similar to: single port voip gateways

Hello list, I'm getting droped calls on an asterisk installation. When on GS phone dials another one, the call is dropped after some (usually random) time but most of the tome within 3 to 20 seconds. I think the cause is stated on the logs, see bellow, and is related with the message "Didn't get a frame from channel: SIP/3805-df43", but I can't figure why. asterisk logs:

I'm looking to build an application which requires a phone to autodial when the handset is lifted off of the hook. Since the system will be available to the general public, I want a solid phone that will withstand abuse, but not be too expensive. I'm leaning towards using a Sipura adapter with an analog outdoor handset. However, I don't nor ever have had a sipura adapter, so I'm

We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing "hello ? Hello ?" and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of

Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul.

Hello, I bought a Grandstream Handytone 486 to forward incoming calls from our old analogue PBX to the asterisk server. My first test was connecting an analogue phone to the Handytone and calling a sip phone - worked. Now I used the same cable to connect the line port of the Handytone to the analogue pbx. When I call the number of the analogue PBX I hear a clicking inside, but the call

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone

Hello, I'm noticing strange behavior in one of our Asterisk nodes where the ACK is always sent to the proxy, but RR is not enabled for calls. The proxy drops the ACK. I'm using the AMI interface to originate a call: Action: login Username: myusername Secret: mypassword Events: on Action: Originate Channel: SIP/<SOMENUMBER>@proxy1 CallerID: <SOMENUMBER> Application: Playback

We are looking to deploy asterisk at one of our locations that will have about 50 phones. I have been buying different phones to test there quality and feature set. So far we have a Grandstream 2000 Grandstream HandyTone 488 Cisco 7912 Polycom SoundPoint IP And we are looking at getting a Linksys SPA-942 Anyone have a favorite? -------------- next part --------------

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

Hi All, I have setup asterisk 1.4.22; so far everything good. Except, I am still searching for voIP phones. Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. I am in the US. thanks, Mike

Hi I have a analog phone connected to my Gateway Handytone and registered to Asterisk 1.4 I have configured my HandyTone 488 (in the section FXS Port) for make and receive calls, however I can not transfer a call when it come via PSTN. But, when a call come from via IP I can transfer it. [phoneanalog] type=friend secret=XXXXXXX context=local nat=no qualify=yes host=dynamic dtmfmode=rfc2833

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

hi, all I have configure tdm400p analog fxo card. that's ok. but how to chek that is working properly or not. i chek with ztcfg -vvvv and zttool . that's ok. i want to dial from my fxo port to another extesion. zaptel.conf ------------------ fxsls=1,2,3,4 defaultzone=in loadzone=in zapata.conf ---------------- context=mycontext signalling=fxl_ls group=1 channel=1-4 thanks' in

Hi, I am quite new to asterisk so I am not sure what is needed to figure out this problem. If more information is needed and not provided I will gladly provide it. I have a very basic asterisk setup. 1 x100p card and a grandstream handytone 286. I can make calls fine to most phone numbers from the handytone device the trouble seems to come when I dial this number 591-1079. It puts me through to

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

uugh : i promise that this will be my last question of the day. i hate to constantly bother this group but it takes me time to get familar with all of these functions, tricks and and manipulations. i appreciate everyone's patience. i used too splus a lot but i've gotten rusty. i have a matrix of say 200 rows and 600 columns. i have a function "getprofit" that takes two series

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

I have problem with fax machine Panasonic DX600. It's connected to Grandstream Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP provider. To some numbers I can't send FAX, and I get following error on CLI. WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 I believe that Panasonic DX600 machine supports T38. And when I have