Re: [Asterisk-Users] Wireless SIP Phones
yet. The only Wireless SIP phone I would use in a productive environment would be the Cisco 7920. I don't see a SIP load for the 7920. Are you sure it is SIP enabled? Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Wireless SIP Phones
yet. The only Wireless SIP phone I would use in a productive environment would be the Cisco 7920. Does it work in SCCP mode with good results in Asterisk? - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hospitality Industry
Anyone connected Asterisk to hospitality packages such as: Micros Fidelio Visual One Jonas We'd be interested in providing bounty on providing a connection to one or more (depending upon what the client selects) if our proposal goes through. Ultimately, about 300 to 600 stations will be provided. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Polycom IP 600 Programmability
I'm looking to program some sort of web-services function: user presses a button and send some info to a web server or scripting program. The web server or script returns some text and/or imagery for the screen. Lather, rinse, repeat. I saw in section 3.7.1 of the manual referenced below that there is a services function. However, it appears to not be enabled. Yet. Any other way of doing this, or has the 3.7.1 function been enabled yet? Ray. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Tuesday, June 15, 2004 11:02 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 600 Programmability Polycom IP 600's are fully programmable, much more so than the Cisco phones. Yes, you can program the phone buttons. That and just about everything else you can imagine is programmable via xml configuration files. Goto http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00. pdf for the admin guide and you can see for yourself how great the difference is. John P.S. Here's the wiki: http://www.voip-info.org/wiki-Polycom+Phones Ray Burkholder wrote: Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Invalid Extensions -- More like traditional PBX systems?
Set up a general pattern match with the message and congestion. Extension pattern matching looks for the most specific match in any one context. So if a specific extension is not found, it will take the general pattern. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Rosebush Sent: Wednesday, June 16, 2004 11:32 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Invalid Extensions -- More like traditional PBX systems? I was wondering if there was a way of setting up the dialplan in a way that if you dial an extension that is NOT in the dialplan then it would play a not-in-service gsm file and then play congestion tones. I would rather like this better than just hearing a busy signal on my phones.. I DID search around on the wiki and using google and could not find anything. Thanks. -- Stephen Rosebush, [EMAIL PROTECTED] http://www.desynched.org/ // Hardline // IP Phone USA: 1-248-724-4452 x201 FWD: 63420 x201 Netherlands: +31-(0)20-6598858 x63420 x201 IAXTEL: 1-700-356-6191 x201 United Kingom:+44-(0)870-3403054 x201 SIP:sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 600 Programmability
Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco SIP Phone Licensing
Cisco has a part number SW-SM-UL-7960 for licensing SIP on their CP-7960 phones. Is this actually required to be purchased to keep everything on the above-board when using Cisco's SIP phone with Asterisk, or is this for something else? Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Cisco SIP Phone Licensing
Cisco has a part number SW-SM-UL-7960 for licensing SIP on their CP-7960 phones. Is this actually required to be purchased to keep everything on the above-board when using Cisco's SIP phone with Asterisk, or is this for something else? I found the answer in the March archives. Yes, according to the list, the license is required. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 via email
You may want to take a look t.38, t.39 which are the fax/ip/smtp standards. If Asterisk could be made to do this, then it would join the mainstream and inter-op with cisco gw's and such handling this sort of thing automagically for the billions of voice/fax minutes served. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, June 09, 2004 00:33 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax via email Steve Underwood wrote: If you want to FAX over IP you need to be *very* careful if you want it to be reliable. You cannot use anything other than A-law or u-law as the codec. However, even using those, any data slips will kill the FAX operation. If the two boxes are on the same LAN it tends to work OK. Yes, I would think that this sort of application would be either local LAN or _extremely_ low latency WAN connections only, and probably not use audio compression at all. If you can't handle a few 64kb/s streams of audio for your FAXing application, then you have other problems to worry about :-) I mean CPU loading. HylaFAX only does 1D coding (unless that changed very recently) and the ECM is brand new. The features you list may be a lot less well tested than you think. :-) Also, only a tiny fraction of FAX machines can even support ECM. As mentioned in the other replies, these are no longer true statements as of HylaFAX 4.2.0 (which is not yet released, but very close). And putting the virtual modem client and HylaFAX on a separate box from Asterisk should eliminate CPU consumption concerns, I'd think. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_rtp_read: Unknown RTP codec
Any one see these? Are they benign, or is some system tuning required to remove them? Can't seem to find a resolution in the archives. If you have a link, it would be appreciated. Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Jun 2 10:58:59 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:00 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:01 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Meetme + Billing
Isn't each call leg represented in the cdr file? If you set up account codes properly, it shouldn't be too difficult to script either a conference duration, or a total call duration to the conference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pablo Endres Sent: Monday, May 31, 2004 10:22 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meetme + Billing Hi, I'm trying to detect and or log the duration a a conference (Meetme). I need it in order to do some billing for theses services. Any ideas on how to do it? I googled around but found nothing. Thanks in advance epablo -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] opinions on oneunified.net as asterisk provider
I hope you don't mind if I step in. I run One Unified. We are in still in 'development' mode. Provisioning stuff has been completed. The auto-invoicing, real-time call detail records, and customer self-management sections are about to be completed any day now. We are hoping to make an intro announcement next week on: * residential broadband services * corporate virtual dial plan capabilities * web based connect me capabilities There's a bunch of other things coming down the pipe, but I'll wait untill they are ready before getting more into them. If you have any questions on our expertise and capabilities, please let me know. I can send some references from some beta customers if you'd like. Ray Burkholder [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Herndon Sent: Thursday, May 27, 2004 09:58 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] opinions on oneunified.net as asterisk provider i'm looking at potential asterisk service providers and came across oneunifed.net i googled for opinions and feedback, but haven't come across anything yet. is anyone using them or does anyone have feedback on their asterisk support and expertise? tia, george ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] AArgh, * and the 7960
It is a royal pain in the butt to manually walk through 2,000 packets calculating timestamp differences, inspecting sequence numbers, etc. I'm in the process of writing a small app to read the ethereal packet capture files and do that stuff on request. Or simply import the trace in to a spreadsheet. Super simplifies everything that way. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] AArgh, * and the 7960
Quoting brian [EMAIL PROTECTED]: You're missing the point, Brian. Those comments were in response to your statement that essentially said there isn't a problem because your system is working fine. And based on your comment, your primary (only?) iax link is to Nufone. No I'm getting it loud and clear. You have some IAX providers that do not want to take care of customers when the software they use to provide service to their customers needs an update they refuse or fail to upgrade. Not our problem if they choose not to. If they update to cvs-head the problem will go away and its backwards compatible with cvs-stable. You can continue to hack rtp.c or ask your providers to upgrade. If they refuse to take care of you then I would consider getting service elsewhere. But as I've mentioned before, this isn't the whole story. There are other repeatable scenarios that still cause problems, and to which some large progressive providers also see as an issue and won't accept termination becuase of it: GW - SIP - * - IAX2 - * - SIP - 79X0 Now, if this scenario has been corrected as well, please accept my apologies for bringing it up. This config, with the absolute latest CVS HEAD, well as of a week or so ago when I last checked, seems to cause issues on the sequencing. I seem to recall comments that there is some work still being done on getting this cross protocol packet sequencing to work properly? I'll have to get Ethereal out again and prove that it is still happening. And why are we blaming Cisco for dropping packets that are mis-sequenced, when we shouldn't be sending them mis-sequenced packets in the first place? Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
This caught me as well. Be aware that if you did any manual mods to rtp.c or related files, you need to delete it and rerun 'make update'. This will bring down the proper file. You should then be all set. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: May 7, 2004 11:13 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway) I've had this too, reported it as a bug last week and got my butt kicked for not being responsive enough in providing support to sort it out. You could file another bug report but be sure to have a thick book ready to stuff down your trousers. Iain --On Friday, May 7, 2004 10:43 am -0400 Brian Cuthie [EMAIL PROTECTED] wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Quick Caller ID and Voicemail ?s
Quoting John Todd [EMAIL PROTECTED]: [apologies for top-posting] I am very interested in what providers typically take CNAM via ISDN. I have some experience with PRI providers, but I've never heard of one offering that service. If you are a PRI provider in the lower 48 who takes CNAM and can pass that off to the PSTN, please get in touch with me off-list, especially if you handle SIP outbound to your gateways. I may not buy service from you (or maybe I will!) but I'd be very interested in hearing what the particulars are about your offering so I can bring those elements to the table with my current providers. I'd be interested in this info as well. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Codec Voodoo: piece of evidence
Andres wrote:
another? We noticed this problem when we upgraded one of
our servers to
the latest CVS and left another one with an older version.
Seems that
the latest changes with rtp.c need to be applied
everywhere. When we
upgraded all servers then the audio returned to normal but the
connection with Nufone started sounding horrible.
We had to roll back to the older version of rtp.c to get
back the good
audio with Nufone.
Codec stuff sure does feel like voodoo sometime. I'd love it
if someone
with a handle on this (particularly the why would a changed
RTP stack
cause some ITSP connections to go down the shi**er deal) could
enlighten us all.
We, too, have noticed issues with sound quality with recent versions of rtp.
I think there is a rounding problem in it. It isn't just something to
Nuphone. We can recreate between Asterisk and a Cisco 7940 with G.711.
In the following ethereal extraction, 10.1.6.2 is the Cisco Phone, 10.1.1.12
is Asterisk. Notice that the cisco phone consistently produces sequence
differences of 160 (the last column), while asterisk produces sequence
differences of 152 and 160. And because the time differences aren't
consistent, the phone probably 'stagger-steps' to get back in sequence, and
therefore sound quality suffers. This probably happens in time calcs for
gsm and other codec types as well.
Here is the snippet from rtp.c that does the processing. How do we fix the
rounding problem.?
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval
*delivery)
{
struct timeval now;
unsigned int ms;
if (!rtp-txcore.tv_sec !rtp-txcore.tv_usec) {
gettimeofday(rtp-txcore, NULL);
}
gettimeofday(now, NULL);
ms = (now.tv_sec - rtp-txcore.tv_sec) * 1000;
ms += (now.tv_usec - rtp-txcore.tv_usec) / 1000;
/* Use what we just got for next time */
rtp-txcore.tv_sec = now.tv_sec;
rtp-txcore.tv_usec = now.tv_usec;
return ms;
}
Here is the ethereal extraction:
114 2.71825910.1.6.2- 10.1.1.12
G.711 Seq 33943, Time37206320
115 2.73100310.1.1.12 - 10.1.6.2
G.711 Seq 11856, Time9472
116 2.7381370.01987810.1.6.2- 10.1.1.12
G.711 Seq 33944, Time37206480160
117 2.7509790.01997610.1.1.12 - 10.1.6.2
G.711 Seq 11857, Time9624152
118 2.7578360.01969910.1.6.2- 10.1.1.12
G.711 Seq 33945, Time37206640160
119 2.7710350.02005610.1.1.12 - 10.1.6.2
G.711 Seq 11858, Time9784160
120 2.490.01991310.1.6.2- 10.1.1.12
G.711 Seq 33946, Time37206800160
121 2.7910710.02003610.1.1.12 - 10.1.6.2
G.711 Seq 11859, Time9944160
122 2.7984190.02067 10.1.6.2- 10.1.1.12
G.711 Seq 33947, Time37206960160
123 2.811 0.01992910.1.1.12 - 10.1.6.2
G.711 Seq 11860, Time10096 152
124 2.8184840.02006510.1.6.2- 10.1.1.12
G.711 Seq 33948, Time37207120160
125 2.8309930.01999310.1.1.12 - 10.1.6.2
G.711 Seq 11861, Time10248 152
126 2.8386820.02019810.1.6.2- 10.1.1.12
G.711 Seq 33949, Time37207280160
--
Scanned for viruses and dangerous content at
http://www.oneunified.net and is believed to be clean.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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] Codec Voodoo: piece of evidence: probable fix
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval
*delivery)
{
struct timeval now;
unsigned int ms;
if (!rtp-txcore.tv_sec !rtp-txcore.tv_usec) {
gettimeofday(rtp-txcore, NULL);
}
gettimeofday(now, NULL);
ms = (now.tv_sec - rtp-txcore.tv_sec) * 1000;
ms += (now.tv_usec - rtp-txcore.tv_usec) / 1000;
/* Use what we just got for next time */
rtp-txcore.tv_sec = now.tv_sec;
rtp-txcore.tv_usec = now.tv_usec;
return ms;
}
This snippet is from old code. Here is a corrected new snippet with proper
rounding that I think fixes the issue (the two lines are marked [sorry
didn't think to do a diff until afterwards]):
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval
*delivery)
{
struct timeval now;
unsigned int ms;
if (!rtp-txcore.tv_sec !rtp-txcore.tv_usec) {
gettimeofday(rtp-txcore, NULL);
rtp-txcore.tv_usec -= rtp-txcore.tv_usec % 2;
}
if (delivery (delivery-tv_sec || delivery-tv_usec)) {
/* Use previous txcore */
=ms = (delivery-tv_sec - rtp-txcore.tv_sec) * 1000;
ms += ((delivery-tv_usec - rtp-txcore.tv_usec) + 500) /
1000;
rtp-txcore.tv_sec = delivery-tv_sec;
rtp-txcore.tv_usec = delivery-tv_usec;
} else {
gettimeofday(now, NULL);
ms = (now.tv_sec - rtp-txcore.tv_sec) * 1000;
=ms += ((now.tv_usec - rtp-txcore.tv_usec) + 500 ) / 1000;
/* Use what we just got for next time */
rtp-txcore.tv_sec = now.tv_sec;
rtp-txcore.tv_usec = now.tv_usec;
}
return ms;
}
--
Scanned for viruses and dangerous content at
http://www.oneunified.net and is believed to be clean.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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 7960 SIP Images
Iain Stevenson wrote: .. not sure this applies outside the US - or I'd reach for the credit card. Iain --On Friday, March 26, 2004 11:29 am -0500 Chris HARIGA [EMAIL PROTECTED] wrote: If you pay 8 USD for 1 year support you can download the image :) Best regards, Chris HARIGA No, you can't use a credit card. You have to send the #$!@@$#'s a check. It's really stupid, but it's the Cisco way. John Or purchase a Smartnet from your local Cisco reseller. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Incoming Fax Call to File
No hardware required. There are one or more pieces of software that do the trick. I can get these installed and trial'd once the calling stuff is in place. For voicemail notification through email, I'm going to need email addresses for everyone. What are you currently using for an email server? I havn't done it yet, but it might be possible to perform some trickery to better integrate the email, voicemail and email. Do you have any virusscanning, spamscanning stuff on your email server? Quoting Ryan Thrash [EMAIL PROTECTED]: I can't seem to find an answer in the archives covering this (or maybe I just missed it)... Setting up * and hope to accomplish the following: 1) Use 5 of our DID numbers from our PRI for inbound fax reception 2) When * receives a call on one of these lines, it digitizes the incoming fax to a multi-page .tif file (ala eFax.com) rather than transferring it to an analog fax machine. 3) Based on the DID number, e-mail the resulting fax to a specific inbox The end result--when coupled with doing the same for voice mail messages--would be a unified inbox, which we really are hoping to have soon. To accomplish Part 2, do we need a fax board or some such piece of external hardware, or is the processing power of a dual Xeon server coupled with some as-yet-to-be-identified-DSP-esque software capable of translating fax-static into an image? Thanks for any ideas or pointers. -- Best regards, Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. Ray Burkholder 704 644 6999 x2002 http://www.oneunified.net [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] LAN card
Take a look at your memory utilization, you should not be paging/caching any memory. Switches are will known not to auto-negotiate properly. All switches, nics, routers, etc should be manually configured for full-duplex. Make sure each connection is set appropriately for 1000/100/10 mpbs, what ever is appropriate for that connection. And yes, you can get full duplex for 10 mpbs connections (in answer to a message a while back on the list). Managed switches are best becuase you can look at them and get an idea of link/packet errors on each port. Obviously you want to completely eliminate errors on each port. Once you've done that, you should be well on your way to a reliable, scalable solution. Quoting T. Chan [EMAIL PROTECTED]: Dear All, Just an experience to run by all you experts out there. I have started to put more VOIP calls into Asterisk, most are pass-through calls and some are terminating on the Digium card to PSTN. Whenever I get to 10 calls or more, I would start to get choppy sound. I tried to ping other IP addresses from the Asterisk and noticed a big packet loss in the vincinity of 7% to 10%, but when there is no call, pinging the same IP addresses reap no packet loss. It seems that the VOIP packets are causing congestion of some kind on the LAN. I am using 100M, full duplex. I tried an autonegotiated switching hub as well as a more sophisticated managed switching hub and forcing the connection to be 100M Full Duplex, non negotiated. However, I reaped the same result. Question is, do you know if it is better to use Managed switch and forcing the Ethernet connection to be 100M Full Duplex, or to use a normal UnManaged switch and let it negotiate. Also, I am using both a normal PCI LAN card as well as trying to use the onboard Intel 100PRO Lan card, and in both situations, I started to get lose packets when the number of calls increased. My colleagues, can anyone tell me if I am doing something wrong here, or is there something I am forgetting, or I simply need to use a more powerful LAN card due to the demand of VOIP packets. Ray Burkholder 704 644 6999 x2002 http://www.oneunified.net [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] LAN card
What else are you running on your server? On my server running asterisk and apache, it has the following: total:used:free: shared: buffers: cached: Mem: 261443584 237064192 243793920 55992320 143912960 Swap: 260104192 11231232 248872960 MemTotal: 255316 kB MemFree: 23808 kB SwapTotal: 254008 kB SwapFree: 243040 kB I've got very little swap usage, even with 256MB total physical. For the switch, have you looked at the statistics? For example on a Cisco: sw2#sho inter f0/1 FastEthernet0/1 is up, line protocol is up Hardware is Fast Ethernet, address is 0005.5e31.5f41 (bia 0005.5e31.5f41) Description: Trunk: r1-skings MTU 1500 bytes, BW 10 Kbit, DLY 100 usec, reliability 251/255, txload 1/255, rxload 1/255 Encapsulation ARPA, loopback not set Keepalive not set Full-duplex, 100Mb/s, 100BaseTX/FX ARP type: ARPA, ARP Timeout 04:00:00 Last input 00:00:40, output 00:00:00, output hang never Last clearing of show interface counters never Queueing strategy: fifo Output queue 0/40, 0 drops; input queue 0/75, 0 drops 30 second input rate 3000 bits/sec, 4 packets/sec 30 second output rate 6000 bits/sec, 8 packets/sec 75312691 packets input, 1770301889 bytes Received 515417 broadcasts, 7622395 runts, 0 giants, 0 throttles 7622399 input errors, 4 CRC, 0 frame, 4 overrun, 92 ignored 0 watchdog, 255441 multicast 0 input packets with dribble condition detected 104212173 packets output, 2775526395 bytes, 0 underruns 0 output errors, 0 collisions, 3 interface resets 0 babbles, 0 late collision, 0 deferred 0 lost carrier, 0 no carrier 0 output buffer failures, 0 output buffers swapped out Looks like I've got some input errors I should be looking into. It should be as close to 0 as possible. An Intel 1000XT are good cards at they do TCP Engine Offload. Or something similar. But voice traffic is measured in kbits/second, which is a very low proportion of 10mbps or even 100mbps. So I'd say take another look at your server and see if an application isn't makeing a mess of your cpu processing. Becuase Asterisk is time senstive, it should really be the only primary process running on your machine. AND, YOU SHOULD NOT BE RUNNING XWINDOWS. The os should have been installed in console mode, and as little as possible relating to X installed. Quoting T. Chan [EMAIL PROTECTED]: Thanks alot, Ray Well, looking at cat /proc/meminfo, I am getting like 250M memory cached, with 512M total RAM, for all the gateways I have, this is quite consistent. Total Memory usages are always low after reboot and then go up to 450M with time. I was informed that this is normal for Linux. Thanks for your input on Managed switch. However as said, I tried both Managed switch and non-Managed switch but have reaped the same result with packet loss when there are more active calls. Do you have any experience whether I need a good PCI LAN card like 3COM or Intel Express due to the demanding VOIP packets or do you think Intel ONBOARD LAN card should be sufficient? Thanks Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ray Burkholder Sent: Sunday, January 25, 2004 4:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LAN card Take a look at your memory utilization, you should not be paging/caching any memory. Switches are will known not to auto-negotiate properly. All switches, nics, routers, etc should be manually configured for full-duplex. Make sure each connection is set appropriately for 1000/100/10 mpbs, what ever is appropriate for that connection. And yes, you can get full duplex for 10 mpbs connections (in answer to a message a while back on the list). Managed switches are best becuase you can look at them and get an idea of link/packet errors on each port. Obviously you want to completely eliminate errors on each port. Once you've done that, you should be well on your way to a reliable, scalable solution. Quoting T. Chan [EMAIL PROTECTED]: Dear All, Just an experience to run by all you experts out there. I have started to put more VOIP calls into Asterisk, most are pass-through calls and some are terminating on the Digium card to PSTN. Whenever I get to 10 calls or more, I would start to get choppy sound. I tried to ping other IP addresses from the Asterisk and noticed a big packet loss in the vincinity of 7% to 10%, but when there is no call, pinging the same IP addresses reap no packet loss. It seems that the VOIP packets are causing congestion of some kind on the LAN. I am using 100M, full duplex. I tried an autonegotiated switching hub as well as a more sophisticated managed switching hub and forcing the connection to be 100M Full Duplex, non negotiated. However, I reaped the same result. Question is, do you know if it is better to use Managed switch
RE: [Asterisk-Users] ADSI phone vs. IP phone (and proper implementation thereof)
Quoting [EMAIL PROTECTED]: Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. Probably because it's well known that these setups are prone to failure of either the PC's connection, the phone's connection, or degredation of one/both. It also breaks switch envirenments where spanning-tree portfast is enabled (not as big of a deal if the deployment is in concert with the infrastructure group, as it should be). Vendors should NEVER have implemented this functionality into phones unless it was working under all conditions. Personal experience shows that it is most definitely not on Cisco and 3Com products. Others have told me their stories with other manufacturer's equipment. None of it was good. It's not a production-stable way to deploy phones. Period. I'm wondering if what you say is actually true. According to recent media releases, Cisco has shipped over 2 million of their IP phones. They must be doing something right. Their phones are _designed_ to function and cooperate with the switch. Obviously, the installer has to be totally familiar with all phone, switch, router and network settings in order to have a successful installation. The switch needs to be configured with specific port, vlan, and class of service settings. Accepted practice is to provide a voice vlan and a data vlan. On the phone side, the phone knows to send voice on the specific vlan told to it by the switch , and to pass through data from the pc through the vlan told to it by the switch. The phone knows to prioritize voice traffic over data traffic. So does the switch. And so on through the connection of switches and routers. This ensures voice quality and precedence through out the network. Voip quality is not necessarily about bandwidth (because it works on T1 data lines as well as GB ports), but about instantaneous bottlenecks in the network. These instantaneous and random bottlenecks can occur in the cad environment mentioned. But with appropriate COS (layer 2) and TOS (layer 3) settings in the phones, switches, and routers, these bottlenecks become non- issues. In addition, what many people forget, or learn by experience, is that you absolutely _must_ have everything running full-duplex, and to physically check errors and statistics on each port of the switch in order to verify that you have error free links. You won't believe how many networks out there are broken because noone checks and fixes these issues. A voip network _must_ have managed switches so you can verify these things. There was mention of a heavy cad environment. Say your computer is connected to the 100mbps port of the phone. A g.711 call comes through. The call takes around 80 kbps. If I've done the math properly, the voice call takes only 0.08% of the bandwidth, hardly something that will interfere with 'heavy cad users'. More likely the opposite, the heavy cad users will interfere with the call, _but_ _only_ if the switch and phone are not configured properly for vlan, cos, tos, speed, and duplex settings. So having said this, you mentioned that you have had personal experience where this functionality is built into, or does not work in Cisco's case. Could you provide some additional info on what did not work for you? Because my experience is opposite: things do work when configured according to manufacturer specifications and using the correct equipment. Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] RE: current version
Will the other chan_oh323 work? Quoting Brian West [EMAIL PROTECTED]: You can't use chan_h323 with call manager. bkw On Sun, 18 Jan 2004, Dan Austin wrote: I tried to use it to create a 'trunk' to Cisco's call manager. The 0.7.1 code worked up to a point. The call would be established, but audio was one-way from the Call Manager. Asterisk with Chan_h323 would not setup the sending rtp stream. The debug results showed the sending stream as using ip:0.0.0.0 I have not checked for a CVS update to see if it is fixed, or if that might just be a quirk when connected to Call Manager. Chan_oh323 works fine with Call Manager. Dan _ From: T. Chan [mailto:[EMAIL PROTECTED] Sent: Sunday, January 18, 2004 5:55 PM To: [EMAIL PROTECTED] Cc: Alan Chan Subject: [Asterisk-Users] RE: current version Dear All, I have been using Asterisk 10 days ago version loaded onto my Redhat 7.3 with kernel 2.4.18-3 running Jeremy's h323 driver. It has been running okay with a bit of problems, like system crashing after certain period of time with 15 simultaneous calls or so. I have tried to load up the current version today again (0.7.1 I guess) and apparently with the new H323 driver as well. I have recompiled the H323 libraries with version Pwlib1.5.2 and Openh3231.12.2 as recommended. However, no call was able to get through at all. I have tried this before when 0.7.0 came out when had the same result, I thought there were bugs, but now I am getting the same thing. I have tried using the same h323.conf configuration as well as trying to change a couple of faststart.parameters, but same result. Is there anyone who has had good experience with the new version of Asterisk 0.7.1 with the most current Jeremy H323 driver? Please suggest, thanks Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. Ray Burkholder 704 644 6999 x2002 http://www.oneunified.net [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] RE: current version
Will either of the h323 channels work with gatekeepers properly? Ie, reregister through * reloads and such? I've had problems with chan_h323 in this regard. Chan_h323 will register fine with a gnugk first time around. But after a reload, it loses its connection. Or should I try downloading the latest version yet again to see if it is fixed? Quoting Brian West [EMAIL PROTECTED]: You can't use chan_h323 with call manager. bkw On Sun, 18 Jan 2004, Dan Austin wrote: I tried to use it to create a 'trunk' to Cisco's call manager. The 0.7.1 code worked up to a point. The call would be established, but audio was one-way from the Call Manager. Asterisk with Chan_h323 would not setup the sending rtp stream. The debug results showed the sending stream as using ip:0.0.0.0 I have not checked for a CVS update to see if it is fixed, or if that might just be a quirk when connected to Call Manager. Chan_oh323 works fine with Call Manager. Dan _ From: T. Chan [mailto:[EMAIL PROTECTED] Sent: Sunday, January 18, 2004 5:55 PM To: [EMAIL PROTECTED] Cc: Alan Chan Subject: [Asterisk-Users] RE: current version Dear All, I have been using Asterisk 10 days ago version loaded onto my Redhat 7.3 with kernel 2.4.18-3 running Jeremy's h323 driver. It has been running okay with a bit of problems, like system crashing after certain period of time with 15 simultaneous calls or so. I have tried to load up the current version today again (0.7.1 I guess) and apparently with the new H323 driver as well. I have recompiled the H323 libraries with version Pwlib1.5.2 and Openh3231.12.2 as recommended. However, no call was able to get through at all. I have tried this before when 0.7.0 came out when had the same result, I thought there were bugs, but now I am getting the same thing. I have tried using the same h323.conf configuration as well as trying to change a couple of faststart.parameters, but same result. Is there anyone who has had good experience with the new version of Asterisk 0.7.1 with the most current Jeremy H323 driver? Please suggest, thanks Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. Ray Burkholder 704 644 6999 x2002 http://www.oneunified.net [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Remote reload Cisco 7960
Quoting B. J. Bomar [EMAIL PROTECTED]: Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. Try telnetting into the phone, and use the ?/help command. You should see a restart or reload command. This will do the trick. Ray Burkholder 704 644 6999 x2002 http://www.oneunified.net [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 7910 phone
Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok Cisco's site shows SIP drivers for 7960, 7940, 7912, 7905 only. If you want to run 7910 in Skinny mode, that may work. I'll leave that up to the chan_sccp and chan_skinny people. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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-Client for Handheld PC
What are the ones you found for PocketPC? I guess you've looked at the Telesym site? They have a SIP flavor coming out shortly for some PDA's. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans-Henrik Andresen Sent: January 12, 2004 05:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP-Client for Handheld PC Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _ Scope out the new MSN Plus Internet Software - optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-uspage=byoa/plusST=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Linux Sip UAs
Maciek Kaminski wrote: Hi, What linux SIP UAs do You successfully use with Asterisk? So far none!!.. The ones I have tried all have issues and very limited functionality.. There is supposed to be a linux version of X-lite/pro coming out but who knows when that will be.. Anyone tried the linphone. Someone tried it here and said it was dead easy to get running. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk success stories in small-mediumoffice environments?
We should get references for these on the Wiki. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey Paul Sent: January 8, 2004 11:09 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk success stories in small-mediumoffice environments? I'm not really looking for working configurations as much as I am looking for people who can say This is a solid product and I trust my business to a solution running Asterisk. As far as pre-sales work... Well, tell that to my consultant. I'm quite excited about *. I've got my company sold on it, they just want some reassurance that it's ready for prime-time production use. I can't think of a better way than printing out an email from John Q. Officemanager saying It works great, I love it!. -j -- Jeffrey Paul - [EMAIL PROTECTED] - (877) 748-3467 Senior Network Administrator, Diamond Financial Products An expert is a man who has made all the mistakes which can be made in a very narrow field. -- Niels Bohr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, January 07, 2004 4:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk success stories in small-mediumoffice environments? While you approached the community in a very polite way and all, your few weeks of *-users list should have told you that most answer should be able to be found on the wiki. http://www.voip-info.org/wiki-Asterisk+hardware+recommendations This has a few listings of working systems. Also, that should really be the type of pre sales work up a consultant would do for you. http://www.voip-info.org/wiki-Asterisk+consultants See, use the Wiki. On Wed, 2004-01-07 at 14:04, Jeffrey Paul wrote: I am the network administrator at a small (20-30 employee) financial company. We are in the process of moving offices and will be obtaining a VoIP phone system when we do. Right now, it's down to the 3com nbx100 series and *. Having lurked on *-user for a few weeks and having seen the nifty features of asterisk, I'm convinced. The price difference has pretty much sold my superiors. However, they're slightly wary of the whole open-source thing. They have no way of knowing, for certain, that asterisk is production-quality until they sign the check and find out. I've been asked by my CTO and CEO to get some testimonials and/or case studies of asterisk in production use in small office / small callcenter environments. We'll be having a contractor configure an IVR, a call center with queues, call detail reporting, and a dialplan for our two inbound groups (our callcenter and our normal office traffic). Does anyone have their own success stories and/or have some verifiable customer testimonials? My CEO and/or CTO might want to call some of these places/people on the phone as well and ask some simple questions about reliability and stability, so please include contact information where permissable. Replies in public or private are okay. I'll summarize the private responses (minus any confidental contact information) to the list once I get them all. Thanks in advance, -j -- Jeffrey Paul - [EMAIL PROTECTED] - (877) 748-3467 Senior Network Administrator, Diamond Financial Products An expert is a man who has made all the mistakes which can be made in a very narrow field. -- Niels Bohr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Trunk two Asterisk boxes.
Quoting Ariel Batista [EMAIL PROTECTED]: [Say hi to Steve for me!]
I need to get 2 Asterisk servers working together. I have been reading
I have tried the command in the 2nd box of
[local]
switch = IAX2/redbox2:[EMAIL PROTECTED]/local
What I am looking for is a real example of someone's working boxes. I
need to share the dialing rules and the trunks. In this case the Wiki
(Names have been changed to protect the innocent)
Hopefully this helps in your quest for knowledge! I think I've got all the
relevant bits included.
machine master iax.conf:
[slave]
type=user
auth=plaintext
context=outbound
context=outbound2 ; (can have multiple if you want)
secret=secret
host=dynamic
callerid=slave
trunk=yes
notransfer=yes
[slave]
type=peer
auth=plaintext
context=outbound-nuphone
secret=secret
host=dynamic
trunk=yes
notransfer=yes
in extensions.conf:
[assigned-dids]
; uncomment a dial mechanism, first one goes to specific extension
; other one goes to dial parameter s.
;exten = 7046446999,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
;exten = 7046446999,1,Dial,IAX2/[EMAIL PROTECTED]
machine slave iax.conf:
register = slave:[EMAIL PROTECTED]
[master]
type=peer
host=iax-gw1.company.net
secret=secret
context=outbound
trunk=yes
canreinvite=no
[master]
type=user
secret=secret
context=acontext
trunk=yes
canreinvite=no
In extensions.conf:
[outbound]
switch = IAX2/master:[EMAIL PROTECTED]/outbound
Ray Burkholder
704 644 6999 x2002
http://www.oneunified.net
[EMAIL PROTECTED]
Asterisk Consulting Services, DID's, Termination, Origination
-
This mail sent through IMP: http://horde.org/imp/
--
Scanned for viruses and dangerous content at
http://www.oneunified.net and is believed to be clean.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Identifying the Originating Cisco SIP Gateway
I have several Cisco SIP gateways sending calls to Asterisk. Because the
gateways don't have user-agents, they don't authenticate with Asterisk. And
because they don't authenticate, they use the default context in the
sip.conf file.
Is there a way to either:
A) identify the inbound gateway with a variable, in channel info, or the
manager interface? If there was a ${SIPDOMAIN} for the originator rather
than the destination, that would be cool, or
B) make the inbound gateway use the sip.conf file section belonging to it
via the host= line in the sip.conf file without user authentication, or
C) some other way I have yet to fathom
I'm trying to differentiate between legitimate gateways that initiate calls
vs other gateways that should get a very limited inbound capability.
Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 644 6999 x2002
--
Scanned for viruses and dangerous content at
http://www.oneunified.net and is believed to be clean.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Java? -- Ming!
Masakazu Nakano Sent: December 31, 2003 21:13 On Wed, 31 Dec 2003 21:19:10 +0200 Stephen Karrington [EMAIL PROTECTED] wrote: We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. snip Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good. Dynamic effective,Easy coding and Fast response :-) Cool. I like the Ming thing. Also works with Perl (many Perl examples available). And has an XML event interface for two way communications with a server. Certainly is way much less overhead than the Java thing I was contemplating. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 P.S. Note, for the message police, I cut out extraneous text, did the attribution at the top, did a bottom post, and made it a single page for zero scrolling. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 API Card Solution
Current Status: http://www.openss7.org/asterix.html Ray Do I need a special Digium Card (E100-SS7) or use my E100P card and compile the new drivers? Daniel Juan J. Sierralta P. wrote: On Sun, 2003-12-21 at 04:10, Ray Burkholder wrote: Is this useful as a bootstrap for getting SS7 to Asterisk? http://www.sangoma.com/api/p-api-ss7.htm You should check http://www.openss7.org the have an stack and works with an special version of digium cards, dunno if is the same HW with special drivers but it looks much more * friendly. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax capabilities of various services
Title: Fax capabilities of various services For the Vonage, Packet8, etc services, are they all able to handle fax machines on their little interconnect boxes? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] OT: SIP vs. Skinny protocol
Skinny phone functionality is 'richer' than SIP phone functionality. First off, on a skinny phone, in hands free mode, you can start dialling and the phone will automatically go off hook. Sip requires you to manually hit the speaker button, hit new call, or pickup the phone before dialling. (One extra confusing key stroke I have a hard time getting over). I don't think SIP will work with the expansion modules on a 7960. Those are a few things I've found. On Asterisk there is a chan_skinny and a chan_sccp available for skinny based phones. Perhaps as more Cisco phones get used with *, more features will get implemented so they respond in a fashion very similar to a Callmanager installation. Maybe Cisco is already doing that in their labs? That would be cool. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Pauly Sent: December 23, 2003 12:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: SIP vs. Skinny protocol I assume there are several people on this list that have Cisco Call Manager implementations under their belt We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two obvious benefits to using SIP: 1. I can get cheaper phones that run SIP, altough Cisco just came out with a 7902G for $130 US. 2. It's an open protocol and is more likely to survive long-term. What functionality do I lose by going with Skinny? Will Cisco eventually go with SIP only and I'll have to convert anyway? Any other pluses or minuses? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SS7 API Card Solution
Title: SS7 API Card Solution Is this useful as a bootstrap for getting SS7 to Asterisk? http://www.sangoma.com/api/p-api-ss7.htm Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] Chan_h323 docs
Title: Chan_h323 docs Jeremy, In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution. Could you post those docs in your download directory? I'm trying to understand the nuances of your driver, gnugk, and extensions. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] Chan_h323 gnugk
Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h323.conf like: [office] type=h323 prefix=9 context=outbound I get a message saying: WARNING[1074403072]: File chan_h323.c, Line 215 (build_alias): Keyword h323 does not make sense in type=h323 Why is that? 3) When making a call from an h323 client such as ohPhone registered with gnugk, I can make the call, but it uses the context from the [general] section, rather than the context in [office]. Is this supposed to be? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iax vs iax2 question
I'm trying to find this 'posting'. For some reason I'm missing it. Can anyone point it out please? - the host= setting (plus deny=/permit=) in particular is what can create the unexpected headaches if used with type=friend (some weeks ago there was an excellent posting on this issue, probably by J Todd), e.g.: you want to fix a host for the peer, but you might not want to fix a host for the user. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Roaming Users
I'm trying to come up with an elegant solution to handle roaming users in a branch office scenario. I have a number of possible scenarios, none of which seem to completely solve the problem. Perhaps someone with a better feel of the interactions can help me out. Is the 'switch' statement useful in some way? What are the ins and outs of the 'switch' statement? Come to think of it, there was a post a while back regarding how * searches the dial plan. Does any one have a handy link to that message Any way, my example: a company has a number of branch offices. A few implementation scenarios include: a) One central hosted * against which all phones register. Phones requiring TFTP loads will need a local server or some sort of VPN'd TFTP connection to the hosted server. The canreinvite parameter on SIP phones can get complicated if we want to keep local calls local to the branch, but yet all off-net calls go to the hosted PBX for transmission to a gateway. This scenario is conducive to a roaming extension that can go from office to office. b) One central hosted * which handles DID routing and corporate Auto Attendant functions. Each branch office has a * for local extensions and voicemail. Each office will have to have a range of extensions assigned to it. This limits portability but effectively makes use of * trunking capability to get to the hosted * for off-net calls. There must be a way of merging the two scenarios so I can get: a) roaming user capability (user can take own extension [logically or physically] between offices) b) local * server at each office to handle trunking to hosted * c) central auto attendant for all offices D) local extension to extension calls stay local (don't cross the WAN to the hosted *) Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk freezing HELP
I'm glad other people are seeing the same problem I've been seeing and posted about a day or so ago. My * is running on rh9 with most recent kernel with up2date. Does someone figure this is a threading issue? Does it need to be debugged using the method presented on the list yesterday? Jeremy, you say that you reload many times a day. Is that through a script or manually? Ray Burkholder Do you type reload at the cli a few times a day? If so try not reloading Asterisk and I'll bet Asterisk stop blocking. Recently I started running reload every 30 minues (to solve a IAX qualify problem). Since then I do see problems that weren't there before, i.e. when I issue reload manually after connecting via -vvvr the CLI client suddenly exits. In other cases a stop now has no effect and only kill -9 does it (just as the original post described). -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk freezing HELP
On Friday 05 December 2003 14:44, mattf wrote: Has anyone out there had the freezing problem(where they have to kill asterisk with kill -9) on any linux distro other than RedHat? What other distros do people out there use with their production Asterisk systems? We have no problem with freezing, and we use Mandrake 9.1 on most production systems. -Tilghman Do you use 'export LD_ASSUME_KERNEL=2.4.1'? How do you have your rc.local configured? Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channelbank Recomendation and GS102 question
I know that Time Warner Telecom (www.twtelecom.com) also provides a mixed data/voice burstable PRI T1 with their VersiPak service. I have installed a number of both channelized/channel banked VersiPaks and one PRI versipak (though it was not with Asterisk, it's connected to an InterTel system). The non-PRI based versipak is a bit cheaper, but you don't get the PRI abilities. They will do channelized hand over via T1 or will break out to POTs for you. It's really a nice product, and well priced. For one client, we picked up a 12 Voice channel/ 768K Internet PRI with data burstable almost all the way to 1.544mbit based on voice channel use. Voice hand over was via a single PRI, and data hand over was via fast eithernet. Pricing with a 3 year contract was below $800/month, in Tucson AZ. What box or boxes do they use on either end to handle the dynamic sizing on the voice/data channels? Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channelbank Recomendation and GS102 question
In what areas are you looking for the hybrid service? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 We have an installation with 9 inbound voice channels (one is the fax) and 768K data. It is a Hybrid PRI. It terminates into a T100P. It is working great! The cost was better than the POTS plus data. Can I ask what Telephone/Internet service provider you are getting this from? Does anybody else have a setup like this? I, too, would be interested in hearing from what vendor you are getting such a service. JT -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 'Stop Now', 'Restart' problems
Title: 'Stop Now', 'Restart' problems I'm not sure where to start looking for a solution on this. I use use Asterisk::Manager to reload Asterisk with a command like: $astman-sendcommand( Action ="" 'Command', Command = 'Reload' ); After a while, when I try to do a manual restart or 'stop now', asterisk will not exit. Any thoughts on where to look for a resolution? Ray -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] TE410P Errors under load
Might you be getting problems because you are using an Ethernet cable? If my memory serves correctly, an Ethernet cable is paired differently than an E1/T1 cable. call generation Perl script for you to try. You would need one E1 crossover cable: (This is simple to construct from a CAT5 Ethernet patch cable). I can make the problem occur with only 30 sending and receiving channels on the same system... THANKS! -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
I think the key idea is to help newbies along as much as possible so they don't have to revert to the list to obtain answers to their questions. This will reduce list bandwidth, possibly significantly. I see that we already have a four line digium footer on each and every message. With judicious re-arrangement, so as not to expand the footer significantly, we put in a pointer to an FAQ, which in turn points to valuable documentation resources such as voip-info and xvoip, plus answers (or links) to common questions such as the moh issue, echo, ... We will then reap a bonus. We reduce the recurring traffic, then we can free up bandwidth for the now being bandied about 'business' list, which in itself, should be content heavy. I'd actually prefer to keep it here, since I obtain all my primary info here anyway, and managing another list is really my idea of a good time. If we start to see 25% or 50% coverage on 'business' related stuff, then would be a good time to slice it off on to its own self-sustaining forum. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 I yammered: of public resources such as this list. put that FAQ in the list subscribe welcome message or the list sig or the asterisk README or handbook or all of the above... er, in case it wasn't obvious: s/that FAQ/a link to that FAQ/ I am all for svng prcs bndwdth. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAQ, Documentation, How-to, etc
whatnot do not help at all when your first exposure to the subject thread is someone saying It's already been answered, check the archive and that message is 6 months old! Worst of all there are no hints on searching for this information. You know in such situations it's helpful Perhaps the solution for this is for the person who is tempted to take the easy way out and say 'check the archives', actually links to the archival message(s) in question. That would be helpful in the max. An added helpful bit would be to include the google search terms they used for finding that item (if applicable). And if we had enough of these references, an index page could be set up to point to these 'well known references', or hidden gems. A reference to this index would be included in the FAQ. Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Card Types for Europe
Title: ISDN Card Types for Europe What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any others? Which driver is appropriate? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] Zultys.
Title: Zultys. Is anyone familiar with http://www.zultys.com/index.htm. Do they use Asterisk? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] D Channel Bonding
Title: D Channel Bonding Are the Digium T1/E1 cards capable of D channel bonding for PRI? As in one D channel can service two more PRI lines? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] Canadian VoIP termination?
By the end of next week, we'll be able to offer IAX2 service for Vancouver, Toronto, Hamilton, Montreal. End of this month or so: Calgary, Edmonton, Ottawa, Winnipeg. Sometime in December: Windsor, Kitchener and London. By mid next week, Charlotte NC should be on line. Other centers, as listed at: http://voice.oneunified.net/coverageareas.html will be available as needed. All with local inbound/outbound with DID service. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dana Martens Sent: November 12, 2003 14:41 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Canadian VoIP termination? 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! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sofphone Recommendation, was Where can i get the g.723 codec?
Speex works perfect with IAX but not that crack headed x-lite stuff. Can anyone make any recommendations, from personal experience, on a good softphone that has good look and feel, and of course reasonable sound quality, and works with Asterisk? Ray [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XTEN-Lite Bad sound!
For some of these noise problems, it is good to do a jitter analysis. I found it to be the cause of the problems I was having with my Cisco phone and a X-Lite client. Also, I found that the X-Pro client is better at voice delivery than the X-Lite client. Anyway, if you have tcpdump available on your asterisk server, use it to collect some timing statistics. For example, tcpdump -i eth0 -T rtp dst host x.x.x.x and dst port 8000 (if I have this wrong, 'man tcpdump' will help you with the various options). Should decode the rtp stream on eth0 to port 8000 on x-lite. Pipe the output to a file. Should look like: 14:44:43.618422 10.1.1.60.8000 sip.oneunified.net.15958: udp/rtp 160 c0 4208 758720 14:44:43.619033 10.1.1.60.8000 sip.oneunified.net.15958: udp/rtp 160 c0 4209 758880 14:44:43.665234 10.1.1.60.8000 sip.oneunified.net.15958: udp/rtp 160 c0 4210 759040 14:44:43.665521 10.1.1.60.8000 sip.oneunified.net.15958: udp/rtp 160 c0 4211 759200 14:44:43.665925 10.1.1.60.8000 sip.oneunified.net.15958: udp/rtp 160 c0 4212 759360 This sample doesn't correspond to the command line shown, but gives you an idea. http://www.erg.abdn.ac.uk/users/alastair/tcpdump.html has some descriptive comments about what you see. Anyway, now import the file into a spreadsheet. Do deltas between the times (first entry on each line, I usually discard the 14:44: stuff and focus on the nn.n portion) on each line (line 2 - line 1, line 3 - line 2, ... ). Make an x-y chart with the numbers in the second last column, 4208, ..., as x and the deltas as y. You may find some interesting results in terms of packet delivery timing. Excellent packet delivery occurs with a max jitter of 20 to 40 ms. Cisco phones have a dynamic jitter buffer up to 150 ms. My problem before was that an X-Lite softphone on a slow machine or on a really bad internet connection was averaging jitter between 80 and 200 ms. It was giving my phone a really hard time. Hence the pops and periods of multi second silence. This may not be at all useful for your problem, I just used it as an excuse to update people on my problems. ;-) Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason A. Pattie Sent: November 3, 2003 16:49 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] XTEN-Lite Bad sound! -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 WipeOut wrote: | Has the bad quality started just recently? Has it ever worked nicely for | you?? It depends. :) | If either of these is yes.. | | What has changed in your setup? Have you recently upgraded to a newer | CVS?? I have a fairly recent version of Asterisk, but X-Lite has always worked like a charm when accessing the PBX functions directly (i.e., leaving voicemail, listening to menus, checking voicemail, etc.). However, whenever I receive or make a call through the X100P card, the people on the other end say that I am garbled and the audio is very choppy and echoy. I also notice an on again, off again kind of beating in the audio stream whenever I am making a call. There is a period of exactly 4 beats that seem to be more or less the same volume, very low and in the background, that span approximately less than a 2 second period. Then there is a period of silence for less than 2 seconds, and then the cycle is repeated. | I don't have an answer for you but at least it may stop others falling | into the same problem if somthing can be identified as the cause.. | | later.. Thanks. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/ps0uuYsUrHkpYtARAkHwAJ4+aVbqZ4cxepsagqYFJC789E6AWACfY53o DmehCt4pF+PvJO2RR+qcN88= =5N/n -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RH9 or RH8?
Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... I recall in the archives somewhere, and through someone's post earlier today, that there is some sort of problem with RH9 with Zaptel (hardware) drivers and that RH8 is preferred. Do you recall what kind of problem? The only problem I have is an annoying echo that I haven't yet gotten rid of. Quoted from Paul Cheng, at 5 pm yesterday: I can also confirm chan_h323 and g.729 work well to 5300s, but we had to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP which works fine--even to i4l interfaces. Quoted from Dustin Wildes, Wed 2003-10-29 10:07: All of the setup is running on RedHat 8.0 - no other router or CSU is needed. Don't use RedHat 9.0 yet in this setup since the ZAPTEL_NETWORK flag will not compile with the new implementation of HDLC in the kernel. -- when discussing T1 card with voice and data transitting on it. Other than that, RH9 is fine. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card
All of the setup is running on RedHat 8.0 - no other router or CSU is needed. Don't use RedHat 9.0 yet in this setup since the ZAPTEL_NETWORK flag will not compile with the new implementation of HDLC in the kernel. I believe that when you use up2date on both RH8 and RH9, you end up with the same version of Kernel. So how do you differentiate RH8 and RH9 in terms of this flag? Or do you not use up2date to get and latest kernel and source? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 and G729: Another sad tale
I can also confirm chan_h323 and g.729 work well to 5300s, but we had to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP which works fine--even to i4l interfaces. I believe that when you use up2date on both RH8 and RH9, you end up with the same version of Kernel. So how do you differentiate RH8 and RH9 in terms of this issue? Or do you not use up2date to get the latest kernel and source? On Friday, October 31, 2003, at 01:57 AM, Jeremy McNamara wrote: John Todd wrote: I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
Title: Message By default X-Lite now has silence supression turned on.. Go to Advanced System Settings Audio Settings Silence Settings and change Transmit Silence to "Yes".. I played with this. Still problems. Where do I check for PT 13 or 19? Could be comfort noice ? Check for PT 13 or 19 Does any one else have problems with huge, random silence breaks between an X-Lite and Cisco 7960 SIP phone? Both are running g.711. Softphone to/from softphone works, softphone to/from iax2 works, iax2 to/.from cisco phone works. -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
However, voice as heard on X-Lite is just fine from Cisco, but voice as heard on Cisco from X-Lite has random silent breaks of one or two or three second duration on a very regular basis. Any ideas on how to get rid of the random silent breaks? X-Lite (build 1082 and possibly later) and choppy sound: In X-Lite go to -- Advanced Setup -- Audio Settings -- Silence Settings -- set Transmit Silence to yes to solve this issue. Sorry, this didn't fix the problem. I put my microphone up to a continuous music source, and the drop outs still occur. I checked, the auto gain controls are off. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Making a Skinny phone talk to Asterisk
Title: Making a Skinny phone talk to Asterisk I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. How do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure about this as the OS79XX.TXT makes newly arriving phones load the SIP image). Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] Quick Question
Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... I recall in the archives somewhere, and through someone's post earlier today, that there is some sort of problem with RH9 with Zaptel (hardware) drivers and that RH8 is preferred. Asterisk works VERY well under RH9. Be sure to install kernel-sources and keep them up-to-date along with the rest of the system. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
Title: Huge silence breaks between Cisco 7960 phone X-Lite Does any one else have problems with huge, random silence breaks between an X-Lite and Cisco 7960 SIP phone? Both are running g.711. Softphone to/from softphone works, softphone to/from iax2 works, iax2 to/.from cisco phone works. However, voice as heard on X-Lite is just fine from Cisco, but voice as heard on Cisco from X-Lite has random silent breaks of one or two or three second duration on a very regular basis. Any ideas on how to get rid of the random silent breaks? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] Absolute Minimum Installation Packages
With development tools, I've had the installation down to about 800 and some odd MB. If you do the install without development tools and without kernel source, you should be able to get it down to 600 MB or less area (kernel source is over 100 MB by itself. The basic way you do this is when you install Redhat, turn off all packages, then start selecting the ones you really need. The dependency generator will then select the related ones for you. And if you've missed some special ones, you can always re-add the rpm's after installation in complete. You are going to need two machines: one with a bunch of extra packages installed so you can compile Asterisk and create an RPM, and another one without the excess stuff on, where you simply install the asterisk rpm. I've kept the Kickstart file that has my typical install. I have a few extra packages in it for network management and troubleshooting. Regards, Ray Burkholder -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen Sent: October 30, 2003 20:44 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Absolute Minimum Installation Packages JR Richardson wrote: I'm trying to get the total Linux/* installation size as small as possible. I'm wondering if anyone has looked at the installed packages list from the Redhat installation [rpm -qa] and has parsed out all packages not needed for * to run. I follow the custom install guide from Andy Powell but the installation yields 948+ Meg with 340 installed packages. I'm sure most of those packages can be eliminated. If the installation can be reduced to below, say 600 Meg, then there's an opportunity to harden * into a KNOPPIX Customization. BTW, has anyone already tried to produce a KNOPPIX * Customization? Wierd that I had actually started to just think about this earlier today... :) Unfortunately this is going to be nothing that I can do to help at this point.. I am really quite budgeted for time, and I can barely work on the other things I have somewhat commited to. I'll be so glad when I'm back in school, and hopefully have some more time to work on this kind of stuff. Keep me posted, I have a couple of idea's that this could be useful for (if anything, just what the minimum packages are for a RH install) Thanks! -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie hardware question
The TA-750's are a bit big. The CarrierAccess Bank I is a 1 U unit, which I understand works nicely with Asterisk. Probably about $750 refurbished. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805) Sent: October 30, 2003 13:29 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Newbie hardware question I have 6 750s attached to my pbx server. The 850s have a lot of functionality you don't really need. -sb -Original Message- From: TC [mailto:[EMAIL PROTECTED] Sent: Thursday, October 30, 2003 1:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie hardware question You will want either a T100P, or a T400P. Then you will want a channel bank that is modular enough to add a FXO card to it. With 5 lines of FXO, the Adtran units will be a good choice as they are in units of 6 lines. hmm what adtran unit is that the most popular adtran cb's used with * are the ta-750/850 and the slots are provisioned with 4 channels per slot/card total 6 slots per unit, 24 channels total ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card
Title: FW: Voice/Data mixed routing over Digium E1/T1 Card The documentation mentions that the Digium channels can be split into some voice channels and the remainder of the channels used for routing IP traffic. Does any one have this in use in conjunction with Asterisk? Does it work well? Would you recommend it for a production server? Obviously, if this works, this makes for a cost effective platform where you obtain one E1/T1 to a provider, and they can provide TDM and data over the one circuit. No separate router required. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] Answering Machine Detection
Might want to write a new energy detector algorithm in dsp.c though based on a wideband/low Q resonator approach (move the pole way in towards the origin) as opposed to narrow band goertzels (pole on the unit circle). More robust for this type of work. Where does one go to learn this terminology and the math to implement it? -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 signaling/Softswitch
I spoke with someone today who is interested in an IP Centrex solution that starts with about 3500 extensions in a multi-tenant application. And growing from there. I'm wondering about scalability of Asterisk. I'm trying to put my head around how to put the whole thing together, if it can be put together. The nice thing about it is that if I can show potential, functionality, and scalability, which is something I'm starting to see (a recent contributor indicated 240 simultaneous calls), the deal will mean more development dollars for adding fine features to Asterisk. If I play my cards right, we might be able to get the engineering info from Cisco we need to make the Skinny phones work in all their true, cool functionality. And to continue with SS7 conversations, I think this gets a good tie in for SS7 for handling numerous distributed gateways and Telco interactions. So a number of questions: 1) so far, I've heard 240 simultaneous calls. Does anyone have systems that are larger? 2) does anyone have suggestions on where to go for making SS7 / Asterisk integration a reality? Obviously on a paid basis. 3) can what I'm proposing work, or am I off my rocker? Obviously there are a bunch of things like redundancy, load balancing, load management, etc that need to be engineered, but I just wanted to be sure I'm going in the right path. For instance, Jeremy, do you have statistics you'd like to publicize in terms of the number of callers you have, number of active extensions in you extensions.conf file, number of minutes/channels/... you put through your system? How much of it is Asterisk based and how much is simply gateway calls? Regards, Ray Burkholder www.oneunified.net 704 576 5101 No, you don't directly send information between PRI and ISUP message... To understand correctly this, I send a complete ISUP trace. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank with E1
I can get a hold of a number of refurbished Carrier Access Bank I (T1/FXS) units for a pretty respectable price. Will they do the job based upon your comment that they support 'Answer Supervision'? How about the Caller ID, MWI stutter that Asterisk provides? Regards, Ray Burkholder -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk online forums Sent: October 29, 2003 09:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channel Bank with E1 Sergi, I would say it depends of your budget. You can find on market different channel banks. Some of them are very expensive and have all fancy features, some of them are not so expensive and of course are missing some features. We are using CAC and New Bridge chanell banks, they are working good and no problem. When you are looking for channel bank, make sure it supports Answer supervision, it is very important feature. But I don't know what exactly are you going to do within your netowrk for 100 phones.. Do they need all features to be trasmitted like Calle ID , from outside world ? Also you can take a look into Adtran or NewBridge. IF you have more specific questions about them, please let me know. Thanks, Alexander *** XVOIP network is lunched, get your +1 777 number today. [EMAIL PROTECTED] *** Unofficial Asterisk Forums *** URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register *** - Original Message - From: Sergio Serrano Revuelto [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 8:44 AM Subject: [Asterisk-Users] Channel Bank with E1 I need connect up to 100 analog phone to a H.323 network through *. I think use TE410P, But I need to know what channel bank is better. I use E1 lines Any idea? Thanks in advance, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de DUSTIN WILDES Enviado el: miércoles, 29 de octubre de 2003 14:30 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Answering Machine Detection Thanks for all the info! So I take it I would need to either build an additional APP to asterisk like (voice_detection) or into an AGI and have that application or AGI run after the call is Answered? Fortunately it's not a telemarketing system! :-) It's an appointment reminder system for some of our employees. Calls them up and reminds them of important tasks like meetings and stuff. -Original Message- From: Michiel Betel [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 8:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Answering Machine Detection See http://resource.intel.com/telecom/support/documentation/unix/S R50_linux/ html _files/vox_feat/contents.html#TopOfPage chapter 2 for a basic insight on Dialogic does it... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: woensdag 29 oktober 2003 3:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Answering Machine Detection Humans tend to say Hello? (short burst of audio followed by silence), and answering machines tend to say I'm sorry I'm not here right now, please leave a message after the beep (long burst of audio followed by a beep and silence). So, basically you need to decide 1) what is audio and what is background noise and 2) how long should there be audio followed by silence. On Tue, 2003-10-28 at 19:25, Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering
RE: [Asterisk-Users] Trouble with 2 NIC cards
I'm just finishing the test of a solution where the Asterisk box acts as a firewall between the outside world and the inside world, but uses only a single network card. It uses the VLAN capabilities built into Redhat 9.0. As a consequence, the switch to which it is connected needs to understand VLAN aka 802.1q as well. I've found an IPTables configuration that locks the box down quite a bit. Now I'm working on installing Asterisk so it can listen on the various sub-interfaces. I think I've successfully resolved the various internal routing issues. I should know the results of this experience later today. I've got some vlan configs on my site: http://www.oneunified.net/support/ under Linux support. I hope to post the iptables config (for NAT, forwarding, and firewall) later on today. Regards, Ray Burkholder The problem should be easy enough to solve for someone who knows the internal guts. As a matter of fact, this is very important to resolve. Asterisk behind firewall is trouble and that is known already. So I decided to use the same linux box as firewall, meaning I need atleast two NICs. I wonder how others are solving this issue. I refuse to believe that no one faced similar problems cause there is no other way for a beginner to plant an Asterisk box but have two NICs or go through the NAT troubles. Moreover, there are news posts all over about SIP phones meaning others are connecting more than one NICs. Wonder why they don't have similar issue. I am sure that there are more out there who are facing similar problems so people who have solved this, please speak up and help us all. Thanks in advance. Ricky -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 signaling/Softswitch
So, the proper answer is that if you really want to implement this PRI - SS7 - PRI message, you should really be talking to your nearest CO Engineer or Telco Enterprise Business Office where they handle this all the time for enterprise call center applications. Hah. I've yet to have any luck talking to anyone that _really_ knows what's going on. Or at least anyone that knows what's going on *and* can think out of the box. I wish I could find a fone phreak that was hired by an RBOC and knows stuff from the inside and out. I would fit the bill, but I didn't get to spend enough time on the inside to get a big enough handle on interconnections. We are doing work with a regional CLEC on the east coast who have their own 5ESS. I'm able to get reasonable access to their engineers. I'm thinking that in the next few months, we may be ramping up to SS7, and will need their assistance, and they will probably be able to supply it. What I mean is that trying to work your way through the labyrinth known as Quest may not be the best way to get the job done. Work with a regional CLEC, you may get better access. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extensions Problem
You may have a file called dialplan.xml being TFTP'd to your phone. It has a number of rules in it for helping the phone to determine when it has complete number. It may need some tuning to bring it in line with what you need. I've found that the phone appears to treat the contents of the file as a hash rather than as a sorted list. That is, certain rules that appear later in the file actually get used before rules earlier in the file. I think the rules get used in a 'shortest match first' scenario. Regards, Ray Burkholder http://www.oneunified.net 704 576 5101 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phillip Jackson Sent: October 26, 2003 18:35 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Extensions Problem Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten = 1637,1,Dial(SIP/100) Exten = _NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten = _NX,2,Congestion Exten = _1NX,1,Dial(SIP/[EMAIL PROTECTED]) Exten = _1NX,2,Congestion These extensions allow us to utilize our SIP provider - ONLY when being dialed from a regular telephone attached to a Cisco ATA-186. Our Cisco 7960 only allows us to dial 4 charachters before it tries dialing. So, I assume we need to implement 9, and the number. However, when I do this, the 9 gets passed on to our SIP provider, which tries to dial 9NXX, and all goes to hell. Question - is there a way to allow 9 in the dialing plan, without having it be passed to the sip provider. Regards, Phillip -- Phillip C. Jackson [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 signaling/Softswitch
From what I've heard and learned, SS7 appears to be a meta meta signalling protocol. First we had analog lines. Then ATT started grouping 24 analog lines to form a T1. Inband signalling was used in each channel. Time studies indicate that these channels can be more effectively used if the signalling (ringing, busy, etc) is removed out of each channel and put into a common data channel, or D Channel as they call it in ISDN parlance. So for a North American PRI line, a T1 with 24 channels is sectioned off into a 1 D channel, which is used for signalling, and 23 B channels (bearer channels) which are used strictly for voice traffic. So ISDN/PRI lines use meta signalling to control the voice channels. SS7 is strictly a signalling and control protocol. It carries no voice traffic, but controls how voice is routed between locations. For instance, some one picks up the telephone and dials a number across the country. The local telco switch signals the telco switch at the other end to ring the destination phone. This signalling is handled by SS7. If the destination party picks up, it is at that moment when the the source telco and the destination telco open up a voice circuit and connect the two parties. If the destination party does not pick up, and the source party hangs up, no voice channels have been opened up, and the telco enjoys a cost saving by not having to dedicate a resource to the conversation, such as like back in the good old days of RBS (Robbed Bit Signalling) T1's. SS7 also handles the CLASS series of value-add signalling services. SS7 is therefore useful for handling the signalling on large channel volumes (loosing a B channel to a D channel in a PRI for every 24 channels is expensive overhead), and is good for geographically distributed dialling plans. So, to wrap up, SS7 is a meta meta signalling protocol. It controls the signals going down a PRI which signals the PBX on what it needs to do with the call. If any others on this list can contribute their thoughts and experiences, it would be greatly appreciated. Regards, Ray Burkholder http://www.oneunified.net 704.576.5101 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brad Waite Sent: October 27, 2003 15:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch CW_ASN - Gus wrote: Anyway, in certanly implemetations you don't need CCS7 to connect to CO. You always can connect with PRI... same speed and same functionalities to user side. In fact, CCS7 is the support for ISDN-PRI avanced features. If you could connect with Lucent 5ESS you can have a PRI treated as route... Gus, I'm not following you here when you say, ...you can have a PRI treated as route... Can you clarify? I'm trying to determine what AIN features may be available on a PRI D channel. I know the D channel is a near extension to SS7, but I don't know what subset of queries/commands are available between the two. Brad Waite W Cubed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on Hold
Some notes can be found at
http://www.oneunified.net/support/asterisk/index.html
Regards,
Ray Burkholder
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: October 27, 2003 15:25
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Music on Hold
I would appreciate it if anyone can give me some instructions
on how to
install mpg123.
Thanks in advance,
Kang
Phillip Jackson, Director
of ITTo:
[EMAIL PROTECTED]
[EMAIL PROTECTED] cc:
Sent by:
Subject: [Asterisk-Users] Music on Hold
[EMAIL PROTECTED]
.digium.com
10/26/2003 02:14 AM
Please respond to
asterisk-users
Having a weird issue with on hold music ... I do have mpg123
installed.
When requesting extension for testing, which is setup as:
exten = ,1,Answer ; Answer the line
exten = ,2,DigitTimeout,5 ; Set Digit Timeout
to 5 seconds
exten = ,3,MP3Player(${MP3ROOT}/sample-hold.mp3)
I recieve this err:
-- Executing MP3Player(SIP/100-26af, /sample-hold.mp3) in
new stack
WARNING[1217602880]: File rtp.c, Line 374 (ast_rtp_read): RTP
Read error:
Resource temporarily unavailable
NOTICE[1217602880]: File app_mp3.c, Line 80 (timed_read):
Selected timed
out/errored out with 0
Not sure what's up...
Phillip
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Scanned for viruses and dangerous content at
http://www.oneunified.net and is believed to be clean.
--
Scanned for viruses and dangerous content at
http://www.oneunified.net and is believed to be clean.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SS7 signaling/Softswitch
You are correct in regards to what SS7 is and does, I thought it would be helpful to bring other users on the list up to speed. ;-) Some additional SS7/VoIP integration info from a 3Com perspective can be found at: http://www.c7.com/ss7/whitepapers/3com_ss7_intelli.pdf What I was inquiring about was Gus' comment about a PRI treated as route on a 5E. I'll have to defer to Gus for an answer on that one. I'm also trying to find out what types of SS7/AIN features may be available over a PRI D channel. For instance, message waiting indication (MWI) signals are sent interoffice over SS7. Could one formulate a packet that's sent over a PRI D channel that would end up in a remote switch via SS7? The MWI you mention is probably part of CLASS services, and is probably a function of AIN on an SS7 SCP (Service Control Point), to which a Telco's switch is connected. For some light reading on AIN, SCP, TPAN and related bits, this page has some interesting info: http://www.ulticom.com/html/products/ss7/ain.asp It doesn't directly answer your question, but I would guess that the Class 5 switch has to make some sort of translation between what happens in the D channel on a PRI and what it needs to communicate over its backend SS7 network. I see two proper solutions: a) implement SS7 directly so you have access to the signaling network for your application, or b) just handle the communications over the ip network in a converged network scenario. By the way, why do you ask the question of the D channel message? What is your application? So, the proper answer is that if you really want to implement this PRI - SS7 - PRI message, you should really be talking to your nearest CO Engineer or Telco Enterprise Business Office where they handle this all the time for enterprise call center applications. On the other hand, maybe Gus could contribute a regular tutorial on how he's got various things interconnected. The more the info, the better. Gus once asked if we want the plethora of info he can provide. I vote yes. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need to partner with someone in Hampstead London on a deal
Title: Need to partner with someone in Hampstead London on a deal I have made a contact with a company in London looking for various voip and ip telephony services. Is there someone local there who may help facilitate this opportunity? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] AGI problem (crash) in RH9
You may wish to upgrade your kernel to 2.4.20-20.9 through 'up2date'. Regards, Ray Burkholder -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Gudino Sent: October 17, 2003 16:17 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AGI problem (crash) in RH9 Hi Ivar, Try putting this line before launching asterisk: export LD_ASSUME_KERNEL=2.4.1 Best regards, On Thu, 2003-10-16 at 06:48, Ívar Ragnarsson wrote: Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm using Netmeeting to test -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP vs SCCP vs XML
As to prior comments about SCCP documentation: if you'd like to help contribute to the SCCP channel project, it seems far from 'aborted' at the moment. Check out http://sourceforge.net/projects/sccp/ and download the channel. Compile, test, send bugs, submit code. The web site indicates that This Project Has Not Released Any Files. Am I not seeing something? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is Asterisk ready for real use?
The Cisco SIP phones have a second voice channel available for a paging type of implementation. Now the problem is simply of finding someone and some time to see if it can be made to work with Asterisk. Ray Burkholder One Unified 519 570 0689 x2002 *'s paging solution is a bad solution in light of today's phone systems. If you need it anywhere but in a barnyard, you should plan on selecting a different phone system. It might work in a Sams, but certainly not in an office. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LAN switches with PoE? PoE phones?
To follow up on this, the Cisco Switches and the Cisco phones will work together to create two vlans: a prioritized vlan for the phone traffic, and a secondary 10/100 link for a computer which can be attached to the phone's second switched ethernet port. Some config is needed in the switch and router to make this happen properly. I have it running well with 79x0 phones, 3550 switch, and 1751-V router. What this means is only one switch port is needed to run both a phone and a computer. This helps on the switch/phone ROI calcs. The PoE can also be used to power wireless access points. Ray Burkholder 519 570 0689 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Brown Sent: August 17, 2003 13:52 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] LAN switches with PoE? PoE phones? Hi Mike, Cisco makes PoE switches, either at the Cat 29xx or the Cat 35xx levels. The 29xx don't have gige uplinks, but the 35xx's do via GBIC interfaces. Meaning you will also need to get a GBIC media converter depending the media type (copper fiber, etc) And of course Cisco makes PoE based phones 7940 7960 which work well with * Grandstream currently requires a wall-wart, but later models are suppose to use PoE as well. I'd personally put the phones on their own subnet so that ACL filtering at the router will be easier, static IP alloc will be easier. hope this helps john brown chagres technologies, inc sip: [EMAIL PROTECTED] ptsn: (01) 505 830 1200 USA On Sun, Aug 17, 2003 at 12:44:43PM -0500, Mike Ciholas wrote: Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of power hubs, but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be special for IP phones in anyway? QoS support? Managed? Also, are there PoE phones that work with *? Most I look at seem to be powered from AC wall blocks. We'd like to centralize the switching and power and provide a UPS so the phone system works when the power goes out. [Apologies, I'm new to this whole concept of IP phones and *.] -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner at One Unified, and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Center RFP
I have an opportunity for a 50 seat call center requiring outbound dialling, inbound call queuing, agent management, call recording, call/skill matching, call list management, reporting, IVR, management call whisper, etc. Are there any * resellers on this list who are capable of handling a sophisticated installation such as this? If so, please contact me off list. Regards, Ray Burkholder 519 570 0689 x2002 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
