Re: [Asterisk-Users] Re: How to best debug SIP registration failure
If you see nothing with full verbosity and SIP debug turned on, the Asterisk SIP channel gets nothing. The reason why we always mix in NAT with questions like yours is that in 90% of the cases, NAT is the problem. It's just a standard response, like when Microsoft support tells you to reinstall windows :-) Do you see any packets going to and from FWD when using SIP debug? You should. If you don't see any packets with SIP DEBUG and can still see that Asterisk registers with FWD, there's a lot of fishing to do in your system :-) If you see those packets going to and from FWD, but not the Grandstream registering, there's a problem between the IP stack and Asterisk. If you have multiple interfaces, the problem might be the IP address Asterisk bind to and the routing between the IP address Grandstreams send packets to. Still an apprentice on how to fish with Asterisk... Eager to know what goes on in your system, so we can document it and maybe fix it. /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] SIP overlap (early dial) 484 response
Hello, I have one question again. I checked archive and I found that somebody before me asked this question already. But no responses for this posting. http://lists.digium.com/pipermail/asterisk-users/2003-September/020065.html So, is it supported or no? If yes, what I need to configure? Thank you. Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ 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] About Grandstream ATA-286 and ring voltage
Dear all, My GS ATA-286, which otherwise work well, seem to be unable to ring a fax (or at least, some kind of fax). The fax basically doesn't detect the ring. I measured with a volt meter about 45V during the ring pulse out of the ATA. This looks fairly low to me (supposed to be in the 70V+ range, isn't it ?). The adapter works with evey kind of phone I tried, but did not work with two different fax machines. Am I simply out of luck with these fax ? Does my ATA look defective (tried two of them, however) ? -- Nicolas Bougues Axialys Interactive ___ 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 questions about ISDN BRI
Hi
I have an ISA Diva 2.0 ISDN card and i am using i4l
as well, and i use the same calling method, it workd for me.
Can u show your modem.conf?
remember to use in modem.conf
driver=i4l
and
group=1
msn=0
incomingmsn=XXX ; your
incoming numbers
device = /dev/ttyI0
device = /dev/ttyI1
Diego
- Original Message -
From: "Tomica Crnek" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 22, 2004 8:08
PM
Subject: RE: [Asterisk-Users] 2 questions about
ISDN BRI
Thanks for stripmsd=0, it helped! It supports groups
also. It is one US Robotics card
[chan_modem.so] = (Generic Voice Modem Driver) ==
Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver
chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem
Driver) == Loading modem driver chan_modem_i4l.so =
(ISDN4Linux Emulated Modem Driver) --
Configured modem /dev/ttyI0 with driver i4l (Linux ISDN)
-- Configured modem /dev/ttyI1 with driver i4l (Linux
ISDN) -- Configured modem /dev/ttyI2 with driver i4l
(Linux ISDN) -- Configured modem /dev/ttyI3 with
driver i4l (Linux ISDN) -- Configured modem
/dev/ttyI4 with driver i4l (Linux ISDN) -- Configured
modem /dev/ttyI5 with driver i4l (Linux ISDN) == Registered
channel type 'Modem' (Generic Voice Modem Channel Driver)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp
von Klitzing Sent: Sunday, February 22, 2004 7:50 PM To:
[EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 questions about ISDN BRI
Hi! I am able to receive calls to Asterisk via
BRI, but when I attempt to dial out from Asterisk to ISDN I always
get-- Executing
Dial("SIP/1001-3fb0", "Modem/g1:6658218") in new stack
-- Called g1:6658218
-- Modem[i4l]/ttyI1 is busy
-- Hungup 'Modem[i4l]/ttyI1'
Change stripmsd=1 in modem.conf to stripmsd=0. It might help if you tell
us which ISDN card you are using. You should also move away from i4l and
use chan_capi instead once you completed your initial tests and
trials. http://www.voip-info.org/wiki-Asterisk+CAPI+Channels Is it possible to emulate ISDN NT device
with Asterisk and connect other ISDN devices directly to Asterisk?
If it is, what do I have to do to switch Asterisk behavior to
NT. Search this list or look here: http://www.voip-info.org/wiki-Asterisk+zaphfc http://www.voip-info.org/wiki-zaptelBRI Cheers, Philipp
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RE: [Asterisk-Users] 2 questions about ISDN BRI
it is solved, thanks!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
dfmSent: Monday, February 23, 2004 9:14 AMTo:
[EMAIL PROTECTED]Subject: Re: [Asterisk-Users] 2
questions about ISDN BRI
Hi
I have an ISA Diva 2.0 ISDN card and i am using i4l
as well, and i use the same calling method, it workd for me.
Can u show your modem.conf?
remember to use in modem.conf
driver=i4l
and
group=1
msn=0
incomingmsn=XXX ; your
incoming numbers
device = /dev/ttyI0
device = /dev/ttyI1
Diego
- Original Message -
From: "Tomica Crnek" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 22, 2004 8:08
PM
Subject: RE: [Asterisk-Users] 2 questions about
ISDN BRI
Thanks for stripmsd=0, it helped! It supports groups
also. It is one US Robotics card
[chan_modem.so] = (Generic Voice Modem Driver) ==
Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver
chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem
Driver) == Loading modem driver chan_modem_i4l.so =
(ISDN4Linux Emulated Modem Driver) --
Configured modem /dev/ttyI0 with driver i4l (Linux ISDN)
-- Configured modem /dev/ttyI1 with driver i4l (Linux
ISDN) -- Configured modem /dev/ttyI2 with driver i4l
(Linux ISDN) -- Configured modem /dev/ttyI3 with
driver i4l (Linux ISDN) -- Configured modem
/dev/ttyI4 with driver i4l (Linux ISDN) -- Configured
modem /dev/ttyI5 with driver i4l (Linux ISDN) == Registered
channel type 'Modem' (Generic Voice Modem Channel Driver)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp
von Klitzing Sent: Sunday, February 22, 2004 7:50 PM To:
[EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 questions about ISDN BRI
Hi! I am able to receive calls to Asterisk via
BRI, but when I attempt to dial out from Asterisk to ISDN I always
get-- Executing
Dial("SIP/1001-3fb0", "Modem/g1:6658218") in new stack
-- Called g1:6658218
-- Modem[i4l]/ttyI1 is busy
-- Hungup 'Modem[i4l]/ttyI1'
Change stripmsd=1 in modem.conf to stripmsd=0. It might help if you tell
us which ISDN card you are using. You should also move away from i4l and
use chan_capi instead once you completed your initial tests and
trials. http://www.voip-info.org/wiki-Asterisk+CAPI+Channels Is it possible to emulate ISDN NT device
with Asterisk and connect other ISDN devices directly to Asterisk?
If it is, what do I have to do to switch Asterisk behavior to
NT. Search this list or look here: http://www.voip-info.org/wiki-Asterisk+zaphfc http://www.voip-info.org/wiki-zaptelBRI Cheers, Philipp
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[Asterisk-Users] Re: How to best debug SIP registration failure (Solved)
Thank you to Olle Johansson, Philipp von Klitzing, and others who suggested approaches to the problem. To summarise what we did and how we ended up solving the problem: Situation: 1. Grandstream phone behind NAT box. 2. Asterisk not behind NAT (with static IP). 3. Phone cannot register. Problem diagnosis: 1. Tested phone with FWD (http://www.pulver.com/fwd/) and was able to register and run echo test successfully. This indicates we have NAT configured properly and know enough about configuring the phone to have some hope to get it working with Asterisk 2. Ran asterisk with verbose debuggery and sip debug. 3. Ran tcpdump and tethereal to see what packets were arriving and departing the Asterisk server. 4. Ran netstat to verify that something was listening on the appropriate ports on the Asterisk box. What we found: (a) tcpdump and tethereal saw traffic from phone but nothing from Asterisk. (b) sip debug reported nothing at all. Putting all of the above together we were left with a number of hypothesises the most likely of which was that the registration packets (which we could see arriving on the net) were being blocked and not being seen by Asterisk. Further research revealed that the system had iptables that were blocking the ports needed. Once we reconfigured the iptables, things worked fine. Sorry to have troubled the list and thank you for helping me towards discovering the solution (and providing valuable advice that will no doubt be invaluable in diagnosing future problems). g. ___ 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] IAX2 Call menu handling problem with Norstar - Partial Solution
Well after a bit more googling, I've found the quick nasty fix to this problem. Users on the Norstar extensions need to dial Feature 808 to enable Long Tones so that when they press a key on their keypad, it's passed correctly to the Analog Terminal Adapter. I call this a partial solution, since this feature only works on a per-call basis. However it would seem to me that this was happening already, just that for some reason the Norstar extension then stops sending/receiving on the voice channel... maybe it's a bug or just a Norstar Feature. I read somewhere that they are one of the worst PBX's to try and integrate with, and my experiences so far definitely concur with that, particularly with the ATA's... no disconnect supervision, can't pass a DTMF properly from digital to analog, argh! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Monday, 23 February 2004 2:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IAX2 Call menu handling problem with Norstar Ok, after much stuffing around with the configs to sort it, I've narrowed the problem down to DTMF passing from the Norstar extension as being what breaks my setup. If I'm on a call with someone on a Norstar extension from my system, and they press a key, I hear a split second of the DTMF signal and the line goes silent. Now I've just got to figure a way to get the Norstar Asterisk to work together in DTMF harmony :-) ___ 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] EMEA and Chagres Technologies
John, You are now advertising your EMEA company in your signature block. Maybe I missed an email that explains the EMEA pricing and availability. Could you please give an update via the list as to the status of your product availablity, pricing and delivery times in Europe? The ordering procedure would be nice too assuming that you are able to deliver to the EU from the EU. Dealing with customs charges for individual shipments from the USA is not desirable. Thank you, Robert John Brown (CV) said: . john brown chagres technologies, inc (Americas) chagres technologies, b.v. (EMEA) ___ 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 extension busy when not available ??
Soren Rathje said: - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 22, 2004 8:52 PM Subject: Re: [Asterisk-Users] SIP extension busy when not available ?? Although the current logic does not require a sip phone to register, it would seem like the asterisk logic should be something like: a. call is attempted to sip x1234, b. if * knows the extn is in use, return busy, or, c. if not busy, asterisk attempts to contact x1234 across the wire, d. if no contact, return Unavailable Or if not registred. In a wireline telephony scenario the above would be the proper method as we do not know if the subscriber have their phone plugged in or not. With Asterisk we experience the same information as mobile operators do: unreachable, unanswered and busy. IMHO we should have the same options. So, the priority for a type=friend would be: a: check if client is registered and/or reachable, if not - return unreachable b: check if client is busy, if call-waiting not active - return busy c: if call is rejected by client, return approriate message d: if call is unanswered, return unavailable or busy with reference to (b). -- Soren I use ChanIsAvail() to check to see if the phone is connected at the top of the dialplan for that extension. This works for IAX2 and SIP channels but not for MGCP. If you are interested in the actual code I can send it to you from home tonight. Robert ___ 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] About Grandstream ATA-286 and ring voltage
The only adapter that I know of that allows you to modify the ring voltage is the Sipura analog SIP adapter. I was able to get my old fax machine to answer after jacking up the ring voltage to 90V. http://www.sipura.com MATT--- -Original Message- From: Nicolas Bougues [mailto:[EMAIL PROTECTED] Sent: Monday, February 23, 2004 3:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] About Grandstream ATA-286 and ring voltage Dear all, My GS ATA-286, which otherwise work well, seem to be unable to ring a fax (or at least, some kind of fax). The fax basically doesn't detect the ring. I measured with a volt meter about 45V during the ring pulse out of the ATA. This looks fairly low to me (supposed to be in the 70V+ range, isn't it ?). The adapter works with evey kind of phone I tried, but did not work with two different fax machines. Am I simply out of luck with these fax ? Does my ATA look defective (tried two of them, however) ? -- Nicolas Bougues Axialys Interactive ___ 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] New Wiki page: Dimensioning an Asterisk system
Title: Re: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system Is the article correct in saying: g729 codecs licenses can be purchased for Asterisk (not for SCSI systems!) I thought people had this working on SCSI now? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: 23 February 2004 04:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system Hi all Sorry for the last post! Not enough sleep combined with inattention caused me to reply to the wrong message. Sean -Original Message- From: Anton Tinchev [mailto:[EMAIL PROTECTED] Sent: Mon 2/23/2004 12:25 AM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system hmm, this pages must be fixed. Looks terrible on all NGlayout based browsers Philipp von Klitzing wrote: Hi there, please comment and adjust or enhance as you find appropriate: http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning Typical questions asked on the mailing asterisk-users are: How fast/big must my machine be in order to serve my needs? How many simultaneous calls can Asterisk handle? Unfortunately there are no simple answers. You'll need work through the following checklist to at least get nearer to an answer or be able to post a meaningful question to asterisk-users: [...] Cheers, Philipp ___ 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: Not Woodpeckers
Jose Quinteiro wrote: I live at sea level, and have never seen a woodpecker going at any telco equipment, but have a 60Hz hum on my POTS line through my Adtran 750. It goes away if I pick up the telephone I have cross-connected on the same line. Could it be the same problem (i.e., tip-ring imbalance?) Thanks, Jose. Lots of phones used to have a separate (3rd) ground wire delivered to the phone which was used in party line set ups. When very young, I discovered that connecting a phone tip to ground or ring to ground would result in a fabulous hum. I suspect that you have inadvertently connected one of the phones tip to ground or ring to ground. Stephen R. Besch ___ 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-Users digest, Vol 1 #2879 - 10 msgs
I think international number dialed through voicepulse have to start with 011... (even if you're located in another countery). I asked them about that once, and that's what works for me (We've been dialing Spain and Germany recently, but never Japan) HTH, Matt --__--__-- Message: 4 Date: Sat, 21 Feb 2004 09:04:43 -0300 From: Daniel Bichara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse Connection Reply-To: [EMAIL PROTECTED] Hi, I have two * connecteds and I wish a phone connected to * #1 calls PSTN via Voicepulse connected to * #2, as follows: telephone --- Asterisk #1 Asterisk #2 Voicepulse When I dial 81-90... (japan), * #1 will route call to Voicepulse at *#2. Everything works between #1 and #2 but when #2 calls Voicepulse I get an error message: -- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED] Feb 21 10:01:34 WARNING[98311]: chan_iax2.c:4445 socket_read: Call rejected by 66.234.228.132: No such context/extension I am clueless!!! What could it be? Follow my confs... Exten.conf - *#1 exten = _81.,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) # Exten.conf - *#2 [outvoicepulse] exten = _.,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _.,2,Congestion # Iax.conf - *#2 [voicepulse] context=VPWS secret=password auth=md5 type=friend host=66.234.228.132 disallow=all allow=speex allow=gsm jitterbuffer=no Daniel --__--__-- ___ 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-Users digest, Vol 1 #2879 - 10 msgs
In the UK it's 00 then the country code. So a call from the UK to my phone would be 0013036742575. Miie Matt Lawson wrote: I think international number dialed through voicepulse have to start with 011... (even if you're located in another countery). I asked them about that once, and that's what works for me (We've been dialing Spain and Germany recently, but never Japan) HTH, Matt --__--__-- Message: 4 Date: Sat, 21 Feb 2004 09:04:43 -0300 From: Daniel Bichara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse Connection Reply-To: [EMAIL PROTECTED] Hi, I have two * connecteds and I wish a phone connected to * #1 calls PSTN via Voicepulse connected to * #2, as follows: telephone --- Asterisk #1 Asterisk #2 Voicepulse When I dial 81-90... (japan), * #1 will route call to Voicepulse at *#2. Everything works between #1 and #2 but when #2 calls Voicepulse I get an error message: -- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED] Feb 21 10:01:34 WARNING[98311]: chan_iax2.c:4445 socket_read: Call rejected by 66.234.228.132: No such context/extension I am clueless!!! What could it be? Follow my confs... Exten.conf - *#1 exten = _81.,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) # Exten.conf - *#2 [outvoicepulse] exten = _.,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _.,2,Congestion # Iax.conf - *#2 [voicepulse] context=VPWS secret=password auth=md5 type=friend host=66.234.228.132 disallow=all allow=speex allow=gsm jitterbuffer=no Daniel --__--__-- ___ 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 Consulting, Inc. 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] Pingtel Opensource PBX Announcement
On Sunday, February 22, 2004 2:04 AM, James H. Thompson [SMTP:[EMAIL PROTECTED] wrote: Other vendors are seeing the benefits of open source: From: http://www.pingtel.com/a_opensource.jsp Announcing the emergence of an enterprise-class open source IP PBX Tapping into open source Pingtel's open source business model will prove to be a fundamentally disruptive force in the $5 billion per year enterprise PBX market. By offering enterprise- class, all SIP-based, open source IP PBX software under a Linux-style subscription license, I do not know what 'Linux-style subscription license' means. Is it GPL? I do know that not all open source is not the same. -- Don Pobanz ___ 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-Users digest, Vol 1 #2879 - 10 msgs
Please cancel my previous post. Matt Lawson wrote: I think international number dialed through voicepulse have to start with 011... (even if you're located in another countery). I asked them about that once, and that's what works for me (We've been dialing Spain and Germany recently, but never Japan) HTH, Matt --__--__-- Message: 4 Date: Sat, 21 Feb 2004 09:04:43 -0300 From: Daniel Bichara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicepulse Connection Reply-To: [EMAIL PROTECTED] Hi, I have two * connecteds and I wish a phone connected to * #1 calls PSTN via Voicepulse connected to * #2, as follows: telephone --- Asterisk #1 Asterisk #2 Voicepulse When I dial 81-90... (japan), * #1 will route call to Voicepulse at *#2. Everything works between #1 and #2 but when #2 calls Voicepulse I get an error message: -- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED] Feb 21 10:01:34 WARNING[98311]: chan_iax2.c:4445 socket_read: Call rejected by 66.234.228.132: No such context/extension I am clueless!!! What could it be? Follow my confs... Exten.conf - *#1 exten = _81.,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) # Exten.conf - *#2 [outvoicepulse] exten = _.,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _.,2,Congestion # Iax.conf - *#2 [voicepulse] context=VPWS secret=password auth=md5 type=friend host=66.234.228.132 disallow=all allow=speex allow=gsm jitterbuffer=no Daniel --__--__-- ___ 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 Consulting, Inc. 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
[Asterisk-Users] Re: Voicepulse Connection
That's what I'm trying to get at. *normally* you expect to dial 00 but
when you're using voicepulse, Asterisk needs to start all international
number with 011. Think of it this way, in VoicePulse's mind, you're
always dialing from the US. Of course the user will try dialing 00
because that's the 'normal' way to do it.
So what you have to do is change your dial plan to intercept a 00
prefix and reformat it using a 011 prefix, something like this:
exten = _00[1-9].,2,Dial,IAX2/[EMAIL PROTECTED]/011${EXTEN:2}
Message: 9
Date: Mon, 23 Feb 2004 07:49:07 -0700
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2879 - 10
msgs
Reply-To: [EMAIL PROTECTED]
In the UK it's 00 then the country code. So a call from the UK to my
phone would be 0013036742575.
Miie
Matt Lawson wrote:
I think international number dialed through voicepulse have to start
with 011... (even if you're located in another countery). I asked them
about that once, and that's what works for me (We've been dialing Spain
and Germany recently, but never Japan)
HTH,
Matt
-- __--__--
Message: 4
Date: Sat, 21 Feb 2004 09:04:43 -0300
From: Daniel Bichara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicepulse Connection
Reply-To: [EMAIL PROTECTED]
Hi,
I have two * connecteds and I wish a phone connected to * #1 calls
PSTN via Voicepulse connected to * #2, as follows:
telephone --- Asterisk #1 Asterisk #2 Voicepulse
When I dial 81-90... (japan), * #1 will route call to Voicepulse at
*#2. Everything works between #1 and #2 but when #2 calls Voicepulse I
get an error message:
-- Called user:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Feb 21 10:01:34 WARNING[98311]: chan_iax2.c:4445 socket_read: Call
rejected by 66.234.228.132: No such context/extension
I am clueless!!! What could it be? Follow my confs...
Exten.conf - *#1
exten = _81.,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
# Exten.conf - *#2
[outvoicepulse]
exten = _.,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = _.,2,Congestion
# Iax.conf - *#2
[voicepulse]
context=VPWS
secret=password
auth=md5
type=friend
host=66.234.228.132
disallow=all
allow=speex
allow=gsm
jitterbuffer=no
Daniel
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Re: [Asterisk-Users] Pingtel Opensource PBX Announcement
Don Pobanz wrote: On Sunday, February 22, 2004 2:04 AM, James H. Thompson [SMTP:[EMAIL PROTECTED] wrote: Other vendors are seeing the benefits of open source: From: http://www.pingtel.com/a_opensource.jsp Announcing the emergence of an enterprise-class open source IP PBX Tapping into open source Pingtel's open source business model will prove to be a fundamentally disruptive force in the $5 billion per year enterprise PBX market. By offering enterprise- class, all SIP-based, open source IP PBX software under a Linux-style subscription license, I do not know what 'Linux-style subscription license' means. Is it GPL? I do know that not all open source is not the same. -- Don Pobanz This is an interesting statement in the press release.. SIPxchange, the industrys first open source based enterprise communications suite, is grounded in the concept that a community of ideas provides a more fertile ground for innovation, progress and product development. I guess they haven't heard of Asterisk.. :) ___ 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: Pingtel Opensource PBX Announcement
Don == Don Pobanz [EMAIL PROTECTED] writes: Don I do not know what 'Linux-style subscription license' means. That one stalled me for a bit, too. Based on their ad copy they are offering annual support contracts for the system, but releasing the code itself under some free/open license. (I forget which one and am going to be too lazy to look it up. :) -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
[Asterisk-Users] Processor load spikes
I always keep a terminal window open with top running for my asterisk servers. Since we've had Asterisk in production, for about 9 months, I've noticed with every platform and every card we've tried that the load average will be going along at about 0.1 to 0.5 with about 30 channels(15 SIP - Zap conversations) going and then at seemingly random times the load average will jump to over 2.0. All the while the processor idle never goes below 50%. Does anyone know what the asterisk process is doing that causes these load jumps? (I have determined that initiating new calls or hanging up calls is not a factor in the timing of these jumps) Does anyone not experience these load jumps? This occurs on all hardware platforms that I've tested: P3 non-SMP, P4 non-SMP, P4 SMP, AMD non-SMP and AMD SMP using all available Digium T1 cards: wct410p, 400p and 100p The only common element is RedHat 9.0 as the OS and the fact the there is no other large service running on the machines(no web, no DB, no X) I have tested other resource-intensive applications(like MySQL in a constant loop of ordered selects of 1 million records) and not seen any other instances of load spikes on these systems. I have loaded up the channels on a test server to see what will happen is the load spikes while it is already at 2.0 and with 100 channels(50 SIP- Zap conversations) it ran for 4 hours with the load averaging around 2.0(on non-SMP P4) and then I got a spike and the load went upto 8.0 and the server crashed. I would like to find out why asterisk is doing this just to satisfy my own curiosity, but if anything can be done about it I could get a lot more out of the servers I have before having to buy more whenever I need to increase capacity. MATT--- ___ 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] An example config for using a Wildcard X100P and a SIP phone?
Hello. I've just recently purchased the Asterisk Developers Kit so we can figure out how to get away from our Nortel system and go to IP based phones. I have a RH 9 box loaded with Asterisk (a very recent cvs download). Either way, I can call the asterisk box and get their demo playing fine. I can even call the SIP phone I've hooked up when I call in from my cell phone to the asterisk box, and that works. I cannot call out with my SIP phone though. It'll dial, ring my cell phone twice and then give up and complain that its busy. Even if I try to answer the cell phone during the first ring. Does anyone have a config they could share with me on how to make this setup work? This sounds like it should be fairly trivial, but I've beaten my head against the wall on this for a few days. =) Thanks alot, Jason ___ 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] OT: SNOM and TAPI
Snom TAPI integration is a joke... Andy *** REPLY SEPARATOR *** On 22/02/2004 at 21:47 Peer Oliver schmidt wrote: Hi, anyone here running SNOM phones with TAPI integration with Outlook? Any other hardware phone with some TAPI integration? rgds pos ___ 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] An example config for using a Wildcard X100P and a SIP phone?
Jason, Include your sip and extensions files so people can take a look. T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Sent: Monday, February 23, 2004 10:25 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone? Hello. I've just recently purchased the Asterisk Developers Kit so we can figure out how to get away from our Nortel system and go to IP based phones. I have a RH 9 box loaded with Asterisk (a very recent cvs download). Either way, I can call the asterisk box and get their demo playing fine. I can even call the SIP phone I've hooked up when I call in from my cell phone to the asterisk box, and that works. I cannot call out with my SIP phone though. It'll dial, ring my cell phone twice and then give up and complain that its busy. Even if I try to answer the cell phone during the first ring. Does anyone have a config they could share with me on how to make this setup work? This sounds like it should be fairly trivial, but I've beaten my head against the wall on this for a few days. =) Thanks alot, Jason ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Pingtel Opensource PBX Announcement
WipeOut wrote: This is an interesting statement in the press release.. SIPxchange, the industrys first open source based enterprise communications suite, is grounded in the concept that a community of ideas provides a more fertile ground for innovation, progress and product development. I guess they haven't heard of Asterisk.. :) They keep saying enterprise class in the blurb, and then they say suitable for up to 400 lines. I seem to remember James T Kirk has 400 people on the Enterprise. I guess that must be their target market. If not, I think they have a scalability issue. :-) Regards, 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
[Asterisk-Users] Thread-safe applications
I'm writing an application for asterisk (really just a set of access commands to the builtin API), and I notice that a lot of existing applications are not thread-safe. Should they be? Should mine be? Thanks, --Ernest ___ 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] OT: SNOM and TAPI
Andy Powell wrote: Snom TAPI integration is a joke... Would you mind elaborating a bit on this? Is the future implemented, but does not work, or is it not implemented at all? Or something else? Thanks rgds pos ___ 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] Thread-safe applications
On Monday 23 February 2004 10:15, Ernest W. Lessenger wrote: I'm writing an application for asterisk (really just a set of access commands to the builtin API), and I notice that a lot of existing applications are not thread-safe. Should they be? Should mine be? Could you elaborate, please? What specific applications are not thread-safe and what aspect makes them not thread-safe? -Tilghman ___ 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] Thread-safe applications
At 08:31 AM 2/23/2004, you wrote: On Monday 23 February 2004 10:15, Ernest W. Lessenger wrote: I'm writing an application for asterisk (really just a set of access commands to the builtin API), and I notice that a lot of existing applications are not thread-safe. Should they be? Should mine be? Could you elaborate, please? What specific applications are not thread-safe and what aspect makes them not thread-safe? Whoops, you're right, the String Manipulation function I was looking at is thread-safe (but some it it's variants aren't). Regardless, do Applications need to be thread safe? Thanks, --Ernest ___ 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] Attended Transfer Question
Hello, I was curious if there was any way to play a tone on Attended transfer once it bridges the party being transferred to the destination? Basically what is happening now is: 1.) A caller calls in using a zap channel 2.) Call is sent to SIP Polycom Phone - Receptionist 3.) Receptionist Forwards calls (mostly attended transfers) 4.) She talks to the party the call is for 5.) Presses transfer and the call is connected to the new party. All of this happens with the party that the call is intended for not knowing if they are still on the line with the receptionist or the new party. Is there some sort of indication I can give, e.g. a beep or some sort of audible indication that the party has been transferred and thus they can say hello? Now they are sitting there and wait a couple of seconds to say hello, when really they shouldn't be? Any input would be greatly appreciated! Thanks, Brent D. Franks Mindworks Internet Services ___ 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] Thread-safe applications
On Mon, 2004-02-23 at 10:55, Ernest W. Lessenger wrote: At 08:31 AM 2/23/2004, you wrote: On Monday 23 February 2004 10:15, Ernest W. Lessenger wrote: I'm writing an application for asterisk (really just a set of access commands to the builtin API), and I notice that a lot of existing applications are not thread-safe. Should they be? Should mine be? Could you elaborate, please? What specific applications are not thread-safe and what aspect makes them not thread-safe? Whoops, you're right, the String Manipulation function I was looking at is thread-safe (but some it it's variants aren't). Regardless, do Applications need to be thread safe? For inclusion in the main tree it should be, and you may get pummeled about the head with a blunt object if someone is using it and it crashes a main machine. Why would you program something that isn't thread safe? From what I can tell, it isn't much extra effort to do things the right way instead of debuging crap later. -- 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] Thread-safe applications
At 09:14 AM 2/23/2004, you wrote: Why would you program something that isn't thread safe? From what I can tell, it isn't much extra effort to do things the right way instead of debuging crap later. I wouldn't, and generally don't. But sometimes (rarely) you need to include functions that aren't thread-safe (ex. specialized operations from vendors who charge a lot of money for poorly-written APIs) and it's good to know what the requirements are. --Ernest ___ 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] Processor load spikes
On Mon, 2004-02-23 at 09:19, mattf wrote: I always keep a terminal window open with top running for my asterisk servers. Since we've had Asterisk in production, for about 9 months, I've noticed with every platform and every card we've tried that the load average will be going along at about 0.1 to 0.5 with about 30 channels(15 SIP - Zap conversations) going and then at seemingly random times the load average will jump to over 2.0. All the while the processor idle never goes below 50%. Does anyone know what the asterisk process is doing that causes these load jumps? (I have determined that initiating new calls or hanging up calls is not a factor in the timing of these jumps) First a word on load averages as opposed to percent idle of CPU. Load average is the average number of processes awaiting cpu service. A process could be idle if it has no real work to complete and has allowed the CPU to skip on to another process. Percent idle is easier to understand as it is how much of the CPU's time is spent waiting for a process to need servicing. The problem of using top to monitor load is much like quantum physics, you change the value when you observe it. So part of your spike may be in timing of the observation. There are many operations that could affect the load average. Any new threads loading would be in a high busy state until the loading period id over and the process starts idle looping. Load mozilla up sometime while watching the load on your system shoot up. Your percent idle may still stay smallish since it is mostly exercising the disk subsystem and the CPU is waiting most of that time. If you are seeing a load average climb, you should identify the processes starting or running at that time. If it is falling, the processes have either completed the busy cycle, or have gone away. It is still likely though that you are seeing some errant behavior in RH9 caused by the new thread library. There may be a broken select function or something similar that is causing your trouble. Maybe you should try an older RH, or a different distribution and see if this happens as well. I have loaded up the channels on a test server to see what will happen is the load spikes while it is already at 2.0 and with 100 channels(50 SIP- Zap conversations) it ran for 4 hours with the load averaging around 2.0(on non-SMP P4) and then I got a spike and the load went upto 8.0 and the server crashed. Did the whole system crash or did just asterisk crash? If it was just asterisk, did you get a core dump and did you do a backtrace on it? -- 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] Thread-safe applications
On Mon, 2004-02-23 at 11:22, Ernest W. Lessenger wrote: At 09:14 AM 2/23/2004, you wrote: Why would you program something that isn't thread safe? From what I can tell, it isn't much extra effort to do things the right way instead of debuging crap later. I wouldn't, and generally don't. But sometimes (rarely) you need to include functions that aren't thread-safe (ex. specialized operations from vendors who charge a lot of money for poorly-written APIs) and it's good to know what the requirements are. Remember that in asterisk we are working with a GPL piece of software. We shouldn't run into the vendor supplied poorly written APIs unless it is for in house work only. Remember that any inclusion of non free software into asterisk requires you either be able to make the software free or it can not be redistributed as part of the software. So if you want anything to become part of the main asterisk tree, it should be thread-safe and free, not to mention the obligatory disclaimer to Digium. -- 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] Processor load spikes
Thanks for the response. I plan on trying Slackware on my backup/test asterisk server when I have a new backup server ready in a few weeks. I've noticed in some database machine testing that Slackware starts up in about half the time of RedHat and doesn't have all of that Redhat junk either. I'll post my results running Slackware after I've had time to test it. When I said crashed I meant that the whole operating system crashed, so no backtrace possible. Thanks, MATT--- -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Monday, February 23, 2004 12:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Processor load spikes On Mon, 2004-02-23 at 09:19, mattf wrote: I always keep a terminal window open with top running for my asterisk servers. Since we've had Asterisk in production, for about 9 months, I've noticed with every platform and every card we've tried that the load average will be going along at about 0.1 to 0.5 with about 30 channels(15 SIP - Zap conversations) going and then at seemingly random times the load average will jump to over 2.0. All the while the processor idle never goes below 50%. Does anyone know what the asterisk process is doing that causes these load jumps? (I have determined that initiating new calls or hanging up calls is not a factor in the timing of these jumps) First a word on load averages as opposed to percent idle of CPU. Load average is the average number of processes awaiting cpu service. A process could be idle if it has no real work to complete and has allowed the CPU to skip on to another process. Percent idle is easier to understand as it is how much of the CPU's time is spent waiting for a process to need servicing. The problem of using top to monitor load is much like quantum physics, you change the value when you observe it. So part of your spike may be in timing of the observation. There are many operations that could affect the load average. Any new threads loading would be in a high busy state until the loading period id over and the process starts idle looping. Load mozilla up sometime while watching the load on your system shoot up. Your percent idle may still stay smallish since it is mostly exercising the disk subsystem and the CPU is waiting most of that time. If you are seeing a load average climb, you should identify the processes starting or running at that time. If it is falling, the processes have either completed the busy cycle, or have gone away. It is still likely though that you are seeing some errant behavior in RH9 caused by the new thread library. There may be a broken select function or something similar that is causing your trouble. Maybe you should try an older RH, or a different distribution and see if this happens as well. I have loaded up the channels on a test server to see what will happen is the load spikes while it is already at 2.0 and with 100 channels(50 SIP- Zap conversations) it ran for 4 hours with the load averaging around 2.0(on non-SMP P4) and then I got a spike and the load went upto 8.0 and the server crashed. Did the whole system crash or did just asterisk crash? If it was just asterisk, did you get a core dump and did you do a backtrace on it? -- 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 ___ 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] Minimum voice mail message limit?
Looking through the Wiki and mailing list, I didn't see an answer to this. Is there a way to set the minimum voice mail message size? Hangups seem to generate 4 to 5 second messages. If I set a min to 6 or 7 that should eliminate most of these. The main voicemail app also seems kind of thin. There are no caller options such as playing back a message you left, deleting it and starting over if you mess up, etc. Voicemailmain also is rather thin - you can't listed to your currently available greetings for example. Is there an alternative voicemail at this time? Patches? FYI, I'm running * from CVS as of Feb 19. ___ 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] An example config for using a Wildcard X100P and a SIP phone?
Timothy,
I have minimally modified the demo files that came with Asterisk, so
what is posted below is most of the comments and the demo section
removed from the config files.
Thanks!
; SIP Configuration for Asterisk
;
[general]
port = 5060; Port to bind to
bindaddr = 0.0.0.0; Address to bind to
context = default; Default for incoming calls
[sipphone]
type=friend
username=sipphone
fromuser=Sipster; Specify user to put in from instead
of callerid
secret=password
host=dynamic
defaultip=192.168.1.201
amaflags=default; Choices are default, omit, billing,
documentation
accountcode=Sipster ; Users may be associated with an
accountcode tp ease billing
mailbox=431
--
extensions.conf
--
[general]
static=yes
writeprotect=no
[globals]
;CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/1; Trunk interface
TRUNKMSD=1; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
[iaxtel700]
exten = _91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[trunkint]
;
; International long distance through trunk
;
exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91800NXX,2,Congestion
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,2,Congestion
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,2,Congestion
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat = 9
include = local
include = trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat = 9
;include = default
;include = parkedcalls
include = trunklocal
include = iaxtel700
include = trunktollfree
include = iaxprovider
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2}); Ring the interface, 20
seconds maximum
exten = s,2,Voicemail(u${ARG1}); If unavailable, send
to voicemail w/ unavail announce
exten = s,3,Goto(default,s,1); If they press #,
return to start
exten = s,102,Voicemail(b${ARG1}); If busy, send to
voicemail w/ busy announce
exten = s,103,Goto(default,s,1); If they press #,
return to start
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include = local
exten = 431,1,Dial,SIP/sipphone
Regovich, Timothy wrote:
Jason,
Include your sip and extensions files so people can take a look.
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Monday, February 23, 2004 10:25 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] An example config for using a Wildcard X100P and a
SIP phone?
Hello.
I've just recently purchased the Asterisk Developers Kit so we can
figure out how to get away from our Nortel system and go to IP based
phones. I have a RH 9 box loaded with Asterisk (a very recent cvs download).
Either way, I can call the asterisk box and get their demo playing fine.
I can even call the SIP phone I've hooked up when I call in from my cell
phone to the asterisk box, and that works.
I cannot call out with my SIP phone though. It'll dial, ring my cell
phone twice and then give up and complain that its busy. Even if I try
to answer the cell phone during the first ring.
Does anyone have a config they could share with me on how to make this
setup work? This sounds like it should be fairly trivial, but I've
beaten my head against the wall on this for a few days. =)
Thanks alot,
Jason
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[Asterisk-Users] Dual Xeon
When compiling Asterisk for a dual XEON based system are there any caveats or "switches" that we need to be aware of?
RE: [Asterisk-Users] asterisk-oh323, new version 0.5.8
Hello, I am interested in running small busines in telecommunication with minimum expenses and investment. Can Windows operating be used for this purpose. Thank you all. Regards, Yaseen From: Michael Manousos [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk-oh323, new version 0.5.8 Date: Mon, 09 Feb 2004 14:07:43 +0200 Hello all, A new version of asterisk-oh323 is now available. It contains numerous minor fixes and updates. Among them, a fix for channels using the G.729 codec (tested with codec_g729b.so codec). Download from: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ 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 _ Say good-bye to spam, viruses and pop-ups with MSN Premium -- free trial offer! http://click.atdmt.com/AVE/go/onm00200359ave/direct/01/ ___ 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] Processor load spikes
On Mon, 2004-02-23 at 18:42, mattf wrote: Thanks for the response. I plan on trying Slackware on my backup/test asterisk server when I have a new backup server ready in a few weeks. I've noticed in some database machine testing that Slackware starts up in about half the time of RedHat and doesn't have all of that Redhat junk either. I'll post my results running Slackware after I've had time to test it. When I said crashed I meant that the whole operating system crashed, so no backtrace possible. Thanks, MATT--- Hi Matt, My RH9 box has never crashed although on some others running RH9 I've seen load spikes also. The only similar situation I vaguely remember from long ago was either related to using a T400P/E400P card on a motherboard with the incorrect pci slot voltage or to a power supply that couldn't cope with the extra load. Don't recall exactly anymore so could be wrong but maybe worth keeping in mind. I always do the following on a RH9 box: * export LD_ASSUME_KERNEL=2.4.1 before you start asterisk. Alternatively you can build a plain vanilla 2.4.2x kernel from kernel.org and use that one * turn off all unnecessary cron jobs. updatedb can have quite a field day with eating up I/O and keeping disks pretty busy and iirc you may want to turn off the fam service also * turn off all unnecessary services and remove all unnecessary modules from /etc/modules.conf If you find the cause, please let us know. Good luck, Patrick ___ 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] SPA 2000 ringing
When placing a call from Sipura SPA 2000 to other extensions, for some reason dialled extension keeps ringing even though SPA 2000 hangs up the call. Asterisk does not end that call until it is not answered by dialled extension. Anyone has experienced similar problem? ___ 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] Queue Modified ACD for Asterisk 0.7.2
Dear All i have modified the app_queue.c with the patch app_queue_patch_1_07 from bug track to use with Asterisk 0.7.2 i have try it and seems to work :-) I hope to help someone Bye Dimi app_queue.c_queue_patch072.tar.gz Description: application/tgz
Re: [Asterisk-Users] asterisk-oh323, new version 0.5.8
On Monday 23 February 2004 12:56 pm, Khalid Yaseen wrote: Hello, I am interested in running small busines in telecommunication with minimum expenses and investment. Can Windows operating be used for this purpose. Thank you all. Regards, Yaseen Haha, that's funny! Unless of course you are serious. Realize that Unix was developed by ATT and has the type of stability needed to run 24/7 mission critical operations. Windows has not been able to become a viable option in that industry. Those who have tried have gone back to Unix (or Unix like operating systems like Linux and BSD). Not that you could not have a windows box that is pretty stable, but there's a big investment in time to get close to what Unix does out of the box. And yet, it's not going to be as dependable and maintenance free... Another note, you replied with a new question into an existing thread. Using windiws you may not see it as it is usually not fully complient. For us however, we have thread in progress and then your email comes in with a new subject. The best way if you have a new subject is to not reply to the list but use new message, this way we don't get all messed up by it. : ) From: Michael Manousos [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk-oh323, new version 0.5.8 Date: Mon, 09 Feb 2004 14:07:43 +0200 Hello all, A new version of asterisk-oh323 is now available. It contains numerous minor fixes and updates. Among them, a fix for channels using the G.729 codec (tested with codec_g729b.so codec). Download from: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ 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 _ Say good-bye to spam, viruses and pop-ups with MSN Premium -- free trial offer! http://click.atdmt.com/AVE/go/onm00200359ave/direct/01/ ___ 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 -- They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] An example config for using a Wildcard X100P and a SIP phone?
Try moving your sip phone into its own context, instead of default (I use
sip) and create a [sip] section in your extensions.conf
Add a sepcific extension to test your outgoing, like :
exten = _5,1,Dial,Zap/1/800551212
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Monday, February 23, 2004 1:02 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] An example config for using a Wildcard X100P
and a SIP phone?
Timothy,
I have minimally modified the demo files that came with Asterisk, so
what is posted below is most of the comments and the demo section
removed from the config files.
Thanks!
; SIP Configuration for Asterisk
;
[general]
port = 5060; Port to bind to
bindaddr = 0.0.0.0; Address to bind to
context = default; Default for incoming calls
[sipphone]
type=friend
username=sipphone
fromuser=Sipster; Specify user to put in from instead
of callerid
secret=password
host=dynamic
defaultip=192.168.1.201
amaflags=default; Choices are default, omit, billing,
documentation
accountcode=Sipster ; Users may be associated with an
accountcode tp ease billing
mailbox=431
--
extensions.conf
--
[general]
static=yes
writeprotect=no
[globals]
;CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/1; Trunk interface
TRUNKMSD=1; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
[iaxtel700]
exten = _91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[trunkint]
;
; International long distance through trunk
;
exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91800NXX,2,Congestion
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,2,Congestion
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,2,Congestion
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat = 9
include = local
include = trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat = 9
;include = default
;include = parkedcalls
include = trunklocal
include = iaxtel700
include = trunktollfree
include = iaxprovider
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2}); Ring the interface, 20
seconds maximum
exten = s,2,Voicemail(u${ARG1}); If unavailable, send
to voicemail w/ unavail announce
exten = s,3,Goto(default,s,1); If they press #,
return to start
exten = s,102,Voicemail(b${ARG1}); If busy, send to
voicemail w/ busy announce
exten = s,103,Goto(default,s,1); If they press #,
return to start
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include = local
exten = 431,1,Dial,SIP/sipphone
Regovich, Timothy wrote:
Jason,
Include your sip and extensions files so people can take a look.
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Monday, February 23, 2004 10:25 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] An example config for using a Wildcard X100P and
a
SIP phone?
Hello.
I've just recently purchased the Asterisk Developers Kit so we can
figure out how to get away from our Nortel system and go to IP based
phones. I have a RH 9 box loaded with Asterisk (a very recent cvs
download).
Either way, I can call the asterisk box and get their demo playing fine.
I can even call the SIP phone I've hooked up when I call in from my cell
phone to the asterisk box, and that works.
I cannot call out with my SIP phone though. It'll dial, ring my cell
phone twice and then give up and complain that its busy. Even if I try
to answer the cell phone during the first ring.
Does anyone have a config they could share with me on how to make this
setup
[Asterisk-Users] Confusion with IAX PBX-PBX
I have been trying to set up three * servers to use IAX between them and am a bit lost as to the finer detail of the config files. I have read the wiki and it has not made things better. Here is my problem; I create a section like this on each machines: [othermachine-1] type=friend host=dynamic secret=password trunk=yes qualify=yes context=incoming-1 [othermachine-2] type=friend host=dynamic secret=password trunk=yes qualify=yes context=incoming-2 Now in my extensions.conf I use the link like this: IAX2/othermachine-1 But my problem comes in with the receiving machine, how does it know which machine the link came from without a username of some kind. Or have I completely missed the point of IAX? Please help I am completely lost. Thanks Chris. ___ 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] Pickup
Title: Pickup The extension for Pickup seems to be *8#, but I cannot find it anywhere in any configuration file. Is this a hard wired extension? Are there other hard wired extensions? If so, is there a list? What priority do they have? Is there any way to change them or map additional extensions to these functions? Thanks. Jim
[Asterisk-Users] Codec Order / Preference
Hi, I wish my IAX connection negotiates codecs in the following order: 1) speex 2) gsm 3) alaw Is it possible? I tried and I detected * selects gsm prior to speex no matter the order I write my iax.conf allow command. Daniel ___ 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 overlap (early dial) 484 response
It works for me for internal calls, and for setting up calls over a PRI. However, there are problems with overlap dialling when using an X100P analogue card as * does not seem to buffer digits correctly. I would recommend not using overlap sending with SIP phones til those issues ar fixed. Rgds Tim Key Aavoja wrote: Hello, I have one question again. I checked archive and I found that somebody before me asked this question already. But no responses for this posting. http://lists.digium.com/pipermail/asterisk-users/2003-September/020065.html So, is it supported or no? If yes, what I need to configure? Thank you. Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ 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] Codec Order / Preference
You cannot specify the order of codec selection with Asterisk On Mon, 2004-02-23 at 13:03, Daniel Bichara wrote: Hi, I wish my IAX connection negotiates codecs in the following order: 1) speex 2) gsm 3) alaw Is it possible? I tried and I detected * selects gsm prior to speex no matter the order I write my iax.conf allow command. Daniel ___ 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 -- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting ___ 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 Groups and outgoing line selection
I have 2 lines setup. One is the house line, the other a business line. What I'd LIKE to do, is if a house extension dials out, it selects the house line to dial out on, but if the house line is busy use the business line. Ditto with the office extension, but reverse. Using distinctive ring on both lines, I can do a CO forward on busy to the alternate number for each line, and maintain correct context depending on which number people called. My guess on how to implement this is to duplicate dial plans for outside numbers, and use macros to dial out on each prime channel, and use the alternate if congestion. This make sense? Anyone do something like this already and have an example? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec Order / Preference
Really? Did you try disallow=all Allow=speex Allow=gsm Allow=alaw ? T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, February 23, 2004 2:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Codec Order / Preference You cannot specify the order of codec selection with Asterisk On Mon, 2004-02-23 at 13:03, Daniel Bichara wrote: Hi, I wish my IAX connection negotiates codecs in the following order: 1) speex 2) gsm 3) alaw Is it possible? I tried and I detected * selects gsm prior to speex no matter the order I write my iax.conf allow command. Daniel ___ 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 -- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] DevLite problem with ztcfg
Hello all, I finally got around to installing my Dev Kit Lite. I did the install yesterday from the latest CVS. I am receiving an error that does not let * start up. When I go through the procedure to load the modules, I get the following error after running ztcfg. Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? I looked in the archives but could not find a valid solution. The config file is: fxoks=1 fxsks=2 loadzone=us defaultzone=us When I did this with only the 100 card, it did work, but that was on a CVS release from last week. I ran the astinstall script and un-tarred the configs into / Any help would be appreciated. Thanks, Tom Schaefer ___ 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] Dual Xeon
Ed Devine wrote: When compiling Asterisk for a dual XEON based system are there any caveats or "switches" that we need to be aware of? Well, for zaptel hardware you need to uncomment the SMP entry in the zaptel Makefile. Also I would turn off Hyperthreading (in the bios). It may cause problems. Kind regards, Geert
[Asterisk-Users] ACD
I've looked through a lot of different pieces of documentation regarding *'s ACD functionality. Is there any one place in particular with a good amount of documentation on it? 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] Queue Modified ACD for Asterisk 0.7.2
On Mon, 23 Feb 2004, reseaux wrote: Dear All i have modified the app_queue.c with the patch app_queue_patch_1_07 from bug track to use with Asterisk 0.7.2 i have try it and seems to work :-) I hope to help someone Bye Dimi What Bug number is this? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Unable to create channem of type 'Zap'
Can anyone
help me, (after a two day search, also on the mailing list)
I have the
following situation:
Asterisk
works fine, until I added a FXO card. (Digium)
When I
tried to call to the pstn I have the following error
Executing
Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
NOTIVE[16401]:
FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE CHANNELOF TYPE
'ZAP'
==
Everyone is busy at this time
When I
start Asterisk I have no error
Only the
following isn't right:
ZAP
SHOW CHANNELS = No channels
modprobe
wcfxo = ok (no errors)
I have
following config.
ZAPATA
[channels]language=encontext=incomingsignalling=fxs_ksusecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1pickupgroup=1immediate=yesmusiconhold=default channel = 1
ZAPTEL
loadzone =
usdefaultzone = usfxsks =
1
EXTENSION
[incoming]exten =
s,1,Dial(SIP/Phone1SIP/Phone3,20,tr)
[outgoing]exten = _0X.,1,Dial,Zap/1/${EXTEN:1}
IN
[SIP]
include
= outgoing
I'm don't
know what I can change to the config.
Anyone an
idea
Thanks,
Wim
Re: [Asterisk-Users] Codec Order / Preference
Regovich, Timothy wrote: Really? Did you try disallow=all Allow=speex Allow=gsm Allow=alaw Yes and it did no work. ? T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, February 23, 2004 2:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Codec Order / Preference You cannot specify the order of codec selection with Asterisk On Mon, 2004-02-23 at 13:03, Daniel Bichara wrote: Hi, I wish my IAX connection negotiates codecs in the following order: 1) speex 2) gsm 3) alaw Is it possible? I tried and I detected * selects gsm prior to speex no matter the order I write my iax.conf allow command. Daniel ___ 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] Unable to create channem of type 'Zap'
On Mon, 2004-02-23 at 17:10, Wim Venneman wrote: Can anyone help me, (after a two day search, also on the mailing list) I have the following situation: Asterisk works fine, until I added a FXO card. (Digium) When I tried to call to the pstn I have the following error Executing Dial(SIP/Phone2-fc49, Zap/1/2355) in new stack [channels] language=en context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 pickupgroup=1 immediate=yes musiconhold=default channel = 1 ^^^ is this a typo? If not, the channel = 1 should go on a line of its own. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ 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] Unable to create channem of type 'Zap'
Make sure you run a ztcfg after you do a modprobe.
ztcfg will configure (or bring up) the zap channels on zaptel interface
cards. Do this before starting * and after the modprobe.
(You may also do a ztcfg -v to see whats configured)
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
Sent: Monday, February 23, 2004 3:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
Can anyone help me, (after a two day search, also on the mailing list)
I have the following situation:
Asterisk works fine, until I added a FXO card. (Digium)
When I tried to call to the pstn I have the following error
Executing Dial(SIP/Phone2-fc49, Zap/1/2355) in new stack
NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
CHANNEL OF TYPE 'ZAP'
== Everyone is busy at this time
When I start Asterisk I have no error
Only the following isn't right:
ZAP SHOW CHANNELS = No channels
modprobe wcfxo = ok (no errors)
I have following config.
ZAPATA
[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
pickupgroup=1
immediate=yes
musiconhold=default channel = 1
ZAPTEL
loadzone = us
defaultzone = us
fxsks = 1
EXTENSION
[incoming]
exten = s,1,Dial(SIP/Phone1SIP/Phone3,20,tr)
[outgoing]
exten = _0X.,1,Dial,Zap/1/${EXTEN:1}
IN [SIP]
include = outgoing
I'm don't know what I can change to the config.
Anyone an idea
Thanks,
Wim
___
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RE: [Asterisk-Users] Codec Order / Preference
That still does not tell Asterisk the ORDER of the codec selection. On Mon, 2004-02-23 at 13:28, Regovich, Timothy wrote: Really? Did you try disallow=all Allow=speex Allow=gsm Allow=alaw ? T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Monday, February 23, 2004 2:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Codec Order / Preference You cannot specify the order of codec selection with Asterisk On Mon, 2004-02-23 at 13:03, Daniel Bichara wrote: Hi, I wish my IAX connection negotiates codecs in the following order: 1) speex 2) gsm 3) alaw Is it possible? I tried and I detected * selects gsm prior to speex no matter the order I write my iax.conf allow command. Daniel ___ 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 -- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting ___ 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 extension busy when not available ??
I use ChanIsAvail() to check to see if the phone is connected at the top of the dialplan for that extension. This works for IAX2 and SIP channels but not for MGCP. If you are interested in the actual code I can send it to you from home tonight. Robert Thank you, yes please... Well, I'm about three weeks into my very first * installation (that sort of works), so basically any info/tips/tricks/word of advice is accepted with appreciation... -- Soren ___ 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] VM: Multilanguage and digits
Hi! I'm trying to record som voiveprompts, and I've created a directory se in /var/lib/asterisk/sounds - in that directory I've put files like vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I hear my own voice prompts! But wehre should I place the digits I've recorded? - I have tried to put them in se/digits but that don't works? Anyone that have an idea about where to put the digits for my own languages - I have read the mult-language section at voip-info.org but that don't really says where to put the sound files. Thanks for any advice! /Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.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] SIP over NAT
Assuming that getting H323 to work over NAT is almost really hard What is about having both SIP clients venid different NATs ¿ is it posible or as hard as H.323? Thanks! Marc. ___ 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] Dual Xeon
Dear Geert I use * with 1 TE400P on Dual Xeon with 1GByte of RAM HT everyday with little 30 channels load of calls at time, can you give me more info about problem in this kind of configuration? thanks Dimitri On Monday 23 February 2004 19:31, Geert Nijpels wrote: Ed Devine wrote: When compiling Asterisk for a dual XEON based system are there any caveats or switches that we need to be aware of? Well, for zaptel hardware you need to uncomment the SMP entry in the zaptel Makefile. Also I would turn off Hyperthreading (in the bios). It may cause problems. Kind regards, Geert ___ 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] VM: Multilanguage and digits
Lars Fredriksson wrote: Hi! I'm trying to record som voiveprompts, and I've created a directory se in /var/lib/asterisk/sounds - in that directory I've put files like vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I hear my own voice prompts! But wehre should I place the digits I've recorded? - I have tried to put them in se/digits but that don't works? Anyone that have an idea about where to put the digits for my own languages - I have read the mult-language section at voip-info.org but that don't really says where to put the sound files. Hej Lars! From the Wiki: (The sound files page) Location of the sound files Asterisk normally looks for a sound file with an extension used for the codec used. If a language is set for the channel with the SetLanguage() application, Asterisk first looks for countrycode/filename where countrycode is the language code (example:. 'fr' for french). Languages and special tones for that country or region are defined in indications.conf. --- Well, this doesn't apply for digits because the source file is patched for english for some reason. The other day, this was removed for norway. Could be done for se as well, don't you agree? From say.c: /* Use english numbers if a given language is supported. */ /* As a special case, Norwegian has the same numerical grammar as English */ if (strcasecmp(language, no)) language = en; Change no to se (who cares about norwegian :-) ) and you'll be ok. And remember to report this to bugs.digium.com - tack! /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] VM: Multilanguage and digits
On Monday, Fedbruary 23rd Olle wrote: Change no to se (who cares about norwegian :-) ) and you'll be ok. And remember to report this to bugs.digium.com - tack! Hey, now!! Thorsten ___ 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] ztmonitor and the x101p
I have a x101p and I can't seem to get ztmonitor to work on it. I've tried it on 2 different machines. One with a SBLive! card and the other with a AMD-768 [Opus] Audio (rev 03) chip. Neither machine give me a graph in ztmonitor 1 -v mode. If I run ztmonitor without the -v I get: Can't turn stereo off :( Sound card won't let me know the input buffering... Is this the problem? ...Jeff ___ 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] Dual Xeon
reseaux wrote: Dear Geert I use * with 1 TE400P on Dual Xeon with 1GByte of RAM HT everyday with little 30 channels load of calls at time, can you give me more info about problem in this kind of configuration? thanks Dimitri I never did experience problems that could be directly linked to HT. However, I was told at #asterisk HT would not give much of a performance gain and can cause problems with sound quality. Also I had a problem with calls having 3 out of 5 calls no sound while the RTP stream did build up correctly, this problem went away after disabling HT with my Xeon proc, but unfortunately I also changed other things in the hardware configuration so I can not point it to the HT stuff. I'm sure that it wasn't a configuration error and also that the memory is working correctly (memtest). Kind regards, Geert ___ 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] Unable to create channem of type 'Zap'
Made changes:
1)
musiconhold= default
channel = 1
2)
reboot
modprobe wcfxo = ok
ztcfg -v
result = 1 channel configured
Try to dial, still the same problem. (error)
Wim
- Original Message -
From: Brent Franks [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 9:19 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
Make sure you run a ztcfg after you do a modprobe.
ztcfg will configure (or bring up) the zap channels on zaptel interface
cards. Do this before starting * and after the modprobe.
(You may also do a ztcfg -v to see whats configured)
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
Sent: Monday, February 23, 2004 3:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
Can anyone help me, (after a two day search, also on the mailing list)
I have the following situation:
Asterisk works fine, until I added a FXO card. (Digium)
When I tried to call to the pstn I have the following error
Executing Dial(SIP/Phone2-fc49, Zap/1/2355) in new stack
NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
CHANNEL OF TYPE 'ZAP'
== Everyone is busy at this time
When I start Asterisk I have no error
Only the following isn't right:
ZAP SHOW CHANNELS = No channels
modprobe wcfxo = ok (no errors)
I have following config.
ZAPATA
[channels]
language=en
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
pickupgroup=1
immediate=yes
musiconhold=default channel = 1
ZAPTEL
loadzone = us
defaultzone = us
fxsks = 1
EXTENSION
[incoming]
exten = s,1,Dial(SIP/Phone1SIP/Phone3,20,tr)
[outgoing]
exten = _0X.,1,Dial,Zap/1/${EXTEN:1}
IN [SIP]
include = outgoing
I'm don't know what I can change to the config.
Anyone an idea
Thanks,
Wim
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[Asterisk-Users] An example config for using a Wildcard X100P and a SIP phone?
|I cannot call out with my SIP phone though. It'll dial, ring my cell |phone twice and then give up and complain that its busy. Even if I try |to answer the cell phone during the first ring. | |Does anyone have a config they could share with me on how to make this |setup work? This sounds like it should be fairly trivial, but I've |beaten my head against the wall on this for a few days. =) | |Thanks alot, |Jason Again most possibily it is codec issue, what sip phone you use and show us your sip.conf. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] calling between two zap points with zaphfc
hi everybody, just went into some trouble (again!!) while I was trying to make a call between two (isdn)phones connected to my hfc-s card. I am running junghanns.net's hfc-bri-driver. the call is terminated after a few seconds. anyone else got this to work? btw: I am using a NTBA as powersource for the two phone. the first phone is an old teles phone. the other one i a siemens cordless phone (with own powersupply). I have not modified my NTBA to have 50 ohm!! making calls to the outside world from these two phones (even two at the same time) via my avm-fritz and chan_capi works perfektly. ony thoughts? bye
Re: [Asterisk-Users] ztmonitor and the x101p
Ah! I just checked out the latest ztmonitor out of cvs and it works just fine. ...Jeff On Mon, 2004-02-23 at 12:51, Jeff Gustafson wrote: I have a x101p and I can't seem to get ztmonitor to work on it. I've tried it on 2 different machines. One with a SBLive! card and the other with a AMD-768 [Opus] Audio (rev 03) chip. Neither machine give me a graph in ztmonitor 1 -v mode. If I run ztmonitor without the -v I get: Can't turn stereo off :( Sound card won't let me know the input buffering... Is this the problem? ...Jeff ___ 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] VM: Multilanguage and digits
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Monday, February 23, 2004 9:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VM: Multilanguage and digits Lars Fredriksson wrote: Hi! I'm trying to record som voiveprompts, and I've created a directory se in /var/lib/asterisk/sounds - in that directory I've put files like vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I hear my own voice prompts! But wehre should I place the digits I've recorded? - I have tried to put them in se/digits but that don't works? Anyone that have an idea about where to put the digits for my own languages - I have read the mult-language section at voip-info.org but that don't really says where to put the sound files. Hej Lars! From the Wiki: (The sound files page) Location of the sound files Asterisk normally looks for a sound file with an extension used for the codec used. If a language is set for the channel with the SetLanguage() application, Asterisk first looks for countrycode/filename where countrycode is the language code (example:. 'fr' for french). Languages and special tones for that country or region are defined in indications.conf. --- Well, this doesn't apply for digits because the source file is patched for english for some reason. The other day, this was removed for norway. Could be done for se as well, don't you agree? From say.c: /* Use english numbers if a given language is supported. */ /* As a special case, Norwegian has the same numerical grammar as English */ if (strcasecmp(language, no)) language = en; Change no to se (who cares about norwegian :-) ) and you'll be ok. And remember to report this to bugs.digium.com - tack! Hi Olle / Hej Olle! Thanks for your answer, but I don't know if I'm doing something wrong because it doesn't make any difference if I change no to se - I'm not a programmer, but I can't see how it should make any difference? Well, I solved it for the moment by replacing the digits with my swedish digits - and I will report it to bugs.digium.com - I think that might be the right way! Thanks! Best regards / Ha det gott Lars __ 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] Unable to create channem of type 'Zap'
Wim, I made some changes to your Zapata.conf and zaptel.conf config
files below.
Hope this helps.
Also, do a less /proc/interrupts and see if the card is on it's own IRQ.
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
Sent: Monday, February 23, 2004 3:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
Can anyone help me, (after a two day search, also on the mailing list)
I have the following situation:
Asterisk works fine, until I added a FXO card. (Digium)
When I tried to call to the pstn I have the following error
Executing Dial(SIP/Phone2-fc49, Zap/1/2355) in new stack
NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
CHANNEL OF TYPE 'ZAP'
== Everyone is busy at this time
When I start Asterisk I have no error
Only the following isn't right:
ZAP SHOW CHANNELS = No channels
modprobe wcfxo = ok (no errors)
I have following config.
ZAPATA
[channels]
language=en
group=1
pickupgroup=1
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=yes
musiconhold=default
channel = 1
ZAPTEL
loadzone = us
defaultzone = us
fxsks = 1
EXTENSION
[incoming]
exten = s,1,Dial(SIP/Phone1SIP/Phone3,20,tr)
[outgoing]
exten = _0X.,1,Dial,Zap/1/${EXTEN:1}
IN [SIP]
include = outgoing
I'm don't know what I can change to the config.
Anyone an idea
Thanks,
Wim
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Re: [Asterisk-Users] Minimum voice mail message limit?
From Posts on this list on Sat. w/ the subject Voicemail brought to light that there is a patch for some more advanced VM features after a message is left. http://bugs.digium.com/bug_view_page.php?bug_id=156 On Mon, 2004-02-23 at 12:56, Walt Reed wrote: Looking through the Wiki and mailing list, I didn't see an answer to this. Is there a way to set the minimum voice mail message size? Hangups seem to generate 4 to 5 second messages. If I set a min to 6 or 7 that should eliminate most of these. The main voicemail app also seems kind of thin. There are no caller options such as playing back a message you left, deleting it and starting over if you mess up, etc. Voicemailmain also is rather thin - you can't listed to your currently available greetings for example. Is there an alternative voicemail at this time? Patches? FYI, I'm running * from CVS as of Feb 19. ___ 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] 12SP
I am trying to get a Cisco 12SP phone to work with *. I do not have call manager. When start * and turn skinny debugging on I get this on the console: -- -- Starting Skinny session from 192.168.1.202 Recieved AlarmMessage Device SEP0010EB003E03 is attempting to register -- Device 'ipme' successfuly registered Requesting capabilities Version Request Received CapabilitiesRes Feb 23 16:29:29 WARNING[794722]: chan_skinny.c:2275 get_input: Skinny Client sent less data than expected. Feb 23 16:29:29 NOTICE[794722]: chan_skinny.c:2333 skinny_session: Skinny Session returned: Success --- The phone indicates that it is programming. The IP address of the phone is correct in the logs. Here is a snippet from my skinny.conf file: --- [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 192.168.1.11 ; Address to bind to dateFormat = M-D-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 ; allow = all ; disallow = ; Typical config for 12SP+ [ipme] device=SEP0010EB003E03 version=P002G204; Thanks critch context=outbound-analog line = 120 ; Dial(Skinny/[EMAIL PROTECTED]) --- Any ideas? -- Cullen Simpson [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
[Asterisk-Users] SIP Codec selection order
I have the following situation: my cisco 7940 attached to asterisk i use asterisk as voice mail, conference room, music on hold and so on. also i have cisco 5350 and use it as PSTN gateway. - As i know asterisk able to forward g729 frame. I enable in asterisk sip,conf options allow=g729 . and on cisco 5350i specified use only G729. (because i have restricted bandwidth to it). also i know that asterisk voice mail working fine with gsm, alaw, ulaw codecs. Now i able to call from 7940 to 5350 . Call established with codec g729. But when i call to voice mail my call failed with the reason specified below: Feb 22 02:21:28 NOTICE[-1336095824]: channel.c:1453 ast_set_write_format: Unable to find a path from GSM to G729A Feb 22 02:21:28 WARNING[-1336095824]: file.c:734 ast_streamfile: Unable to open vm-login (format G729A): No such file or directory Feb 22 02:21:28 WARNING[-1336095824]: app_voicemail.c:2714 vm_execmain: Couldn't stream login file any idea how i can make voicemail working with 7940 and g729 forwarding enabled. Alexey _ Store more e-mails with MSN Hotmail Extra Storage 4 plans to choose from! http://click.atdmt.com/AVE/go/onm00200362ave/direct/01/ ___ 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] Unable to create channem of type 'Zap'
Thanks for the help !
Made changes, still the same message.
I have two NIC's with IRQ 11
The FXO card has IRQ10 (and no other card has IRQ10)
Wim
- Original Message -
From: Brent Franks [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 10:21 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
Wim, I made some changes to your Zapata.conf and zaptel.conf config
files below.
Hope this helps.
Also, do a less /proc/interrupts and see if the card is on it's own IRQ.
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
Sent: Monday, February 23, 2004 3:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
Can anyone help me, (after a two day search, also on the mailing list)
I have the following situation:
Asterisk works fine, until I added a FXO card. (Digium)
When I tried to call to the pstn I have the following error
Executing Dial(SIP/Phone2-fc49, Zap/1/2355) in new stack
NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
CHANNEL OF TYPE 'ZAP'
== Everyone is busy at this time
When I start Asterisk I have no error
Only the following isn't right:
ZAP SHOW CHANNELS = No channels
modprobe wcfxo = ok (no errors)
I have following config.
ZAPATA
[channels]
language=en
group=1
pickupgroup=1
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=yes
musiconhold=default
channel = 1
ZAPTEL
loadzone = us
defaultzone = us
fxsks = 1
EXTENSION
[incoming]
exten = s,1,Dial(SIP/Phone1SIP/Phone3,20,tr)
[outgoing]
exten = _0X.,1,Dial,Zap/1/${EXTEN:1}
IN [SIP]
include = outgoing
I'm don't know what I can change to the config.
Anyone an idea
Thanks,
Wim
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Re: [Asterisk-Users] SIP extension busy when not available ??
Soren Rathje said:
I use ChanIsAvail() to check to see if the phone is connected at the
top
of the dialplan for that extension. This works for IAX2 and SIP channels
but not for MGCP.
If you are interested in the actual code I can send it to you from home
tonight.
Robert
Thank you, yes please...
Well, I'm about three weeks into my very first * installation (that sort
of
works), so basically any info/tips/tricks/word of advice is accepted
with
appreciation...
-- Soren
I use a macro to define the extensions. In this way I only have to enter 1
line per actual extension.
The Macro is:
[macro-stdexten]
exten = s,1,ChanisAvail(${ARG2})
exten = s,2,Dial(${ARG2},20,Ttr)
exten = s,102,GoTo(voicemail,s,1)--Note A
exten = s,103,Hangup
exten = s,104,GoTo(voicemail,s,1)--Note B
exten = s,105,Hangup
The extensions are defined as:
exten = 10,1,Macro(stdexten,10,MGCP/aaln/[EMAIL PROTECTED])
exten = 11,1,Macro(stdexten,11,SIP/11)
exten = 12,1,Macro(stdexten,12,IAX2/12)
The 2nd argument in the () is the voicemailbox number, 3rd argument is the
Channel to dial.
Note A: If the Channel is not available then control comes here. You can
put a Voicemail2 statement here with the u option or whatever you want to
use.
Note B: This is where the Busy/Timeout comes from the Dial command.
In my case I have a voicemail context that handles the 2 mailboxes we use
here in the house. That is: an announcement is played and the caller
selects the mailbox to get the message.
Its not perfect and for sure can be improved.
Robert
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[Asterisk-Users] codec translation
The route of my call is: gs101--asterisk--iaxtel--asterisk--gs101 I have 2 g729 from Digium and calls to iaxtel can only be in gsm format. The GS101 phones are set to use g729, then 711ulaw. However when the called GS phone is picked up the connection is terminated. These are the console messages: -- SIP/1003-f8e1 is ringing -- SIP/1003-f8e1 answered [EMAIL PROTECTED]:4569]/6 Error Opening channel:2 not available, see va_g729_init_global(..)Feb 24 08:47:55 WARNING[1242768320]: codec_g729b.c:179 lintog729_new: No available g729 resources for channel 2 Feb 24 08:47:55 WARNING[1242768320]: translate.c:111 ast_translator_build_path: Failed to build translator step from 6 to 8 Feb 24 08:47:55 WARNING[1242768320]: chan_sip.c:1322 sip_write: Asked to transmit frame type 2, while native formats is 256 (read/write = 2/256) == Spawn extension (macro-stdexten, s, 4) exited non-zero on '[EMAIL PROTECTED]:4569]/6' in macro 'stdexten' == Spawn extension (incoming, 1003, 1) exited non-zero on '[EMAIL PROTECTED]:4569]/6' -- Executing Hangup([EMAIL PROTECTED]:4569]/6, ) in new stack == Spawn extension (incoming, h, 1) exited non-zero on '[EMAIL PROTECTED]:4569]/6' -- Hungup '[EMAIL PROTECTED]:4569]/6' -- Hungup 'IAX2[69.73.19.178:4569]/5' == Spawn extension (local, 17001813482, 1) exited non-zero on 'SIP/1002-4360' -- Executing Hangup(SIP/1002-4360, ) in new stack My questions are: Although both gs101 are set to use g729 is the actual communication from gs to asterisk using g729 and asterisk to iaxtel using gsm and asterisk to the called gs using g729. Do anyone make sense out of the console messages since I have 2 g729 licence. It should be able to handle 2 g729 channel one receive and one send. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
RE: [Asterisk-Users] OT: SNOM and TAPI
I remember we had something one or two years ago, but I remember that was not what I was dreaming of. Sorry we are not so good in implementing Windows-stuff... Maybe has someone out there a template for TAPI? Something for someone who never did something with COM or DCOM or .net or whatever... BTW click-to-dial can be initiated with a REFER request. That's 100 % SIP. CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Monday, February 23, 2004 4:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: SNOM and TAPI Snom TAPI integration is a joke... Andy ___ 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] A missing argument
Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2 with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3 the most 10.0 fails at this point chan_zap.c: In function `handle_init_r2_event': chan_zap.c:4773: error: too few arguments to function `zt_new' make[1]: *** [chan_zap.o] Error 1 line 4773 has chan = zt_new(i, AST_STATE_RING, 0, SUB_REAL, 0); but greping shows that the declaration and other instances have 6 arguments. ML 9.2 is using gcc-3.3.1 whilst 10.0 is using gcc-3.3.2 What worries me is how many other programs in the world have the same type of error and the compiler has missed it? -- 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] OT: SNOM and TAPI
On Mon, 23 Feb 2004, Christian Stredicke wrote: I remember we had something one or two years ago, but I remember that was not what I was dreaming of. Sorry we are not so good in implementing Windows-stuff... Maybe has someone out there a template for TAPI? Something for someone who never did something with COM or DCOM or .net or whatever... BTW click-to-dial can be initiated with a REFER request. That's 100 % SIP. Someone also mentioned that you could submit Dial requests to the SNOM via it's Web interface. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] SPA 2000 ringing
Senad Jordanovic wrote: When placing a call from Sipura SPA 2000 to other extensions, for some reason dialled extension keeps ringing even though SPA 2000 hangs up the call. Asterisk does not end that call until it is not answered by dialled extension. Anyone has experienced similar problem? Yes, I have a SPA2000 as well, and noticed this on CVS from 2-3 months ago. I have pulled the newest CVS a week or so ago, but not tested this scenario since then. I will pull a new CVS tonight and test again. I've been meaning trace and see if I can watch the activity for what's actually happening. - Andrew Thompson http://aktzero.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] A missing argument
On Mon, 23 Feb 2004, Dave Cotton wrote: Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2 with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3 the most 10.0 fails at this point chan_zap.c: In function `handle_init_r2_event': chan_zap.c:4773: error: too few arguments to function `zt_new' make[1]: *** [chan_zap.o] Error 1 line 4773 has chan = zt_new(i, AST_STATE_RING, 0, SUB_REAL, 0); but greping shows that the declaration and other instances have 6 arguments. ML 9.2 is using gcc-3.3.1 whilst 10.0 is using gcc-3.3.2 What worries me is how many other programs in the world have the same type of error and the compiler has missed it? Dave, You if have libr2 installed. I don't believe much work has been done on R2 in quite some time, so it might not be up to date. The channels/Makefile looks for the existance of /usr/lib/libmfcr2.so.1 to set ZAPATA_R2 which is causing those sections of code to be compiled in James ___ 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] Pickup
Jim Sneeringer wrote: The extension for Pickup seems to be *8#, but I cannot find it anywhere in any configuration file. Is this a hard wired extension? Are there other hard wired extensions? If so, is there a list? What priority do they have? Is there any way to change them or map additional extensions to these functions? Thanks. Jim See: http://www.voip-info.org/tiki-index.php?page=Asterisk+channels+zap Yes, they should be .conf changeable(IMHO), but they're not. - Andrew Thompson http://aktzero.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] Asterisk and Multicast
Hi, Could anyone tell me if asterisk supports multicast? And if so, what type? And if not, are there any plans to implement one in the forseeable future? Thanks, Jason ___ 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] OT: SNOM and TAPI
I tend to agree with Christian, imho if there is something a joke then it is TAPI. There are lot of service creation techniques, be distributed REFER-based or centralized B2BUA-based which take no additional .*APIs. -jiri On Mon, 23 Feb 2004, Christian Stredicke wrote: I remember we had something one or two years ago, but I remember that was not what I was dreaming of. Sorry we are not so good in implementing Windows-stuff... Maybe has someone out there a template for TAPI? Something for someone who never did something with COM or DCOM or .net or whatever... BTW click-to-dial can be initiated with a REFER request. That's 100 % SIP. CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Monday, February 23, 2004 4:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: SNOM and TAPI Snom TAPI integration is a joke... Andy ___ 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] OT: SNOM and TAPI
hi christian, have a look at http://www.julmar.com/. TSP++ version 2 is a opensource, GPLed library for creating a tapi service provider. I think this is a good point to start. I was just dreaming of having such a baby for use with asterisk* via it's manager function. bye thorsten - Original Message - From: Christian Stredicke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 23, 2004 10:58 PM Subject: RE: [Asterisk-Users] OT: SNOM and TAPI Sorry we are not so good in implementing Windows-stuff... Maybe has someone out there a template for TAPI? Something for someone who never did something with COM or DCOM or .net or whatever... ___ 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 Codec selection order
You can't do what you're trying to do. Asterisk isn't forwarding the g729 when you check voicemail. The voicemail is part of Asterisk so, it has to be able to speak the codec that you're using. John Alex Ovcharenko wrote: I have the following situation: my cisco 7940 attached to asterisk i use asterisk as voice mail, conference room, music on hold and so on. also i have cisco 5350 and use it as PSTN gateway. - As i know asterisk able to forward g729 frame. I enable in asterisk sip,conf options allow=g729 . and on cisco 5350i specified use only G729. (because i have restricted bandwidth to it). also i know that asterisk voice mail working fine with gsm, alaw, ulaw codecs. Now i able to call from 7940 to 5350 . Call established with codec g729. But when i call to voice mail my call failed with the reason specified below: Feb 22 02:21:28 NOTICE[-1336095824]: channel.c:1453 ast_set_write_format: Unable to find a path from GSM to G729A Feb 22 02:21:28 WARNING[-1336095824]: file.c:734 ast_streamfile: Unable to open vm-login (format G729A): No such file or directory Feb 22 02:21:28 WARNING[-1336095824]: app_voicemail.c:2714 vm_execmain: Couldn't stream login file any idea how i can make voicemail working with 7940 and g729 forwarding enabled. Alexey _ Store more e-mails with MSN Hotmail Extra Storage 4 plans to choose from! http://click.atdmt.com/AVE/go/onm00200362ave/direct/01/ ___ 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] *8# and zaphfc in NT-mode
hi everybody, does the zaphfc driver support the *8#, *78#, *72#, ... functions when running in NT-mode? thanks... bye thorsten
[Asterisk-Users] Nested include statements in extensions.conf?
Is it possible to have nested include statements in iax.conf? Example: [access-1] ; Can only use resouces in this context. [access-2] ;can use resources in this context and access-1 include = level-1 [access-3] ;can use resources in all three contexts. include = level-2 If we can't do this, would someone be interested in writing a patch so it is possible? It sure would help clean up some very messy extensions.conf files. John ___ 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] cdr_addon_mysql problem linking
I have Suse 9.0 with gcc3.3.1 (didn't have any problem with the previous version of gcc )and when I run make install I get the following error: /usr/lib/gcc-lib/i586-suse-linux-/3.3.1/../../../.../i586-suse-linux/bin/ld: cannot find -lz Any help would be appreciated. Dan
[Asterisk-Users] ATA 186 Registration!!!!
I'm tring to register my ATA to * and I getting the following message: Feb 23 18:13:04 NOTICE[1125329600]: chan_sip.c:5405 handle_request: Registration from 'sip:[EMAIL PROTECTED] user=phone' failed for 'xxx.xxx.xxx.xxx' I don't know what's wrong an why it register as user=phone??? Coul some one help me Thanks Erick ___ 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_addon_mysql problem linking
Make sure that zlib is installed and its location is in your LD_LIBRARY_CONFIG path (or /etc/ld.so.conf, at least on RH it's that file, I assume that SuSE is the same). This package would be on your SuSE CD(s), it's pretty much a base Linux package. Greg - Original Message - From: Dan Fernandez [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Monday, February 23, 2004 6:07 PM Subject: [Asterisk-Users] cdr_addon_mysql problem linking I have Suse 9.0 with gcc3.3.1 (didn't have any problem with the previous version of gcc )and when I run make install I get the following error: /usr/lib/gcc-lib/i586-suse-linux-/3.3.1/../../../.../i586-suse-linux/bin/ld: cannot find -lz Any help would be appreciated. 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
Re: [Asterisk-Users] FIXED : cdr_addon_mysql problem linking
I finally figured it out. Had to install zlib-devel package. sorry for the posting, but it was driving me nuts. - Original Message - From: Dan Fernandez To: [EMAIL PROTECTED] ; [EMAIL PROTECTED] Sent: Monday, February 23, 2004 8:07 PM Subject: [Asterisk-Users] cdr_addon_mysql problem linking I have Suse 9.0 with gcc3.3.1 (didn't have any problem with the previous version of gcc )and when I run make install I get the following error: /usr/lib/gcc-lib/i586-suse-linux-/3.3.1/../../../.../i586-suse-linux/bin/ld: cannot find -lz Any help would be appreciated. Dan
Re: [Asterisk-Users] A missing argument
Dave Cotton wrote: Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2 with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3 the most 10.0 fails at this point chan_zap.c: In function `handle_init_r2_event': chan_zap.c:4773: error: too few arguments to function `zt_new' make[1]: *** [chan_zap.o] Error 1 line 4773 has chan = zt_new(i, AST_STATE_RING, 0, SUB_REAL, 0); but greping shows that the declaration and other instances have 6 arguments. ML 9.2 is using gcc-3.3.1 whilst 10.0 is using gcc-3.3.2 What worries me is how many other programs in the world have the same type of error and the compiler has missed it? The keeps comming up. DO NOT INSTALL libr2. IT IS A HALF FINISHED PIECE OF JUNK I WILL NEVER FINISH. I now have a complete implementaion of R2, but I started agin when making that. It has nothing to do with the libr2 you have, and does not even use chan_zap. Regards, 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
Re: [Asterisk-Users] Re: [Asterisk-Users] Dual Xeon
Costa Tsaousis wrote: Also I would turn off Hyperthreading (in the bios). It may cause problems. What problems? Are these digium H/W specific, asterisk specific or generaly Linux problems? I don't know if The HT problems are generic, or something quirky in the Zaptel drivers. However, if you look at bug #828, it goes away when HT is switched off on the dual Xeon + 7505 chip set board from Tyan (model 2665) that I use. Regards, 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
Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?
Andres, thanks for your reply. I beg to disagree, here are the arguments: 1) Having INFO is imho a useful thing: it allows elements out of the media path to control DTMF-based service logic. Otherwise, you will end up processing media which affects bandwidth and latency noticably and does not scale. 2) Apart from the out-of-order argument, reprocessing retransmissions is a bug worth fixing. It is responsibility of transaction layer to absorb UDP retransmissions and never let app see them. (Similarly like TCP does not pass retranmissions to apps.) I think there are more cases for proper transaction processing other than just DTMF/INFO. 3) out-of-order delivery may or may not be an issue: gnerally, one would need to mainain a kind of playout buffer like for RTP. O-o-o delivery does not matter to me personaly since I send DTMF/INFO in stop-and-go mode. (BTW, I think the text in the RFC is not entirely correct, re-INIVITE should not cause CSeq gaps. Nevertheless, the RFC does not prevent anybody from implementing an INFO playout buffer). -jiri On Sun, 22 Feb 2004, Andres wrote: Hi Jiri, Been there. We switched from INFO to RFC2833 for this same reason. Take a look at: http://bugs.digium.com/bug_view_page.php?bug_id=0001033 Not only retransmissions are affected but out of order packets too. This behaviour can be partly blamed on the RFC: In addition, the INFO method does not define additional mechanisms for ensuring in-order delivery. While the CSeq header will be incremented upon the transmission of new INFO messages, this should not be used to determine the sequence of INFO information. This is due to the fact that there could be gaps in the INFO message CSeq count caused by a user agent sending re-INVITES or other SIP messages. Regards, Andres Jiri Kuthan wrote: I'm wondering whether people know if there could be a problem with * receiving retransmissions of INFO/DTMF requests. I'm trying to play DTMF via INFO to *. If it takes a 200 reply too long to come back, the request is retransmitted. Whenever this happens, the IVR down in PSTN reports that the number sequence is incorrect. This makes me guess that * turns INFO retransmissions into new DTMF digits on the PSTN part. Does anybody have the same experience? Is it a known problem? Are there any patches? Thanks, -jiri -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ 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 -- Andres Network Admin http://www.telesip.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 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] Pickup
Hi! The extension for Pickup seems to be *8#, but I cannot find it anywhere in any configuration file. Is this a hard wired extension? Yes, but you can override it in extensions.conf. Are there other hard wired extensions? If so, is there a list? What priority do they have? Is there any way to change them or map additional extensions to these functions? http://www.voip-info.org/wiki-Asterisk+PBX+functions http://www.voip-info.org/wiki-CLASS Philipp ___ 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 extension busy when not available ??
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 10:45 PM
Subject: Re: [Asterisk-Users] SIP extension busy when not available ??
I use a macro to define the extensions. In this way I only have to enter 1
line per actual extension.
The Macro is:
[macro-stdexten]
exten = s,1,ChanisAvail(${ARG2})
exten = s,2,Dial(${ARG2},20,Ttr)
exten = s,102,GoTo(voicemail,s,1)--Note A
exten = s,103,Hangup
exten = s,104,GoTo(voicemail,s,1)--Note B
exten = s,105,Hangup
Hey, that works pretty cool...
I've changed it a bit... (the DND stuff I found elsewhere)
[macro-stdexten]
exten = s,1,DBget(temp=DND/${ARG1}); DND set ?
exten = s,2,Goto(104) ; Yes.
exten = s,102,ChanisAvail(${ARG2}) ; Channel up?
exten = s,103,Dial(${ARG2},20,tr) ; Ring the interface, 20 seconds
maximum
exten = s,104,Voicemail(u${ARG1}) ; Send to voicemail w/ unavail
announce
exten = s,105,Hangup ; Doh...
exten = s,203,NoOp ; Nada...
exten = s,204,Voicemail(b${ARG1}) ; Send to voicemail w/ busy announce
exten = s,205,Hangup ; Doh...
[dnd]
; *61# turns it on
; *60# turns it off
;
exten = _*61,1,DBput(DND/${CALLERIDNUM}=YES})
exten = _*61,2,Playback(vm-goodbye)
exten = _*61,3,SoftHangup
exten = _*60,1,DBdel(DND/${CALLERIDNUM})
exten = _*61,2,Playback(vm-goodbye)
exten = _*60,3,SoftHangup
Now I just have to figure out a way to tell if it's on or not..
-- Soren
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