similar to: Russian and French sounds

Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel: >-- Executing Dial("Zap/1-1", "SIP/6137451576@6477235412||t|") in new stack >Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such host: 6477235412 >Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create >channel of type

After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a version of the Asterisk core sounds in English done by June Wallack? Some folks here prefer her voice to Allison's, but we'd like consistency. And is there a version of the Cepstral software with her voice?

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

I'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

Hello, I've been trying to setup a Fax2Email mecanism on my Asterisk box. I have been using the following: 1) An incoming IAX line on Unlimitel (Im not even sure if it's worth mentionning the provider, but I do just in case) 2) NVBackGroundDetect from Newman Telecom 3) The following extension to test: exten => fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten

Does anyone have any recommendations for a SIP/IAX provider I can use for inbound callls? The plan is to have a 1800 number people can call and reach my Asterisk. The only provider I'm familiar with is Vonage, but they don't really like the idea of Asterisk according to their terms of service and they would probably notice if there are for example 3 calls coming in at the same time.

Hi, I'm looking for a new Internet Telephony Service Provider for my company in Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in Otawa, Toronto, NY & San Jose. Anyone out ther who can help me with a recommendation? Vonage seemed clueless when I called them. Broadvoice is good but no Canadian DIDs... Thanks, Hugh -------------- next part -------------- An

hi Can anyone recommend a good DID provider offering numbers in Toronto ? ( 1 very stable 2 support porting numbers from Bell, primus, telus.. ) Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070702/cc39db9d/attachment.htm

Just out of curiosity, I''m running shorewall on a machine that has 4 nic''s and 4 different VPN tunneled subnets. When I want to define a service that is available from any source to a certain destination, instead of making a matrix of all the different combinations possible, is there an easier way? Something like,: ACCEPT any loc tcp ssh Which

Hello, I have a server to which is defined with static nat in Shorewall, and on that server, I''m running a http on a non-standard port (lets say, port 1234). I would like to use on of my free IP addresses, and map port 80 on the public side to port 1234 on the private side (forget about binding my services on a separate IP on the server, if it was feasible, I would have done that).

I've had a heck of a time getting a SIP channel to work in Asterisk 1.2.7.1 (Redhat 9.0). I've done it successfully a number of times on pre 1.2 versions of Asterisk so perhaps it's version related. Any insights are welcome! At first I wasn't able to dial out on the SIP channel _whenever_ I started Asterisk (i.e. not just when the box was booted). I always had to do a reload

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve a large assortment of issues reported by the community. For a summary of the changes in these releases, please see the release summaries:

Thanks. I know that line quality is a factor, and I know I could get a 50$ fax with a PSTN line (that is what I have now). But I have my reasons to want to setup a fax over IP, and I want to keep going. Where do I find info on this debug mode? Is there a detaild log in Asterisk that show exactly what happens when the fax is trying to come in? Also, could this console output help? - Executing

Hi, Red Hat 9.0 Asterisk 1.2.7.1 I'm having a bit of an intermittent problem with my SIP account. Often (but not always) when I start * or RELOAD my dial plan from the CLI I get this message: >Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822 add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 >Apr 30 11:01:21 WARNING[12785]: acl.c:244

Hi, This is a reminder about our next meeting. It will be held from 6pm to 8pm, February 21 at Modulis offices which are at 360 Notre Dame ouest bureau 104, H2Y1T9, Old Montreal. Thanks to Claude Patry, we will be having a 20 minute conference call with Mark Spencer. If you'd like to ask Mark a question, please send it to me by email. We are limited to 5 questions, and will do our best to

Greetings- First, my apologies for the OT post. Yes, I understand this is not the FreePBX-users mailing list. But, there are a large number of people that use FreePBX and I'm hoping they can be of assistance. I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX 2.6.0. There are a large number of inbound routes configured for the various DID's coming in via PRI, SIP,