Re: [asterisk-users] Asterisk 1.6.2.12 segfault
Grant, Perhaps it's time to upgrade? I used to see tons of unexplained segfaults in 1.6.X, haven't seen any in 1.8.22.0 (I'm afraid to upgrade since I finally found a stable version) You should, also, have you heard of FreeSWITCH? IMO much more stable PBX software. Thanks On Thu, Aug 28, 2014 at 5:45 AM, Grant Bagdasarian g...@cm.nl wrote: Hello, Could someone explain to me what this means? asterisk[30269]: segfault at 0008 rip 2aaac8b388f2 rsp 40a75910 error 4 Also, would this segfault crash the whole Asterisk process or will Asterisk continue to run? Is it possible this would affect/disconnect “SOME” DAHDI channels, but not all? At this point, upgrading is not an option, even though I agree we should. Regards, Grant -- _ -- 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 sends SIP/2.0 481 Call/Transaction Does Not Exist to INVITE
To: sip:8009499...@x.yyy.32.10:5060 ;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 In your call sample To has a tag. if this is the first Invite it can't have a tag at the end, otherwise Asterisk will look for an existing dialog in its database and will show an error, if can't find any. It looks like the other end is never closing the previous dialog?.. is Asterisk sending a proper 200 OK after receiving a BYE? NAT problem? Thanks, I think you are correct on that... there are no NAT problems... the dialog ends with Asterisk sending a 481 because the dialog does not exist. Im going to try to have the customer remove that tag from the To header. -- _ -- 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 sends SIP/2.0 481 Call/Transaction Does Not Exist to INVITE
Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499...@x.yyy.32.3:5060;transport=udp SIP/2.0 Record-Route: sip:X.YYY.32.10;lr=on;ftag=247898 To: sip:8009499...@x.yyy.32.10 :5060;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: Scott Thompson sip:7166359...@x.yyy.32.10;tag=247898 Via: SIP/2.0/UDP X.YYY.32.10;branch=z9hG4bK542e.5042d534.0 Via: SIP/2.0/UDP X.YYY.33.178:5060;rport=5060;received=X.YYY.33.178;branch=z9hG4bK57b720cccb00f8498662f48e8 Call-ID: 94f80f866e877490729548a079abe371@192.168.101.5 CSeq: 2 INVITE Contact: sip:7166359...@x.yyy.33.178:5060 Max-Forwards: 69 x-inin-crn: 2001471530;loc=Amherst;ms=STAMPEDE-MS Supported: join, replaces User-Agent: ININ-TsServer/3.13.11.12748 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, SUBSCRIBE Accept: application/sdp Accept-Encoding: identity Content-Type: application/sdp Content-Length: 252 Proxy-Authorization: Digest username=909003660716,realm=X.YYY.32.10,nonce=523755911a22ed0fae66765d46ef9131e311fbb9d2fb,uri=sip:8009499...@x.yyy.32.10 :5060,response=cb6306569b3047ac35064dcb5aee6db4 X-Enswitch-RURI: sip:8009499...@x.yyy.32.10:5060 X-Enswitch-Source: X.YYY.33.178:5060 The only problem I see with this INVITE is the VIAs are not right after the INVITE line... although in https://www.ietf.org/rfc/rfc3261.txt, it explicitly states the the order of the headers is not a requirement, it seems Asterisk does make it one... The relative order of header fields with different field names is not significant. However, it is RECOMMENDED that header fields which are needed for proxy processing (Via, Route, Record-Route, Proxy-Require, Max-Forwards, and Proxy-Authorization, for example) appear towards the top of the message to facilitate rapid parsing. The relative order of header field rows with the same field name is important. -- _ -- 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 audio from meetme conference bridge
Asterisk intermittently does not send audio back to the callers in the meetme conference bridge. If the caller hangs up and calls back sometimes the audio will work and sometimes it does not. We have taken packet captures and reviewed the SIP and SDP, both are correct and you can actually hear the audio being transmitted from the callers to the conference bridge but no audio is sent back to the callers. We've also verified the meetme conference flags are correct so the callers are not set to be muted. I should mention this only happens when the call is going from one asterisk box to another (the second box being where the conference is living). If all conference participants are on single asterisk machine the problem does not occur. Has anybody had this problem in past? I could not find the issue doing a google. -- _ -- 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 audio from meetme conference bridge
I may have discovered what the issue was... we had *rfc2833compensate=yes* on version 1.8 of asterisk, I believe the extra RTP asterisk was sending with this enabled caused a problem. Can anyone confirm this? I'm not exactly sure (in technical terms) what this setting does...I only found a description that it's necessary for asterisk version=1.4 snip -- _ -- 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] DTMF digits are coming through twice
Any ideas? On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa vipki...@gmail.com wrote: Call was to 7167436110 -- _ -- 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] DTMF digits are coming through twice
The trace is attached 3 emails back. -- _ -- 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] DTMF digits are coming through twice
Sorry the attachment was too big. here is link: http://www.2shared.com/file/Ola640Pn/doubledigit.html On Fri, Oct 12, 2012 at 9:24 AM, SamyGo govoi...@gmail.com wrote: Why am I feeling like I'm the only one here who is not able to see any pastebin link or attachments in this thread ! -- _ -- 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] DTMF digits are coming through twice
Only callers calling from Earthlink internet connection On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly d...@donkelly.biz wrote: Is this happening for all callers, or just iPhone callers? --Don -- _ -- 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] DTMF digits are coming through twice
I've been running an Asterisk server (1.6.2.17.2) for over a year without any major issues. All of a sudden people are unable to login to their voicemail because Asterisk is seeing DTMF twice for each digit the caller pushes. We've noticed the problem only consistently happens to callers from specific locations. All the locations having the issue use te same ITSP and internet provider. The ITSP swears it's not them because they tested outside their phone system, straight from a d-mark. We've tinkered and played with all options in Asterisk related to DTMF with no success. Aside from upgrading Asterisk, I'm at a loss as to how to fix this. This system has worked for over a year without this issue and all of a sudden it appears. My thought is that it's the internet provider (Earthlink) but they say it can not possibly be them. Here is my DTMF settings in sip.conf: dtmfmode=rfc2833 relaxdtmf=yes Any input appreciated! 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] DTMF digits are coming through twice
I'm not sure I follow, the packet capture on the asterisk server shows double digits being entered. Does that mean it's the source? On Wed, Oct 10, 2012 at 11:55 AM, SamyGo govoi...@gmail.com wrote: Hi, Not exactly a solution, but I'm sure you must've taken pcap traces of a few such sample calls. See in their RTPs that you are receiving repeatedly same RTPs which will tell you that any DTMF packet is coming in twice by the source or not ! just one such simple pcap will help you identify at whose end the issue lies. If the source is your vendor sending you RTPs twice for DTMFs then send them the capture and ask them to fix it however they can. BR, Sammy On Wed, Oct 10, 2012 at 8:44 PM, Vik Killa vipki...@gmail.com wrote: I've been running an Asterisk server (1.6.2.17.2) for over a year without any major issues. All of a sudden people are unable to login to their voicemail because Asterisk is seeing DTMF twice for each digit the caller pushes. We've noticed the problem only consistently happens to callers from specific locations. All the locations having the issue use te same ITSP and internet provider. The ITSP swears it's not them because they tested outside their phone system, straight from a d-mark. We've tinkered and played with all options in Asterisk related to DTMF with no success. Aside from upgrading Asterisk, I'm at a loss as to how to fix this. This system has worked for over a year without this issue and all of a sudden it appears. My thought is that it's the internet provider (Earthlink) but they say it can not possibly be them. Here is my DTMF settings in sip.conf: dtmfmode=rfc2833 relaxdtmf=yes Any input appreciated! 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 -- _ -- 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] DTMF digits are coming through twice
After comparing packet captures of good and bad calls. It looks like the double digit is coming from rfc2833 and dtmf inband. It looks like the inband tone is splitting the rfc2833 in two? Is there some way to resolve this??? On Wed, Oct 10, 2012 at 12:28 PM, Vik Killa vipki...@gmail.com wrote: I'm not sure I follow, the packet capture on the asterisk server shows double digits being entered. Does that mean it's the source? On Wed, Oct 10, 2012 at 11:55 AM, SamyGo govoi...@gmail.com wrote: Hi, Not exactly a solution, but I'm sure you must've taken pcap traces of a few such sample calls. See in their RTPs that you are receiving repeatedly same RTPs which will tell you that any DTMF packet is coming in twice by the source or not ! just one such simple pcap will help you identify at whose end the issue lies. If the source is your vendor sending you RTPs twice for DTMFs then send them the capture and ask them to fix it however they can. BR, Sammy On Wed, Oct 10, 2012 at 8:44 PM, Vik Killa vipki...@gmail.com wrote: I've been running an Asterisk server (1.6.2.17.2) for over a year without any major issues. All of a sudden people are unable to login to their voicemail because Asterisk is seeing DTMF twice for each digit the caller pushes. We've noticed the problem only consistently happens to callers from specific locations. All the locations having the issue use te same ITSP and internet provider. The ITSP swears it's not them because they tested outside their phone system, straight from a d-mark. We've tinkered and played with all options in Asterisk related to DTMF with no success. Aside from upgrading Asterisk, I'm at a loss as to how to fix this. This system has worked for over a year without this issue and all of a sudden it appears. My thought is that it's the internet provider (Earthlink) but they say it can not possibly be them. Here is my DTMF settings in sip.conf: dtmfmode=rfc2833 relaxdtmf=yes Any input appreciated! 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 -- _ -- 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] Voicemail crashs asterisk
http://asterisk-sucks.blogspot.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] Voicemail crashs asterisk
#disasterisk fail #freeswitch win Hi guys, I have the following problem: My System: asterisk 1.8.11.0 on debian squeeze I login to my mailbox from voicemailmain. Once I am logged in, I get my number of messages announced. I change the directory of old messages 2 times in a row: 2 1 2 1. Asterisk exits completely. Can your help please? 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] Exceptionally long voice queue length
I guess that's why people call it *disasterisk* On Mon, Jan 16, 2012 at 8:10 AM, Vik Killa vipki...@gmail.com wrote: Anybody? I've read this might be a deadlock On Thu, Jan 12, 2012 at 8:09 AM, Vik Killa vipki...@gmail.com wrote: Asterisk 1.6.1.22 On Thu, Jan 12, 2012 at 2:08 AM, Sammy Govind govoi...@gmail.com wrote: which version of Asterisk are you using !. AFAIK this issue has been in asterisk for queue calls and I'm not sure if this has ever been resolved fully and stabilized. Not binding to Local channel only, I've seen this on SIP and IAX channels as well ! -- _ -- 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] Exceptionally long voice queue length
Anybody? I've read this might be a deadlock On Thu, Jan 12, 2012 at 8:09 AM, Vik Killa vipki...@gmail.com wrote: Asterisk 1.6.1.22 On Thu, Jan 12, 2012 at 2:08 AM, Sammy Govind govoi...@gmail.com wrote: which version of Asterisk are you using !. AFAIK this issue has been in asterisk for queue calls and I'm not sure if this has ever been resolved fully and stabilized. Not binding to Local channel only, I've seen this on SIP and IAX channels as well ! -- _ -- 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] Exceptionally long voice queue length
Asterisk 1.6.1.22 On Thu, Jan 12, 2012 at 2:08 AM, Sammy Govind govoi...@gmail.com wrote: which version of Asterisk are you using !. AFAIK this issue has been in asterisk for queue calls and I'm not sure if this has ever been resolved fully and stabilized. Not binding to Local channel only, I've seen this on SIP and IAX channels as well ! -- _ -- 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] Exceptionally long voice queue length
I'm seeing this error thousands of times per minute and it's causing the CPU to sky rocket WARNING[16095]: channel.c:1039 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/*7...etc... Any idea what could be causing this? -- _ -- 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
