHDI Telephony Depends on: res_smdi(M), dahdi(E), tonezone(E) Can use: pri(E), ss7(E), openr2(E) Yours says 'depends on' for all of these items, which means you *must* have them installed. Have you made any changes to the Asterisk source code? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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
Re: [asterisk-users] asterisk hints
On Wednesday 06 April 2011 10:09:07 satish patel wrote:
I used following hint dialplan and i ran show hints but its showing only
one extension what about other 200 phones status ?
exten = _7[456]XX,hint,SIP/${EXTEN}
exten = _7[456]XX,1,Macro(stdexten,${EXTEN},sip/${EXTEN})
shirley*CLI core show hints
-= Registered Asterisk Dial Plan Hints =-
_7[456]XX@ora-cam-extensions : SIP/${EXTEN}
State:IdleWatchers 0
- 1 hints registered
It's actually just showing the pattern, which is not any. In order for the
pattern to generate individual items, something must query an individual
hint state. The usual method of doing this would be for a SIP phone to
subscribe to that extension state, but you can also use EXTENSION_STATE
in the dialplan to query individual extensions.
Just note that if you query an extension that comes back with an invalid
devicename, you've still queried that extension, so the Invalid state will
be preserved in your hint list. The pattern match is intended to be a
shortcut for configuring a lot of phones (and allowing new ones to be
populated on the fly), not a shortcut for making a pretty list for the
command line.
--
Tilghman
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Re: [asterisk-users] Question About Codecs
On Wednesday 06 April 2011 09:49:17 Jon Farmer wrote: Hi I have a call into a MeetMe conference that when I do a core show channel returns NativeFormats: 0x4 (ulaw) WriteFormat: 0x1000 (g722) ReadFormat: 0x1000 (g722) Can someone explain what the differences between Native, Wite and Read are? Your native format is the format that the phone actually uses (on the wire). The read and write formats are what Asterisk expects to send to and receive from the application, because Asterisk has set up a translation path to ensure that the application gets a format that is more conducive to its purpose. Internally to Asterisk, when you ast_read() a frame from the channel, you should expect that, when the frame is a voice frame, the frame will be in the ReadFormat. And, when you ast_write() a voice frame to that channel, it should be in the WriteFormat. -- Tilghman -- _ -- 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
Re: [asterisk-users] realtime mysql for 1.8
On Wednesday 06 April 2011 14:53:00 Hans Witvliet wrote: I'm going to have a go with realtime mysql. Just wondering, most examples i came across while googling, was with 1.6 systems. So any drastic changes with 1.8.3, table-layout? other pitfalls? This isn't a pitfall that comes with the upgrade, but you should set wait_timeout internal to the MySQL server to 864000 or higher. This will prevent a number of mysterious crashes that are otherwise possible (and difficult to diagnose) with the threaded MySQL client driver. This is the case, whether you use the native res_config_mysql or the abstract res_config_odbc driver. The usual symptom of this problem is that Asterisk crashes on the first call of the day on Monday morning and then is fine (either for the rest of the week, or until the next morning, depending upon how active calls are on your system). -- Tilghman -- _ -- 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
Re: [asterisk-users] Call duration problem or maybe calls not hanging up problem
Very weird mate...I would have replied sooner, but in reality there's a LOT of troubleshooting to be done and it would require working with your provider. It sounds like (if you're sending a bye when your calls disconnect) you never receive an actual 200 OK stating the call is picked up and so your system is sending a CANCEL ? Just spitballing here -- _ -- 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
Re: [asterisk-users] No ringback even though progressinband=yes is set
On 4/7/2011 11:02 AM, Douglas Mortensen wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- 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 If you are referring to a ringback tone when they first dial your system, meaning that they immediately hear your IVR when they dial your PBX's number, it's because that's how it's supposed to work. Unless you tell your PBX to use the Ringing() app and wait for a period of time, Asterisk normally picks up at the beginning of the IVR (since the first thing you have to do to send audio via Background or Playback is issue the command Answer() to start sending actual audio. (Note: The Ringing app just signals RINGING to the remote party) -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- 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
[asterisk-users] asterisk login to voicemail
Is there a way to login to a voicemail box when someone pushes '#' during greeting? -- _ -- 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
Re: [asterisk-users] asterisk login to voicemail
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Thursday, April 07, 2011 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk login to voicemail Is there a way to login to a voicemail box when someone pushes '#' during greeting? [Danny Nicholas] Here is one way: [greeting] Exten = s,1,background(greeting) Exten =s,n,hangup Exten = #,1,voicemailmain(100@default) -- _ -- 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
Re: [asterisk-users] asterisk login to voicemail
i'm afraid my setup is more complex than that [inbound] exten = _X.,1,agi(route.pl) after some logic using mysql, route.pl then does: $AGI-exec(VoiceMail, $options); at that point, I would like the caller to be able to push '#' and be prompted for Password for that particular mailbox On Thu, Apr 7, 2011 at 2:58 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Thursday, April 07, 2011 1:56 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] asterisk login to voicemail Is there a way to login to a voicemail box when someone pushes '#' during greeting? *[Danny Nicholas] * *Here is one way:* *[greeting]* *Exten = s,1,background(greeting)* *Exten =s,n,hangup* *Exten = #,1,voicemailmain(100@default)* -- _ -- 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
Re: [asterisk-users] asterisk login to voicemail
If you add the exten = #,1 line to the end of the inbound context, that should do it for you. If not, change $AGI-exec(VoiceMail,$options) to go to a context instead of running Voicemail directly. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Thursday, April 07, 2011 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk login to voicemail i'm afraid my setup is more complex than that [inbound] exten = _X.,1,agi(route.pl) after some logic using mysql, route.pl then does: $AGI-exec(VoiceMail, $options); at that point, I would like the caller to be able to push '#' and be prompted for Password for that particular mailbox On Thu, Apr 7, 2011 at 2:58 PM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Thursday, April 07, 2011 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk login to voicemail Is there a way to login to a voicemail box when someone pushes '#' during greeting? [Danny Nicholas] Here is one way: [greeting] Exten = s,1,background(greeting) Exten =s,n,hangup Exten = #,1,voicemailmain(100@default) -- _ -- 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
Re: [asterisk-users] asterisk login to voicemail
I'm sorry I'm new to AGI programming but i did this:
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
and in extensions.conf i have:
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = #,n,VoiceMailMain(${EXTEN}@4)
I keep getting 603 declined when i call the number...
On Thu, Apr 7, 2011 at 3:09 PM, Danny Nicholas da...@debsinc.com wrote:
If you add the exten = #,1 line to the end of the inbound context, that
should do it for you. If not, change $AGI-exec(“VoiceMail”,$options) to go
to a context instead of running Voicemail directly.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:04 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
i'm afraid my setup is more complex than that
[inbound]
exten = _X.,1,agi(route.pl)
after some logic using mysql, route.pl then does:
$AGI-exec(VoiceMail, $options);
at that point, I would like the caller to be able to push '#' and be
prompted for Password for that particular mailbox
On Thu, Apr 7, 2011 at 2:58 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 1:56 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] asterisk login to voicemail
Is there a way to login to a voicemail box when someone pushes '#' during
greeting?
*[Danny Nicholas] *
*Here is one way:*
*[greeting]*
*Exten = s,1,background(greeting)*
*Exten =s,n,hangup*
*Exten = #,1,voicemailmain(100@default)*
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Re: [asterisk-users] asterisk login to voicemail
You're on the right track, but # is going to blow away ${EXTEN} so you are
going to have to hard-code that value or use a different variable that
contains what should have been in ${EXTEN}. Also, #,n has to be #,1 (each
part of a context has to have line 1 - not my rule, Asterisk's)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 2:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk login to voicemail
I'm sorry I'm new to AGI programming but i did this:
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
and in extensions.conf i have:
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = #,n,VoiceMailMain(${EXTEN}@4)
I keep getting 603 declined when i call the number...
On Thu, Apr 7, 2011 at 3:09 PM, Danny Nicholas da...@debsinc.com wrote:
If you add the exten = #,1 line to the end of the inbound context, that
should do it for you. If not, change $AGI-exec(VoiceMail,$options) to go
to a context instead of running Voicemail directly.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk login to voicemail
i'm afraid my setup is more complex than that
[inbound]
exten = _X.,1,agi(route.pl)
after some logic using mysql, route.pl then does:
$AGI-exec(VoiceMail, $options);
at that point, I would like the caller to be able to push '#' and be
prompted for Password for that particular mailbox
On Thu, Apr 7, 2011 at 2:58 PM, Danny Nicholas da...@debsinc.com wrote:
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk login to voicemail
Is there a way to login to a voicemail box when someone pushes '#' during
greeting?
[Danny Nicholas]
Here is one way:
[greeting]
Exten = s,1,background(greeting)
Exten =s,n,hangup
Exten = #,1,voicemailmain(100@default)
--
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Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
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Re: [asterisk-users] asterisk login to voicemail
As I see it, callednum and vmbox should not be the same. Vmbox is a good
mailbox you're going to reach if the user doesn't hit #, callednum is the
fallback number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
--
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Re: [asterisk-users] asterisk login to voicemail
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox (VoiceMailMain)
I'm totally lost as to how to get this done... any suggestions?
On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas da...@debsinc.com wrote:
As I see it, callednum and vmbox should not be the same. Vmbox is a
“good” mailbox you’re going to reach if the user doesn’t hit #, callednum is
the “fallback” number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:41 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
--
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Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Re-opening this issue. If i dial number which doesn't existing then i am getting following error. So is there anyway i can fix my dialplan to check whether that number exist or not or i can check channel status. shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0032, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0032, sip/7623sip/7624IAX2/7623,20,t) in new stack [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown) == Using SIP RTP CoS mark 5 [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-13525 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-13525' [Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0032' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0032' [Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011 20:22:55 + Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Thanks for reply! I found this problem only with X-lite version of softphone. Other phones are working fine without any WARNING! look like X-lite has some short of SIP issue. -S From: mden...@gmail.com Date: Mon, 4 Apr 2011 15:59:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer for SIP/7527-0008 -- SIP/7623-0009 answered SIP/7527-0008 [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on
Re: [asterisk-users] asterisk login to voicemail
Here's your solution
[vmtest]
exten = s,1,background(vm-Family,3)
exten = s,n,waitexten(3)
exten = s,n,Voicemail(${callnum}@default)
exten = *,1,VoicemailMain(${callnum}@default)
exten = #,1,VoicemailMain(${callnum}@default)
exten = i,1,Voicemail(${callnum}@default)
exten = t,1,Voicemail(${callnum}@default)
vm-family plays when you come in. you have 3 seconds to hit * or #. If
not, you go to regular voicemail. If so, you go to admin and get prompted
for the password.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk login to voicemail
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox (VoiceMailMain)
I'm totally lost as to how to get this done... any suggestions?
On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas da...@debsinc.com wrote:
As I see it, callednum and vmbox should not be the same. Vmbox is a good
mailbox you're going to reach if the user doesn't hit #, callednum is the
fallback number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
--
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Re: [asterisk-users] asterisk login to voicemail
Unfortunately, that solution will not work for me... The user must be able
to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
On Thu, Apr 7, 2011 at 4:18 PM, Danny Nicholas da...@debsinc.com wrote:
Here’s your solution
[vmtest]
exten = s,1,background(vm-Family,3)
exten = s,n,waitexten(3)
exten = s,n,Voicemail(${callnum}@default)
exten = *,1,VoicemailMain(${callnum}@default)
exten = #,1,VoicemailMain(${callnum}@default)
exten = i,1,Voicemail(${callnum}@default)
exten = t,1,Voicemail(${callnum}@default)
vm-family plays when you come in. you have 3 seconds to hit * or #. If
not, you go to regular voicemail. If so, you go to admin and get prompted
for the password.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:52 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox (VoiceMailMain)
I'm totally lost as to how to get this done... any suggestions?
On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas da...@debsinc.com wrote:
As I see it, callednum and vmbox should not be the same. Vmbox is a “good”
mailbox you’re going to reach if the user doesn’t hit #, callednum is the
“fallback” number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:41 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
--
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Re: [asterisk-users] asterisk login to voicemail
One more thought - assuming that your users all have greetings recorded, you
could change vm-family to
/var/spoo/asterisk/voicemail/default/${callnum}/busy and keep the same plan.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk login to voicemail
Unfortunately, that solution will not work for me... The user must be able
to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
On Thu, Apr 7, 2011 at 4:18 PM, Danny Nicholas da...@debsinc.com wrote:
Here's your solution
[vmtest]
exten = s,1,background(vm-Family,3)
exten = s,n,waitexten(3)
exten = s,n,Voicemail(${callnum}@default)
exten = *,1,VoicemailMain(${callnum}@default)
exten = #,1,VoicemailMain(${callnum}@default)
exten = i,1,Voicemail(${callnum}@default)
exten = t,1,Voicemail(${callnum}@default)
vm-family plays when you come in. you have 3 seconds to hit * or #. If
not, you go to regular voicemail. If so, you go to admin and get prompted
for the password.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk login to voicemail
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox (VoiceMailMain)
I'm totally lost as to how to get this done... any suggestions?
On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas da...@debsinc.com wrote:
As I see it, callednum and vmbox should not be the same. Vmbox is a good
mailbox you're going to reach if the user doesn't hit #, callednum is the
fallback number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
--
_
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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] asterisk login to voicemail
Indeed, that is what i would do except many users will not have a greeting.
so those without a greeting will not be able to login unless i generate a
canned greeting which i think i will have to do.
On Thu, Apr 7, 2011 at 4:33 PM, Danny Nicholas da...@debsinc.com wrote:
One more thought – assuming that your users all have greetings recorded,
you could change vm-family to
/var/spoo/asterisk/voicemail/default/${callnum}/busy and keep the same plan.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 3:25 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Unfortunately, that solution will not work for me... The user must be able
to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
On Thu, Apr 7, 2011 at 4:18 PM, Danny Nicholas da...@debsinc.com wrote:
Here’s your solution
[vmtest]
exten = s,1,background(vm-Family,3)
exten = s,n,waitexten(3)
exten = s,n,Voicemail(${callnum}@default)
exten = *,1,VoicemailMain(${callnum}@default)
exten = #,1,VoicemailMain(${callnum}@default)
exten = i,1,Voicemail(${callnum}@default)
exten = t,1,Voicemail(${callnum}@default)
vm-family plays when you come in. you have 3 seconds to hit * or #. If
not, you go to regular voicemail. If so, you go to admin and get prompted
for the password.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:52 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox (VoiceMailMain)
I'm totally lost as to how to get this done... any suggestions?
On Thu, Apr 7, 2011 at 3:47 PM, Danny Nicholas da...@debsinc.com wrote:
As I see it, callednum and vmbox should not be the same. Vmbox is a “good”
mailbox you’re going to reach if the user doesn’t hit #, callednum is the
“fallback” number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
*Sent:* Thursday, April 07, 2011 2:41 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] asterisk login to voicemail
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this error
pbx.c:4413 __ast_pbx_run: Channel 'SIP/8.224.32.3-0002' sent into
invalid extension 'XX' in context 'voicemail', but no invalid
handler
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Just a guess but is it possible one of your SIP peers (7623 or 7624) has an invalid IP address of 0.0.29.200? I wonder what sip show peers shows. On Thu, Apr 7, 2011 at 4:03 PM, satish patel satish...@hotmail.com wrote: Re-opening this issue. If i dial number which doesn't existing then i am getting following error. So is there anyway i can fix my dialplan to check whether that number exist or not or i can check channel status. shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0032, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0032, sip/7623sip/7624IAX2/7623,20,t) in new stack [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown) == Using SIP RTP CoS mark 5 [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-13525 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-13525' [Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0032' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0032' [Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011 20:22:55 + Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Thanks for reply! I found this problem only with X-lite version of softphone. Other phones are working fine without any WARNING! look like X-lite has some short of SIP issue. -S From: mden...@gmail.com Date: Mon, 4 Apr 2011 15:59:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-0009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-5537 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-5537' [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- SIP/7623-0009 connected line has changed. Saving it until answer
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
They are on valid IP address range and working properly but when i switched off that phone and dialing number from other phone i am getting following WARNING!! So i would like to have some thing like who check CHANNEL first and then say Phone is not register or If phone is available it will ring phone. I guess ChanIsAvail will fix my issue. http://www.asteriskguru.com/tutorials/chanisavail_image60455.jpg But now my asterisk saying i don't have cut application :( How to compile app_cut.so i didn't find anything related to this in asterisk source. -- User entered nothing. [Apr 7 16:36:53] WARNING[14134]: pbx.c:4055 pbx_extension_helper: No application 'Cut' for extension (macro-stdexten, s, 3) == Spawn extension (macro-stdexten, s, 3) exited non-zero on 'SIP/7527-003a' in macro 'stdexten' Date: Thu, 7 Apr 2011 16:40:12 -0400 From: p...@dugasenterprises.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Just a guess but is it possible one of your SIP peers (7623 or 7624) has an invalid IP address of 0.0.29.200? I wonder what sip show peers shows. On Thu, Apr 7, 2011 at 4:03 PM, satish patel satish...@hotmail.com wrote: Re-opening this issue. If i dial number which doesn't existing then i am getting following error. So is there anyway i can fix my dialplan to check whether that number exist or not or i can check channel status. shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0032, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0032, sip/7623sip/7624IAX2/7623,20,t) in new stack [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown) == Using SIP RTP CoS mark 5 [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response -- IAX2/0.0.29.199:4569-13525 is circuit-busy -- Hungup 'IAX2/0.0.29.199:4569-13525' [Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument [Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response [Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/7527-0032' in macro 'stdexten' == Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0032' [Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 4 Apr 2011 20:22:55 + Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit Thanks for reply! I found this problem only with X-lite version of softphone. Other phones are working fine without any WARNING! look like X-lite has some short of SIP issue. -S From: mden...@gmail.com Date: Mon, 4 Apr 2011 15:59:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit On Mon, Apr 4, 2011 at 3:51 PM, satish patel satish...@hotmail.com wrote: Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI == Using SIP RTP CoS mark 5 -- Executing [7623@from-sip:1] Macro(SIP/7527-0008, stdexten,7623,sip/7623sip/7624) in new stack -- Executing [s@macro-stdexten:1] Dial(SIP/7527-0008, sip/7623sip/7624iax2/7623,20,t) in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
That should be CUT all caps I think -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 7 Apr 2011 20:45:21 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit -- _ -- 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
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Yes! You are right! Its working. Now issue is we have SIP extension for
local office users and same number has IAX extension for remote
traveling users. How could i use ChanIsAvail with best action ?
I did following
exten = s,1,ChanIsAvail(${ARG2}IAX2/${ARG1},20,t)
exten = s,n,NoOp(${AVAILCHAN})
exten = s,n,Set(NewVar=${CUT(AVAILCHAN,,1)})
exten = s,n,NoOp(${NewVar})
exten = s,n,Dial(${NewVar}/${EXTEN})
exten = s,n,Hangup()
And in result i got following: Why its looking at IAX2/0.0.29.199 what is
0.0.29.199?
shirley*CLI
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro(SIP/7527-004c,
stdexten,7623,SIP/7623) in new stack
-- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-004c,
SIP/7623IAX2/7623,20,t) in new stack
-- Hungup 'IAX2/0.0.29.199:4569-2707'
-- Executing [s@macro-stdexten:2] NoOp(SIP/7527-004c,
IAX2/0.0.29.199:4569-2707) in new stack
-- Executing [s@macro-stdexten:3] Set(SIP/7527-004c,
NewVar=IAX2/0.0.29.199:4569) in new stack
-- Executing [s@macro-stdexten:4] NoOp(SIP/7527-004c,
IAX2/0.0.29.199:4569) in new stack
-- Executing [s@macro-stdexten:5] Dial(SIP/7527-004c,
IAX2/0.0.29.199:4569/s) in new stack
-- Called 0.0.29.199:4569/s
[Apr 7 16:59:21] NOTICE[13915]: chan_iax2.c:4643 __auto_congest:
Auto-congesting call due to slow response
-- IAX2/0.0.29.199:4569-3390 is circuit-busy
-- Hungup 'IAX2/0.0.29.199:4569-3390'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-stdexten:6] Hangup(SIP/7527-004c, ) in new
stack
== Spawn extension (macro-stdexten, s, 6) exited non-zero on
'SIP/7527-004c' in macro 'stdexten'
== Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-004c'
To: asterisk-users@lists.digium.com
From: isr...@gmail.com
Date: Thu, 7 Apr 2011 20:49:04 +
Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
That should be CUT all caps I think
-Original Message-
From: satish patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 7 Apr 2011 20:45:21
To: asterisk-usersasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
--
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[asterisk-users] Any way to temporarily disable a registered SIP PEER in Asterisk?
Hi Everyone, We want to be able to momentarily or temporarily provide CONGESTION or DE-REGISTER a SIP PEER to asterisk through WEB GUI. I don't want to indulge into dial-plan and write changes to .conf file every-time. Is there any way that a SIP PEER can be de-registered for an amount of time or maybe deactivated? or there isn't such facility available in asterisk? Thanks -- _ -- 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
Re: [asterisk-users] Any way to temporarily disable a registered SIPPEER in Asterisk?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Thursday, April 07, 2011 4:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Any way to temporarily disable a registered
SIPPEER in Asterisk?
Hi Everyone,
We want to be able to momentarily or temporarily provide CONGESTION or
DE-REGISTER a SIP PEER to asterisk through WEB GUI. I don't want to indulge
into dial-plan and write changes to .conf file every-time. Is there any way
that a SIP PEER can be de-registered for an amount of time or maybe
deactivated? or there isn't such facility available in asterisk?
Thanks
[Danny Nicholas]
You could have an entry in ASTDB that pseudo-disables the peer from
dialing that you could turn on/off from a gui using AMI. Where your
dialplan does the dial, just wrap some logic around it like this
[dialout]
exten = s,1,noop(start to dial)
exten = s,n,Set(dialval=${DB(dialok/${ARG2})})
exten = s,n,Gotoif($[ ${dialval} != yes]?dialout,s-BUSY,1)
exten = s,n,Dial(DAHDI/1,${ARG1})
exten = s,n,hangup
exten = s-BUSY,1,playback(busy)
exten = s-BUSY,n,hangup
--
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[asterisk-users] Occasional call from asterisk
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight? Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
--
Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
--
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Re: [asterisk-users] No ringback even though progressinband=yes is set
I have inbound calls going directly to a ring group. When callers call in, they (the callers) hear complete silence even though the phones that are part of the ring group ARE ringing properly. Employees can answer the calls when their phones ring, and everything works fine. The problem is simply that the external caller never hears any ringing. Even if the SIP phones in the ring group ring for 5 rings, it is total silence even though there is ringing going on inside of the office. I'm pretty sure it is a ringback issue. I'm going to try to turn on SIP debugging see what I can figure out that way. I do appreciate your help. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: Sherwood McGowan [mailto:sherwood.mcgo...@gmail.com] Sent: Thursday, April 07, 2011 12:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set On 4/7/2011 11:02 AM, Douglas Mortensen wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- 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 If you are referring to a ringback tone when they first dial your system, meaning that they immediately hear your IVR when they dial your PBX's number, it's because that's how it's supposed to work. Unless you tell your PBX to use the Ringing() app and wait for a period of time, Asterisk normally picks up at the beginning of the IVR (since the first thing you have to do to send audio via Background or Playback is issue the command Answer() to start sending actual audio. (Note: The Ringing app just signals RINGING to the remote party) -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- 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
Re: [asterisk-users] No ringback even though progressinband=yes is set
Steve. Thanks for the insight. I won't pretend to know what early-audio is, but I guess I'm about to find out :-). Also, I believe that I have a nearly identical setup like this with the exact same SIP provider w/o any trouble. However, I think that system must be running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to confirm). Is there a significant difference between 1.2/1.4 1.6 in this scenario? Thanks a million!! :-) - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -Original Message- From: Steve Davies [mailto:davies...@gmail.com] Sent: Thursday, April 07, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, Steve -- _ -- 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
Re: [asterisk-users] No ringback even though progressinband=yes is set
On 4/7/2011 4:54 PM, Douglas Mortensen wrote: I have inbound calls going directly to a ring group. When callers call in, they (the callers) hear complete silence even though the phones that are part of the ring group ARE ringing properly. Employees can answer the calls when their phones ring, and everything works fine. The problem is simply that the external caller never hears any ringing. Even if the SIP phones in the ring group ring for 5 rings, it is total silence even though there is ringing going on inside of the office. I'm pretty sure it is a ringback issue. I'm going to try to turn on SIP debugging see what I can figure out that way. I do appreciate your help. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 If you're using an interface (I believe you said AsteriskNOW), you might want to check the Dial Options...Make sure that 'r' is one of the options. The reason you're not hearing ringing is probably due to Asterisk not sending a RINGING signal. If you have 'r' defined in the dial options in your interface, then AsteriskNOW is probably using a Dial command that is NOT using your global dial options. -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- 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
Re: [asterisk-users] asterisk login to voicemail
Unfortunately, that solution will not work for me... The user must be able to
hit * during the greeting of any voicemail and be prompted for the Password
to that particular mailbox looks like i got a lot of programming to do to
create a work around for this... thanks for your help...
Forgive me if i'm wrong, but you guys seem to be over complicating things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail context.
Example:
Exten = a, 1, VoicemailMain(@default)
Exten = a, 2, Hangup
When using the star '*' it's important to note that the context you placed the
application voicemail in is irrelevant, it's the context for the voicemail box
that we're looking for in the dialplan for the jump to the 'a' extension.
So this is what i do...
Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to the
correct context, and i set ${MAILBOXID} to the mailbox name.
Then, in extensions.conf, I added this:-
[voicemail]
exten = a,1,Playback(astcc-please-enter-your)
exten = a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})
When the user presses *, they are passed to the 'a' extension above and into
VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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Re: [asterisk-users] Occasional call from asterisk
We were getting a lot of those. We installed IPTables with blocking of
everything outside of North America and they all but vanished.
No direct evidence, but a pretty good empirical guess that they were related
to hackers trying to get paths to the US.
CF
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Henning
Sent: Thursday, April 07, 2011 4:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Occasional call from asterisk
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight? Here's
relevant snippets from my extensions.conf and Master.csv log:
This line shows up in Master.csv:
,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,
Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup
The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings. Works great most of the time.
I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there. (The
preceding line of the log is a call from almost an hour earlier). In that
case though I'd expect to see 1-CANCEL in the log instead. Perhaps if the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Thanks in advance for any and all advice!
~Brian
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Brian Henning, Software Engineer
/\Pine Research Instrumentation
//\\ 5908 Triangle Drive
///\\\ Raleigh, NC 27617
USA
||
||phone: 919.782.8320
fax: 919.782.8323
email: bhenn...@pineinst.com
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Re: [asterisk-users] Occasional call from asterisk
We experience exact same thing on DAHDI with Sangoma USB FXO device on short
circuited lines. Phantom calls are actually due to a short in the lines that
happen occasionally.
-Bruce
On Thu, Apr 7, 2011 at 7:16 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, Apr 7, 2011 at 4:53 PM, Brian Henning bhenn...@pineinst.comwrote:
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight?
Here's
relevant snippets from my extensions.conf and Master.csv log:
snip
I've seen this on cases where a phantom call comes in on a DAHDI channel
- these calls were the results of faulty wiring on the part of the telco.
Check your logs for any errors on your DAHDI channels around the time of the
ghost calls.
It could also be a case of someone calls in and then hangs up before the
call is actually passed to asterisk, and the telco is just slow to hangup
the call.
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to
prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line). Unfortunately the 9 does not get prepended.
Your Set() syntax is wrong. Try this:
exten = s,n,Set(CALLERID(num)=9${CALLERID(num)})
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio
First, I'm pretty sure avaya peer needs to friend. Try adding the below to sip.conf and do a reload. [general] externip = the.wan.ext.ip localnet = 192.168.1.0/255.255.255.0 If that doesn't work, add nat=yes to avaya peer=friend Skyler From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Thursday, April 07, 2011 6:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Avaya SIP Trunking One Way Audio On 04/07/11 03:00, Shariq Khan wrote: I am facing one way audio problem in sip trunking between asterisk and avaya. +-+ ++ | avaya sip |---| P1 | +-+ ++ | | | +-+ | Asterisk | WAN - | | LAN +-+ | / ++ / | P2 |--+ ++ When P1 dial P2, P2 hears voice clear but P1 could not hear any voice. My sip.conf is [avaya] type=peer fromdomain=xx.xx.xx.xx host=xx.xx.xx.xx disallow=all allow=ulaw dtmfmode=rfc2833 canreinvite=yes -- Regards, Shariq Khan 0333-3501125 -- _ -- 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 Turn off reinvite on all extensions and SIP trunks involved and try again. Lyle Giese LCR Computer Services, Inc. -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1204 / Virus Database: 1498/3523 - Release Date: 03/22/11 Internal Virus Database is out of date. -- _ -- 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
