similar to: Cisco IP 8841 asterisk integration

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

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.

True agree, problem is somehow the people purchased.... am supporting to overcome that. Trying level best... around 20 phones has been purchased.... ?? On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, <mefhigoseth at gmail.com> wrote: > With all the money you plan to invest in firmware, licenses, etc., you > have bought a Grandstream IP phone or Yealink... > -- >

Hello; I am facing a problem to let Cisco IP Phone 7942G register on Asterisk. The firmware has been downloaded from the TFTP successfully and currently I am running this load SIP42.9-3-1SR2-1S* I feel that there is a problem in the SEPMAC.cnf.xml but really I do not know which one to be used exactly. Basically, there is some effect that appears on the Phone (for example, it is appearing the

Hi, I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and downloaded Asterisk 1.8 current version, after installing Asterisk, while starting Asterisk it hangs at the stage of loading modules.conf, I checked the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still after updating through yast also i am facing the issue. Have anybody faced this type of issue with

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

I've just recieved a copy of the new SIP firmware for the Cisco 7970, those of you with Cisco accounts may wish to try it (shock horror I'm sticking with SCCP). This coincides with the release of v8 firmware for all Cisco phones (and for those of you running Sergio's chan_sccp v8 works fine) The firmware is now also (and for the 7970 SIP, only) distributed in ".cop" files,

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Am looking for a service provider who can provide enterprise SIP trunk with 100 channels concurrent sessions. I see some like Inphonex, Broadvoice... and etc.... Is there any suggestions for the service providers. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello. I am trying to find a way to retrieve data from Harvard Dataverse website. I usually don't have problem in web-scraping data but the problem here is that there are a bunch of data formats such as .tab, .7z and so and I just can't find a way to retrieve the data I am interested in woth an unique solution. Any hint? [[alternative HTML version deleted]]

Hi, I have a weird problem woth the time zone on my virtual (goDaddy) dedicated server I had to setup the timezone using system-config-date to GMT+7 i am in NY (GMT-5) when i do GMT-5 i get all the times 2 hours ahead seems like there's something im missing here If i add an event to my web app let's say 7:30 - 8:15 , the only way for me to show it right is GMT+7 can anyone help me

Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is

Hi users, I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file to G729 using file convert, but I am facing error as follows, file convert /tmp/welcome.gsm /tmp/welcome.g729 Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729! Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed. [Dec 20 17:24:18] WARNING[2221]: translate.c:256

I'm trying to upgrade a very old sparcstation running Solaris 8 which is running dovecot 1.x for few users. All I have for the task is good old gcc 2.95.2. The poor sod complains because it can't compute the sizeof(unsigned char prefix_text[]) at line 13 of log-error-buffer.c. Can I help it by - say - putting a constant between the '[]'? Or is it unfair? :-) I don't think

When I create a chained file with a mono and then a stereo substream, winamp crashes when it gets to the second stream. in_vorbis v1.16c. And when vcedit writes comments from a chained file, it only saves the first substream and throws subsequent streams away. Perhaps this is woth mentioning somewhere... Matthijs --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project

Hello, I''m using scriptaculous library (in symfony) v. 1.6.1 Great library !!!!! I have a problem. While in IE6 it''s all ok (the autocomplete field works perfectly) in FF 1.5.0.2 the field apparently isn''t working. If I try the example in the scriptaculous homepage (v. 1.4.0_pre4 of Prototype) doesn''t work with FF, while the simple online demo, that uses

Hi, How to create dialplan restriction for particular extensions.. -- Thank you with regards, Gopalakrishnan A.N, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100917/a4bc96f6/attachment.htm

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any