similar to: Congestion outbound only with ATA boxes

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

Hi All, I am using Asterisk for one of my projects in OpenBTS. I am having the age old problem of "extension not found" when try to make a call from one registered SIP phone to other registered SIP phone (two mobile phones connected to Asterisk via OpenBTS). The exact error thrown on Asterisk CLI is *"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to

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

Incoming calls ring SIP users who have |Ttr in their dial plan, but outgoing calls are done through a macro as follows: [macro-diallink2voip] exten => s,1,Dial(SIP/${ARG1}@link2voip-sw2,120) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-ANSWER,1,Hangup exten => s-CONGESTION,1,Dial(SIP/${ARG1}@link2voip-sw1,120) exten => s-CONGESTION,2,Goto(ss-${DIALSTATUS},1) exten =>

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

the legs separately as if they were not related to the same call. So the ingress leg negotiates ulaw, and despite it knowing that the peer also supports g729 fails the call since it's already decided on ulaw and the egress leg only accepts g729. If this is design intent I'm wondering if there is demand enough to justify a feature request? Any advice on how I can work around this issue?

Hello, I'm having trouble getting finding a company that provides DID in Western Canada. More specifically in Edmonton, Alberta. We have tried getting in contact with Link2Voip and Calgary Telecom but neither seems to be answering their phones or email. I would appreciate it if anyone can point me in the right direction. Thank you, Nelson

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support

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

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

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'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 a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

I have a pair of Asterisk servers which are happily routeing VoIP calls. I want to hook one of them to the PSTN. Given that I am in the UK, what is a reasonably easily-available device to provide an FXO interface from a Linux box, with a minimum of faffing around with drivers? Just one line is needed, though in theory two might eventually be useful. My usual white-box hardware suppliers don't