similar to: Problem with outbound calls

Greetings. I have a scenario I would like some advice on. I have a 100,000 square foot building that we will be moving some work crews into. It has offices on each end of the building and a fiber line between them. I currently have an asterisk 1.2 system in place and about 30 phones. My problem is they want a few fax machines out in the warehouse area where I currently have no wiring for

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hello all, I'm a newbie in * and i want to start by making internall calls between ip phones (Grandstream BT100, and HT286), if someone can help me with an ewample of sip.conf file specially with the "register" field in [general] defintion. Thanks D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail sur

Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I've tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can recommend an ATA that might do the trick? Thanks, Alex -------------- next part -------------- An HTML

Hello, fax/ata(ht286) -> asterisk/tdm04b -> pstn fax machine I can fax out from the sip side, but I can't fax in from the PSTN side. When I try to send a fax, asterisk sees the call and show me this: "Redirecting Zap/1-1 to fax extension" "Timeout on Zap/1-1" TCPDUMP doesn't show any activity to the extension that I configured to be the fax machine.

A DID number was dedicated to receive fax, but i have the problem when getting fax call, which call will become a normal phone call and no fax was printed. When fax is detected, the fax extension is executed and dial the extension of the HT486 device (firmware 1.0.5.22). Somehow sending fax out working well. In the mailing lists, i notice some are using HT286 and it work. Could someone share

I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

I just checkout the cvs code for asterisk...... when I use my grandstream phone (that worked on the old code that was about 2 months old) I do not hear anything at all... I get this error: Sep 27 23:20:27 WARNING[1142127920]: File chan_sip.c, Line 444 (retrans_pkt): Maximum retries exceeded on call 0765c89e-9d67-3c0a-b9b9-2e7f3cd1d9ef@192.168.50.248 for seqno 58430 (Response) here is my

I'm having no end of trouble with some Siemens Gigaset phones and GS HT286s. Gigaset 100 and 3010 phones work perfectly, but a 4010 only rings once then it goes off and then just flashes it's LEDs and displays "incoming call" on the LCD with no further ringing. According to the manual it is CTR37 but the only setting on the GSs is CTR21, I've tried different cables but some

I've dealing with this issue for a while, and I'd really like to know if anybody has experienced the same pain before :-) I've a lot of Grandstream HandyTone 286, loaded with the latest firmware (1.0.8.16) from the GS website. In my sip.conf, this ATA's are configured as: [05] type=friend username=05 secret=XXXX callerid="User 05" host=dynamic nat=yes qualify=yes

I recently reconfigured a box running FreeBSD 5.3 with Samba v. 3.0.7 running as PDC. It was servicing 2 networks, processing logons for 192.168.200.0/24 and 192.168.115.0/24. The reason for 2 networks was security, 115.0/24 had some stuff on it that the users on 200.0/24 shouldn't have access to. Before the reconfiguration, the system was working 100%. To make everything better I flattened

Trying to add a new driver using add printer wizard. A panic action occur : m403 (10.217.5.209) connect to service print$ initially as user spu (uid=0, gid=221) (pid 8619) [2009/02/16 10:40:35, 0] lib/fault.c:fault_report(40) =============================================================== [2009/02/16 10:40:35, 0] lib/fault.c:fault_report(41) INTERNAL ERROR: Signal 11 in pid 8619 (3.2.4)

Hello, I'm using the Samba server 3.0.33 that exports volume from a GPFS. The GPFS strongly dislikes unlinking files that are locked (resulting in permission denied) using fcntl F_SETLEASE. It seems that the Samba *sometimes* tries to unlink a file that is oplocked. Why? Is this a bug? Why it does not happen always but only sometimes? I have strace logs showing: Wrong case: 8711

Not sure if this is the correct subject line but my recently installed Centos build (Linux localhost.localdomain 3.10.0-229.14.1.el7.x86_64 #1 SMP Tue Sep 15 15:05:51 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux) periodically just freezes - completely locks up, no activity, nothing in the logs, just stops dead requiring a power off and reboot. I've really looked around to try and find the

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just