similar to: sip.conf configuration for internal calls

Hi everybody, I have some problems making calls from a sip user (HT286) to the pstn trough Digium Wildcard TE110P, i allways have an error : SIP 403 INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:0170708959@192.168.1.4;user=phone>

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:

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

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,

Hello my friends, My asterisk is going down randomly, following you will find some errors that i could see in the /var/log/asterisk/message at the moment of the crash: [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Maximum retries exceeded on transmission 1850202354 at 10.4.1.152 for seqno 21 (Critical Response) [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Hanging up call 1850202354 at 10.4.1.152 -

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, 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, 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 =>

Hi I have reading the voiip side i found some document says " The conference bridge runs Ulaw codec by default. If you let people connect with GSM or other codecs, Asterisk will use CPU power to convert audio between codecs " iam using vicidial and meetme for callcenter application. iam geting choppy voice, and voice breaks. iam using connecting VoIP SIP provider using g729 codec,

http://bugzilla.netfilter.org/show_bug.cgi?id=570 Summary: iptables save saves broken rules for -m owner Product: iptables Version: unspecified Platform: All OS/Version: All Status: NEW Severity: major Priority: P1 Component: iptables-save AssignedTo: laforge at netfilter.org ReportedBy:

Hello, I have an #include file containing user voicemail configurations. This works fine for the most part, but when a user changes their password via the phone the #include file is not updated. Is there a way to do this? I have something like this: In voicemail.conf: [default] #include "vmail_accounts.conf" 5551234567 => 0000, Customer, email@domain.com 5552234567 => 0000,

Hi, I am using the reldist package and having problems determining the p-value for the entropy value from the reldist function. I am able to properly determine the entropy value, but cannot figure out what function to use to find the p-value. I have tried using rpy, rpluy (which provides p-values for the polarization values) and investing the results from reldist(). Thus, far I cannot find the

My friends, I'm having some problems in my Asterisk, the thing is that Asterisk seem to be crashed (or dead) sometimes (2 times in 3 weeks) I noticed this today, when i could not make any internall call, tha calls to the voicemail (*1) did not work it just don't say nothing, nothing appears in console; i tried to make a CLI>stop now but nothing happens, i could not stop the asterisk

When firstboot is used from virt-v2v the scripts, if not fast enough, can get killed by Windows. After windows installs virtio drivers injected by virt-v2v it performs some internall reboot, stopping all the running services and killing any running firstboot script. This is problem mostly for MSI installs (like qemu-ga that was added recently) that can take several seconds to finish. This change

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while