RE: [Asterisk-Users] What's in ${EXTEN} ? Why does voicemail prompt for an extension?
Excellent answer. Thank you very much.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andreas Frackowiak
Sent: Saturday, May 15, 2004 1:32 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] What's in ${EXTEN} ? Why does
voicemail prompt for an extension?
Why does voicemail prompt me for an extension instead of
just asking
my password?
Because there is no Voicemailbox 99 in that context in your
configuration.
[voice-mail]
exten = 99,1,VoicemailMain([EMAIL PROTECTED])
exten = 99,2,Hangup
In your example, $EXTEN will always be 99, because that is
the extension.
If you would like to have the 99 as a prefix for the
following voicemailbox number you could do something like:
exten = _99.,1,VoicemailMain(${EXTEN:[EMAIL PROTECTED])
exten = _99.,2,Hangup
And then 99123 would go directly to Mailbox 123 (if it exists).
regards
Andreas
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[Asterisk-Users] indications.conf
Hello I am looking for Czech (Czech Republic) country support to indications.conf Have you ever seen it anywhere ? We are a small country in middle Europe :) thank you -- Vit Bohacek ___ 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] 2.05a firmware
Please take a look at http://www.snom.com/faq/FAQ-04-04-28-ut.pdf. It describes the hand/headset policy! It was supposed to be an improvement... CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Thursday, May 13, 2004 7:35 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware They also made a bad (for me) change. In 2.05a the phone would ring normally and I could press OK for headset or pick up the handset for handset. Now, when headset is enabled the phone only rings in the headset (i.e. not through speakerphone). --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ 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 ___ 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] 2.05a firmware
8 kHz 16 bit/sample (linear) mono WAV files. CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Thursday, May 13, 2004 7:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Does anyone know what kind of file needs to be uploaded for the custom ring tone? --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ 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 ___ 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] Grandstream phone from speaker phone back to handset
I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible to switch back to handset mode. -- David Kwok Tel: 612 82315701 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
RE: [Asterisk-Users] 2.05a firmware
While we're at the 2.05 firmware - the DTMF handling on the Codec configuration page have disappeared. I assume this is because the phone now got some kind of default behaviour based on the codec. Can you describe that behaviour? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christian Stredicke Sent: 17 May 2004 15:21 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware 8 kHz 16 bit/sample (linear) mono WAV files. CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Thursday, May 13, 2004 7:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Does anyone know what kind of file needs to be uploaded for the custom ring tone? --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ 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 ___ 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
Re: [Asterisk-Users] Grandstream phone from speaker phone back to handset
Hi David, I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible to switch back to handset mode. Simply pickup the handset, works fine for me. :) -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ 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] Tones...
Hi all. I was wondering about how to set different tones, in the Asterisk I use indications.conf, in the Cisco ATA-186 I use the webinterface. How do I set tones in the Grandstream, handytone, Cisco 7960 ? The US tones does not apply to all countries. (Unfortunatley) /Mike ___ 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] cdr-csv / Zaptel BRI / full number not stored (missing leading zeroes)
hi, do you have nationalprefix=0 internationalprefix=00 in your zapata.conf? best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am So, 2004-05-16 um 16.21 schrieb Frederic Olivie: Hi, I'm using a ZaptelBRI card. It works fine. But I have a small problem with call logs. The leading zeroes of the external calling party are not stored (e.g. : 0140302010 will be stored as 140302010). Same for international numbers for which 00 will be stripped out. I would not mind if the cdr record would give me an indication of the call's origin (national or international), but it does not. The goal here is to implement a basic missed call web service that would allow my users to generate a call back. -- Frdric Olivi (Alf) @ Club-Internet Don't SCREAM, It hurts my eyes ! Ne CRIEZ pas, a fait mal aux yeux ! Alf, March 2001 ___ 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
[Asterisk-Users] Asterisk Proxy Type
Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace ___ 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] X100P Ireland Red Alarm (AR Tarzi)
On 16 May 2004, at 23:17, Aaron Clauson wrote: maybe they just do it different here in Ireland. They do it differently in Ireland. To get a functioning modem cable you need to have a cable that takes the outside two wires and crosses them to the inside two wires. We had hundreds of these made for us when we were selling computers for internet access. How this relates to the X100P is not something I know anything about but this information may at least confirm that Eircom's RJ11 wiring is 'an Irish solution to an Irish problem. :-) Good luck. Stephan ANU Internet Services Galway, Ireland http://www.anu.net ___ 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: Asterisk Proxy Type
may not correct but i tought * is not a proxy. Ignace CARIA wrote: Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace ___ 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
Re: [Asterisk-Users] CAPI-SIP broken incoming audio
G'day list, Follow-up post to the one I sent last week regarding bad calls between SIP and ISDN. On Fri, 14 May 2004, Vic Cross wrote: To me, it looks like a variation of the SIP RTP timestamp problem (yes, my 7960 is at 6.3 code), but the problem exists on the ATA-186 too and I don't have any other issues with that (not even SIP-IAX, where the 7960 is really bad). I've done some crawling over ethereal traces, and have found the problem to indeed be bad timestamps in the RTP payload from *. I was advised (by JerJer on the IRC channel, thanks!) to update to latest CVS-HEAD, but the problem has recurred. What's happening is that the RTP frames from * are all going out with the same timestamp, which is causing my timestamp-sensitive 7960 to barf and ignore the incoming audio stream (it's interesting that X-Lite is largely trouble-free in this scenario!). On calls that work well, the RTP frames from * have a timestamp starting at 160 and incrementing normally (160,320,480...). On the bad calls, the timestamp is a very large number (4015105112, for example) and does not increment. So, next step is to look thru the code and find where the timestamp is initialised and incremented. In rtp.c, I found a couple of instances of a variable declared as signed int being used to hold the return value of the unsigned int function calc_txstamp(), but only time will tell if this fixes the problem (as it still takes anything up to a couple of days after an * restart before the problem occurs). What bugs me the most is that I can call SIP-SIP to the 7960 (from my ATA, for example) and the RTP timestamp is incremented correctly. Immediately afterwards I will call from ISDN, and I get bad timestamp. Which would imply that the generation of the timestamp is related to the source of the call, but I'm %$*ed if I can find where -- it seems to be time-of-day dependent, but nothing else (I can see where the codec seems to affect timestamps, but in my test case I'm using the same codec as the ISDN call). Finally, should I take this to asterisk-dev? Cheers, Vic Cross PS: Now that I can show a real problem I'm going to file a bug report too. ___ 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] 2.05a firmware
(Forwarded:) - Original Message - From: Usman Tahir [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: Christian Stredicke [EMAIL PROTECTED] Sent: Monday, May 17, 2004 11:25 AM Subject: Re: [Asterisk-Users] 2.05a firmware Hi, In firmware release 2.05c and later, the user has an option to select the ringer device when headset is in use. The default is Speaker as before. But for call center and other closed environments, you can also select Headset. In such a case ringing will be played on the headset. You'll find the appropriate setting in Settings/Miscellaneous/Audio. Try the latest beta from http://www.snom.com/download/share/snom200-2.05c-SIP.bin. Regards, Usman. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Thursday, May 13, 2004 7:35 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware They also made a bad (for me) change. In 2.05a the phone would ring normally and I could press OK for headset or pick up the handset for handset. Now, when headset is enabled the phone only rings in the headset (i.e. not through speakerphone). --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ 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 ___ 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] OMG THE SKY IS FALLING!! NOT!!!
The gun issue highlights the absurd nature of Louderback's opinions. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Friday, May 14, 2004 3:20 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OMG THE SKY IS FALLING!! NOT!!! Although I agree with eveything you said, I would still leave out the gun part. Its too controversial, almost like bringing aborting into the issue. Totally unrelated. Just my 2 cents and COMPLETELY OFF TOPIC - Original Message - From: Ronald R. McDaniel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 14, 2004 11:55 AM Subject: RE: [Asterisk-Users] OMG THE SKY IS FALLING!! NOT!!! I sent the following to the great Mr. Louderback: Mr. Louderback, If I was your friend, I would request to remain nameless too. I would like to reply to a couple of your statements in the article Security Holes Make VOIP a Risky Business. Security holes make any type of technology risky business. A small company using wireless two-line analogue sets are risky since the conversation can be picked up with a device purchased from Radio Shack. As in most situations, the technology isn't the problem, it's the untrained individual that doesn't incorporate security measures when implementing technology. Many of the VOIP solutions that we are putting in place are running over private networks that are not open to the public ( ie frame relay and point-point leased line). You probably believe that guns kill people too. One last note, VOIP isn't new technology, you are a few years late with your announcement. If your article has any positive outcome, I hope that it may encourage those implementing any solution to seek professional assistance. Sincerely, Ronald R. McDaniel Southern Computer Services, Inc. [EMAIL PROTECTED] (251) 444-3136 office (251) 446-3137 fax (251) 294-1202 cell Ronald R. McDaniel Southern Computer Services, Inc. [EMAIL PROTECTED] (251) 444-3136 office (251) 446-3137 fax (251) 294-1202 cell ___ 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 ___ 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] Power alarm on module 1, resetting.
Yup Juan J. Sierralta P. wrote: On Sat, 2004-05-15 at 12:22, Michael Welter wrote: I've gotten several Power alarm on module 1, resetting since I installed a quad FXS TDM400 card. Dell 400sc. Does your motherboard have the A-B-C-D LEDS above the keyboard/mouse connectors? I suppose you plugged the power conector from the power supply to the TDM400 board to feed +12V. ___ 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 -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ 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] CAPI-SIP broken incoming audio
Follow-up post to the one I sent last week regarding bad calls between SIP and ISDN. On Fri, 14 May 2004, Vic Cross wrote: To me, it looks like a variation of the SIP RTP timestamp problem (yes, my 7960 is at 6.3 code), but the problem exists on the ATA-186 too and I don't have any other issues with that (not even SIP-IAX, where the 7960 is really bad). I've done some crawling over ethereal traces, and have found the problem to indeed be bad timestamps in the RTP payload from *. I was advised (by JerJer on the IRC channel, thanks!) to update to latest CVS-HEAD, but the problem has recurred. What's happening is that the RTP frames from * are all going out with the same timestamp, which is causing my timestamp-sensitive 7960 to barf and ignore the incoming audio stream (it's interesting that X-Lite is largely trouble-free in this scenario!). On calls that work well, the RTP frames from * have a timestamp starting at 160 and incrementing normally (160,320,480...). On the bad calls, the timestamp is a very large number (4015105112, for example) and does not increment. So, next step is to look thru the code and find where the timestamp is initialised and incremented. In rtp.c, I found a couple of instances of a variable declared as signed int being used to hold the return value of the unsigned int function calc_txstamp(), but only time will tell if this fixes the problem (as it still takes anything up to a couple of days after an * restart before the problem occurs). What bugs me the most is that I can call SIP-SIP to the 7960 (from my ATA, for example) and the RTP timestamp is incremented correctly. Immediately afterwards I will call from ISDN, and I get bad timestamp. Which would imply that the generation of the timestamp is related to the source of the call, but I'm %$*ed if I can find where -- it seems to be time-of-day dependent, but nothing else (I can see where the codec seems to affect timestamps, but in my test case I'm using the same codec as the ISDN call). Finally, should I take this to asterisk-dev? Vic, It sounds like you've nailed the problem with the signed int statement. However, I'd suggest you open a bug report on this (rather than using list mail only) to get it some attention and tracking. It is very likely that other rtp channel drivers have the same issue as well. (We know it was a problem for iax/gsm - sip/rtp.) As for the Cisco dropping packets with uneven timestamps, that issue is totally unrelated to which codec is used; it affects all. In researching the Cisco bug tracking list, it would appear this particular problem is rated as a Sev 6 (lowest) and unless folks with smartnet maintenance contracts start pushing to increase the Sev level, its not likely to get fixed anytime soon. Sev 6 is basically considered cosmetic and not service impacting. Rich ___ 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] recommended hardware for quad E1 system
Hi All, Could anyone tell me which is the recommended hardware to a system running voicemail and conference, attending four E1 trunks and, another, attending only one E1? Can I use a PIII 850Mhz? Thanks in advance. Robert Almeida
[Asterisk-Users] Some thougts about implementing native 3-way calling and attended transfer
As I understood, Asterisk has a lot of features but lacks native 3-way calling and attended transfer. It would be great to have these features available to a simple IAX phone. I wonder how this could be implemented in Asterisk without asking for a patch. It should be possible with parking, conferencing, AGI and the manager interface. The extension 77 could be used by the attendant to blindly park the call. #77 would launch the command exten = 77,1,ParkAndAnnounce(|7200||). This seems to work fine. Let's imagine we have a program (MIP) connected to the manager interface. A proper parsing of the event messages sent by the manager interface should find out which extension initiated the parking, who is parked, in which slot. Now the attendant would ring the third party and transfer him either to 76 (attended transfer) or either to 75 (3-way calling). 76 would launch an AGI script that would tell the program (MIP) to redirect the third party to the second party (parked one). MIP would identify the parked call as the last call parked by the attendant. 75 would tell the program (MIP) through AGI to create a dynamic conference and drop in the three parties. This conference should turn down to a normal call when the number of participants turns down to two. I'm not sure this last thing is feasible. Mwell, I'm not sure the whole thing is feasible tan bien. But if it works it could be also used for many other things, like auto callback on failed transfer, DND, auto redial ... ___ 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] 4 X100P + 1 TDMP400(4 FXS): Only by miracle?
Hi folks, I'm trying to make an * PBX for a customer using 4 X100Ps and 1 TDM400p(4FXS). The problem I'm facing is to make one unique IRQ for each PCI slot/board since shared IRQs create all kind of weird noises and echos. Anybody got any workaround for that? Any recommended motherboard to accomplish that ? Currently, I'm playing with an ASUS A7V600. Thanks for any tip, Isamar ___ 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 Proxy Type
Ignace CARIA wrote: Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Asterisk is a no-SIP-proxy-at-all :-) Again: Asterisk is not a SIP proxy. A SIP proxy doesn't handle media streams and doesn't terminate or originate calls. Asterisk does. Asterisk is a stateful SIP UAS (User Agent Server) or UAC (User Agent Client) that has some characteristics of a stateful SIP proxy, but in most cases no similarity at all. Asterisk is a SIP registration server, though. /O ___ 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] CAPI-SIP broken incoming audio
On Mon, 17 May 2004, Rich Adamson wrote: It sounds like you've nailed the problem with the signed int statement. markster doesn't think so. Apparently this is normal ;) However, I'd suggest you open a bug report on this (rather than using list mail only) to get it some attention and tracking. Did so, got bounced. I should have checked the bugtracker before I lodged a bug report. It didn't take me long to find 1284, which is a similar report and describes a problem in chan_capi. Kapejod, any status on chan_capi 0.3.2? Cheers, Vic Cross ___ 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 7940/7960 users
Those of us that use Cisco 7940/7960 sip phones know that we've been impacted by two very different changes that have occurred over the last couple of months. First, when cisco created sip v6.x code, they implemented a new DSP (as well as other software changes) that effectively drops any incoming rtp packet that does not have even timestamps within the rtp packets. The dropped packets cause choppy audio and significantly impacts quality. The issue is unrelated to which codec one might use. Second, an ongoing programming effort has been happening within * to tie the rtp timestamps together when the rtp traffic crosses channel types (iax/gsm to sip/rtp, isdn/capi to sip/rtp, etc). It would appear the iax-sip timestamps are now functional, however Vic just found where isdn-sip is creating the same type of problem. I'm not sure why handling the timestamps in this way is needed, however only those users of Cisco phones are actually impacted by the irregular timestamps. In searching the Cisco TAC problem list, it would appear the timestamp issue has not been reported. (A search based on timestamp turned up nothing, and a search on rtp found nothing relevant.) If someone that has experienced these problems has a SmartNet service agreement could open a TAC case, it would help all 7960 users. (Now I wish I had a SmartNet agreement!) Rich ___ 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] openbsd compilation fails for recent checkout of v1-0_stable
This has been mentioned before on this list, but in order for md5.c to
compile successfully (OpenBSD 3.3), the following must change in md5.c:
#if defined( __FreeBSD__ ) || defined( __OpenBSD__ )
# include sys/endian.h
Change this to be:
#if defined( __FreeBSD__ ) || defined( __OpenBSD__ )
# include machine/types.h
# include machine/endian.h
And -E is an invalid linker option, so the Makefile needs to be changed:
ifeq (${OSARCH},Darwin)
OBJS+=poll.o dlfcn.o
ASTLINK=-Wl,-dynamic
SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
+else
+ifeq (${OSARCH},OpenBSD)
+ASTLINK=-Wl
+SOLINK=-shared -Xlinker -x
else
ASTLINK=-Wl,-E
SOLINK=-shared -Xlinker -x
endif
+endif
Also, for OpenBSD, asterisk's use of gethostbyname_r doesn't work out of the
box, so needs to follow FreeBSD's fixi, by changing the
#if defined(__FreeBSD__)
to
#if defined(__FreeBSD__) || defined (__OpenBSD__)
Also, it is likely that PROC needs to be set manually for your architecture
in the top level Makefile.
Thanks,
Tor
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Re: [Asterisk-Users] Grandstream v1.0.4.68 firmware
Duane wrote: Grandstream v1.0.4.68 firmware http://www.hellofone.com/downloads.html Seems to have loaded ok on my BT100.. Hmmm well I need to kinda figure out how to get the custom ringtones to ring on the phone... :-) ___ 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] recommended hardware for quad E1 system
On Mon, 17 May 2004, Robert Almeida waxed: Could anyone tell me which is the recommended hardware to a system running voicemail and conference, attending four E1 trunks and, another, attending only one E1? Can I use a PIII 850Mhz? Maybe for a single port E1 card, maybe. You'll definitely have problems with 4 trunks, voicemail, conference, etc., on a P3. Need at least a P4 for all of that. Maybe dual if you are doing lots of codec translation. If you are really going to put asterisk to work with 120 lines, buy the fastest box your budget will allow, 1 gig ram, SCSI, the works. If you've already got the P3, then by all means, try that out first before spending your money. You have to buy the card either way. Might as well test your configs on that. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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: Asterisk Proxy Type
Asterisk is not a SIP Proxy, It's a soft PBX. But it is a SIP registrar, and forwarding is stateful, i think. I could be wrong. Regards, Girish From: nicolas [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk Proxy Type Date: Mon, 17 May 2004 11:31:16 +0200 may not correct but i tought * is not a proxy. Ignace CARIA wrote: Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Ignace _ Post Classifieds on MSN classifieds. http://go.msnserver.com/IN/44045.asp Buy and Sell on MSN Classifieds. ___ 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] Snom200 Firmware: I only see 2.04g
Hello all, I've noticed several messages about the latest firmware on Snom's site, 2.05b, and today I see that another update is listed, 2.05c. However, when I go to the download page (http://www.snom.com/support_dl_en.php), the latest firmware version available for the Snom200 is 2.04g. Are the newest firmware releases not yet available, or am I doing something stupid? Thanks, Kanwar Systems Aligned Inc. www.systemsaligned.com ___ 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] Snom200 Firmware: I only see 2.04g
Try this one. Took me a while too. http://www.snom.com/download/share/snom200-2.05c-SIP.bin -Original Message- From: M3 Freak [mailto:[EMAIL PROTECTED] Sent: Monday, May 17, 2004 11:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom200 Firmware: I only see 2.04g Hello all, I've noticed several messages about the latest firmware on Snom's site, 2.05b, and today I see that another update is listed, 2.05c. However, when I go to the download page (http://www.snom.com/support_dl_en.php), the latest firmware version available for the Snom200 is 2.04g. Are the newest firmware releases not yet available, or am I doing something stupid? Thanks, Kanwar Systems Aligned Inc. www.systemsaligned.com ___ 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
[Asterisk-Users] Chan_capi and modem-fax
Hi all, I have just put a message from a few days with a problem with CAPI hangup. I have noticed that line with 97% of hangs, is a line connected with a ATA286 with a modem-fax. Could it be the problem? Regards, srsergio ___ 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] 2.05a firmware
Christian, That's the wonderful thing about VoIP phones... Just upload new firmware and we can have the best of both worlds! (Thanks for making the change in 2.05c.) Great phones, by the way :) --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Monday, May 17, 2004 12:20 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Please take a look at http://www.snom.com/faq/FAQ-04-04-28-ut.pdf. It describes the hand/headset policy! It was supposed to be an improvement... CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Thursday, May 13, 2004 7:35 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware They also made a bad (for me) change. In 2.05a the phone would ring normally and I could press OK for headset or pick up the handset for handset. Now, when headset is enabled the phone only rings in the headset (i.e. not through speakerphone). --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ 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 ___ 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
RE: [Asterisk-Users] ** Asterisk Sunday Morning News: Contribute to the community
You know what would be cool? A Show Variables command in the cli. It could
return something like this...
VariableScope Channel
=
CallerIDC ZAP/1-1
EPOCH G
EXTEN C ZAP/1-1
...
--Ernest
* Dial plan tips of the week: Discover the variables!
-
When creating a dial plan, there's a lot of logic to help
you. One thing that
takes time to discover is the use of variables.
Asterisk has a range of globally defined variables that you can use to
configure extensions the way you want. Here's a list:
${CALLERID} Caller ID
${CALLERIDNAME} Caller ID Name only
${CALLERIDNUM} Caller ID Number only
${EXTEN}Current extension
${CONTEXT} Current context
${PRIORITY} Current priority
${CHANNEL} Current channel name
${ENV(VAR)} Environmental variable VAR
${LEN(VAR)} String length of VAR (integer)
${EPOCH}Current unix style epoch
${DATETIME} Current date time in the format: -MM-DD_HH:MM:SS
${TIMESTAMP}Current date time in the format: MMDD-HHMMSS
${UNIQUEID} Current call unique identifier
${DNID} Dialed Number Identifier
${RDNIS}Redirected Dial Number ID Service
${HANGUPCAUSE} Asterisk hangup cause
${ACCOUNTCODE} Account code (if specified)
${LANGUAGE} Current language
${SIPDOMAIN}SIP destination domain of an inbound call (if
appropriate)
${SIPUSERAGENT} SIP user agent
${SIPCALLID}SIP Call-ID: header verbatim (for logging or
CDR matching)
Applications that works with variables
* set your own variables with the setvar() and the
setglobalvar() application.
* the gotoif() app lets you can make conditional tests on
variables and
jump to various extensions or priorities of your dial plan.
* the cut() app lets you divide a variable in two or more parts
To learn more, read README.variables in your docs/ directory.
Or visit the Wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+variables
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Re: [Asterisk-Users] X100P problem with PSTN from BOLIVIA
The silence last 60s (aprox) On Sun, 2004-05-16 at 21:38, Juan J. Sierralta P. wrote: On Sun, 2004-05-16 at 15:15, Jorge Verastegui wrote: Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. How much time ? At least in Chile when the call is originated from some call leg and the other call legs hangups (the one that didn't start the call) it must pass at least 30s to signal busy to the calling leg. ___ 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 --NOTA DE REDCETUS S.R.L. : La informacin contenida en este E-mail y sus anexos, slo puede ser utilizada por el individuo o la compaa a la cual est dirigido. Si no es el receptor autorizado, cualquier retencin, difusin, distribucin o copia de este mensaje es prohibida y sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo y borrar el mismo
Re: [Asterisk-Users] cdr-csv / Zaptel BRI / full number not stored(missing leading zeroes)
Yes. I do. Maybe though do I have a problem with my dialplan. I have : pridialplan = unknown prilocaldialplan=national This is for France. Would it explain ? Thanks. - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 11:13 AM Subject: Re: [Asterisk-Users] cdr-csv / Zaptel BRI / full number not stored(missing leading zeroes) hi, do you have nationalprefix=0 internationalprefix=00 in your zapata.conf? best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am So, 2004-05-16 um 16.21 schrieb Frederic Olivie: Hi, I'm using a ZaptelBRI card. It works fine. But I have a small problem with call logs. The leading zeroes of the external calling party are not stored (e.g. : 0140302010 will be stored as 140302010). Same for international numbers for which 00 will be stripped out. I would not mind if the cdr record would give me an indication of the call's origin (national or international), but it does not. The goal here is to implement a basic missed call web service that would allow my users to generate a call back. -- Frdric Olivi (Alf) @ Club-Internet Don't SCREAM, It hurts my eyes ! Ne CRIEZ pas, a fait mal aux yeux ! Alf, March 2001 ___ 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 ___ 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] speex
Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. As a quick example, here are the show translation outputs from * on a 2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5 (compiled with CFLAGS=-march=pentium4 and --enable-sse). Note how encoding from slin went from 25 to 15 ms. That is from the re-write of the sse optimized routines in libspeex. The % change is similar to what I saw on my p3 notebook, where both 1.0.3 and 1.1.5 were compiled with --enable-sse and -march=pentium3. (As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) spx103*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -2619 ULAW - 3 - 1 2 2 1 6 -2619 ALAW - 3 1 - 2 2 1 6 -2619 G726 - 3 2 2 - 2 1 6 -2619 ADPCM - 3 2 2 2 - 1 6 -2619 SLINR - 2 1 1 1 1 - 5 -2518 LPC10 - 4 3 3 3 3 2 - -2720 G729A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - -19 ILBC - 5 4 4 4 4 3 8 -28 - spx115*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -1619 ULAW - 3 - 1 2 2 1 6 -1619 ALAW - 3 1 - 2 2 1 6 -1619 G726 - 3 2 2 - 2 1 6 -1619 ADPCM - 3 2 2 2 - 1 6 -1619 SLINR - 2 1 1 1 1 - 5 -1518 LPC10 - 4 3 3 3 3 2 - -1720 G729A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - -19 ILBC - 5 4 4 4 4 3 8 -18 - -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ 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] Redhat 7.3 compiling problem
Firstly, amazing software, props to all the developers. I'm trying to compile the latest asterisk cvs checkout and keep getting an error which I can't solve, any help would be much appreciated - make[1]: Leaving directory `/usr/src/asterisk/stdtime' if [ -d CVS ] ! [ -f .version ]; then echo CVS-HEAD-05/17/04-16:45:34 .version; fi for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-05/17/04-16:45:34\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_crypto.o res_crypto.c res_crypto.c:25:25: openssl/ssl.h: No such file or directory res_crypto.c:26:25: openssl/err.h: No such file or directory res_crypto.c:74: parse error before `RSA' res_crypto.c:74: warning: no semicolon at end of struct or union res_crypto.c:84: parse error before `}' res_crypto.c: In function `pw_cb': res_crypto.c:101: dereferencing pointer to incomplete type res_crypto.c:103: dereferencing pointer to incomplete type res_crypto.c:103: dereferencing pointer to incomplete type res_crypto.c:104: dereferencing pointer to incomplete type res_crypto.c:106: dereferencing pointer to incomplete type res_crypto.c:108: dereferencing pointer to incomplete type res_crypto.c:109: dereferencing pointer to incomplete type res_crypto.c:115: dereferencing pointer to incomplete type res_crypto.c: In function `ast_key_get': res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:127: dereferencing pointer to incomplete type res_crypto.c:129: dereferencing pointer to incomplete type res_crypto.c: In function `try_load_key': res_crypto.c:163: dereferencing pointer to incomplete type res_crypto.c:165: dereferencing pointer to incomplete type res_crypto.c:188: dereferencing pointer to incomplete type res_crypto.c:189: dereferencing pointer to incomplete type res_crypto.c:191: dereferencing pointer to incomplete type res_crypto.c:195: dereferencing pointer to incomplete type res_crypto.c:204: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:220: dereferencing pointer to incomplete type res_crypto.c:220: dereferencing
RE: [Asterisk-Users] speex
(As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) Major improvements in speex... I'm impressed you did what I was going to work on today :P I started on this quest Saturday since grandstream support iLBC I thought I would improve it if I could and 5ms is a small improvement but still an improvement. bkw ___ 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] span_dsp faxing: segmentation fault
Hi all, I use span_dsl (opencall.org) for faxing, libtiff 3.5.7-12 is installed. But when trying to receive a fax, i get the following error: [...] Fast carrier up Coarse carrier frequency 1699.97 (66) Training error 7.516780 Training succeeded (constellation mismatch 9.892637) Fast carrier trained Segmentation fault Any ideas...? Regards, Thomas ___ 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] Redhat 7.3 compiling problem
On Mon, 2004-05-17 at 17:35 +0100, Nicholas Ruddick wrote: I'm trying to compile the latest asterisk cvs checkout and keep getting an error which I can't solve, any help would be much appreciated - res_crypto.c:25:25: openssl/ssl.h: No such file or directory res_crypto.c:26:25: openssl/err.h: No such file or directory It appears that openssl can't be found from the above. -- Dave Cotton [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
RE: [Asterisk-Users] Redhat 7.3 compiling problem
Install openssl-devl bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nicholas Ruddick Sent: Monday, May 17, 2004 11:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Redhat 7.3 compiling problem Firstly, amazing software, props to all the developers. I'm trying to compile the latest asterisk cvs checkout and keep getting an error which I can't solve, any help would be much appreciated - make[1]: Leaving directory `/usr/src/asterisk/stdtime' if [ -d CVS ] ! [ -f .version ]; then echo CVS-HEAD-05/17/04-16:45:34 .version; fi for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-05/17/04-16:45:34\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_crypto.o res_crypto.c res_crypto.c:25:25: openssl/ssl.h: No such file or directory res_crypto.c:26:25: openssl/err.h: No such file or directory res_crypto.c:74: parse error before `RSA' res_crypto.c:74: warning: no semicolon at end of struct or union res_crypto.c:84: parse error before `}' res_crypto.c: In function `pw_cb': res_crypto.c:101: dereferencing pointer to incomplete type res_crypto.c:103: dereferencing pointer to incomplete type res_crypto.c:103: dereferencing pointer to incomplete type res_crypto.c:104: dereferencing pointer to incomplete type res_crypto.c:106: dereferencing pointer to incomplete type res_crypto.c:108: dereferencing pointer to incomplete type res_crypto.c:109: dereferencing pointer to incomplete type res_crypto.c:115: dereferencing pointer to incomplete type res_crypto.c: In function `ast_key_get': res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:127: dereferencing pointer to incomplete type res_crypto.c:129: dereferencing pointer to incomplete type res_crypto.c: In function `try_load_key': res_crypto.c:163: dereferencing pointer to incomplete type res_crypto.c:165: dereferencing pointer to incomplete type res_crypto.c:188: dereferencing pointer to incomplete type res_crypto.c:189: dereferencing pointer to incomplete type res_crypto.c:191: dereferencing pointer to incomplete type res_crypto.c:195: dereferencing pointer to incomplete type res_crypto.c:204: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete type res_crypto.c:218: dereferencing pointer to incomplete
Re: [Asterisk-Users] Redhat 7.3 compiling problem
I can't believe you are still on 7.3, but whatever. On Mon, 2004-05-17 at 11:35, Nicholas Ruddick wrote: res_crypto.c:25:25: openssl/ssl.h: No such file or directory res_crypto.c:26:25: openssl/err.h: No such file or directory Those two entries alone solve whats wrong. When you don't have .h files, you are missing the -dev or -devel packages for whatever it is they belong to. In this case, you are missing the openssl-devel package. Good luck finding the one for such a vintage version of RedHat. -- Steven Critchfield [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
RE: [Asterisk-Users] Redhat 7.3 compiling problem
Looks like you need to install the openssl development packages and make sure you are doing an 'export LANG=C'. Some of the stuff did not compile on my redhat system (9.0) because it was set as LANG=en_US.UTF-8. The first problem is indeed it cannot find your openssl headers. -- Matthew Billings | Affordable WWW Internet Solutions foreThought.net | for Small Business [EMAIL PROTECTED] | 910 16th Street, #1220 (303) 228-0070 x821 --The Future is Now!--| Denver, CO 80202(303) 228-0077 fax -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nicholas Ruddick Sent: Monday, May 17, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Redhat 7.3 compiling problem Firstly, amazing software, props to all the developers. I'm trying to compile the latest asterisk cvs checkout and keep getting an error which I can't solve, any help would be much appreciated - make[1]: Leaving directory `/usr/src/asterisk/stdtime' if [ -d CVS ] ! [ -f .version ]; then echo CVS-HEAD-05/17/04-16:45:34 .version; fi for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-05/17/04-16:45:34\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC -c -o res_crypto.o res_crypto.c res_crypto.c:25:25: openssl/ssl.h: No such file or directory res_crypto.c:26:25: openssl/err.h: No such file or directory res_crypto.c:74: parse error before `RSA' res_crypto.c:74: warning: no semicolon at end of struct or union res_crypto.c:84: parse error before `}' res_crypto.c: In function `pw_cb': res_crypto.c:101: dereferencing pointer to incomplete type res_crypto.c:103: dereferencing pointer to incomplete type res_crypto.c:103: dereferencing pointer to incomplete type res_crypto.c:104: dereferencing pointer to incomplete type res_crypto.c:106: dereferencing pointer to incomplete type res_crypto.c:108: dereferencing pointer to incomplete type res_crypto.c:109: dereferencing pointer to incomplete type res_crypto.c:115: dereferencing pointer to incomplete type res_crypto.c: In function `ast_key_get': res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:126: dereferencing pointer to incomplete type res_crypto.c:127: dereferencing pointer to incomplete type res_crypto.c:129: dereferencing pointer to incomplete type res_crypto.c: In function `try_load_key': res_crypto.c:163: dereferencing pointer to incomplete type res_crypto.c:165: dereferencing pointer to incomplete type res_crypto.c:188: dereferencing pointer to incomplete type res_crypto.c:189: dereferencing pointer to incomplete type res_crypto.c:191: dereferencing pointer to incomplete type res_crypto.c:195: dereferencing pointer to incomplete type res_crypto.c:204: sizeof applied to an incomplete type res_crypto.c:210: sizeof applied to an
RE: [Asterisk-Users] speex
http://asterisk.bkw.org/diff/translate.patch.txt If you try that patch out it adds a nice feature... show translation recalc [xx] You can throw more than 1 sample thru it and recalculate your translation matrix. It also allows you to see TRUE translation under a load or just when ever you feel like seeing them updated. When * loads the codec it shoot one frame thru and times it. Now under real world scenarios you will be shooting more than one frame thru so LETS have the option to update the matrix with these types of tests. 200 is the max. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Monday, May 17, 2004 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] speex Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. As a quick example, here are the show translation outputs from * on a 2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5 (compiled with CFLAGS=-march=pentium4 and --enable-sse). Note how encoding from slin went from 25 to 15 ms. That is from the re-write of the sse optimized routines in libspeex. The % change is similar to what I saw on my p3 notebook, where both 1.0.3 and 1.1.5 were compiled with --enable-sse and -march=pentium3. (As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) spx103*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -2619 ULAW - 3 - 1 2 2 1 6 -2619 ALAW - 3 1 - 2 2 1 6 -2619 G726 - 3 2 2 - 2 1 6 -2619 ADPCM - 3 2 2 2 - 1 6 -2619 SLINR - 2 1 1 1 1 - 5 -2518 LPC10 - 4 3 3 3 3 2 - -2720 G729A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - -19 ILBC - 5 4 4 4 4 3 8 -28 - spx115*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -1619 ULAW - 3 - 1 2 2 1 6 -1619 ALAW - 3 1 - 2 2 1 6 -1619 G726 - 3 2 2 - 2 1 6 -1619 ADPCM - 3 2 2 2 - 1 6 -1619 SLINR - 2 1 1 1 1 - 5 -1518 LPC10 - 4 3 3 3 3 2 - -1720 G729A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - -19 ILBC - 5 4 4 4 4 3 8 -18 - -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ 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
RE: [Asterisk-Users] SIP in the UK
Voiptalk provide an excellent service and great support. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Potkin Sent: 10 May 2004 23:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP in the UK On Mon, May 10, 2004 at 08:58:23AM +0100, Gavin Hamill wrote: http://www.voiptalk.org/ - this is the service-side of TelAppliant, official UK Digium resellers. I've written to VoIPTalk a couple of times and never got any response from them, and their outbound calling rates aren't fantastic. I would be concerned about their quality of customer service were I to be considering using them for business use. The comment on VoIPTalk's calling rates is interesting as I came to a different conclusion. To instance the two main destinations I call, the UK and Spain, as an example, the rates are 1.6p and 2p per minute respectively. This appears to me to be very competitive with other offerings. 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
[Asterisk-Users] dial without answer ?
hi, dialing without answer is descriped in serveral docs. But if i try to do so * sends an invite message ever and ever to my phone. My phone (snom) sends a busy here back and the call is cancled. When i answer before i dial all is ok. Where is the problem ? make i everything wrong ? nicolas ___ 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] Redhat 7.3 compiling problem
Steven Critchfield wrote: I can't believe you are still on 7.3, but whatever. Keeping it old school :-) Those two entries alone solve whats wrong. When you don't have .h files, you are missing the -dev or -devel packages for whatever it is they belong to. In this case, you are missing the openssl-devel package. Good luck finding the one for such a vintage version of RedHat. Fedora / Legacy Project and Freshrpms with good old failthful apt-get solves all my packaging needs. Lagacy project still updates pacages for security needs. Asterisk 0.5 is even on freshrpms! I'll update some day to Fedora Core 2, it's whats running on the rest of my machines. I really need more ram and hardisk space in my serverf first, it's doing way to much as it is. Compiling. Nicholas Ruddick ___ 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] *8 problem still there?
I see this (but not using a recent asterisk version) I had put it down to a software bug in the grandstream phones that I'm using - are you sure its an asterisk bug or are you using grandstream also? Steve On Mon, 17 May 2004, John Vogel wrote: I upgraded to the latest stable version of 1.0 today and am still seeing the *8 problem where the phone that was originally dialed keeps on ringing even after another phone picks up. Are other people also seeing this? Has somebody figured out how to make this go away? Thanks! ___ 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] Zap callwaiting hookflash idiosyncracy/flaw?
Don't know what else to call this. Googling and some time on the IRC channel haven't gotten me anywhere. Here's the sitch, which is a bit complicated but is something my customers are in fact encountering on an everyday basis: 1. Bob is on a Zap channel talking through the PSTN to Carol. Both have the misfortune, like so many of us, of having LECs who do not offer disconnect supervision. 2. While that conversation is going on, Ted calls in on another line and selects the ACD option for Bob, and Bob sees Ted's CLID on his phone and hears the CW tones. 3. Bob wants to hang up his call with Carol to talk to Ted. *** Now the possible scenarios, and their apparent resolutions *** A. Bob hookflashes and takes the call from Ted. BUT NOW CAROL'S PHONE IS STUCK OFFHOOK LISTENING TO MUSIC ON HOLD FROM THE THREE-WAY CALL CAUSED BY THE HOOKFLASH. That surely isn't what we want. . . . B. Bob hangs up for longer than a hookflash period. Picks up the phone and gets a fresh dialtone. But now there seems to be no connection between his phone and the incoming call; Ted hears continuous ringing, and then gets voicemail. Bob really wants to be able to take the call from Ted, but Brian can't figure out how to configure Bob's dialplan to do it!! Someone suggested shutting off 3-way calling, but that feature seems to be a nice one that everyone makes a lot of use of. Thanks in advance. B. ___ 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 Proxy Type
On Mon, 2004-05-17 at 09:57, Olle E. Johansson wrote: Ignace CARIA wrote: Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Asterisk is a no-SIP-proxy-at-all :-) Again: Asterisk is not a SIP proxy. A SIP proxy doesn't handle media streams and doesn't terminate or originate calls. Asterisk does. Asterisk is a stateful SIP UAS (User Agent Server) or UAC (User Agent Client) that has some characteristics of a stateful SIP proxy, but in most cases no similarity at all. Would be correct to say that * SIP channel is a B2BUA plus it handle the media stream ? -- Juanjo sin .sig ___ 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] iax2 and ethereal
If you are using ethereal to decode packet traces that include iax2 packets, you may have noticed that codecs such as ilbc were being shown as unkown. I've had a patch accepted into the ethereal cvs that corrects that, updating packet-iax2.[ch] to match asterisk cvs HEAD. I presume it will be in the next release, and is now available in ethereal's anon cvs tree. -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ 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] Astricon 2004 - the developer's meeting ** CALL FOR PAPERS
During Astricon 2004, we'll have the first Asterisk developer's meeting. The Asterisk developer's meeting is a one day meeting with discussions, brainstorms and tests. For each session, we need a white paper produced that outlines the topic to be discussed. If controversial, several whitepapers on the same topic will be contributed. This is a list of suggested topics - please feel free to add topics. Topics will be jointly reviewed by the Asterisk.org bug marshals and Digium developers. Please observe that this meeting will be held in a very professional way, with a schedule and a painful chairman that will make the meeting stay in order. We will not solve every topic, but will be able to see pros, cons and discuss ways forward, maybe initiate some work. This is just a list of ideas - PLEASE feel free to suggest your own. * Asterisk roadmap - after 1.x (Digium) * Channel specific discussions o SIP, SRTP and TCP/TLS o The new H.323 architecture * Configuration architectures o Dynamic and static data - how to separate o res_config and others * A common Authentication architecture * Extension and peer/user/friend configs * AGI and alternatives * Manager API development * Multiprotocol presence and messaging architecture for Asterisk * Documentation - the Wiki and the Doc project * Sexy things done with Asterisk - Bluetooth presence and others (showcase, including Mark's Wifiaxy ) * Voice: Vxml, Speech recognition and TTS modules * Instant messaging (Jabber, SIMPLE and IAX) This is not an educational forum for beginners. It is planned for active Asterisk developer's and contributors. A conference call number will, if possible, be available for those wishing to listen. HOW TO SUBMIT A WHITE PAPER * Send your white paper to [EMAIL PROTECTED] no later than june 15th * If accepted, we will schedule it for the dev meeting and work with it * All white papers is released to the community. There should be no NDA's or other clauses attached to your paper. If there are any limitations on how we can use the document or the ideas or solutions based on the paper, it will not be considered for submission to the conference. * Attach all your contact data - Name, Adress, Phone, E-mail so we can contact you * All white papers will be published in a central repository, regardless if they are accepted as a basis for this first meeting or not. If you have any questions, please feel free to send them to us on [EMAIL PROTECTED] * OFF LIST * /Olle and Steven ___ 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 Proxy Type
Juan J. Sierralta P. wrote: On Mon, 2004-05-17 at 09:57, Olle E. Johansson wrote: Ignace CARIA wrote: Perhaps stupid question but, is Asterisk a statefull or stateless proxy? Asterisk is a no-SIP-proxy-at-all :-) Again: Asterisk is not a SIP proxy. A SIP proxy doesn't handle media streams and doesn't terminate or originate calls. Asterisk does. Asterisk is a stateful SIP UAS (User Agent Server) or UAC (User Agent Client) that has some characteristics of a stateful SIP proxy, but in most cases no similarity at all. Would be correct to say that * SIP channel is a B2BUA plus it handle the media stream ? Yes, a B2BUA handles the media stream, so you are 100% correct. I forgot that part of the SIP terminology :-) I would like to add that even though Asterisk from a SIP proxy standpoint is a bit limited, it's a very clever B2BUA for connections to the PSTN as well as other VOIP protocols. If someone adds IPv6 connectivity, we'll be a really wonderful IPv4 to IPv6 VoIP gateway as well. /O ___ 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: question on domains requiring SRV lookups within asterisk
I was wondering if someone could shed some light on using SRV records in conjunction with *. I have a domain in my sip.conf file that utilizes SRV records to direct SIP traffic when using the regular domain name (i.e. sip:[EMAIL PROTECTED]) in the SIP uri to a SNOM 4S proxy to which this particular end user is registered (have to do it there and not the asterisk). However when I look at an ethereal trace from the * server on an inbound call from the PSTN (over a SIP interconnect), it looks like the * is doing just a regular DNS lookup when it sends the INVITE out even though I have srvlookup=yes in the sip.conf file. My assumption is that the srv record in the DNS is set up correctly (which could be the wrong assumption), since the SNOM proxy and my outbound proxy are able to resolve the SRV lookup correctly. Is there any specific command or port designation that needs to be added in the host= line in a given context? Any suggestions would be welcome. Tom Schroer attachment: winmail.dat
Re: [Asterisk-Users] X100P problem with PSTN from BOLIVIA
On Mon, 2004-05-17 at 12:10, Jorge Verastegui wrote: The silence last 60s (aprox) So maybe is the timeout used by your Telco (Entel?) Here at Chile we use 30s to let called people to be able to hang and get the call on another phone plugged to the same line. So I think its better to consult your telco. Does happen the same when the called party is a common phone not asterisk ? -- Juanjo sin .sig ___ 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] Enhanced voicemail Externnotify
Has anyone implemented the externnotify feature that is mentioned on the wiki with the enhanced voicemail? I have tried to invoke the command both in the general section as well as a part of the user mailbox definition with no luck. The explanation of the feature is as follows: Externnotify Want to run an external program whenever a caller leaves a voice mail message for a user? This is where the externnotify command comes in handy. Externnotify takes a string value which is the command line you want to execute when the caller finishes leaving a message. Does anyone know the proper usage of the feature and can it be defined for individual mailboxes ? In this example: 4069 = 6522,Matt Brooks,[EMAIL PROTECTED],,|tz=central|attach=yes|saycid=yes|dialout=fromvm| callback=fromvm|review=yes|operator=yes, ___ 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] Enhanced voicemail Externnotify
On Mon, May 17, 2004 at 05:11:40PM -0400, Kevin wrote: Has anyone implemented the externnotify feature that is mentioned on the wiki with the enhanced voicemail? I have (I wrote it :-) ). Externnotify Want to run an external program whenever a caller leaves a voice mail message for a user? This is where the externnotify command comes in handy. Externnotify takes a string value which is the command line you want to execute when the caller finishes leaving a message. Does anyone know the proper usage of the feature and can it be defined for individual mailboxes ? Externotify is not implemented for individual mailboxes. It is a one use general variable that is set in the general voicemail configuration. The way it works is basically any time that somebody leaves a voicmail on the system (regardless of mailbox number) the command specified for externnotify is run with the arguements (in this order): context, extension, and number of voicemails in that mailbox. These arguements are passed to the program that you set in the externnotify variable. Matthew Fredrickson ___ 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] DTMF transmitted over IAX2 coming out as clicks at the other end
I'm having a weird problem with IAX2 in today's CVS HEAD. I have two boxes with T100P cards connected via IAX2. Calls between them work fine, but when I press a key at one end, it comes out the other end as a click, with no tone. I've tested the DTMF on the T1 using SendDTMF with an outgoing call, and it sounds fine in that case; it seems to be only IAX2 that has the problem. iax2 debug confirms that the digits are arriving as type DTMF, with the right subclass, so I'm thinking it's a problem with the way Asterisk is synthesizing the digits. I've poked through the chan_iax2 code, but I can't find the DTMF synthesis - has anyone had this problem, or can someone point me towards the right place to look? Thanks, Mark ___ 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] Enhanced voicemail Externnotify
Thanks for the quick response. I am trying to implement a solution for voicemail outcall notification for individual users. There is a suggested solution posted in the wiki but it has some limitations thus I was looking for an alternate solution. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, May 17, 2004 5:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Enhanced voicemail Externnotify On Mon, May 17, 2004 at 05:11:40PM -0400, Kevin wrote: Has anyone implemented the externnotify feature that is mentioned on the wiki with the enhanced voicemail? I have (I wrote it :-) ). Externnotify Want to run an external program whenever a caller leaves a voice mail message for a user? This is where the externnotify command comes in handy. Externnotify takes a string value which is the command line you want to execute when the caller finishes leaving a message. Does anyone know the proper usage of the feature and can it be defined for individual mailboxes ? Externotify is not implemented for individual mailboxes. It is a one use general variable that is set in the general voicemail configuration. The way it works is basically any time that somebody leaves a voicmail on the system (regardless of mailbox number) the command specified for externnotify is run with the arguements (in this order): context, extension, and number of voicemails in that mailbox. These arguements are passed to the program that you set in the externnotify variable. Matthew Fredrickson ___ 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
[Asterisk-Users] Something weird
Ever since I updated to CVS-head from 10 May, something weird has been happening... Every night at 1:10 AM (Eastern Australian time) my phone rings, there is no callerid, and it results in a message in Voicemail which is just the disconnect beeps (due to the inability of being able to detect disconnect). Is there any way that some test code has been left behind that causes this? Puzzled, Simon ___ 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] Music on hold
I have set up an extension so I can dial it and listen to my MusicOnHold from any handset. This is what is in the extensions.conf: exten = 997,1,MusicOnHold() exten = 997,2,Hangup After 180 seconds of playing, the call terminates. Why does this happen? Simon ___ 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] 4 X100P + 1 TDMP400(4 FXS): Only by miracle?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Isamar Maia wrote: | Hi folks, | | I'm trying to make an * PBX for a customer using 4 X100Ps | and 1 TDM400p(4FXS). | The problem I'm facing is to make one unique IRQ for each | PCI slot/board since shared IRQs create all kind of weird noises | and echos. | Anybody got any workaround for that? | Any recommended motherboard to accomplish that ? | Currently, I'm playing with an ASUS A7V600. Have you looked at possibly using the TDM400P with 4 FXO modules? Then you would only need to have 2 cards (currently) in the system and possibly have room for expansion in the future, if needed. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFAqTDQuYsUrHkpYtARAkH4AKCBVA3Jhi3TFgWQkcCNcJvu+In4RQCeKDvq ZJ00qFa6tcVM2jXxtkBAc3Q= =OX7D -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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 error
when asterisk has more than 50 h323 calls it craps out on me. Can anyone help? May 17 10:45:35 WARNING[1769581]: chan_sip.c:1114 create_addr: No such host: 19149191120-- Executing Dial(H323/ip$66.238.200.224:32943/16164, H323/[EMAIL PROTECTED]/1957408) in new stack-- Called [EMAIL PROTECTED]-- Executing ChanIsAvail(H323/ip$66.238.200.224:32944/16165, Sip/19149191120) in new stackMay 17 10:45:35 WARNING[1818736]: chan_sip.c:1114 create_addr: No such host: 19149191120-- Executing Dial(H323/ip$66.238.200.224:32944/16165, H323/[EMAIL PROTECTED]/1957408) in new stack-- Called [EMAIL PROTECTED] 0:15.753 H225 Caller:41c14748 assert.cxx(105) PWLib Assertion fail: Invalid array element, file /root/pwlib/include/ptlib/array.h, line 1116, Error=115 Abort, Core dump, Ignore? ___ 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] call announce?
I want to do this same thing, does anyone have an example of how to do it? using a zap fxo and zap fxs card how can I set up caller announce? like this. 1 call comes in and a prompt asks the called to identify themselves. 2 the system would then put the caller on hold and pick up the FXS and play the message for the users prompting them to hit 1 to accept the call and have it connected or hit 2 to dump the live caller to voicemail. Can this be done with * Dave ___ 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] recommended hardware for quad E1 system
Thanks Chris, but I will use only voicemail and conference, I think that is better 4 Pentium III boxes that one dual pentium box only. Do you think that it can attend 30 channels? regards Robert Almeida On Mon, 2004-05-17 at 17:23, Robert Almeida wrote: Could anyone tell me which is the recommended hardware to a system running voicemail and conference, attending four E1 trunks and, another, attending only one E1? Can I use a PIII 850Mhz? Maybe for a single port E1 card, maybe. You'll definitely have problems with 4 trunks, voicemail, conference, etc., on a P3. Need at least a P4 for all of that. Maybe dual if you are doing lots of codec translation. If you are really going to put asterisk to work with 120 lines, buy the fastest box your budget will allow, 1 gig ram, SCSI, the works. If you've already got the P3, then by all means, try that out first before spending your money. You have to buy the card either way. Might as well test your configs on that. --Chris Robert Almeida
RE: [Asterisk-Users] Music on hold
Issue an Answer first! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Monday, May 17, 2004 4:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Music on hold I have set up an extension so I can dial it and listen to my MusicOnHold from any handset. This is what is in the extensions.conf: exten = 997,1,MusicOnHold() exten = 997,2,Hangup After 180 seconds of playing, the call terminates. Why does this happen? Simon ___ 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
RE: [Asterisk-Users] call announce?
Hint use app_parkandannounce with a twist of app_record. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gavin Hollinger Sent: Monday, May 17, 2004 4:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] call announce? I want to do this same thing, does anyone have an example of how to do it? using a zap fxo and zap fxs card how can I set up caller announce? like this. 1 call comes in and a prompt asks the called to identify themselves. 2 the system would then put the caller on hold and pick up the FXS and play the message for the users prompting them to hit 1 to accept the call and have it connected or hit 2 to dump the live caller to voicemail. Can this be done with * Dave ___ 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
[Asterisk-Users] problems compiling h323 support
Hi all I am trying to compile Asterisk on RH9
I am have installed
pwlib_1.5.2
readline-4.3-5
readline-devel-4.3-5
openssl-devel-0.9.7a-2
openssl-0.9.7a-2
openh323_1.12.2
I read that it has to do with pwlib not being installed correctly so i made
sure it was in /usr/local correctly
When i try to compile i get the follow errors
I was wondering can anyone shed in light on this issue
Thanks
g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT -DP_HAS_SEMAPHORES
-O3 -
DNDEBUG -DP_SSL -I../include -I/include -I/crypto
-DPNX_VERSION=\CVS-05/17/04-1
7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -I/include/ptlib/unix
-I/inc
lude -I/home/dictate/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c
chan_h323
.cxx -o chan_h323.o
In file included from /usr/include/ptlib/contain.h:222,
from /usr/include/ptlib.h:139,
from chan_h323.cxx:37:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
[ snip ]
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RE: [Asterisk-Users] *8 problem still there?
Not sure it is an Asterisk bug - I am using GrandStreams. Will upgrade their software and also try it on my Snom. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen J. Wilcox Sent: Monday, May 17, 2004 12:45 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] *8 problem still there? I see this (but not using a recent asterisk version) I had put it down to a software bug in the grandstream phones that I'm using - are you sure its an asterisk bug or are you using grandstream also? Steve On Mon, 17 May 2004, John Vogel wrote: I upgraded to the latest stable version of 1.0 today and am still seeing the *8 problem where the phone that was originally dialed keeps on ringing even after another phone picks up. Are other people also seeing this? Has somebody figured out how to make this go away? Thanks! ___ 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
RE: [Asterisk-Users] Music on hold
Where? Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, 18 May 2004 8:04 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Music on hold Issue an Answer first! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Monday, May 17, 2004 4:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Music on hold I have set up an extension so I can dial it and listen to my MusicOnHold from any handset. This is what is in the extensions.conf: exten = 997,1,MusicOnHold() exten = 997,2,Hangup After 180 seconds of playing, the call terminates. Why does this happen? Simon ___ 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 ___ 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] problems compiling h323 support
Try 'export LANG=C' then 'make clean make'
--
Matthew Billings | Affordable WWW Internet Solutions
foreThought.net | for Small Business
[EMAIL PROTECTED] | 910 16th Street, #1220 (303)
228-0070 x821
--The Future is Now!--| Denver, CO 80202(303)
228-0077 fax
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jer
Sent: Monday, May 17, 2004 4:07 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] problems compiling h323 support
Hi all I am trying to compile Asterisk on RH9
I am have installed
pwlib_1.5.2
readline-4.3-5
readline-devel-4.3-5
openssl-devel-0.9.7a-2
openssl-0.9.7a-2
openh323_1.12.2
I read that it has to do with pwlib not being installed correctly so i
made
sure it was in /usr/local correctly
When i try to compile i get the follow errors
I was wondering can anyone shed in light on this issue
Thanks
g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT
-DP_HAS_SEMAPHORES
-O3 -
DNDEBUG -DP_SSL -I../include -I/include -I/crypto
-DPNX_VERSION=\CVS-05/17/04-1
7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN
-I/include/ptlib/unix
-I/inc
lude -I/home/dictate/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c
chan_h323
.cxx -o chan_h323.o
In file included from /usr/include/ptlib/contain.h:222,
from /usr/include/ptlib.h:139,
from chan_h323.cxx:37:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration
`int
[ snip ]
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Re: [Asterisk-Users] 4 X100P + 1 TDMP400(4 FXS): Only by miracle?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Isamar Maia wrote: | Hi folks, | | I'm trying to make an * PBX for a customer using 4 X100Ps | and 1 TDM400p(4FXS). | The problem I'm facing is to make one unique IRQ for each | PCI slot/board since shared IRQs create all kind of weird noises | and echos. | Anybody got any workaround for that? | Any recommended motherboard to accomplish that ? | Currently, I'm playing with an ASUS A7V600. Have you looked at possibly using the TDM400P with 4 FXO modules? Then you would only need to have 2 cards (currently) in the system and possibly have room for expansion in the future, if needed. That's an excellent idea, and maybe the unique way out. But, what do I do with all my X100Ps that I bought from Digium? Give them back and get my money back and buy a TDM400P(4FXO) ? :-) Isamar ___ 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] problems compiling h323 support
At 06:23 PM 5/17/2004, you wrote:
gives the same error...
g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT -DP_HAS_SEMAPHORES
-O3 -
DNDEBUG -DP_SSL -I../include -I/include -I/crypto
-DPNX_VERSION=\CVS-05/17/04-1
7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -I/include/ptlib/unix
-I/inc
lude -I/root/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c chan_h323.cxx -o
chan_h323.o
In file included from /usr/include/ptlib/contain.h:222,
from /usr/include/ptlib.h:139,
from chan_h323.cxx:37:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
Try 'export LANG=C' then 'make clean make'
--
Matthew Billings | Affordable WWW Internet Solutions
foreThought.net | for Small Business
[EMAIL PROTECTED] | 910 16th Street, #1220 (303)
228-0070 x821
--The Future is Now!--| Denver, CO 80202(303)
228-0077 fax
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jer
Sent: Monday, May 17, 2004 4:07 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] problems compiling h323 support
Hi all I am trying to compile Asterisk on RH9
I am have installed
pwlib_1.5.2
readline-4.3-5
readline-devel-4.3-5
openssl-devel-0.9.7a-2
openssl-0.9.7a-2
openh323_1.12.2
I read that it has to do with pwlib not being installed correctly so i
made
sure it was in /usr/local correctly
When i try to compile i get the follow errors
I was wondering can anyone shed in light on this issue
Thanks
g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT
-DP_HAS_SEMAPHORES
-O3 -
DNDEBUG -DP_SSL -I../include -I/include -I/crypto
-DPNX_VERSION=\CVS-05/17/04-1
7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN
-I/include/ptlib/unix
-I/inc
lude -I/home/dictate/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c
chan_h323
.cxx -o chan_h323.o
In file included from /usr/include/ptlib/contain.h:222,
from /usr/include/ptlib.h:139,
from chan_h323.cxx:37:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration
`int
[ snip ]
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RE: [Asterisk-Users] Music on hold
exten = 997,1,Answer exten = 997,2,MusicOnHold() exten = 997,3,Hangup -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Monday, May 17, 2004 5:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Music on hold Where? Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, 18 May 2004 8:04 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Music on hold Issue an Answer first! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Simon Brown Sent: Monday, May 17, 2004 4:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Music on hold I have set up an extension so I can dial it and listen to my MusicOnHold from any handset. This is what is in the extensions.conf: exten = 997,1,MusicOnHold() exten = 997,2,Hangup After 180 seconds of playing, the call terminates. Why does this happen? Simon ___ 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 ___ 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
RE: [Asterisk-Users] *8 problem still there?
John Vogel [EMAIL PROTECTED] wrote: I upgraded to the latest stable version of 1.0 today and am still seeing the *8 problem where the phone that was originally dialed keeps on ringing even after another phone picks up. Are other people also seeing this? Has somebody figured out how to make this go away? I was getting this but I think it's gone away now (using CVS head, which is much better than stable, in my opinion). I haven't used *8 in a while, so I'm not 100% sure that it's been fixed. I'll test that out tomorrow. I'm using a mixture of Sipura SPA-2000 and Cisco 7960G devices. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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
Re: [Asterisk-Users] call announce?
Does parkandannounce create a variable with the parked extension number that I could use in later linking the calls? Hint use app_parkandannounce with a twist of app_record. ___ 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] total newbie sanity check
Im a total newbie at this telephony stuff but I'm putting together a low cost PBX for my small company and wanted a check on the h/w Im planning on ordering and my system configuration. Any input is appreciated. Take it offline and email me directly if appropriate ([EMAIL PROTECTED]). Heres what Im planning: === Parts List === 1 Digium Wildcard TDM400P w/ 4-port FXO bundle Im planning on using this to connect to a few CO POTS lines. A mid-range computer (600MHz or so w/ 512 MB RAM) Some form of Linux (fedora?) Asterisk 10/100 Ethernet card (in the computer) 10/100 Ethernet Switch (8 port or so) A few SIP capable phones === End Parts List === And now a few questions: 1) Is this a feasible system? Am I missing any important hardware? 2) What is a good Linux to use? Im reasonably proficient w/ Linux. 3) Do I need to tell the phone company anything special or do I just have them connect up standard phone lines? 4) Can the phone company usually roll the calls onto a spare incoming CO line? (e.g. if the first line is busy - route it to the second line, if the 2nd is busy route to the 3rd, etc.) Is there a special name for feature this that the phone company will recognize? 5) Id like to support a special feature Id like to have 2 different incoming phone numbers (on all lines) and have asterisk multiplex to the right voice menu system based on the incoming phone number. Is this possible? Does it require special features from the phone company? Thats all for now (though Im sure this wont be my last post. Thanks again for any help. - Mike Stupak
Re: [Asterisk-Users] total newbie sanity check
Looks good to me. You will want hunting or call forward busy on the phone lines you order. Mine costs $1.15 per month - Original Message - From: Mike Stupak To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 4:59 PM Subject: [Asterisk-Users] total newbie sanity check Im a total newbie at this telephony stuff but I'm putting together a low cost PBX for my small company and wanted a check on the h/w Im planning on ordering and my system configuration. Any input is appreciated. Take it offline and email me directly if appropriate ([EMAIL PROTECTED]). Heres what Im planning: === Parts List === 1 Digium Wildcard TDM400P w/ 4-port FXO bundle Im planning on using this to connect to a few CO POTS lines. A mid-range computer (600MHz or so w/ 512 MB RAM) Some form of Linux (fedora?) Asterisk 10/100 Ethernet card (in the computer) 10/100 Ethernet Switch (8 port or so) A few SIP capable phones === End Parts List === And now a few questions: 1) Is this a feasible system? Am I missing any important hardware? 2) What is a good Linux to use? Im reasonably proficient w/ Linux. 3) Do I need to tell the phone company anything special or do I just have them connect up standard phone lines? 4) Can the phone company usually roll the calls onto a spare incoming CO line? (e.g. if the first line is busy - route it to the second line, if the 2nd is busy route to the 3rd, etc.) Is there a special name for feature this that the phone company will recognize? 5) Id like to support a special feature Id like to have 2 different incoming phone numbers (on all lines) and have asterisk multiplex to the right voice menu system based on the incoming phone number. Is this possible? Does it require special features from the phone company? Thats all for now (though Im sure this wont be my last post. Thanks again for any help. - Mike Stupak
Re: [Asterisk-Users] total newbie sanity check
Question 5 will be harder. What you need is DID ( Direct Inward Dialing) Not available in my area with regular phone lines. Perhaps it could be done with distinctive ring? Gavin - Original Message - From: Mike Stupak To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 4:59 PM Subject: [Asterisk-Users] total newbie sanity check Im a total newbie at this telephony stuff but I'm putting together a low cost PBX for my small company and wanted a check on the h/w Im planning on ordering and my system configuration. Any input is appreciated. Take it offline and email me directly if appropriate ([EMAIL PROTECTED]). Heres what Im planning: === Parts List === 1 Digium Wildcard TDM400P w/ 4-port FXO bundle Im planning on using this to connect to a few CO POTS lines. A mid-range computer (600MHz or so w/ 512 MB RAM) Some form of Linux (fedora?) Asterisk 10/100 Ethernet card (in the computer) 10/100 Ethernet Switch (8 port or so) A few SIP capable phones === End Parts List === And now a few questions: 1) Is this a feasible system? Am I missing any important hardware? 2) What is a good Linux to use? Im reasonably proficient w/ Linux. 3) Do I need to tell the phone company anything special or do I just have them connect up standard phone lines? 4) Can the phone company usually roll the calls onto a spare incoming CO line? (e.g. if the first line is busy - route it to the second line, if the 2nd is busy route to the 3rd, etc.) Is there a special name for feature this that the phone company will recognize? 5) Id like to support a special feature Id like to have 2 different incoming phone numbers (on all lines) and have asterisk multiplex to the right voice menu system based on the incoming phone number. Is this possible? Does it require special features from the phone company? Thats all for now (though Im sure this wont be my last post. Thanks again for any help. - Mike Stupak
Re: [Asterisk-Users] total newbie sanity check
DID is inbound only. DID will not work if you plan to use the same trunks for outbound calls. Do the Digium cards support DID? Normally DID lines require an external power supply which connects to the card. I don't remember seeing anything like that on the Digium analog cards. Distinctive ring would work if * can support it. Bill Doll Jr Gavin Hollinger [EMAIL PROTECTED] Gavin Hollinger [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/17/2004 04:14 PM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users] total newbie sanity check Question 5 will be harder. What you need is DID ( Direct Inward Dialing) Not available in my area with regular phone lines. Perhaps it could be done with distinctive ring? Gavin - Original Message - From: Mike Stupak To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 4:59 PM Subject: [Asterisk-Users] total newbie sanity check Im a total newbie at this telephony stuff but I'm putting together a low cost PBX for my small company and wanted a check on the h/w Im planning on ordering and my system configuration. Any input is appreciated. Take it offline and email me directly if appropriate ([EMAIL PROTECTED]). Heres what Im planning: === Parts List === 1 Digium Wildcard TDM400P w/ 4-port FXO bundle Im planning on using this to connect to a few CO POTS lines. A mid-range computer (600MHz or so w/ 512 MB RAM) Some form of Linux (fedora?) Asterisk 10/100 Ethernet card (in the computer) 10/100 Ethernet Switch (8 port or so) A few SIP capable phones === End Parts List === And now a few questions: 1) Is this a feasible system? Am I missing any important hardware? 2) What is a good Linux to use? Im reasonably proficient w/ Linux. 3) Do I need to tell the phone company anything special or do I just have them connect up standard phone lines? 4) Can the phone company usually roll the calls onto a spare incoming CO line? (e.g. if the first line is busy - route it to the second line, if the 2nd is busy route to the 3rd, etc.) Is there a special name for feature this that the phone company will recognize? 5) Id like to support a special feature Id like to have 2 different incoming phone numbers (on all lines) and have asterisk multiplex to the right voice menu system based on the incoming phone number. Is this possible? Does it require special features from the phone company? Thats all for now (though Im sure this wont be my last post. Thanks again for any help. - Mike Stupak inline: graycol.gifinline: pic15724.gifinline: ecblank.gif
Re: [Asterisk-Users] total newbie sanity check
Mike Stupak wrote: Im a total newbie at this telephony stuff but I'm putting together a low cost PBX for my small company and wanted a check on the h/w Im planning on ordering and my system configuration. Any input is appreciated. Take it offline and email me directly if appropriate ([EMAIL PROTECTED] mailto:[EMAIL PROTECTED]). Heres what Im planning: === Parts List === 1 Digium Wildcard TDM400P w/ 4-port FXO bundle Im planning on using this to connect to a few CO POTS lines. That should be fine if you have 4 incoming pots lines A mid-range computer (600MHz or so w/ 512 MB RAM) Should do fine too! We run a similar setup with a P3-1Ghz and it's most of the time in idle. Some form of Linux (fedora?) I run fedora FC1 too. Runs quite well. Just remember to install the kernel and kernel headers. This is required for asterisk and all zaptel stuff to compile right. The tool you want to use is yum. yum install kernel-header and so on. There is an yum.conf file. Yum is also good if you want to up date your system. Asterisk Yup. Install asterisk from CVS and NOT from rpm. The rpm version that I installed back then was causing me nothing but trouble. 10/100 Ethernet card (in the computer) 10/100 Ethernet Switch (8 port or so) A few SIP capable phones === End Parts List === And now a few questions: 1) Is this a feasible system? Am I missing any important hardware? Yes 2) What is a good Linux to use? Im reasonably proficient w/ Linux. I think most people will suggest you to use debian. Fedora works quite well here at my office. 3) Do I need to tell the phone company anything special or do I just have them connect up standard phone lines? 4) Can the phone company usually roll the calls onto a spare incoming CO line? (e.g. if the first line is busy - route it to the second line, if the 2^nd is busy route to the 3^rd , etc.) Is there a special name for feature this that the phone company will recognize? We do it that way. We have 4 incoming pots lines with 4 numbers. When number 1 is busy it will forward to the next line. If that one is busy it will forward to the next. The only thing I have run into as a problem sometimes people dial back the number they got called from. If that is the last number we have and it is busy even if line 1/2/3 are free line 4 will not forward back to 1. For some reason our phone provider will not do that. 5) Id like to support a special feature Id like to have 2 different incoming phone numbers (o n all lines) and have asterisk multiplex to the right voice menu system based on the incoming phone number. Is this possible? Does it require special features from the phone company? I guess that is not that easy. If you can just group 2 of the 4 pots lines together I guess you can create a dialplan like that. But then you could only have 2 incoming lines per number but 4 outgoing lines. Thats all for now (though Im sure this wont be my last post. Thanks again for any help. - Mike Stupak -- Thomas ___ 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] X100P problem with PSTN from BOLIVIA
When i make a call from Asterisk everything goes Ok, I do have a problem: when a call from the PSTN originates, the extension in Asterisk hangs up and I only hear silence in the PSTN for approximately 60 seconds. On Mon, 2004-05-17 at 16:56, Juan J. Sierralta P. wrote: On Mon, 2004-05-17 at 12:10, Jorge Verastegui wrote: The silence last 60s (aprox) So maybe is the timeout used by your Telco (Entel?) Here at Chile we use 30s to let called people to be able to hang and get the call on another phone plugged to the same line. So I think it´s better to consult your telco. Does happen the same when the called party is a common phone not asterisk ? -- Jorge Verastegui [EMAIL PROTECTED] RedCetus S.R.L. --NOTA DE REDCETUS S.R.L. : La informacin contenida en este E-mail y sus anexos, slo puede ser utilizada por el individuo o la compaa a la cual est dirigido. Si no es el receptor autorizado, cualquier retencin, difusin, distribucin o copia de este mensaje es prohibida y sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo y borrar el mismo
Re: [Asterisk-Users] speex
Actually it encodes a second of data, which with a 20ms codec would be 50 frames. The timing shows better than expected results due to caching. -Adam brian wrote: http://asterisk.bkw.org/diff/translate.patch.txt If you try that patch out it adds a nice feature... show translation recalc [xx] You can throw more than 1 sample thru it and recalculate your translation matrix. It also allows you to see TRUE translation under a load or just when ever you feel like seeing them updated. When * loads the codec it shoot one frame thru and times it. Now under real world scenarios you will be shooting more than one frame thru so LETS have the option to update the matrix with these types of tests. 200 is the max. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Monday, May 17, 2004 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] speex Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. As a quick example, here are the show translation outputs from * on a 2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5 (compiled with CFLAGS=-march=pentium4 and --enable-sse). Note how encoding from slin went from 25 to 15 ms. That is from the re-write of the sse optimized routines in libspeex. The % change is similar to what I saw on my p3 notebook, where both 1.0.3 and 1.1.5 were compiled with --enable-sse and -march=pentium3. (As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) spx103*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -2619 ULAW - 3 - 1 2 2 1 6 -2619 ALAW - 3 1 - 2 2 1 6 -2619 G726 - 3 2 2 - 2 1 6 -2619 ADPCM - 3 2 2 2 - 1 6 -2619 SLINR - 2 1 1 1 1 - 5 -2518 LPC10 - 4 3 3 3 3 2 - -2720 G729A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - -19 ILBC - 5 4 4 4 4 3 8 -28 - spx115*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -1619 ULAW - 3 - 1 2 2 1 6 -1619 ALAW - 3 1 - 2 2 1 6 -1619 G726 - 3 2 2 - 2 1 6 -1619 ADPCM - 3 2 2 2 - 1 6 -1619 SLINR - 2 1 1 1 1 - 5 -1518 LPC10 - 4 3 3 3 3 2 - -1720 G729A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - -19 ILBC - 5 4 4 4 4 3 8 -18 - -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ 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 ___ 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] X100P problem with PSTN from BOLIVIA
Paste your extensions.conf Check the answer command if you're running IVR of special services. - Original Message - From: Jorge Verastegui To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 8:46 PM Subject: Re: [Asterisk-Users] X100P problem with PSTN from BOLIVIA When i make a call from Asterisk everything goes Ok, I do have a problem: when a call from the PSTN originates, the extension in Asterisk hangs up and I only hear silence in the PSTN for approximately 60 seconds. On Mon, 2004-05-17 at 16:56, Juan J. Sierralta P. wrote: On Mon, 2004-05-17 at 12:10, Jorge Verastegui wrote: The silence last 60s (aprox) So maybe is the timeout used by your Telco (Entel?) Here at Chile we use 30s to let called people to be able to hang and get the call on another phone plugged to the same line. So I think it´s better to consult your telco. Does happen the same when the called party is a common phone not asterisk ? -- Jorge Verastegui [EMAIL PROTECTED] RedCetus S.R.L. -- NOTA DE REDCETUS S.R.L. : La información contenida en este E-mail y sus anexos, sólo puede ser utilizada por el individuo o la compañía a la cual está dirigido. Si no es el receptor autorizado, cualquier retención, difusión, distribución o copia de este mensaje es prohibida y sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo y borrar el mismo ___ 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] speex
Yes I realized my error in my wording but it was early :P It doesn't improve alot but does give you some ways to get a better idea of translation times if your box is loaded up with calls. bkw PS this patch was added to CVS-HEAD - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 5:49 PM Subject: Re: [Asterisk-Users] speex Actually it encodes a second of data, which with a 20ms codec would be 50 frames. The timing shows better than expected results due to caching. -Adam brian wrote: http://asterisk.bkw.org/diff/translate.patch.txt If you try that patch out it adds a nice feature... show translation recalc [xx] You can throw more than 1 sample thru it and recalculate your translation matrix. It also allows you to see TRUE translation under a load or just when ever you feel like seeing them updated. When * loads the codec it shoot one frame thru and times it. Now under real world scenarios you will be shooting more than one frame thru so LETS have the option to update the matrix with these types of tests. 200 is the max. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Monday, May 17, 2004 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] speex Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. As a quick example, here are the show translation outputs from * on a 2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5 (compiled with CFLAGS=-march=pentium4 and --enable-sse). Note how encoding from slin went from 25 to 15 ms. That is from the re-write of the sse optimized routines in libspeex. The % change is similar to what I saw on my p3 notebook, where both 1.0.3 and 1.1.5 were compiled with --enable-sse and -march=pentium3. (As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) spx103*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -26 19 ULAW - 3 - 1 2 2 1 6 -26 19 ALAW - 3 1 - 2 2 1 6 -26 19 G726 - 3 2 2 - 2 1 6 -26 19 ADPCM - 3 2 2 2 - 1 6 -26 19 SLINR - 2 1 1 1 1 - 5 -25 18 LPC10 - 4 3 3 3 3 2 - -27 20 9A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - - 19 ILBC - 5 4 4 4 4 3 8 - 8 - spx115*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -16 19 ULAW - 3 - 1 2 2 1 6 -16 19 ALAW - 3 1 - 2 2 1 6 -16 19 G726 - 3 2 2 - 2 1 6 -16 19 ADPCM - 3 2 2 2 - 1 6 -16 19 SLINR - 2 1 1 1 1 - 5 -15 18 LPC10 - 4 3 3 3 3 2 - -17 20 9A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - - 19 ILBC - 5 4 4 4 4 3 8 - 8 - -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] Rate Engine Application
Hello, Have done some research on the Wiki and via Google, but have not found anything that describes what the tables and columns in the rate engine database really mean. Does anyone have any documentation on those? Thanks, Darrin Johnson Systems Engineer IS Domain Inc. ___ 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] speex
Btw, Good work. 5ms is a huge different, espically in optimizing terms. I've added a few flags and shaved off another ms here's my flags: (only for p4/xeon) -fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse -msse2 -mfpmath=sse keep up the good work, Adam brian k. west wrote: Yes I realized my error in my wording but it was early :P It doesn't improve alot but does give you some ways to get a better idea of translation times if your box is loaded up with calls. bkw PS this patch was added to CVS-HEAD - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 5:49 PM Subject: Re: [Asterisk-Users] speex Actually it encodes a second of data, which with a 20ms codec would be 50 frames. The timing shows better than expected results due to caching. -Adam brian wrote: http://asterisk.bkw.org/diff/translate.patch.txt If you try that patch out it adds a nice feature... show translation recalc [xx] You can throw more than 1 sample thru it and recalculate your translation matrix. It also allows you to see TRUE translation under a load or just when ever you feel like seeing them updated. When * loads the codec it shoot one frame thru and times it. Now under real world scenarios you will be shooting more than one frame thru so LETS have the option to update the matrix with these types of tests. 200 is the max. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Monday, May 17, 2004 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] speex Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. As a quick example, here are the show translation outputs from * on a 2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5 (compiled with CFLAGS=-march=pentium4 and --enable-sse). Note how encoding from slin went from 25 to 15 ms. That is from the re-write of the sse optimized routines in libspeex. The % change is similar to what I saw on my p3 notebook, where both 1.0.3 and 1.1.5 were compiled with --enable-sse and -march=pentium3. (As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) spx103*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -26 19 ULAW - 3 - 1 2 2 1 6 -26 19 ALAW - 3 1 - 2 2 1 6 -26 19 G726 - 3 2 2 - 2 1 6 -26 19 ADPCM - 3 2 2 2 - 1 6 -26 19 SLINR - 2 1 1 1 1 - 5 -25 18 LPC10 - 4 3 3 3 3 2 - -27 20 9A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - - 19 ILBC - 5 4 4 4 4 3 8 - 8 - spx115*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -16 19 ULAW - 3 - 1 2 2 1 6 -16 19 ALAW - 3 1 - 2 2 1 6 -16 19 G726 - 3 2 2 - 2 1 6 -16 19 ADPCM - 3 2 2 2 - 1 6 -16 19 SLINR - 2 1 1 1 1 - 5 -15 18 LPC10 - 4 3 3 3 3 2 - -17 20 9A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - - 19 ILBC - 5 4 4 4 4 3 8 - 8 - -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED]
[Asterisk-Users] *, Sipura, Call-Waiting, X100P, 2 ZAP Calls
Does call-waiting work for anyone that recieves 2 pstn calls on a X100P using a Sipura? I have modified the dialplan in the Sipura such that the *0 is definately getting sent to the Asterisk server now. When the phone beeps and I flash hook I get tone, then dial *0# and the sip debug shows that *0 was acknowledged but I get a rapid busy back from Asterisk.
Re: [Asterisk-Users] speex
I toyed with -msse and -mmmx and others too but couldn't put any of those in. :P bkw - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 6:20 PM Subject: Re: [Asterisk-Users] speex Btw, Good work. 5ms is a huge different, espically in optimizing terms. I've added a few flags and shaved off another ms here's my flags: (only for p4/xeon) -fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse -msse2 -mfpmath=sse keep up the good work, Adam brian k. west wrote: Yes I realized my error in my wording but it was early :P It doesn't improve alot but does give you some ways to get a better idea of translation times if your box is loaded up with calls. bkw PS this patch was added to CVS-HEAD - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 5:49 PM Subject: Re: [Asterisk-Users] speex Actually it encodes a second of data, which with a 20ms codec would be 50 frames. The timing shows better than expected results due to caching. -Adam brian wrote: http://asterisk.bkw.org/diff/translate.patch.txt If you try that patch out it adds a nice feature... show translation recalc [xx] You can throw more than 1 sample thru it and recalculate your translation matrix. It also allows you to see TRUE translation under a load or just when ever you feel like seeing them updated. When * loads the codec it shoot one frame thru and times it. Now under real world scenarios you will be shooting more than one frame thru so LETS have the option to update the matrix with these types of tests. 200 is the max. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Monday, May 17, 2004 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] speex Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. As a quick example, here are the show translation outputs from * on a 2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5 (compiled with CFLAGS=-march=pentium4 and --enable-sse). Note how encoding from slin went from 25 to 15 ms. That is from the re-write of the sse optimized routines in libspeex. The % change is similar to what I saw on my p3 notebook, where both 1.0.3 and 1.1.5 were compiled with --enable-sse and -march=pentium3. (As a side note, these were captured before Brian's ilbc Makefile patch made it to the anon cvs tree; that optimization shaved 5ms off the time to encode to iLBC on that box.) spx103*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -26 19 ULAW - 3 - 1 2 2 1 6 -26 19 ALAW - 3 1 - 2 2 1 6 -26 19 G726 - 3 2 2 - 2 1 6 -26 19 ADPCM - 3 2 2 2 - 1 6 -26 19 SLINR - 2 1 1 1 1 - 5 -25 18 LPC10 - 4 3 3 3 3 2 - -27 20 9A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - - 19 ILBC - 5 4 4 4 4 3 8 - 8 - spx115*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC 23 - - - - - - - - - - - GSM - - 2 2 2 2 1 6 -16 19 ULAW - 3 - 1 2 2 1 6 -16 19 ALAW - 3 1 - 2 2 1 6 -16 19 G726 - 3 2 2 - 2 1 6 -16 19 ADPCM - 3 2 2 2 - 1 6 -16 19 SLINR - 2 1 1 1 1 - 5 -15 18 LPC10 - 4 3 3 3 3 2 - -17 20 9A - - - - - - - - - - - SPEEX - 3 2 2 2 2 1 6 - - 19 ILBC - 5 4 4 4 4 3 8 - 8 - -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com/voip ___ Asterisk-Users mailing
RE: [Asterisk-Users] CAPI-SIP broken incoming audio
Follow-up post to the one I sent last week regarding bad calls between SIP and ISDN. On Fri, 14 May 2004, Vic Cross wrote: To me, it looks like a variation of the SIP RTP timestamp problem (yes, my 7960 is at 6.3 code), but the problem exists on the ATA-186 too and I don't have any other issues with that (not even SIP-IAX, where the 7960 is really bad). I've done some crawling over ethereal traces, and have found the problem to indeed be bad timestamps in the RTP payload from *. I was advised (by JerJer on the IRC channel, thanks!) to update to latest CVS-HEAD, but the problem has recurred. What's happening is that the RTP frames from * are all going out with the same timestamp, which is causing my timestamp-sensitive 7960 to barf and ignore the incoming audio stream (it's interesting that X-Lite is largely trouble-free in this scenario!). On calls that work well, the RTP frames from * have a timestamp starting at 160 and incrementing normally (160,320,480...). On the bad calls, the timestamp is a very large number (4015105112, for example) and does not increment. So, next step is to look thru the code and find where the timestamp is initialised and incremented. In rtp.c, I found a couple of instances of a variable declared as signed int being used to hold the return value of the unsigned int function calc_txstamp(), but only time will tell if this fixes the problem (as it still takes anything up to a couple of days after an * restart before the problem occurs). What bugs me the most is that I can call SIP-SIP to the 7960 (from my ATA, for example) and the RTP timestamp is incremented correctly. Immediately afterwards I will call from ISDN, and I get bad timestamp. Which would imply that the generation of the timestamp is related to the source of the call, but I'm %$*ed if I can find where -- it seems to be time-of-day dependent, but nothing else (I can see where the codec seems to affect timestamps, but in my test case I'm using the same codec as the ISDN call). Finally, should I take this to asterisk-dev? Vic, It sounds like you've nailed the problem with the signed int statement. However, I'd suggest you open a bug report on this (rather than using list mail only) to get it some attention and tracking. It is very likely that other rtp channel drivers have the same issue as well. (We know it was a problem for iax/gsm - sip/rtp.) As for the Cisco dropping packets with uneven timestamps, that issue is totally unrelated to which codec is used; it affects all. In researching the Cisco bug tracking list, it would appear this particular problem is rated as a Sev 6 (lowest) and unless folks with smartnet maintenance contracts start pushing to increase the Sev level, its not likely to get fixed anytime soon. Sev 6 is basically considered cosmetic and not service impacting. Rich For what it is worth, the un-even rtp timestamp also impacted SCCP phones when used with Cisco's own conferencing solution. After almost two years of trying to fix the conferencing software, we were informed of the issue and it was suggested we update the phones to the 6.0.3 code. Since then we've seen a marked improvement. Now that Cisco has the fix in the SCCP code, maybe it'll find its way into the SIP code. Dan ___ 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
[Asterisk-Users] mgcp with busy tone
Hi there, ::: i have an mgcp phone (iph-90) using the sample mgcp.conf for it (the dlink section). i've tried both asterisk stable and development release but i'm getting the following error when i lift the receiver: . .. in stable branch: -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down while the phone is giving me busy tone . .. in development release: chan_mgcp.c:2227 handle_response: Terminating on result 502 from aaln/[EMAIL PROTECTED] while the phone after a short beep giving me busy tone too... any ideas? wiking ___ 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] User picks up phone, hears another call, not dialtone
Steve Creel wrote: snip I have heard complaints that once every couple weeks, when a user picks up their analog phone (t1 span off of a TE410P into an Adtran 750 with 6 FXS cards), they don't get dialtone, but instead, hear another conversation. I'm under the impression that they can only hear and not speak into that conference. We got the same problem after start * in few hours only restart the * can solve that problem... stable-1.0-branch is good idea to instead but the less functions than cvs version... System information: This has happened with zaptel and asterisk from November 2003, as well as after an upgrade in mid April 2004 (most recent occurance was with asterisk CVS from March 26, Zaptel from April 24th) System is a dual Xeon 2.4GHz, 1GB RAM Load avg: 0.04, 0.04, 0.00 Single TE410P with a channelized T1 from clec providing timing, 2 channelized T1s out to two Adtran 750s. Handsets are ATT/Lucent/Avaya 2554 sets asterisk-CVS-05-17-04, zaptel-CVS-May-17 Pentium4 2.8GHz with 1GB Ram T100P * 2 CarrierAccess ABI * 2 snip Many, many thanks. Steve ___ 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 -- Jim Kou Malico Inc. No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643 Tel : 886-3-472-8155#218 Fax : 886-3-472-5979 Site: http://www.malico.com.tw ___ 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] Dropped calls
I'm having a problem with outgoing dropped calls. They symptom is, when I place a call from a sip extension to the outside, the call is connected properly, but then abruptly disconnects anywhere from 10 to 60 seconds later. This happens when the outgoing call is through a POTS line (TDM) as well as over a sip gateway. Calls between sip extensions do not have this problem. Has anyone ever experienced this? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ 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] Dropped calls
do a 'sip debug' and make sure all looks good. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bruce Komito Sent: Monday, May 17, 2004 10:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dropped calls I'm having a problem with outgoing dropped calls. They symptom is, when I place a call from a sip extension to the outside, the call is connected properly, but then abruptly disconnects anywhere from 10 to 60 seconds later. This happens when the outgoing call is through a POTS line (TDM) as well as over a sip gateway. Calls between sip extensions do not have this problem. Has anyone ever experienced this? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 ___ 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
Re: [Asterisk-Users] indications.conf
Hi http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/ this is a useful link, (but not specific for Czech Republic ) I am looking this support for Bolivia (South America) On Mon, 2004-05-17 at 03:20, Dudlik wrote: Hello I am looking for Czech (Czech Republic) country support to indications.conf Have you ever seen it anywhere ? We are a small country in middle Europe :) thank you --NOTA DE REDCETUS S.R.L. : La informacin contenida en este E-mail y sus anexos, slo puede ser utilizada por el individuo o la compaa a la cual est dirigido. Si no es el receptor autorizado, cualquier retencin, difusin, distribucin o copia de este mensaje es prohibida y sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo y borrar el mismo
Re: [Asterisk-Users] problems compiling h323 support
MattB wrote: Try 'export LANG=C' then 'make clean make' Huh? Jeremy McNamara ___ 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] *8 problem still there?
I'm not seeing this - using stable CVS from 14-05-2004. Phone types are Grandstream Budgetone 101 using firmware 1.0.4.68, and Cisco 7940 using SIP 6.2. -Shaun On Mon, 17 May 2004, John Vogel wrote: I upgraded to the latest stable version of 1.0 today and am still seeing the *8 problem where the phone that was originally dialed keeps on ringing even after another phone picks up. Are other people also seeing this? Has somebody figured out how to make this go away? Thanks! ___ 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] 2.05a firmware
Now it's always proposing out of band DTMF. If there should be a user agent which does not support RFC2833 it will answer the SDP accordingly and then the phone automatically falls back to inband DTMF. The setting was previously necessary because some equipment could not deal with this negotiation process. However, it seems that such equipment does not exist on the market any more. VoIP is getting more mature! Christian -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lars Boegild Thomsen Sent: Monday, May 17, 2004 9:54 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware While we're at the 2.05 firmware - the DTMF handling on the Codec configuration page have disappeared. I assume this is because the phone now got some kind of default behaviour based on the codec. Can you describe that behaviour? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christian Stredicke Sent: 17 May 2004 15:21 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware 8 kHz 16 bit/sample (linear) mono WAV files. CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Thursday, May 13, 2004 7:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Does anyone know what kind of file needs to be uploaded for the custom ring tone? --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Huff Sent: Thursday, May 13, 2004 10:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 2.05a firmware Whoohoo, they added a way to upload ring tones! My life is now complete. They also added the 'Name+Number' callerID display mode, yay! Way to go SNOM! --Justin ___ 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 ___ 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 ___ 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] failed compile
Hi all I am trying to compile Asterisk on RH9
I am have installed
pwlib_1.5.2
readline-4.3-5
readline-devel-4.3-5
openssl-devel-0.9.7a-2
openssl-0.9.7a-2
openh323_1.12.2
I read that it has to do with pwlib not being installed correctly so i made
sure it was in /usr/local correctly
When i try to compile i get the follow errors
I was wondering can anyone shed in light on this issue
Thanks
g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT -DP_HAS_SEMAPHORES
-O3 -
DNDEBUG -DP_SSL -I../include -I/include -I/crypto
-DPNX_VERSION=\CVS-05/17/04-1
7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -I/include/ptlib/unix
-I/inc
lude -I/home/dictate/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c
chan_h323
.cxx -o chan_h323.o
In file included from /usr/include/ptlib/contain.h:222,
from /usr/include/ptlib.h:139,
from chan_h323.cxx:37:
/usr/include/ptlib/object.h:585: parse error before `(' token
/usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1201: parse error before `(' token
/usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field
/usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL'
/usr/include/ptlib/object.h:1201: conflicts with previous declaration `int
[ snip ]
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[Asterisk-Users] Corrupt Callerid Data
Hi, The incoming caller id on the X101P always comes up scrambled except when there is no name, just a number. Usually a cellphone would do this, and the number is perfect. I was reading posts about using ztmonitor to capture the spill and listening to it. The resulting file is alway 0 bytes... Also, how do you interpet the output of ztmonitor, is it suppose to be near the middle? Thanks, -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
[Asterisk-Users] Re: speex
brian == brian k west [EMAIL PROTECTED] writes: brian I toyed with -msse and -mmmx and others too but couldn't put brian any of those in. :P The options -msse, -msse2, -mmmx et al are all implied by the relevant -march options. uname only reports i686, so you have to use some other construct to get things like -march=pentium3 (implies -msse and -mmmx), -march=pentium4 (also implies -msse2) etc for k7 and k8. On amd64 when compiling -m64 almost everything useful is already implied (k8 does not support -msse3 and the intel amd64 chips lack one thing that -march=k8 implies). Similarly, -O3 implies -funroll-loops. Since there is no portable way to automatically decide -march for pentium3, pentium4, k7, k8, et al the starndard practice is to specify them manually at configure or compile time. Perhaps the top Makefile should have an MARCH= line that defaults to $(shell uname -m) but can be changed once by the user and is propagated down to all subdirs in a useful way (including as part of CFLAGS for all configure calls). Then one could either edit the top Makefile or run make MARCH= -JimC ___ 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
