similar to: Grandstream GXE2504A codec disable option

Hi i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

Hi guys, I setup an IPPBX and some IP Phones to register to it. IPPBX has 6 "register=> xxx@xxx.x.x.x.." sip accounts. When these sip accounts register timeout due to the WAN network problem all the extensions got logon failed. It seems IPPBX register to its own sip proxy server first. If it failed it cannot take the sip clients' request for registration. Anybody has

Hi , I am now running squid with ncsa_auth. I have bound ip addresses to usernames. So users now can access Internet from their ips. Now I want a few users to prevent from accessing all the sites. But Instead, I want them to allow to access a few sites scuh as google.com,cnn.com, bbc.com. I want to limit in that way. I have wriiten below rules. But those users still can access all the sites.

Hi, What's the correct way to disble TSO (TCP Segmentation Offload) in RHEL5? I have tried adding those options in ifcfg-ethX configuration file: # grep ETHTOOL /etc/sysconfig/network-scripts/ifcfg-eth0 ETHTOOL_OPTS="tso off" And also with: ETHTOOL_OPTS="-K eth0 tso off" But when restating the server TSO is enabled: # ethtool -k eth0 tcp segmentation offload: on As

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

to start with i am new to asterisks and i am also a telcom idiot. with that said i have one vonage line i would like to hook up in my soon to be built Asterisk ippbx server. Now with the one Vonage (with call waiting) line can i receive more one call using an auto attendant route the call the approiate extention? thanks mike

*Hi Everybody* ** *I am building a small ippbx network for my office* *I have 6 hard ip phone's and asterisk server but * *now for outging and incoming calls i want to use* *gsm router instead of x100p card ... or pstn* *I want my calls will go out and come through mobile sim card (gsm router).* *My mobile service provider Simultaneously 64 lines conference...* *How many calls can i receive

I have a linux server connecto into a 100mps LAN, i use iptables implementation of shorewall (shorewall.net) and then I used the wondershaper. When I adjusted the values below, I got a download speed for vsftp at a whooping 1187.91 KB/s but a upload speed is only a painful 27 KB/s. I dont fully understand the underpinng codes behind the wondershaper htb stuff. can someone guide me on how can i

After moving from samba 4.0.0rc5 to 4.3.11. There seems to be a fair amount of disk activity. It seems to center around the msg.sock messaging. IS there away to disble the messaging temporarily? To verify that it is causing the problem? Thanks

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Anyone in need of some of these? Garrett Smith Sales Executive garrett.smith@b2llc.com B2 Technologies 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix,

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

Hi All We have a private network setup (no nat) with three types of phones connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco 7940 IP phones. When we ring polycom to grandstream or grandstream to polycom then both phones can send and receive voice fine and all is well. When we dial any combination of Cisco and either Polycom, or Granstream the Cisco, no voice is being sent

I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,email@mail Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with

Hi list, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream? for example: exten => 0,1,Playback(pls-wait-connect-call) same=> n,SIPAddHeader(Alert-Info:;info=ring3) same=> n,Dial(SIP/310&SIP/318,30,t) can not get it to work any idea o tips? regardss -- rickygm http://gnuforever.homelinux.com

I have setup Grandstream to connect to my Asterisk Server. All the digits 0-9 are accepting dtmf. But When I try to send the call by Pressing # Key, nothing happens. Does anyone has a standard configuration for Asterisk and Grandstream as a PDF file or something to see. How do you send the connect signal? Seshu Kanuri -----Original Message----- From: asterisk-users-admin@lists.digium.com

On the Granstream 102 box that I have in front of me, there is a "feature list" on the side. One of the features has grabbed my attention: " - optional voice encryption (model 102D)" Now, digging through Grandstream's site, I see that it's not offered quite yet. However, sending mail to their standard "information" email address has resulted in no

Hi Mark We setup a small IPPBX using Asterisk. So far we've been using it for voip network. It's been working fine. We also have a Dialogic D/4PCI four port FXO board with us. We've found some references in the user guide available on your web site and couple of other mailing transaction threads that Asterisk provides pay-for-add-on support for Dialogic h/w. Can you please tell us

Hi, I am using asterisk 1.2.1 for building an ippbx for my setup. I am having problem in accessing advanced options option that comes on pressing 5. It says to leave a msg and asks for extension. Now when i dial the extension the IVR silently goes into the top IVR menu. And the asterisk console show the message: "No entry for <EXTEN> in voicemail config file" As i