similar to: Canadian Content: Telus and Shaw...

In the latest CAVP conference call, the membership body voted to restrict membership to VoIP LEC's and to create a seperate membership body for any other parties interested in contributing to the CAVP's efforts in CRTC lobbying and providing a unified industry presence in the Canadian telco industry with a view to VoIP services. Accordingly, I would like to propose an adjutant association

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

Hi, We ran into a little problem recently with our phone provider (Telus Canada): we are unable to dial numbers outside North America. This is what happens: the phone number 011... is sent out over our T1, Telus sees the correct number on their switch. However the switch thinks it's a North American phone number (and thus has to have 10 digits) and rejects the overseas number as

Good morning and Happy New Year 2003! Let it be prosperous for both you and me! Below you see a very short version of my CV as of January 2003 for your kind consideration and update. Word, TXT, and PDF version of the complete CV are available upon request. Enjoy the economic recovery! Make it a great day: ZAR ================================================================= ZAR PETKOV,

We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's something specific that I need to do to make it work, since many PBXs can do this. The switch is a

Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks!

Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign

I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David >>> asterisk-users@eol.ca 1/21/2004 6:39:34 AM >>> I'm not sure Primus

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the "great white north" (sources, services, telcos, etc.), I created the asterisk-canada mailing list: http://lists.syonex.com/mailman/listinfo/asterisk-canada or asterisk-canada-subscribe@lists.syonex.com Cheers! John

If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP.

I would like to participate in Xen Debian package development. I am not sure if this email should go to this list or somewhere else. According to the wiki this is the development list for Xen Debian package development. I have just subscribed to the list and have not yet been authorized as a list member so any response should be sent to thilts33@telus.net. My name is Ted Hilts and I am

Hi, Does anyone has configured Asterisk to connect to BabyTel (a SIP Provider in Canada) ? Here is my sip.conf (I'm behind a firewall and I already opened port 5060 and 5065 (udp and tcp) to my Asterisk server): [general] port = 5065 context = Test insecure = very register => 1514XXXXXXX:password@sip.babytel.ca When starting Asterisk, the sip registration failed after 5 connecting

I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system. What I'd like to do is from the CLI "ping"

Turn on sip debugging for this peer and watch for the options sending and response. If you are getting a response to your options asterisk shouldn't be marking the peer as unavailable. is your asterisk behind a firewall? On 20 March 2015 at 13:42, thufir <hawat.thufir at gmail.com> wrote: > I wasn't able to get much out of babytel, beyond the fact that I was, > apparently,

On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote: > This is just screaming "configuration mismatch" -- or, possibly, "latent bug > whereby things parsed in separate places should be treated the same, but are > actually getting treated differently". I really don't want to be the "my system isn't working so there must be a bug in Asterisk" guy

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

Hello ! I am running samba-2.2.0a-0 on Suse 7.3 and attempting to use smbfs to mount a share on a dos 6.x system running, pctcp + nebios + powerlan When I attempt to mount the share ie in fstab //ARNLSPEC/E /wheelspec/ARNL smbfs noauto,debug=10,ip=192.168.80.101,workgroup=WORKGROUP 0 0 ie mount /wheelspec/ARNL I receive a : opts: ip=192.168.80.101 opts: workgroup=WORKGROUP passthrough

It looks as if that is more of a question/issue with your router, rather than Asterisk. I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut" You SHOULD be able to communicate between devices on the

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active