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Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound

I am trying to dial through my asterisk machine from phone A to phone B. My DID is registered properly with the SIP provider. When I dial from A to B it looks fine so far. A rings B and B can pick up and the call is bridged. However, I don't hear any audio so therefor it is not working. I am running Asterisk 1.8 on a cloud server. I have had the same configuration running on a physical

Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include => stdexten exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks)) exten => s,n,Wait(2)

I have just started working with Asterisk and have run into a road block concerning IAX and an inbound DID from callwithus.com. I am getting nowhere and I don't really know how to isolate the problem. The asterisk version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can connect and make a call to other internal extensions using zoiper and iax. When I try and use the number,

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

I am trying to use a simple tool in the Dial plan so that if the first number does not connect the logic will go to the second and/or third. Basically, I want the call to ring and connect to the first number Then, if it is not answered I want another number to try to get connected Then, if second number does not answer I want the third to be tried i only list the scenario for the first two

Hello list, I read on voip-info.org that Asterisk 1.4 support T38 passthrough. So I guess this means that I can have a Grandstream HT503 with T38 support and an analogue faxmachine on the other side of my Asterisk and a T38-account with a ITSP on the other side of my Asterisk machine, right ?! The fax coming from the faxmachine passes the HT503 to my Asterisk and my Asterisk sends the fax to

I have an Asterisk server it runs great with SIP phones, soft SIP phones (twinkle) and a soft SIP phone app on my Android phone but I am having problems getting two ATA boxes working. I have a Linksys PAP2T, it is unlocked and I have used them before with no problems. I was able to receive calls with from any local SIP phone or from my Link2VoIP connection via the Internet but it could not call

i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at "Updating initial configuration..." screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the

Hello list ! I'm having one way audio on incoming and outgoing calls. Outgoing audio works fine, incoming audio is not working. My setup is the following : incoming calls : PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the same) -- FXSport -- DECTphone outgoing calls : DECTphone -- FXSport -- HT503 -- WAN-port -- Asterisk -- internet (VoIPprovider) I've done a

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

What is wrong with IAX2 protocol? If IAX2 is so much better than SIP so why providers discontinuing support for IAX2 I was with provider "callwithus" but they discontinue IAX2 I switched to "checkbox.cc" but they discontinued it as well. What is wrong with IAX2? -- Joseph

On Wed, 4 Oct 2017, Jose Maria Terry Jimenez wrote: > El 4/10/17 a las 17:45, david escribi?: > >> Folks >> >> A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83.? It interfaces >> to my land-line (POTS) telephone line in the United States.? On Windows, I >> had a good answering machine package (Ventafax) that reported CallerID, >> recorded

Please DO NOT EVER CONTACT me or anyone off list if it's an answer to something on list unless specifically asked to do so. On Sat, Jun 26, 2010 at 10:43 PM, Thomas Perron <thomas.perron at gmail.com> wrote: > It kinda did not work. > > exten => s,n,Set(CALLERID(name)=label${CALLERID(name)}) > exten => s,n,Dial(SIP/callwithus/12025551212,120,A,(demo-thanks)) > >

El 4/10/17 a las 17:45, david escribi?: > Folks > > A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83.? It > interfaces to my land-line (POTS) telephone line in the United > States.? On Windows, I had a good answering machine package (Ventafax) > that reported CallerID, recorded messages, sent/received fax, and had > a scripting language that let me say "To

At 10:20 AM 10/4/2017, you wrote: >On Wed, 4 Oct 2017, Jose Maria Terry Jimenez wrote: > >>El 4/10/17 a las 17:45, david escribi?: >> >>>Folks >>>A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83.? It >>>interfaces to my land-line (POTS) telephone line in the United >>>States.? On Windows, I had a good answering machine package

On Wed, 4 Oct 2017, hw wrote: > Jose Maria Terry Jimenez wrote: >> El 4/10/17 a las 17:45, david escribi?: >> >>> Folks >>> >>> A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83. It interfaces >>> to my land-line (POTS) telephone line in the United States. On Windows, I >>> had a good answering machine package (Ventafax)

Jose Maria Terry Jimenez wrote: > El 4/10/17 a las 17:45, david escribi?: > >> Folks >> >> A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83. It interfaces to my land-line (POTS) telephone line in the United States. On Windows, I had a good answering machine package (Ventafax) that reported CallerID, recorded messages, sent/received fax, and had a scripting

Hi, I was trying to register a VoIP trunk in Asterisk , where its keep on sending Register message to the server, where I am not getting any response from server. But whereas if i register in Xlite softphone the account is getting registered. I suspect it could be network related issue, but since in softphone it is getting registered from the same network. Any ideas to isolate things would be

me at tdiehl.org wrote: > On Wed, 4 Oct 2017, hw wrote: > >> Jose Maria Terry Jimenez wrote: >>> El 4/10/17 a las 17:45, david escribi?: >>> >>>> Folks >>>> >>>> A have a PCIe modem (Conexant ChipSet, PCI id = 14f1:2f83. It interfaces to my land-line (POTS) telephone line in the United States. On Windows, I had a good answering