similar to: Ring but now audio on answer

Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do

I am looking for a voip provider that provides good rates (below 5 cents/min or unlimited) to UK NCFA numbers. Braodvoice advertises they do unlimited to NCFA but does not have the ability to actually termiate those calls as per the CTO Nathan Stratton, and last he said they dont even have contracts in place to get service provisioned for that. As such I am looking for another provider to take

We have Polycom IP500's, and just starting recently (after the broadvoice patch I might add) after about 1-2 days these phones ring, and answer, but we get no audio on the phones. The caller can hear us, but we cannot hear the caller. Its happened 4-5 times and is only intermittent. No errors on the console, using g.711u. Any ideas? Tim Jackson Network Engineer Angelina County, Texas

I have a SIP connection to Broadvoice and sometimes when I make outgoing calls from a SIP ATA-188 (could be the same number) (the ATA-188, is currently the only extension), there is no audio passed for 5-10 secs. I have set all the codec the same to 711u and also ensured canreinvite is set to no. Any suggestions? Places to look for? -------------- next part -------------- An HTML

I've been reading this forum for over the past 4 years and have gained a wealth of knowledge. - Thanks to all! I don't post very often but I've just ran into a problem/condition that I simply can't figure out. - Hopefully some kind soul will help me. I've got an Asterisk server in a lab environment on my home LAN. - The LAN is "talking" to the Internet via a Linksys

I've got a problem with DTMF, again. My asterisk box is connected with the outside world (PSTN) via a sip proxy. The problem is that for some reason, I need to use rfc2833 for signaling digits to the gateway and inband to accept digits from outside (eg. when someone dials one of our DIDs). It's possible to do this? I've ever tried splitting 'peer' and 'user' part in

Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone "Allow Auto Answer by Call-Info: yes Dialplan: exten => 501,1,SIPAddHeader(Call-Info: answer-after=2) exten => 501,n,Page(SIP/140&SIP/110,d) exten

I've got all three CODECs the 300 supports -- G.711u, G.177A, and G.729AB -- enabled, I've changed the order, I've got them all in allow lines in my sip.conf, as follows: disallow=all allow=ulaw allow=alaw allow=G729 From "sip debug" I get the following snippets: =================================================================================== Found description format

Hello all - This is basically an updated re-posting of one I've posted a few days ago. Thanks to the kind help provided but I still can't make it work. But I'm moving a little further down the line (thanks to you folks). Basically, I've got an Asterisk server in a LAB ENVIRONMENT on my home LAN. The server has a Wildcard TDM400 installed but has no POTs lines/phones connected.

Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823

We are trying to determine how to build out an IVR system we are working on. The system needs to be able to handle probably at most 5-10 concurrent calls at peak times. Other times we are just looking for a reliable service. For incoming calls we've been using BroadVoice VOIP and before that VoicePulse VOIP. VoicePulse's IAX service started dropping DTMF inputs that we were processing

I have a sipphone.com account, with asterisk set to answer incoming calls, using the following settings (phone number and password omitted) in the Peer Details for the SIP Trunk: allow=ulaw context=from-pstn dtmfmode=rfc2833 fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx host=proxy01.sipphone.com insecure=very secret=xxxxx type=peer username=1747xxxxxxx The Asterisk machine is

Hi, is there a definite answer if asterisk can pass calls between SIP and h.323 protocols? Thanks, Yiannis.

Hello everyone, Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I am now running into a frustrating problem...when a call comes in to the BV number via a cell phone (tested with 3 different cell phones; albeit all on T-Mobile) the beginning of the IVR welcome audio is cut off. A call placed via a landline phone over the PSTN to the BV number does not exhibit the problem.

Hi list, I'm having problems transferring certain calls made by the attendant between the PSTN and to an internal extension. Although, transfers between the majority of the calls ends successfully. Debugin this, I've found that calls made to certain "numbers" (Telephony Providers), aren't detected as ANSWERED in the CDR, so they are not properly accounted (for billing),

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've just received a couple of the Grandstream GXP-2000 enterprise phones for evaluation. When a line on the phone is configured for auto answer, it connects silently. Has anyone been successful in havein a "beep" sound played to alert the user that he has an autoanswer call? Thanks Bill

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

By randomization of functions I mean the manner in whch they are called , so that has to do with address.I wish to randomize the order of calls made to functions when a program is run. Reid Spencer <reid at x10sys.com> wrote:Can you explain a little bit more about what you mean by "randomize" functions or global variables? What aspect of them do you want to randomize? Their