search for: davies147

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

Can't I upload the 3PCC firmware ? available from the Cisco website? Actually it came with sip88xx.... firmware. Regards . On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies147 at gmail.com> wrote: > 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. >...

Hi list, There are many Polycom experts on this list -- hopefully someone has a solution. With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk causes the Polycom 601 phones to start dumping these messages to the CLI. -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.x They continue on until we force the devices to reboot from the

Hi, I will confess immediately that this is only tested on 1.2.24, and I would be interested to know if it happens on 1.4, but I cannot find a bug-tracker entry which represents this issue. Consider a PSTN call which comes into asterisk, and is bridged to a SIP phone. The phone operator then places the call on hold (hold music plays) and a second call is made from this handset to a Queue...

Anyone interested in this feature? I have a version 0.1 patch, which is currently against 1.2.25-bristuffed, but which should port trivially to almost any version. I am away until Tuesday 8th April, but if there is enough interest, I will open a "new-feature" ticket and upload the patch to the bugtracker so that more capable programmers can laugh at it ;-) It should work reasonably on

...or not)? Could be wrong, but I'm guessing there may be an incorrect DMARC policy somewhere - although this is the only fail I could find in the headers. bounces at lists.digium.com; dmarc=fail (p=NONE sp=NONE dis=NONE) header.from=gmail.com On 12 June 2017 at 09:12, Steve Davies <davies147 at gmail.com> wrote: > 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...

...st) - A few people are seeing a Cause 34 (congestion) from ISDN installs, where there clearly is an available channel. This was originally related to Bristuff as it happens to ISDN2 users, but there is at least one report of an unpatched 1.6.x user seeing the same issue. 2009/4/23 Steve Davies <davies147 at gmail.com>: > I think I have a site where this is happening, but all I see is a > series of outbound calls, which look perfectly normal, but at some > "random" point, ISDN channels stop being available, until they run > out. It can go anywhere from weeks down to a couple...

...m guessing there may be an incorrect DMARC > policy somewhere - although this is the only fail I could find in the > headers. > bounces at lists.digium.com; > dmarc=fail (p=NONE sp=NONE dis=NONE) header.from=gmail.com > > > > On 12 June 2017 at 09:12, Steve Davies <davies147 at gmail.com> <davies147 at gmail.com> wrote: > > 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...

Hi ALL, After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf. Do i need to do anything else to fix my MWI on polycom 501 ? It was working with 1.2 asterisk. pollmailboxes=yes -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e1f59a92/attachment.htm>

Hello, There are 3 PRI's connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Zapata.conf -------------------------- [channels] context=default switchtype=euroisdn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes

Actually now I have the phones with SIP firmware. I will try with 3pcc firmware along with XML files. Or any idea if we have CUCM application can we change the firmware. am ready to buy the developer edition. Regards . On Mon, 5 Dec 2016, 3:34 p.m. Steve Davies, <davies147 at gmail.com> wrote: > I tried... repeatedly... I failed. I bought some 3PCC phones, and they > just worked. > > If you have the relevant Cisco telephony server product you might be able > to trick it into doing what you want, as that has the proper upgrader for > that model o...

Hi, To solve echo problems, I'm considering 2 alternatives. 1> Sangoma A104d - I can't find support for asterisk 1.2.1 2> Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command "reload" can change rx_gain and

I want to use extension numbers that begin with the # key in my dialplan, but I can't get my Aastra phone (6731i) to transmit the # key to asterisk. It works fine for the * key. I've tried numerous Local Dial Plan patterns in the aastra web configuration but none of them worked. My current Local dial Plan pattern is "x+#|xx+*|#x+". Any help would be appreciated. -- Marvin

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

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 using penalty: queues.conf: member => SIP/dev1,0,Agent1 member => SIP/dev2,0,Agent2 member =>

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 <xserverlinux at gmail.com> wrote: > > > On Wednesday, March 4, 2015, ricky gutierrez

Hi, I unsuccessfully tried to download AstManProxy, clicking over download button in http://github.com/davetroy/astmanproxy/tree/master . It failed with "XML error". How can you download AstManProxy ? Has the project moved to somewhere else ? Have its features been deprecated and replaced by something embedded in Asterisk code or elsewhere ? Regrads -------------- next part

Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent "generic" board out there that I could be investigating? It would be such a shame to waste a PCI slot that

Hi, On one of our asterisk systems that is quite busy, we are seeing the following from 'netstat -s': Udp: 17725210 packets received 36547 packets to unknown port received. 44017 packet receive errors 17101174 packets sent RcvbufErrors: 44017 <--- this When this number increases, we see SIP errors, and in particular Qualify packets are lost, and