search for: link2voip

...16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type : little Reverse Nibbles: no Reverse Bits : no Comment : 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rvvvvv): == Using SIP RTP CoS mark 5 -- Executing [xxxxxxx at phones:1] Goto("SIP/ATA-xxxxxxxxxx-L1-024b6d8...

...SI404864430002302) ; check for local extensions first include => sip-local =============================== *sip.conf* ============================== [general] ; Comment these out if no backhaul is available. ; Use the pair with the shortest latency. ;register => kestrel0:v01ptest at sip.ca1.link2voip.com:5060 ;register => kestrel0:v01ptest at sip.ca2.link2voip.com:5060 ;register => kestrel0:v01ptest at sip.us1.link2voip.com:5060 ;register => kestrel0:v01ptest at sip.us2.link2voip.com:5060 ;register => kestrel0:v01ptest at sip.nl1.link2voip.com:5060 ;register => kestrel0:v01ptest...

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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around h...

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 => s-CANCEL,1,Hangup exten => s-BUSY,1,Busy(30) e...

...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:${LINE})}) [pbx_config] 104. Macro(dial${LINE}|${EXTEN}) [pbx_config] 105. Hangup() [pbx_config] Include => 'toll-free-override' [pbx_config] and "toll-free-overri...

...t 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 out. It could not call out to the Link2VoIP or any of the SIP phones. I spent a lot of time going over the configureation for this Asterisk server and the settings in the Linksys PAP2T box but could not get it to work. I removed the Linksys PAP2T an...

...ff. This happens regardless of encoding of the file (ulaw/gsm) and regardless of the incoming codec. However when using Echo() both tones & voice are flawlessly echoed back to me, as are the Packet2Packet bridging calls connected to remote phones. I tested this issue with 3 other providers (Link2VoIP/Babytel/Junction Networks) and I'm not experiencing this issue with them, despite having identical peer configurations across for all 4. Though with Teliax I'm using SIP, I did try to use IAX2 for the heck of it and the same problem seems to exists, so it's not specific to SIP. Add...

Does anybody know anyone who offers encrypted IAX termination at reasonable rates? I googled, searched the WIKI, but didn't find a whole lot of information. Thanks, David

Hi, Anyone know any decent trunk provider in Canada that can offer multiple channel trunks (16channels) via IAX or Sip trunking? Having some pleasant experience with IAXTEL out of Denver, though they don't offer services into Canada. Please let me know S.

...ittle more complex but because this didn't even work, there's no point in posting. Does anyone have any idea why this works fine when I was using 1.2 but suddenly with 1.4.18 it isn't? This is on a server connected directly to the internet, no NAT. Nothing else has changed on it, and Link2Voip (SIP) and Vittelity (IAX) works flawlessly. Any help would be GREATLY appreciated. Thanks in advance!

I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple