RE: [Asterisk-Users] fax with asterisk
Some of the issues I have had with fax with IP Centrex (my team was responsible for an IP Centrex product at a CLEC) that sounds like you are having, had to do with a number of factors: - Network: latency, jitter, packetloss... fax is intolerant - Gateways and IADs: echo cancellers incorrectly set and NSE incorrectly set, attenuation on the PSTN side incorrectly set - Fax type: G3 or superG3, the echo-can settings and NSE settings are different for each since they are completely different ;-) Some of the symptoms were: - complete inability to send/receive faxes - partial faxes (x pages received/sent out of x+y total) I have never heard of a noise component though... which would account for the scrambling you mentioned. The RTP wouldn't be modified in transit or the packets checksum would barf. Its sounds like it maybe getting scrambled by a terminal which in * case could be the transmit/receive fax machines or * itself. I hate fax over VoIP. E-fax anyone? :-) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Justin Richards Sent: Thursday, February 17, 2005 9:08 AM To: Keith Burns Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] fax with asterisk I'm not using any Digium cards. I'm actually using SpanDSP and app_rxfax to process incoming faxes. After drilling into it for about 8 hours yesterday I come to realize that there is a lot more to it than the asterisk upgrade. I patched my FC3 box, which means libtiff is now 3.6.1 which according to Steven (spandsp) is broken for this application. First, i compiled the latest spandsp which didn't make a difference. I re-compiled the rxfax and txfax apps, got nowhere. Finally started trying to revert to an older libtiff, but apparently went too far back (3.5.7) and ended up getting core dumps. I am need to downgrade libtiff to 3.6.0, and everything dependant on libtiff to try again. simply downgrading libtiff on its own doesn't work well.. :-( Thats what I get for patching.. :-) This setup worked nearly flawless until I upgraded, so I'm pretty sure I can get it back again after i downgrade the right stuff in the right order. If anyone has that list, please share!! On Wed, 16 Feb 2005 19:45:51 -0700, Keith Burns [EMAIL PROTECTED] wrote: Are you both using Digium cards? Do you know if you are using G3 (standard) or SuperG3 (like a modem) fax machines? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Justin Richards Sent: Wednesday, February 16, 2005 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] fax with asterisk I'm getting a lot of this too :-( my fax stuff worked great under 1.0 but after upgrading to 1.0.5 i've been broken.. Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 496 (got 912, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 497 (x 470). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 497 (got 470, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 498 (got 3195, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 500 (got 2471, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 501 (x 1595). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 501 (got 1595, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 503 (got 2583, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 504 (got 1877, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 506 (x 22). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 506 (got 22, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 507 (got 1818, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 509 (got 1729, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 510 (got 1738, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 511 (got 2228, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 514 (got 1824, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 515 (got 2466, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 516 (got 1730, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 517 (got 2534, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 518 (got 1949, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 519 (got 1830, expected 1728
RE: [Asterisk-Users] Sipura g729 call quality to PSTN
The only problem is that it is bandwidth inefficient and may cause a CPU hit on your IAD (since you have effectively doubled the pps for a call). The packets should be 10ms apart. Perhaps the timestamp is not in ms. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Thursday, February 17, 2005 5:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN That does not sound right at all. The difference between the two Time= values should have been 10 (milliseconds). Did you reboot the Sipura after making the change? There are some values in the Sipura that don't take effect until after the next reboot; I don't have a clue whether this happens to be one of them. Yes - sipura was rebooted. Actually, the changes did seem to take affect even before the reboot (verified by call quality improvement and ethereal traces). So in your opinion, instead of 80, it should be a difference of 10? If so - then you are saying that the timestamp is in miliseconds? I am as puzzled as you - really does not seem logical, but call quality is finally decent and it does not seem to bother asterisk at all. Do you see any potential problems with this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER/Asterisk consultants in Denver
Hi, I am looking for SER/Asterisk consultants in Denver, please contact me at [EMAIL PROTECTED] attachment: winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura g729 call quality to PSTN
Title: RE: [Asterisk-Users] Sipura g729 call quality to PSTN Next thing I would check are the de-jitter buffers if possible on the Sipura, or jitter in general. Do you have control of the PSTN gateway ? Measure the jitter on ingress to the gateway. You can do this crudely by using Ethereal and looking at the delta between timestamps on RTP packets from Sipura to PSTN. -Original Message- From: Pedro [mailto:[EMAIL PROTECTED]] Sent: Wednesday, February 16, 2005 1:37 PM To: Keith Burns Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Jeffrey Chan Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality slightly, but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns [EMAIL PROTECTED] wrote: What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura g729 call quality to PSTN
Hmmm, that worked? Interesting that you can change the sample size to 10ms since the standard is 20ms that most people don't go below. I know you *can* do below 20 but if you are doubt the technical ability of the box it seems strange they are capable of that. This seems to smack of bad de-jitter buffers on the egress gateway... are you receiving 20ms sampled RTP ? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Wednesday, February 16, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN FYI - Seems the latest firmware in conjunction with changing the packet size to 10ms improved the call quality to usable. The Cisco 7960 is stell superior, but now at least the SPA-2100 is acceptable (and with 2 working g729 channels including 3-way calling). On Wed, 16 Feb 2005 15:44:58 -0500, Pedro [EMAIL PROTECTED] wrote: Forgot to mention that when I set the RTP Packet Size to 20ms that the difference was 160 (like the Cisco) but call quality was much worse. On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote: Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality slightly, but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns [EMAIL PROTECTED] wrote: What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk
RE: [Asterisk-Users] fax with asterisk
Are you both using Digium cards? Do you know if you are using G3 (standard) or SuperG3 (like a modem) fax machines? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Justin Richards Sent: Wednesday, February 16, 2005 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] fax with asterisk I'm getting a lot of this too :-( my fax stuff worked great under 1.0 but after upgrading to 1.0.5 i've been broken.. Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 496 (got 912, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 497 (x 470). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 497 (got 470, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 498 (got 3195, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 500 (got 2471, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 501 (x 1595). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 501 (got 1595, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 503 (got 2583, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 504 (got 1877, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 506 (x 22). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 506 (got 22, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 507 (got 1818, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 509 (got 1729, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 510 (got 1738, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 511 (got 2228, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 514 (got 1824, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 515 (got 2466, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 516 (got 1730, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 517 (got 2534, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 518 (got 1949, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 519 (got 1830, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 521 (got 3401, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 522 (x 775). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 522 (got 775, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 523 (got 2413, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 524 (got 2279, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 526 (got 2015, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 530 (got 2677, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 531 (x 1220). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 531 (got 1220, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 532 (got 1729, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 534 (x 0). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 534 (got 0, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 536 (got 2454, expected 1728). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 537 (got 0, expected 1728). Page 4 of /var/spool/asterisk/fax//1108596104.3.tif: 538 rows received 0 total bad rows 0 max consecutive bad rows ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura g729 call quality to PSTN
What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk performance monitoring
Title: Asterisk performance monitoring Hello, Has anyone used any 3rd party web based software to get performance information out of Asterisk? Looking for CPS, call setup times, voicemail database utilization etc Cheers Keith. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!
I think most people have spent more time complaining about the AUTO-RESPONDERS than it takes to hit the delete key. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Sunday, February 06, 2005 4:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!! They can always check the archives to read up on missed posts, and it would save us all the trouble in the mean time ;-) Isn't it obvious that with a choice of hundreds of free email providers, anyone who wants to avois this problem need only use a throwaway account like gmail with mailing lists and avoid checking the away message stuff. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-lite to Cisco ATA - no RTP
Title: X-lite to Cisco ATA - no RTP Hi there, I have X-lite and a Cisco ATA on the same hub (i.e. no NAT, no ACLs) as my Asterisk box. Ethereal shows normal SIP signaling when I call from X-lite to the ATA. Ethereal also shows RTP is passed from X-lite to Asterisk, and RTP is passed from the ATA to Asterisk, but no RTP from Asterisk to either device. (Note that Ethereal does show the SIP signaling packets originating from Asterisk, so nothing funky with my Ethereal filter either) Has anyone run into anything similar? Any pointers? I set up all the extensions using AMP. If you need specific configs, I am happy to provide. Cheers Keith. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk user with a goal
Hi Ryan, Assuming you have looked at the WIKI at www.voip-info.org, there is some good info there. Not sure of your background, mine is mainly VoIP, telephony and networking, and definitely not strong on Linux, so if you would like to email me directly offline ([EMAIL PROTECTED]) with some of your issues, as long as they are VoIP and IP, I may be able to help. On the Linux stuff, I can only share with you my experience which is limited at best :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ryan Coates Sent: Friday, February 04, 2005 2:09 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] New Asterisk user with a goal Hi All, I am rather new to the asterisk world, and new to VoIP in general, my question seems rather simple compared to some of the topics under discussion here :) I have done quite a bit of reading and fiddling trying to get a system set up, to no avail yet basically myself and a friend (both behind NAT and Firewalls, but able to set up the firewall rules/port mappings ourself) are interesting in setting up a PBX each, initially just for IP based calls over the net, so that we can call each other for free eventually we will look at plugging this into the POTS system and repacing all our phones with IP phones and rouitng the calls appropriately depending on destination at present I am trying to test with some software phones and an asterisk box on a virtual network (no nat, no firewalls) to see if we can get anything working (client 1 calling client 2), before we splash out a fair ammount on some decent IP Phones, but am not having much joy if anyone could give me some help/advice on the matter I would be greatful if you need any more details please do not hesitate to ask, ill try to answer whatever I can -- Regards, Ryan Phoenix Coates [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-lite to Cisco ATA - no RTP
Title: X-lite to Cisco ATA - no RTP Interesting, SUSE firewall allows SIP but not RTP out of the box. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Keith Burns Sent: Friday, February 04, 2005 8:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] X-lite to Cisco ATA - no RTP Hi there, I have X-lite and a Cisco ATA on the same hub (i.e. no NAT, no ACLs) as my Asterisk box. Ethereal shows normal SIP signaling when I call from X-lite to the ATA. Ethereal also shows RTP is passed from X-lite to Asterisk, and RTP is passed from the ATA to Asterisk, but no RTP from Asterisk to either device. (Note that Ethereal does show the SIP signaling packets originating from Asterisk, so nothing funky with my Ethereal filter either) Has anyone run into anything similar? Any pointers? I set up all the extensions using AMP. If you need specific configs, I am happy to provide. Cheers Keith. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP with SUSE9.2 (Apache2)
Title: AMP with SUSE9.2 (Apache2) Hi all, After pinging the AMP userlist at SourceForge, I got a great step by step explanation as to how to set up AMP for Apache2 (some maybe obvious stuff that wasnt in the Newbie Guide). Thanks to Jason Becker of Coalescent Systems. If anyone needs me to post Jasons instructions here, I can, but they are in a thread called AMP noob issues with Apache2/Suse9.2 at SourceForge. Again my thanks Jason, looking forward to using your software. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP with SUSE 9.2
Title: AMP with SUSE 9.2 Hi, I have the newbie guide from AMPs website and (fair enough) it is all about whitebox linux. Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ? Any help appreciated. Cheers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AMP with SUSE 9.2
Cool, will do, thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Becker Sent: Tuesday, January 25, 2005 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP with SUSE 9.2 Keith Burns wrote: *Hi,* *I have the newbie guide from AMP**'**s website and (fair enough) it is all about whitebox linux.** Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ?* Please post to the amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or Help forum: http://sourceforge.net/forum/?group_id=121515 SUSE does some things differently - the main difference is the apache2 (httpd) configuration. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AMP with SUSE 9.2
Ok, I signed up a few hours ago for the AMP mailing list, and no confirmation. If anyone on this list has installed AMP with SUSE 9.2, if you wouldn't mind emailing me with any gotchas at [EMAIL PROTECTED] I sure would appreciate it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Keith Burns Sent: Tuesday, January 25, 2005 9:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] AMP with SUSE 9.2 Cool, will do, thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Becker Sent: Tuesday, January 25, 2005 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP with SUSE 9.2 Keith Burns wrote: *Hi,* *I have the newbie guide from AMP**'**s website and (fair enough) it is all about whitebox linux.** Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ?* Please post to the amportal mailing list: http://lists.sourceforge.net/lists/listinfo/amportal-users or Help forum: http://sourceforge.net/forum/?group_id=121515 SUSE does some things differently - the main difference is the apache2 (httpd) configuration. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AMP with SUSE 9.2
Yeah, I got the AMP part working but in the process messed up my * install WRT to the ZAP stuff and /dev (I *think* made some changes to the ZAP Makefile to support SUSE 9.2 and udev last time I installed, but didn't make those changes this time - I am a complete Linux noob). It was interesting trying to get the described Apache changes done when you are using Apache2 and have never touched Apache before... :-) Fun and games... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Charles D'Englere Sent: Tuesday, January 25, 2005 4:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP with SUSE 9.2 One thing for sure it was a rela headache for me... I finaly did get it working... Don't for get to follow the installation guide to a t... Charles On Tue, 25 Jan 2005, Keith Burns wrote: Hi, I have the newbie guide from AMP's website and (fair enough) it is all about whitebox linux. Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ? Any help appreciated. Cheers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Network Test Tool?
Smartbits ? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Me Sent: Monday, January 24, 2005 12:33 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Network Test Tool? We have been having WAY too many issues lately with our VOIP calls. I suspect it may be the particular T1 we are pushing these calls out through from our office. Is there a decent tool out there that I can stick on the network that will measure things like Jitter, ping times and overall network quality for say a 24 hour period and stick it in a human readable report. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Damn DTMF Beeps on my calls
It could be a number of things... if you are running RFC2833 then some sounds on the line (high-pitched voice?) can be interpreted as DTMF. Also beware of some gateways and using SUB/NOTIFY, had a couple of instances when the DTMF was supposed to be stripped from the in-band (RTP) but wasn't... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Me Sent: Monday, January 24, 2005 12:39 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Damn DTMF Beeps on my calls Can someone give me a clue as to why I keep hearing DTMF type beeps on my phone calls. It sounds exactly like someone on the other end is pushing a key on their phone but they are not! Has anyone ever heard of this before? It use to happen once in a while, today it's been happening a LOT and it's driving me batty.. -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 EM vs PRI question
Depending on the switch they are using, there are a limited number of D-channels (or D-channel licenses). CAS signaling needs RBS (its the winking in this case). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Beebe Sent: Monday, January 24, 2005 2:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 EM vs PRI question Ok, I'm about to take the plunge, and am trying to decide between Channelized T1 EM and PRI. I'm getting an Integrated T1 which will have data and voice capability, all plugged directly into my digium single T1 card. In either case the data piece looks pretty straighforward, just setup the channel properly, hand it off to the linux hdlc layer, and route away the voice side seems a little more complex -- I'm looking for clarification and/or advice: It seems to me that the major differences between the two different voice delivery mechanisms (other than cost)is caller id functionality and call setup delay. With the PRI, I'll have practically instant call setup and the ability to pass CNAM (caller name) and CID (caller ID) informationinBOTH directions. The PRI willgive me the ability to have additional directory numbers (typically called DIDs) assigned against myvoice trunks and will provide the full ANI (automatic numberidentification) and DNIS (dialed number identificaton service) over the PRI signalling trunk. Eachvoice channel will also be 64k clear channel, so I could (theoretically) provide 56k dial-in modem service from the same box (anyone actually doing this?? seems like a neat application for the dsp software guys) I also lose one 64k channel to signalling. Sounds like the way to go, but basically the PRI ends up being$100/month more expensive than the Channelized T1 EM. The T1 EM approach will still give me CID (but not CNAM???) over the in-band call setup mechanism (ie: quick DTMF tones during the wink). Each voice channel will actually be 56k because it uses RBS (robbed bit signalling -- not sure what its using this for, as the call setup is delivered via wink???). As a result, this approach would also keep me from implementing a 56k dial-in modem service, but I could still use an ordinarymodem or fax dsp to provide 33.6k dial-in. Thissetup can support DID, but its appended (or prepended, depending on the provider) to the DTMF call setup (which extends the time for calls toactually connect).Not sure if CID or CNAM can be provided foroutgoing calls(I think some providers canenable me to be able to wink to them the numberto pass as caller id??) I believe in either case, the normal call features (3-way, forwarding, etc) can be provisioned. Do I have it about right?? Is it pretty normal for providers to charge a premium for the PRI? Any thoughts/clarifications to my above assumptions?? Are there other pros/cons of each setup? Thanks in advance! -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-T Support (I got my head in an SS7 cloud)
I think of *, Broadworks, Vocaldata, Sylantro as line side feature servers, and SS7 signaling with say IMTs/PRIs more for the class5 network side soft-switch (NexVerse, SONUS etc). Typically they handle the LERG, complex translations etc and do it quite well (although typically they take in native A-links for SS7 or some degree of the SS7-o-IP standards). I'm not sure I would want a line side feature server trying to be all things to all people... kinda gets like Cisco IOS Enterprise :-o -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, January 24, 2005 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP-T Support (I got my head in an SS7 cloud) Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am I missing here.. ?? Hmm, but outbound calls would be more complicated I think.. Let see, SIP user dials a number, we'll eventually place a dial out on the MGCP line, but we need to first send a few SIP-T messages to find out where to put it.. Just swiming around in it here.. Any thoughts? It seems to me that you MUST use something like MGCP or H.248 to connect the call to the PSTN (media gateway) since the specific DS0 to be utilized will be included in the ISUP messages.. -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-T Support (I got my head in an SS7 cloud)
Yep, still lineside... you can do it with SIP too. If it was going to do MGCP, it only makes sense if it does it properly and IS aware of the channels on the other side of the gateway (multi-chassis trunk failover etc) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Monday, January 24, 2005 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP-T Support (I got my head in an SS7 cloud) [EMAIL PROTECTED] wrote: Just swiming around in it here.. Any thoughts? It seems to me that you MUST use something like MGCP or H.248 to connect the call to the PSTN (media gateway) since the specific DS0 to be utilized will be included in the ISUP messages.. No, you can just do what you are doing now, and use SIP to talk to your gateway. The SIP user (Asterisk) has no concept of how many channels exist on the TDM side, or their arrangement, or anything like that. If Asterisk could be an MGCP gateway controller (whatever the right term for that is) it's possible that it could control MGCP gateways directly, but it would still need to speak some sort of signaling with the PSTN to setup/teardown the calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 EM vs PRI question
Correct, CAS can supply DNIS but the call set up times are significantly longer. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of David Boyd Sent: Monday, January 24, 2005 7:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] T1 EM vs PRI question Responses embedded below! On Mon, 2005-01-24 at 18:49, Keith Burns wrote: Depending on the switch they are using, there are a limited number of D-channels (or D-channel licenses). CAS signaling needs RBS (it's the winking in this case). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Beebe Sent: Monday, January 24, 2005 2:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 EM vs PRI question Ok, I'm about to take the plunge, and am trying to decide between Channelized T1 EM and PRI. I'm getting an Integrated T1 which will have data and voice capability, all plugged directly into my digium single T1 card. In either case the data piece looks pretty straighforward, just setup the channel properly, hand it off to the linux hdlc layer, and route away the voice side seems a little more complex -- I'm looking for clarification and/or advice: PLease no Flame, just a correction if required. There seemed to be issue using Data/Voice on the digium cards, but I believe it is a setup issue not a technical limitation on the card itself. It seems to me that the major differences between the two different voice delivery mechanisms (other than cost) is caller id functionality and call setup delay. With the PRI, I'll have practically instant call setup and the ability to pass CNAM (caller name) and CID (caller ID) information in BOTH directions. The PRI will give me the ability to have additional directory numbers (typically called DIDs) assigned against my voice trunks and will provide the full ANI (automatic number identification) and DNIS (dialed number identificaton service) over the PRI signalling trunk. Each voice channel will also be 64k clear channel, so I could (theoretically) provide 56k dial-in modem service from the same box (anyone actually doing this?? seems like a neat application for the dsp software guys) I also lose one 64k channel to signalling. Actually DNIS can be provisioned over em trunking also, the separation of digits is done with *'s or KP/ST. So the digiti dump would be something like: DTMF OH - - Wink digit dump *703727131229*8004231212*- -wink -Answer The breakdown of the digits is ani + Info digits then DNIS The *'s would be replaced with KP/ST pulses if MF. KP start sequence, ST stop sequence. Sorry for the crude drawing, and the disclaimer is its been 4 years since I have looked at the digit sequence for an EM t1 :) Sounds like the way to go, but basically the PRI ends up being $100/month more expensive than the Channelized T1 EM. The T1 EM approach will still give me CID (but not CNAM???) over the in-band call setup mechanism (ie: quick DTMF tones during the wink). Each voice channel will actually be 56k because it uses RBS (robbed bit signalling -- not sure what its using this for, as the call setup is delivered via wink???). As a result, this approach would also keep me from implementing a 56k dial-in modem service, but I could still use an ordinary modem or fax dsp to provide 33.6k dial-in. This setup can support DID, but its appended (or prepended, depending on the provider) to the DTMF call setup (which extends the time for calls to actually connect). Not sure if CID or CNAM can be provided for outgoing calls (I think some providers can enable me to be able to wink to them the number to pass as caller id??) I don't know of a way for outbound or inbound CNAM to be provided on a T1 unless you are using SS7 or some like control protocol. The setup time is in milliseconds for PRI and potentially could be 1.2 seconds in EM including wink times, and outpulse dump. This can be decreased if the carrier can accept fast outpulse, and also be decreased if you use MF with KP ST pulses instead of DTMF. Robbed bit allows for the current channel condition to be maintained in the signalling stream. When a channel hangs up the onhook condition has to be able to be passed to the other end of the t1 for disconnect. The wink and digits dump at the start of the call only provides call setup capability. I believe in either case, the normal call features (3-way, forwarding, etc) can be provisioned. Additional features are usually handled within the switching/* system once the call has been setup. There are some
RE: [Asterisk-Users] Zapata in Australia
Yep, I could buy it in Australia, install it in a * box, and deploy it in, Fiji for instance (if legal there) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dave Green Sent: Monday, January 24, 2005 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zapata in Australia Howard Lowndes wrote: On Tue, 2005-01-25 at 03:23, Andrew Yager wrote: As a general rule, the X100P should not be used in Australia as it is set to an incorrect impedence and can't be changed. The TDM series of cards with FXO/FXS modules can be set to work in AU. ... You should also be aware that the PSTN connect cards do not have Austel approval as yet, and so they shouldn't be connected the the public phone network. Another example of a situation where the sale and use of an article in Australia by an Australian business is legal, but the use of the article in Australia can be illegal. How do you spell telco cartel? Sorry, I don't follow .. many things can be sold/purchased legally but put to a use that is not legal. Dave ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco IP Phones
I think you need to look at a few other factors. 1. Some IP phones are really flakey (had some serious issues with a couple of vendors MGCP Business line package). 2. Line power - Cisco uses one standard, other phones use another... but Cisco is the 900# gorilla in the powered switch market... your call... 3. Feature sets. Cisco puts a lot into their SCCP image... cos... well, its their (ok, Selsius') standard, but not a great deal into their SIP image (can anyone say 7914 ?) 4. Perception. Yep, it matters... want to put a Freedom Fries phone on a customer's desktop when they have all Cisco switches and routers... if they are so technically obtuse they need someone to put a telephony system in for them, they will probably believe the hype and want Cisco. Anyway, my 2c (and given the value of the Euro vs the USD, I guess my opinion ain't worth that much) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Glenn Powers Sent: Friday, January 21, 2005 5:25 PM To: Mike Dent; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco IP Phones Mike Dent wrote: Hi Glenn, What do you mean by provisioning? loading the config files, with proxy servers, usernames, passwords, etc. cheers, glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Becoming a VOIP provider
Be careful of LI requirements in Australia. You MAY be able to put the onus for this on your upstream (PRI/IMT) provider, but if you have many, this could be messy. Best bet would be to have a solution yourself... when I was looking into this the good news was that the enforcement agencies (which at last count was around 47, any of whom could hit you for their own real-time feed of the conversation) were considering taking the VoIP feed (RTP) and the logs of the signaling. (Things may have changed, your mileage may vary, yada, yada, yada). Also, after a little kiddy died of an asthma attack in rural Victoria because Telstra (the lazy @[EMAIL PROTECTED] - I digress) hadn't fixed their phone, lifeline services (E911 in the US) are more and more important to have nailed.. you don't want that on your conscience (your service not working causing harm to someone) nor would your business appreciate the lawsuits. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ed Robbins Sent: Wednesday, January 19, 2005 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Becoming a VOIP provider Ty Carter wrote: Ed: I think you must have some bad information here.VoIP is an Information service and not subject to CALEA regulations. Whether it's a subject to those regulations or not I still know first hand it's a big issue with broadband voip providers. I work for a company that develops VoiP for the broadband market and it's something we had to develop for our customers. I don't know all the details of it and what is going on behind the scenes in terms of regulations but my thinking is that voip providers have to tie into the PSTN somewhere and the FCC can most likely tap into(no pun intended), meaning require you meet the guidlines put forth in CALEA, from that legal point of view. I had never thought about this before but I should talk to my buddy who got a CLEC a few years ago, I'm wondering if there is something in there that spells it out. Ed According to the calea website: In a Notice of Proposed Rulemaking FCC 02-42 released on February 15, 2002, the FCC initiated a proceeding to establish rules and regulations regarding the classification of wireline broadband Internet access under the Telecommunications Act. Digital Subscriber Line (DSL) service is an example of wireline broadband Internet access. In this document, the FCC tentatively decided that wireline broadband Internet access is an information service. In a Declaratory Ruling and Notice of Proposed Rulemaking FCC 02-77 released on March 15, 2002, the FCC made a declaratory ruling that cable modem service (Internet access through cable TV lines) is an information service under the Telecommunication Act and initiated a proceeding to establish rules and regulations based on that finding. Therefore, the FCC's pending wireline broadband Internet access proceeding is CC Docket Nos. 02-33, 95-20, and 98-10 and the cable modem broadband Internet access proceeding is CS Docket No. 02-52 (collectively the FCC Broadband Proceedings). It should be noted that the FCC is not primarily focusing on CALEA in these proceedings, rather its emphasis is on the economic and policy concerns involved in regulation of these services under the Communications Act. However, since CALEA exempts information service from the surveillance capability requirements of Section 103, these FCC decisions have the potential to exclude broadband DSL and cable modem service from CALEA compliance. The FBI filed the following comments in the Broadband -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Robbins Sent: Wednesday, January 19, 2005 3:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Becoming a VOIP provider Manjit Riat wrote: That was a really nice description... Can you do 1-14 and I'll do 15 and 16?? Just kiddin. -Original Message- From: Ty Carter [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 19, 2005 10:58 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Becoming a VOIP provider 1. You must have some type of business model / plan 2. Be well capitalized, starting out is going to be a cash draining experience. 3. Have access to (U.S.) PRI or Channelized T1 and High speed Internet connection 4. For U.S. it always helps on the bottom line if you're a CLEC 5. Have a test server, if you want to play in the enterprise market, buy a test 1U server and a 1 T1 PRI card 6. Forumlate your POPS 7. Get a ANCP Code from Telcordia, then apply for a CIC, Part A code (commly reffered to as a PIC code (10-10-987) 8. Arrange for a LD carrier, preferabably one that can
RE: [Asterisk-Users] E911 Testing !
What do you want to test? Call routing under certain failure scenarios or CAMA trunking? We tested 911 to a PRI not connected to the PSTN that terminates on another gateway (back to back PRI) and make a dedicated handset ring using a dedicated pass through dial-peer. That way you can do the Q931 debugging on the far end gateway to make sure you have all the right ISDN signaling in place (assuming you are using ISDN, which makes sense if you are an office PBX) CAMA trunking will require. CAMA trunks As for 911 design there are a number of ways of doing this depending on the hazard/failure you are trying to protect yourself from. You could go as far as dedicated 911 IADs using, for instance, Ciscos SRST and if you are using IP Phones, set up the SRST gateway as the secondary call manager etc etc etc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Manjit Riat Sent: Wednesday, January 19, 2005 2:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] E911 Testing ! I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Wont they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden bringing the whole office PBX down. Since it would be not be possible to place a call and emergency matter becomes more serious, who would be held responsible? The person who installed the PBX for not implementing a redundant and reliable system? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E911 Testing !
Oh well... at least no one here thought E911 was 911 for IM or email (yes... someone once asked me that) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Wednesday, January 19, 2005 3:43 PM To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] E911 Testing ! 911 Testing is a very complicated issue. For a clec it typically involves scheduling with them so they will expect your call. Also we frequently use false addresses (that are MSAG resolvable) and some very sophisticated PSAPs even have fake addresses that MSAG resolve to a testing ESN. Translated in english: 1. I put in a special address mapped to a phone number into the 911 location database. This is in the ALI database. The primary source of data that the 911 centers map phone number to address. 2. MSAG (The master street address guide) maps actual street addresses to ESNs an ESN is an Emergency Service Number (or something like that, feel free to correct me). It is basically a specific collection of Police, Fire and EMS. For example, Your house might use Police A, Fire B and EMS B, but the people on the other side of the street might use Police C, Fire B, EMS B (maybe it's jurisdictionally a different town). The PSAPs make up a fake address like 1234 Network Testing Blvd and they make it resolve to ESN 555 which will route to a testing center (joe) who only recieves test calls. Ok.. so too much information.. right? Definitely. Unless you happen to be doing a CLEC's office, none of it has any bearing on the original question. :-) here's the short answer. Please don't call 911 unless you have an emergency. False. Local policies vary widely. Our 911 service here in Milwaukee is the preferred method for reporting debris on the freeway to the Sheriff's Department, for example - a dispatcher once scolded me for *not* calling 911, though admittedly this was only a few years after a truck dropped some debris on I-94 that ultimately punctured the gas tank of a minivan containing a large family and lots of people died, so people have been more sensitive to debris on the highway. In fact, around here, it's fairly common for installers to test 911 service, because there's a danger in 911 *not* working as advertised under ordinary conditions (someone forgot this or that, not too hard on a PRI). Find out who your local PSAP is and call the administative number for it and talk to them. Sometimes it is hard to find this number, but it's out there. Look for Emergency services in ACME town or ACME Town 911 Dispatch etc,etc. Some very small towns actually have their administrative lines forward to the 911 centers for those areas. Call the police department's non-emergency number and they can help track down who to contact, if all else fails. Also be aware that if you are a carrier, you are required by law to have a signed contract with the 911 agency. This is typically so they can collect on the federally mandated 911 end user line fees. Most offices aren't phone carriers. Even most offices for carriers won't have an installer putting in phones that knows anything about some contract locked up half a dozen states away in the Legal Department vault at LEC Headquarters. So that's not too useful to the guy who just wants to verify correct operation of 911 services for an office install. The short form: *ASK* your local 911 center what they prefer you to do. In general, they *want* 911 to work right, and there will be some way to get you what you need. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to change the packet size
Just beware of the effects of changing sample size for any codec. We found that a sample size of 2 for G.711 (ie 2x20ms) allowed for pretty robust interoperability between vendors. Not specifically with Asterisk, but we did find that using a mixed CPE/gw environment with a couple of Call Agent vendors that Smartbits PSQM scores varied wildly with changed sample sizes but 2 samples yielded pretty consistent multi-vendor results. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Luki Sent: Wednesday, January 19, 2005 6:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to change the packet size Although this probably isn't the right way of doing it, you can change in the source code, globally for all calls using a codec: See the smooter creation statement in the function ast_rtp_write: rtp-smoother = ast_smoother_new(4 * 50); (I changed mine to 50 ms for G726 which did wonders for those slow DSL users to reduce the number of packet/sec, and the latency increase is virtually not noticeable to me). I'm sure we could make a patch to set it on a per-call basis from the dialplan... if someone cares to do so. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax and PRI
In a previous company, we had issues with selling Fax-o-IP services varying from the ability of CPE to support the correct NSE values and recognize the difference between G3 (standard fax) and Super G3 (more like a modem fax) through to the gateway's ability to turn echo-can's on and off based on the fax type. Fax is SO sensitive to packet-loss, NSE response, the disabling/enabling of echo-can's based on G3/Super-G3... I have a pretty decent white paper I found on this subject I can forward to those that are interested offline ([EMAIL PROTECTED]) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Wednesday, January 19, 2005 6:46 PM To: Jon Radon Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fax and PRI On 2005.01.19 17:06 Jon Radon wrote: I use a Multi-Tech Multimodem ZPX and a Sipura SPA-2000 along with Hylafax. On the Asterisk end I'm using X100P's. This has worked flawlessly for me thus far. Some tips of my success. 1.) Fax detect on the Sipura breaks faxing. Strange, I know. 2.) Tuning your txgain and rxgain both on the Sipura and on the Zaptel line is imperative. Originally with this setup I was getting all kinds of bad frames. After tweaking my faxes are squeaky clean. Thanks for these tips it reminded me that I still hadn't looked in on the FXS port (line) settings on the SPA. (I don't receive many faxes on this setup, so it hasn't been a priority.) As it turns out, disabling silence suppression, echo cancellation, echo suppression, and all of the fax features did the trick. Now I'm receiving faxes fine. I did also have to disable V.34-Fax on the HylaFAX-controlled modem. I didn't have to adjust any of the gain controls, although I already had adjusted the FXS port impedence a long time ago. Lee. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
