[asterisk-users] Asterisk 1.8.3: Started but no SIP talking
Hi All; I installed Asterisk on a new Server, it is a Dell Server and has 4 Ethernet ports. I gave IP address 192.168.0.3 for one Ethernet port. I am able to login for asterisk using /usr/sbin/asterisk -rvvv and from there (in the command line) I can type a commands. I have an Polycom IP Phone that is able to register for other Asterisk boxes (and some of them is 1.8.3) but with this new server, I do not see any messages coming to the consol when I give the IP address of this new asterisk server !! What could be? Actually, in the sip.conf file, it is hearing for all IPs 0.0.0.0 and the IP phone sending on port 5060 UDP (every thing default). What could be I am missing? Even if username and password wrong, I should be able to see traffic but without registration ... By the way: on the same network, there is another Asterisk box running with IP address 192.168.0.2 .. does it effect? It should not. I am missing any thing? Should I do any thing? How can I know if my new asterisk is running sip well? Regards Bilal -- _ -- 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: Started but no SIP talking
On Sun, Apr 17, 2011 at 1:46 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; snip I am missing any thing? Should I do any thing? How can I know if my new asterisk is running sip well? Regards Bilal Check if you've got a software firewall running, check if SELinux is running, etc. Try running a packet capture using tcpdump to see if your asterisk box is getting any traffic from the phone, etc. Basic network troubleshooting at this point. Can you ping the box from your network, can you ping the phone from your box, etc? -- Thanks, --Warren Selby, dCAP http://www.selbytech.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
From: Chris Owen ow...@hubris.net Sent: Thursday, April 07, 2011 9:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: 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 I have not been able to get multi-tenant parking stable there either. I gave up yesterday on 1.8.3.2 as I could not get it stable with any number of patches I could find. I fell back to 1.8.2.3 as that is the last version that I have been able to run production with. My customers have now been happy for the last 24 hours. I also tried 1.8.4 rc and the stability did not appear to be much better then 1.8.3.2 I hope they don't release 1.8.4 until the stability issues are addressed more rc version with fixes would be ideal. The longer these items drag out the worse it gets for users to know what to use. I would ask the developers to hold 1.8.4 until some of these items can be fixed and rolled in. Bryant -- _ -- 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] 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 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 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 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
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
Re: [asterisk-users] Asterisk 1.8.3
On Tue, 2011-04-05 at 21:10 -0400, Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. Thanks Bryant -- Could it be this issue? https://issues.asterisk.org/view.php?id=18818 Mind you, this one will only affect you if you use RealTime architecture -- 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 also see the random freeze of asterisk 1.8.3.2. We do use realtime. I have just applied the patch and will see how our environment holds. I will report back to the issue mentioned by Ishfaq Michel Verbraak *InterCommIT bv* ** On 06-04-11 09:44, Ishfaq Malik wrote: On Tue, 2011-04-05 at 21:10 -0400, Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. Thanks Bryant -- Could it be this issue? https://issues.asterisk.org/view.php?id=18818 Mind you, this one will only affect you if you use RealTime architecture -- _ -- 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 4/5/11 6:10 PM, Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. We've upgraded our system over the weekend from 1.4.35 to 1.8.3.2 For the past couple of days, we had several random hangs(most of the time core stop now didn't work, I had to kill -9 the process) Also the PRI behavior seems to be slightly different, we can't hear any early media sounds on 800 numbers that goes through ATT. I finally downgraded it back to 1.6.2.17, now everything work. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.3
On 4/5/11 6:10 PM, Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. From: Edwin Lam edwin@officegeneral.com Sent: Wednesday, April 06, 2011 5:37 PM We've upgraded our system over the weekend from 1.4.35 to 1.8.3.2 For the past couple of days, we had several random hangs(most of the time core stop now didn't work, I had to kill -9 the process) Also the PRI behavior seems to be slightly different, we can't hear any early media sounds on 800 numbers that goes through ATT. I finally downgraded it back to 1.6.2.17, now everything work. Edwin 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. For Now 1.8.3..2 is very bad. Thanks Bryant -- _ -- 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 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 0x8fd2298 on 0x90d9cf0 TEI/SAPI 0/0, call-pri is 0x90d9cf0 TEI/SAPI 0/0 -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) -- PROGRESS with cause code 127 received -- DAHDI/34-1 is making progress passing it to
[asterisk-users] Asterisk 1.8.3
I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. Thanks Bryant -- _ -- 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 Tuesday 05 April 2011 20:10:48 Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. This sounds like a deadlock of some kind. Asterisk has a debugging facility built-in for finding this type of problem, but you will need to compile in DONT_OPTIMIZE and DEBUG_THREADS. Also, it would be helpful, but not entirely necessary, to compile in BETTER_BACKTRACES. Once the problem occurs with the recompiled binary, issuing a core show locks should turn up an indication of where the problem lies. -- 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] Asterisk 1.8.3-rc3 1.8.3 and one way audio
On Mon, 2011-02-28 at 13:40 +, Ishfaq Malik wrote: I've just installed 1.8.3-rc3 on a test server as we really needed that deadlock involving REFER fix on our server but now I'm having an odd issue with one way audio with a specific type of call. If I do extension to extension calls there is full 2 way audio. If I route in an incoming call through numbers provided by our SIP provider there is no inbound audio (mobile to * SIP extension) but there is outbound audio (* SIP extension to mobile). If I route a call through our production server (1.4.17 debian) to a second identity on the same SIP phone as the previous condition there is perfect 2 way audio. I did suspect it might have been the firewall on the test server but I did the same call with the firewall turned off and still only had one way audio. Has anyone else experienced anything like this? Seeing that 1.8.3 had been released I updated our main test server to that from 1.8.2.2 using the digium yum repo. All audio had been working fine on this server before the update but after the update I experienced the same as I did with rc3. Internal ext to ext calls are fine. Outbound calls to mobile networks via our SIP provider are fine. Inbound calls via our SIP provider have one way audio. The servers are CentOs 5.5 and we are using RealTime architecture. Any thoughts would be appreciated Ish -- 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-rc3 1.8.3 and one way audio
On Mar 1, 2011, at 4:19 AM, Ishfaq Malik wrote: Seeing that 1.8.3 had been released I updated our main test server to that from 1.8.2.2 using the digium yum repo. All audio had been working fine on this server before the update but after the update I experienced the same as I did with rc3. Internal ext to ext calls are fine. Outbound calls to mobile networks via our SIP provider are fine. Inbound calls via our SIP provider have one way audio. The servers are CentOs 5.5 and we are using RealTime architecture. Any thoughts would be appreciated One-way audio is almost always NAT related. Does the Asterisk server have a public IP?-- _ -- 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-rc3 1.8.3 and one way audio
On Tue, 2011-03-01 at 10:08 -0600, Terry Wilson wrote: On Mar 1, 2011, at 4:19 AM, Ishfaq Malik wrote: Seeing that 1.8.3 had been released I updated our main test server to that from 1.8.2.2 using the digium yum repo. All audio had been working fine on this server before the update but after the update I experienced the same as I did with rc3. Internal ext to ext calls are fine. Outbound calls to mobile networks via our SIP provider are fine. Inbound calls via our SIP provider have one way audio. The servers are CentOs 5.5 and we are using RealTime architecture. Any thoughts would be appreciated One-way audio is almost always NAT related. Does the Asterisk server have a public IP? I think this is NAT related. The server has it's own public IP and is in a data centre. The extension is in out office and behind a NAT. If I do a sip show peer then the Addr-IP shows our office real world IP address but the reg contact shows my phones internal network IP Address. The really strange thing is that there is no section for NAT which I would expect to show as always. The sip peer data is stored in a MySQL table with the column name nat and a value of yes. I so far have not been able to find a definitive table definition for asterisk 1.8 and have been using the one for 1.6 Does anyone know if the nat column has been changed in the sip realtime table? I'd even happily have a rummage in the source code if anyone would point me in the direction of the right files to look at. Thanks Ish -- 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
[asterisk-users] Asterisk 1.8.3-rc3 and one way audio
I've just installed 1.8.3-rc3 on a test server as we really needed that deadlock involving REFER fix on our server but now I'm having an odd issue with one way audio with a specific type of call. If I do extension to extension calls there is full 2 way audio. If I route in an incoming call through numbers provided by our SIP provider there is no inbound audio (mobile to * SIP extension) but there is outbound audio (* SIP extension to mobile). If I route a call through our production server (1.4.17 debian) to a second identity on the same SIP phone as the previous condition there is perfect 2 way audio. I did suspect it might have been the firewall on the test server but I did the same call with the firewall turned off and still only had one way audio. Has anyone else experienced anything like this? -- 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
[asterisk-users] Asterisk 1.8.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Resolve duplicated data in the AstDB when using DIALGROUP() (Closes issue #18091. Reported by bunny. Patched by tilghman) * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. (Closes issue #18464. Reported, patched by IgorG) * Reworking parsing of mwi = lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. (Closes issue #18350. Reported by gbour. Patched by Marquis) * When using cdr_pgsql the billsec field was not populated correctly on unanswered calls. (Closes issue #18406. Reported by joscas. Patched by tilghman) * Resolve memory leak in iCalendar and Exchange calendaring modules. (Closes issue #18521. Reported, patched by pitel. Tested by cervajs) * This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. (Patched by tilghman) * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) * Resolve a memory leak when the Asterisk Manager Interface is disabled. (Reported internally by kmorgan. Patched by russellb) * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported internally. Patched by mnicholson) * Fix regression that changed behavior of queues when ringing a queue member. (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) * Resolve deadlock involving REFER. (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.) Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3 Thank you for your continued support of Asterisk! -- _ -- 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're installing from maintained packages (rpm) rather than compiling code. On Fri, 2011-02-11 at 17:47 +, satish patel wrote: Here is the patch did you apply it ? https://issues.asterisk.org/file_download.php?file_id=28206type=bug Date: Fri, 11 Feb 2011 08:46:36 -0200 From: vinic...@canall.com.br To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.3 That makes two of us. I tried asking on asterisk-dev but had no reply. - Mensagem original - Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of 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 -- _ -- 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 -- 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
[asterisk-users] Asterisk 1.8.3
Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of 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
On 11.02.2011 12:37, Ishfaq Malik wrote: Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 Have you tried issue18403.patch ? -- _ -- 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
That makes two of us. I tried asking on asterisk-dev but had no reply. - Mensagem original - Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of 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
I thought it has been resolved in 1.8.2 version Thanks, Satish Date: Fri, 11 Feb 2011 08:46:36 -0200 From: vinic...@canall.com.br To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.3 That makes two of us. I tried asking on asterisk-dev but had no reply. - Mensagem original - Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of 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 -- _ -- 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
So did I but I'm having the same problem in Asterisk 1.8.2.2 and the asterisk issue number I pasted the link of has Target Version 1.8.3 On Fri, 2011-02-11 at 15:59 +, satish patel wrote: I thought it has been resolved in 1.8.2 version Thanks, Satish Date: Fri, 11 Feb 2011 08:46:36 -0200 From: vinic...@canall.com.br To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.3 That makes two of us. I tried asking on asterisk-dev but had no reply. - Mensagem original - Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of 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
On 11-02-11 04:37 AM, Ishfaq Malik wrote: Does anyone have any rough idea how far away 1.8.3 is? If you are hard up for a release, you can use the latest 1.8.3 RC[1]. [1] http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.3-rc2.tar.gz -- 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
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Friday, February 11, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8.3 On 11-02-11 04:37 AM, Ishfaq Malik wrote: Does anyone have any rough idea how far away 1.8.3 is? If you are hard up for a release, you can use the latest 1.8.3 RC[1]. [1] http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.3-rc2.tar. gz -- Paul Belanger Isn't that a little Bleeding edge? Guess that's why I'm still using the 1.4X 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
On Fri, Feb 11, 2011 at 7:59 AM, satish patel satish...@hotmail.com wrote: I thought it has been resolved in 1.8.2 version Issue 18403 was not resolved in 1.8.2, but in 1.8.3-rc1. Release 1.8.3-rc2 was cut on 1/20/2011, so hopefully the full release will be out soon. You can see where the issue was merged here: http://svn.asterisk.org/svn/asterisk/tags/1.8.3-rc1/ChangeLog -Jonathan -- _ -- 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 11-02-11 11:14 AM, Danny Nicholas wrote: Isn't that a little Bleeding edge? Guess that's why I'm still using the 1.4X set. No? OP asked for asterisk 1.8.3, currently asterisk-1.8.3-rc2 is the latest RC. If no regressions are found with the new patches, it will become 1.8.3. Basically, if you are waiting for asterisk-1.8.x to be released, start testing asterisk-1.8.x-rcx and report feedback. -- 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 BLF stopped working
I am running 1.8.3 and my BLF lights have stopped working. The hints appear to be intact when I use core show hints. But none of the phones are getting the BLF updates. This has happend in the past and I have had to restart my server. What could be causing this to occur. It did not do this with the 1.6.x builds. Is there a way to reload the hints or force a refresh without re-starting Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- 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
I have asterisk 1.8.2 in development and i can blind transfer from A to C without any issue. Or may be i am doing wrong thing? How do i reproduce this error ? -S Date: Fri, 11 Feb 2011 11:41:37 -0500 From: pabelan...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.3 On 11-02-11 11:14 AM, Danny Nicholas wrote: Isn't that a little Bleeding edge? Guess that's why I'm still using the 1.4X set. No? OP asked for asterisk 1.8.3, currently asterisk-1.8.3-rc2 is the latest RC. If no regressions are found with the new patches, it will become 1.8.3. Basically, if you are waiting for asterisk-1.8.x to be released, start testing asterisk-1.8.x-rcx and report feedback. -- 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 -- _ -- 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
Here is the patch did you apply it ? https://issues.asterisk.org/file_download.php?file_id=28206type=bug Date: Fri, 11 Feb 2011 08:46:36 -0200 From: vinic...@canall.com.br To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.3 That makes two of us. I tried asking on asterisk-dev but had no reply. - Mensagem original - Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of 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 -- _ -- 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 BLF stopped working
On 12/02/11 04:02, Bryant Zimmerman wrote: I am running 1.8.3 and my BLF lights have stopped working. The hints appear to be intact when I use core show hints. But none of the phones are getting the BLF updates. This has happend in the past and I have had to restart my server. What could be causing this to occur. It did not do this with the 1.6.x builds. Is there a way to reload the hints or force a refresh without re-starting Does a restart actually fix the problem? If not, compare the hint context in core show hints and the Subscr.Cont. line in sip show peer xxx, where xxx is one of the extensions attempting to subscribe to hints. Make sure the two match. I've had this problem before, and that was the cause. -- _ -- 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 1.8.3 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * 'sip notify clear-mwi' needs terminating CRLF. (Closes issue #18275. Reported, patched by klaus3000) * Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (Closes issue #18031. Reported by rain. Patched by bbryant) * Fix cache of device state changes for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. (Closes issue #18342. Reported, patched by nivek.) * Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 so res_jabber doesn't think there is already an XMPP connection sending device state. Also clean up CLI commands a bit. (Closes issue #18272. Reported by klaus3000. Patched by Marquis42) * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer-cdr = NULL, set it to not post. (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) * Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2 Thank you for your continued support of Asterisk! -- _ -- 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
