Re: [asterisk-users] Asterisk 1.4.x segfaulting daily

On 14 December 2011 12:56, Paulo Santos paulo.r.san...@sapo.pt wrote: Hello list, An Asterisk installation that was doing fine suddenly stared segfaulting a couple of times per day. I enabled all the logging and debugging to try to find a pattern but there was too much information to see

Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

On 28 December 2011 03:02, Joseph syscon...@gmail.com wrote: No, it makes no difference, on the other end is asterisk 1.4.39 and 1.8.8 is still giving me:  Executing [4@internal:1] Dial(SIP/11-0003, IAX2/home_server:@192.168.141.1/4,30,rw) in new stack    -- Called

[asterisk-users] SIP and NAT best practices since recent changes?

Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done in the past is: Global: nat=no SIP handsets that are local: nat=no SIP handsets that are remote: nat=yes ITSP SIP trunks: nat=yes

Re: [asterisk-users] SIP and NAT best practices since recent changes?

On 11 January 2012 15:43, Kevin P. Fleming kpflem...@digium.com wrote: On 01/11/2012 05:29 AM, Steve Davies wrote: Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done

Re: [asterisk-users] Asterisk CLI unresponsive

On 3 February 2012 12:12, Jonas Kellens jonas.kell...@telenet.be wrote: On 02/03/2012 01:05 PM, Mikhail Lischuk wrote: Jonas Kellens писал 03.02.2012 12:09: using asterisk 1.6.2.22 What is wrong with Asterisk when the CLI becomes unresponsive ?! Greetings. I am using the same version, 

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be wrote: ** Hello, is there anyone that can give me some more information on these deadlocks ?! How can these deadlocks occur and what is good practise to avoid these problems ?? Jonas. The only way to avoid deadlocks is

Re: [asterisk-users] Asterisk ACL

On 2 April 2012 14:06, Mark Farmer mark.far...@gagenetworks.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: 02 April 2012 13:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

[asterisk-users] CONNECTEDLINE() updated during SIP events?

Hi, I have read the excellent information here: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information and believe I have an understanding of what is offered. I have a couple of questions: - Is it possible to update COLP/COLR when a SIP redirect occurs, or when a SIP

Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

On 25 April 2012 16:55, Richard Mudgett rmudg...@digium.com wrote: [snip] - Is it possible to have the COLP/COLR information updated when a SIP attended transfer is completed? If so how? Transfers generate connected line update events automatically.  The connected line interception macros

Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

On 25 April 2012 18:05, Kevin P. Fleming kpflem...@digium.com wrote: On 04/25/2012 11:54 AM, Steve Davies wrote: A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms

[asterisk-users] SIP Question - Early audio one-way or 2-way?

Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer. If a SIP call is initiated (INVITE) and receives either a 180 with SDP, or a 183 with SDP, then the remote party will start to

Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

On 24 August 2012 15:34, Faisal Hanif fai...@vopium.com wrote: Steve Davies davies...@gmail.com wrote: Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer. If a SIP call is initiated

Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

On 1 September 2012 09:08, Olle E. Johansson o...@edvina.net wrote: 31 aug 2012 kl. 13:13 skrev Steve Davies davies...@gmail.com: On 31 August 2012 07:49, Olle E. Johansson o...@edvina.net wrote: 24 aug 2012 kl. 16:18 skrev Steve Davies davies...@gmail.com: Hi SIP Gurus, I've tried

Re: [asterisk-users] Congestion() forcing PRI channels to be not available

On 19 December 2012 21:54, Christopher Harrington ch...@acsdi.com wrote: You probably already know this, but 1.4x is very old (released in 2006) and is officially end-of-life. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions You might get more help or better behavior by updating

Re: [asterisk-users] Fundemental changes to CDR within single asterisk family

On 4 April 2013 09:05, Ishfaq Malik i...@pack-net.co.uk wrote: On Tue, 2013-03-26 at 07:26 -0500, Matthew Jordan wrote: On 03/26/2013 05:22 AM, Ishfaq Malik wrote: Hi In asterisk 1.8.7.0, an inbound call that was transferred to another peer would have 2 cdr entries. In

Re: [asterisk-users] IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6

Hi Xavier, The issue you are seeing is an old Asterisk/Bristuff bug that was fixed years ago. Basically ISDN is unable to understand a call going from RING state to BUSY state, so Asterisk converts the BUSY into a HANGUP/Normal Clearing, and warns that this is happening. Sadly, in that old

Re: [asterisk-users] FW: IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6

Xavier, DoNotDisturb generates a Busy indication. Insert that into my earlier response, and you have an explanation of why the call tries to go from RING to BUSY, and confirms my theory. No you cannot replace the Zaptel card driver on its own (and the problem was bigger than that anyway), as

Re: [asterisk-users] ignore 183 session progress in parallel call scenarios

I am sure I submitted the following alternative behaviour to the bug-tracker in the past, but cannot find any reference to it. Here is the patch I use to IMHO improve this behaviour. In case it is not officially uploaded, I will state here that this code is disclaimed and unencumbered as if

[asterisk-users] Connected Line presentation in 1.8.x upwards

Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some untrusted trunks. On these

Re: [asterisk-users] Connected Line presentation in 1.8.x upwards

On 29 July 2013 16:55, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: From:Steve Davies davies...@gmail.com To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date:07/29/2013 10:53 AM Subject:[asterisk-users

Re: [asterisk-users] hangup call gw FXO

Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs Regards, Steve On Thu, 5 Mar 2015 at 11:41 ricky gutierrez

Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

Hi, In my experience, all Yealink phones work just fine with Asterisk, we have hundreds (perhaps even low-thousands) out there with customers on Asterisk 1.2, 1.6.2, 1.8 and 11. If you are accurately representing the SIP trace on the phone and the SIP trace on Asterisk, then I would strongly

Re: [asterisk-users] pedantic=yes in sip.conf

Alan, A little more context would be useful. Where are you putting the '#' and why? ( If all else fails, print it out and mail it to them ;-) ) %23 is the correct encoding for a hash '#' symbol in many SIP contexts, and should be decoded by a properly functioning far-end. Regards, Steve On

Re: [asterisk-users] Cisco IP 8841 asterisk integration

If you need to know what the provisioning XML should look like for a 3PCC build of a Cisco 78xx or 88xx phone, then let it boot without provisioning, and then log in to its web interface. Select admin mode and log-in if necessary. Then edit the URL in the browser from: http://ip-address/admin/ to

Re: [asterisk-users] Cisco IP 8841 asterisk integration

able from the Cisco website? > > Actually it came with sip88xx firmware. > > Regards . > > > On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies...@gmail.com> wrote: > > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > the

Re: [asterisk-users] Cisco IP 8841 asterisk integration

Hi, You have to buy the 3PCC version for this to work. Once you have this, they work very much like the Cisco SPA handsets. I also ended up with a non-3PCC handset and it is useless, and as far as I can tell they cannot be re-flashed. Cheers, Steve On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan

Re: [asterisk-users] Asterisk sip_autodestruct messages - extensions locked

Based on the line number of that error in chan_sip.c, it looks like you're running Asterisk 1.8 or earlier. AFAIK, The issue you are seeing was fixed years ago, but not THAT many years ago! If I'm right, you should upgrade to fix that issue. Cheers, Steve On Fri, 30 Jun 2017 at 13:39 Stefan

Re: [asterisk-users] ** in extensions.conf

On Wed, 26 Apr 2017 at 20:29 Jerry Geis wrote: > I just tried this in my extensions.conf > > exten => **,1,Noop(Testing) > exten => **,n,Playback(demo-congrats) > > Did a reload... and the above does not happen. > I created as 12 instead of the ** and that works fine. > >

[asterisk-users] Using queue priorities to add agents

Hi, I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace... 1) (this bit works fine) I want a queue caller to have access to the basic set of agents initially, with an overflow to additional agents if they are busy - This is done

Re: [asterisk-users] Using queue priorities to add agents

member => SIP/104.2,4,Debbie > member => SIP/105.2,4,Luci > member => SIP/106.2,5,Sheila > member => SIP/107.2,6,Mike > > So every 20 seconds it jumps up to the next Penalty and every few minutes > it resets the penalty back down to 1 and starts again. > > > On Thu, Ma

Re: [asterisk-users] Using queue priorities to add agents

t queue to timeout after 60 secs. Then send to the overflow > queue with all agents/members as same priority. > > > > > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Davies > *Sent:* Thur

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

I am also getting this, three or four times in the last month after years of no problems. I agree that Gmail is the likely common factor, but I would love to have access to these bounce messages to know whether it is actually an overly-paranoid list server! Steve On Mon, 12 Jun 2017 at 09:09

Re: [asterisk-users] RTP Timestamp rewind

Mark, You have cropped the image you inserted above and removed a very important part of the line you highlighted. I think is says ",Mark" after the time value - You can even see the un-cropped comma in your picture. RTP timestamps can be reset mid-stream if needed - It is part of the spec, and

Re: [asterisk-users] Queues and penalties

I'm fairly sure the patch to App Queue that was added to Asterisk 13+ should do the job... It causes agent priorities to "float up" over time so that new agents are included without excluding old agents. I can't find it right now but there can't be that many app_queue patches to ast 13 in the

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