[asterisk-users] Destroy a zombie sip channel
I am unable how to get a zomebie sip channel to hangup. I've tried the following in the manager but it doesn't work. Action: Status Response: Success Message: Channel status will follow Event: Status Privilege: Call Channel: SIP/2003-09e2bbe8ZOMBIE CallerID: 093611168 CallerIDName: unknown Account: State: Up Link: SIP/2003-09e719f0 Uniqueid: 1171346560.592 Event: StatusComplete Action: Hangup Channel: SIP/2003-09e2bbe8ZOMBIE Response: Success Message: Channel Hungup Action: Status Response: Success Message: Channel status will follow Event: Status Privilege: Call Channel: SIP/2003-09e2bbe8ZOMBIE CallerID: 093611168 CallerIDName: unknown Account: State: Up Link: SIP/2003-09e719f0 Uniqueid: 1171346560.592 Event: StatusComplete Any other suggestions for how to kill this thing (ideally without restarting asterisk) would be appreciated. Cameron ___ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?
Benny Amorsen wrote: LA == Larry Alkoff [EMAIL PROTECTED] writes: LA If it's not a security issue I might as well have all phones with LA context=default in sip.conf even though voip-info specifically LA warns against that. Wonder why? Random SIP calls from the internet could end up in context default, if that is the default context mentioned in sip.conf. Then anyone on the Internet can use your outgoing lines. /Benny Hi /Benny Is the context default a 'special' context? That is, does Asterisk recognize it as unique in some way? How would anyone on the internet go about using my outgoing lines? Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 22 Feb 2007, at 07:25, Tzafrir Cohen wrote: I tried to use the 1.2.x RPMs and they would not work for me attempting to use them with an Eicon Diva Server card and Melware's chan_capi. Only by looking at the SRPM did I notice that they are patched with BRIStuff patches, which I have assume causes incompatibilities. Why is the a problem? The bristuff zaptel patch is a really small and non-intrussive one. The bristuff Asterisk patch, though, includes a complete reimplementation of chan_capi (the Junghanns' original chan_capi), which I heard noone really uses. Specifically, some simple AGI script I run to send and receive faxes with chan_capi did not work anymore. Both you and Axel are right about rebuilding the RPM of course. Matter of fact I always strongly prefer packages that come from (trusted) yum repositories. However, in this special case if I have to rebuild the package every time I don't see much advantage over a standard source install, which is very quick and simple. I just don't want to spend time analyzing a spec file to see which patches are applied and, if needed, back them out and build again and see if my stuff works again. It would be worth it if I had more than a single server running Asterisk. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF3VEDRAx5nvEhZLIRAgcTAJ0YXMtebIMdeuGPJ4rr3yilbEYDrgCgowKp 6Or+CuV7NIxLIGVp/ApIwHs= =J7Fe -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue information into db
Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cannot get whole DNID with ISDN line
Hi, I have an Asterisk 1.2.9.1 with chan_misdn 0.3.1-rc23 and a beronet octoBRI on a Debian box I have to set up instead of an old legacy PBX. My problem is I get only the base DNID and not the extensions (the last two digits) in Asterisk but the old PBX got all the DNID number so I think it is the card. Is there anybody experiencing a problem like this? How can I solve it? Any ideas? TIA Giorgio Incantalupo Follows my misdn.conf: [general] debug = 0 tracefile = /var/log/asterisk/misdn.trace bridging = yes stop_tone_after_first_digit = yes append_digits2exten = yes dynamic_crypt = no crypt_prefix = ** crypt_keys = test,muh ; users sections: ; ; inbound group ; [inbound] ports = 1ptp,2ptp,3ptp,4ptp,5ptp,6ptp,7ptp,8ptp context = outbound_isdn msns=* musicclass = native method = standard need_more_infos=yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Answer() command?
hi, is there anyway to Answer() the caller channel after the called number pickedup the phone. when an outside caller calls * system just continue ringing and not pick up the line and just dial an extension and then answer the caller channel after the called extension picked up the phone. is this possible in *? something like this: [incoming] exten = s,1,NoOp() exten = s,n,Dial(SIP/120) i've done this but when 120 extension picks up the phone just a noise will be heard and call won't be bridged to caller channel. i have also used Dial(SIP/120|M(answerme)) which runs a macro with Answer() command when called party picksup but it just re-answers the called extension!! thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?
context 'default' has not any special, it's context, that will be used if your peers/users definition doesn't contain any specific context if you have permited 'anonymous' calls to your asterisk, i.e allowguest=yes, unautenticated calls (calls, that will not match any specific user in sip.conf) will land to that context, that is defined in your [general] section in sip.conf you can use something like context=from-guest in [general] in extensions.conf you must define in [from-guest] section only your internal (ie. tool free) patterns to dial never put here (in [from-guest] in extensions.conf) patterns to dial outgoing lines (pstn), directly or indirectly via 'include=' statement PJ Larry Alkoff wrote: Is the context default a 'special' context? That is, does Asterisk recognize it as unique in some way? How would anyone on the internet go about using my outgoing lines? Larry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fax support
Hi all, I have read many forums and discussion groups talking about fax support in asterisk. Some of them conclude that asterisk doesn't support fax. However, some of them conclude that there is no relationship between fax and asterisk as asterisk will only pass the fax signal to the fax machine. I have tried the fax in asterisk before but failed. Anyone can give me some guideline how to make fax support with asterisk? ango ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Paradise Dove wrote: hi, is there anyway to Answer() the caller channel after the called number pickedup the phone. when an outside caller calls * system just continue ringing and not pick up the line and just dial an extension and then answer the caller channel after the called extension picked up the phone. is this possible in *? something like this: [incoming] exten = s,1,NoOp() exten = s,n,Dial(SIP/120) i've done this but when 120 extension picks up the phone just a noise will be heard and call won't be bridged to caller channel. i have also used Dial(SIP/120|M(answerme)) which runs a macro with Answer() command when called party picksup but it just re-answers the called extension!! thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax support
On Thu, 22 Feb 2007, Rilawich Ango wrote: Hi all, I have read many forums and discussion groups talking about fax support in asterisk. Some of them conclude that asterisk doesn't support fax. However, some of them conclude that there is no relationship between fax and asterisk as asterisk will only pass the fax signal to the fax machine. I have tried the fax in asterisk before but failed. Anyone can give me some guideline how to make fax support with asterisk? Search the archives for some scripts I posted a few weeks ago. But, as shipped, asterisk doesn't have native fax support, but it can be patched in via the spandsp code, giving you 2 new applications: RxFax and TxFax. You can plumb an incoming answered call to RxFax and it will decode the incoming fax stream into a TIFF file which you can then process as required. Asterisk does have the capability to listen to the incoming line for the fax startup tones though, so you can use this as part of an auto-attendant dialplan script to answer the line, listen for fax, if fax, then call RxFax or connect it to an outgoing analogue port with a real fax machine on it, or if not, play a message (if you know the extension, dial it now, etc.). You really want the incoming stream to be a PSTN line though, trying to encode an analogue fax call over the interweb is problematic and prone to failure. If going down that route, I'd get your DID supplier to do the fax to email conversion for you. (Theres a plethora of them in the UK, don't know about elswhere though) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax support
Hello ango, Try asterfax, I use it and it works fine. If you want to use a regular fax machine you can use an iaxt device to connect your fax machine to the LAN. Regards, mjmarrio On Thu, 2007-02-22 at 16:32 +0800, Rilawich Ango wrote: Hi all, I have read many forums and discussion groups talking about fax support in asterisk. Some of them conclude that asterisk doesn't support fax. However, some of them conclude that there is no relationship between fax and asterisk as asterisk will only pass the fax signal to the fax machine. I have tried the fax in asterisk before but failed. Anyone can give me some guideline how to make fax support with asterisk? ango ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] fax support
But, as shipped, asterisk doesn't have native fax support, but it can be patched in via the spandsp code, giving you 2 new applications: RxFax and TxFax. You can plumb an incoming answered call to RxFax and it will decode the incoming fax stream into a TIFF file which you can then process as required. You can also use a combination of iaxmodem and hylafax to add fax capabillities to asterisk. Although this is harder to configure it was more reliable on our systems. Kind regards, Ardjan Zwartjes, Telecats. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: How to separate outgoing extens from thecontexts from s
From: Larry Alkoff [EMAIL PROTECTED]
Date: Wed, 21 Feb 2007 20:00:52 -0600
...
You should consider that if any channel, incoming line, etc can enter an
extension context that it has the capability of accessing any extension
within that context.
Therefore, you should NOT allow access to outgoing or toll services in
contexts that are accessible (especially without a password) from incoming
channels
Doesn't that mean that
1. I have to have context=toll-access]
in any phone that can make toll calls
2, There is no way to give access to all internal phones unless I violate
voip-info's security directive above?
Not really. The voip-info warning is about incoming channels. But
definition they exclude any of your internal phones. The key is to use a
one context for your phones and a different one for your incoming line.
For example, suppose all your internal phones are SIP phones, and you use an
FXO channel for incoming. Then your sip.conf would include
context=toll-access with all devices, but the general section would have
context=incoming. Your zapata.conf would also include context=incoming.
Your extensions.conf may look like:
[general]
sippy1=SIP/phone1; living room
sippy2=SIP/phone2; kitchen
sippy3=SIP/phone3; bedroom
sippy4=SIP/phone4; laundry room
[incoming]
exten = s,1,NoOp(no dialing out allowd)
exten = s,n,Answer()
exten = s,n,Background(press-1-for-living-roompress-2-for-kitchen...)
exten = s,n,Dial(${sippy1}${sippy3},15); ring living room and bedroom
first
exten = s,n,Dial(${sippy1}${sippy3}${sippy2}${sippy4}); ring 'em all
exten = s,n,Hangup
exten = 1,1,Dial(${sippy1}); 1 is for living room
exten = 2,1,Dial(${sippy2}); 2 for kitchen
exten = 3,1,Dial(${sippy3}); 3 rings bedroom
exten = 4,1,Dial(${sippy4}); 4 rings laundry room
exten = 0,1,Dial(${sippy1}${sippy3}${sippy2}${sippy4}); ring 'em all
[toll-access]
; allow toll access and internal calls
exten = _Z.,1,Dial(Zap/1/${EXTEN}); anything other than [0-4] will go to
toll
exten = _[0-4],1,Goto(incoming,${EXTEN},1); internal extensions
Since I can give a password from sip.conf, is there an easy way to
automatically give that password in calls made from my internal phones
in such a way that external callers won't know the password even if they
breach the system?
Once you separate the contexts, there is no need for internal password.
How do people breach a system anyway? I've heard about hitting an
For example, if instead of separate contexts, your sip.conf has general
context and device context all in [default] (and zapata.conf has FXO channel
also in [default] context). Your [default] will look something like:
[default]
exten = s,1,Answer()
exten = s,n,Background(press-1-for-living-roompress-2-for-kitchen...)
exten = s,n,Dial(${sippy1}${sippy3},15); ring living room and bedroom
first
exten = s,n,Dial(${sippy1}${sippy3}${sippy2}${sippy4}); ring 'em all
exten = s,n,Hangup
exten = _Z.,1,Dial(Zap/1/${EXTEN}); anything other than [0-4] will go to
toll
exten = 1,1,Dial(${sippy1}); 1 is for living room
exten = 2,1,Dial(${sippy2}); 2 for kitchen
exten = 3,1,Dial(${sippy3}); 3 rings bedroom
exten = 4,1,Dial(${sippy4}); 4 rings laundry room
exten = 0,1,Dial(${sippy1}${sippy3}${sippy2}${sippy4}); ring 'em all
Now, some random SIP dialers on the net may land on your Asterisk SIP
address. This will invoke extension [EMAIL PROTECTED] If the caller dials 1
during your announcement after Asterisk answers, only living room rings.
But if the caller starts to dial 011315158005, Asterisk will transfer to
that extension, which will be matched by _Z. and dials out from your FXO
(Zap/1). Even if you don't have a lengthy announcement like illustrated
above, there's still a possibility that Asterisk intercepts the toll number
the caller dials in between priorities before priorities in s extension.
Even if you don't use Answer at all, there's a possibility that Asterisk
intercepts the toll number after you hang up but before the dial plan is
taken to h priority. The less IVR functions you implement, the lower the
risk. But there's always this possibility.
This is my understanding. More knowledgeable please correct me if I'm
wrong.
Yuan Liu
'*' as soon as the connection is made but don't understand it.
Or much else apparently g.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] SIP interface status and calllimit
21 feb 2007 kl. 15.50 skrev James Fromm: Anybody seen this behavior? To determine if it's my config or a bug, could I trouble someone running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP interface as a test? After a few hours a 'sip show inuse' should indicate the interface is on calls that it isn't. The incorrect count can be cleared up by ringing the interface for how ever many calls are incorrect. Beware, removing call-limit will require a restart to take effect. Thanks in advance for any help. A good way to check is to visit the bug tracker at bugs.digium.com If you do, you will find a few bug reports and also notice a few that has been resolved in Asterisk 1.4 svn, which is the base for the coming 1.4.1 release. Please try with latest 1.4 from subversion to test if the behaviour is fixed. Thanks, /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Asterisk use SIP info command
21 feb 2007 kl. 21.58 skrev Yuan LIU: What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. None. What do you want to do with SIP INFO? /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) The noise may indicate other problems. Yuan Liu Paradise Dove wrote: hi, is there anyway to Answer() the caller channel after the called number pickedup the phone. when an outside caller calls * system just continue ringing and not pick up the line and just dial an extension and then answer the caller channel after the called extension picked up the phone. is this possible in *? something like this: [incoming] exten = s,1,NoOp() exten = s,n,Dial(SIP/120) i've done this but when 120 extension picks up the phone just a noise will be heard and call won't be bridged to caller channel. i have also used Dial(SIP/120|M(answerme)) which runs a macro with Answer() command when called party picksup but it just re-answers the called extension!! thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
22 feb 2007 kl. 08.24 skrev Davy Chan: **I have one Asterisk box registering to another via SIP and on the registar **console I keep getting: ** **-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx ** **Anyone know how to turn off this feature? Look at: http://lists.digium.com/pipermail/asterisk-users/2007-February/ 179168.html The message is popping up because Asterisk's new behavior to SIP NOTIFY messages carrying Message Waiting Indication (MWI) info. See ya... Why enable MWI notification when you don't need it? /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
22 feb 2007 kl. 08.24 skrev Davy Chan: **I have one Asterisk box registering to another via SIP and on the registar **console I keep getting: ** **-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx ** **Anyone know how to turn off this feature? These messages also only show up if you have high verbosity. Taking verbosity down should remove the messages. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax support
22 feb 2007 kl. 10.05 skrev Gordon Henderson: On Thu, 22 Feb 2007, Rilawich Ango wrote: Hi all, I have read many forums and discussion groups talking about fax support in asterisk. Some of them conclude that asterisk doesn't support fax. However, some of them conclude that there is no relationship between fax and asterisk as asterisk will only pass the fax signal to the fax machine. I have tried the fax in asterisk before but failed. Anyone can give me some guideline how to make fax support with asterisk? Search the archives for some scripts I posted a few weeks ago. But, as shipped, asterisk doesn't have native fax support, but it can be patched in via the spandsp code, giving you 2 new applications: RxFax and TxFax. You can plumb an incoming answered call to RxFax and it will decode the incoming fax stream into a TIFF file which you can then process as required. Asterisk does have the capability to listen to the incoming line for the fax startup tones though, so you can use this as part of an auto-attendant dialplan script to answer the line, listen for fax, if fax, then call RxFax or connect it to an outgoing analogue port with a real fax machine on it, or if not, play a message (if you know the extension, dial it now, etc.). You really want the incoming stream to be a PSTN line though, trying to encode an analogue fax call over the interweb is problematic and prone to failure. If going down that route, I'd get your DID supplier to do the fax to email conversion for you. (Theres a plethora of them in the UK, don't know about elswhere though) And for fax over VOIP, sometimes called FOIP, Asterisk 1.4.x supports T.38 passthrough. However, the 1.4.0 release is buggy, so either use 1.4 from subversion or wait for 1.4.1. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: fax support
OEJ == Olle E Johansson [EMAIL PROTECTED] writes: OEJ And for fax over VOIP, sometimes called FOIP, Asterisk 1.4.x OEJ supports T.38 passthrough. However, the 1.4.0 release is buggy, OEJ so either use 1.4 from subversion or wait for 1.4.1. T.38 passthrough is not very exciting unless you happen to speak SIP to a provider which doesn't use Asterisk. Not that any of the other free software PBX's have trouble free T.38-to-PSTN, according to their mailing lists. Perhaps in a few months T.38 in free software will be more mature. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
Well, but isn't lines that begin with -- on the same verbosity level? So lowering the verbosity would in this case mean that you also stop displaying the dialplan execution steps. I have a similar problem regarding the -- SIP Seeding peer from astdb messages. I get a lot of these, so I tried to lower the verbosity, but to stop seeing these messages I had to go to verbosity level 2, and therefore got no dialplan statements. So shouldn't these informational statements be on a higher level? // Tobbe Olle E Johansson wrote: 22 feb 2007 kl. 08.24 skrev Davy Chan: **I have one Asterisk box registering to another via SIP and on the registar **console I keep getting: ** **-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx ** **Anyone know how to turn off this feature? These messages also only show up if you have high verbosity. Taking verbosity down should remove the messages. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE110P
genzaptelconf you mean? 2007/2/22, Paul Hales [EMAIL PROTECTED]: genzaptel is _not_ your friend when setting up E1. PaulH On Thu, 2007-02-22 at 00:46 +, younss azzayani wrote: this is my zaptel.conf:: [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 # channel 1, WCT1, unhandled for now # channel 2, WCT1, unhandled for now # channel 3, WCT1, unhandled for now # channel 4, WCT1, unhandled for now # channel 5, WCT1, unhandled for now # channel 6, WCT1, unhandled for now # channel 7, WCT1, unhandled for now # channel 8, WCT1, unhandled for now # channel 9, WCT1, unhandled for now # channel 10, WCT1, unhandled for now # channel 11, WCT1, unhandled for now # channel 12, WCT1, unhandled for now # channel 13, WCT1, unhandled for now # channel 14, WCT1, unhandled for now # channel 15, WCT1, unhandled for now # channel 16, WCT1, unhandled for now # channel 17, WCT1, unhandled for now # channel 18, WCT1, unhandled for now # channel 19, WCT1, unhandled for now # channel 20, WCT1, unhandled for now # channel 21, WCT1, unhandled for now # channel 22, WCT1, unhandled for now # channel 23, WCT1, unhandled for now # channel 24, WCT1, unhandled for now # channel 25, WCT1, unhandled for now # channel 26, WCT1, unhandled for now # channel 27, WCT1, unhandled for now # channel 28, WCT1, unhandled for now # channel 29, WCT1, unhandled for now # channel 30, WCT1, unhandled for now # channel 31, WCT1, unhandled for now # Span 2: ZTDUMMY/1 ZTDUMMY/1 1 # Global data loadzone= us defaultzone = us ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
22 feb 2007 kl. 11.19 skrev Torbjörn Abrahamsson: Well, but isn't lines that begin with -- on the same verbosity level? So lowering the verbosity would in this case mean that you also stop displaying the dialplan execution steps. I have a similar problem regarding the -- SIP Seeding peer from astdb messages. I get a lot of these, so I tried to lower the verbosity, but to stop seeing these messages I had to go to verbosity level 2, and therefore got no dialplan statements. So shouldn't these informational statements be on a higher level? The SIP Seeding peer from Astdb I think has no value at all, should propably be a debug message. I would still like to see SIP errors at verbosity 3. /O___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE110P
when i m running genzaptelconf [EMAIL PROTECTED] ~]# genzaptelconf STOPPING ASTERISK STOPPING FOP SERVER safe_opserver: no process killed FOP Server Stopped Generating '/etc/zaptel.conf' Generating '/etc/asterisk/zapata-auto.conf' Unloading zaptel hardware drivers: Unloading rxt1: ERROR: Module rxt1 does not exist in /proc/modules [FAILED] Unloading r1t1: ERROR: Module r1t1 does not exist in /proc/modules [FAILED] Unloading r4fxo: ERROR: Module r4fxo does not exist in /proc/modules [FAILED] Unloading ztdummy: [ OK ] Unloading wctdm: [ OK ] Unloading wcte11xp:[ OK ] Removing zaptel module:[ OK ] Loading zaptel framework: [ OK ] Waiting for zap to come online:[ OK ] Loading zaptel hardware modules: Loading wcte11xp: [ OK ] Loading wctdm: [ OK ] Loading ztdummy: [ OK ] Loading r4fxo: FATAL: Error inserting r4fxo (/lib/modules/2.6.9-34.0.2.EL/extra/r4fxo.ko): Unknown symbol in module, or unknown parameter (see dmesg) [FAILED] Loading r1t1: FATAL: Error inserting r1t1 (/lib/modules/2.6.9-34.0.2.EL/extra/r1t1.ko): Unknown symbol in module, or unknown parameter (see dmesg) [FAILED] Loading rxt1: FATAL: Error inserting rxt1 (/lib/modules/2.6.9-34.0.2.EL/extra/rxt1.ko): Unknown symbol in module, or unknown parameter (see dmesg) [FAILED] Running ztcfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP server is already running Binary file (standard input) matches ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] b410p + fax (echo cancellation)
We have recently purchased a B410P Digium 4* ISDN-2 card with hardware EC. On the same server, I also have a regular Digium 4-channel PSTN-card (TDM410P ?), used to interface to some analog devices, a.o. 2 fax machines. For faxing, EC needs to be off (or so I understand from the archives). How can I switch EC off for an ISDN B-channel if a fax is coming in? Z. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
I do need MWI notifcation, just not on this particulary trunk. Is there anyway to to turn off MWI on a particular trunk or can it only be done globally? On 2/22/07, Olle E Johansson [EMAIL PROTECTED] wrote: 22 feb 2007 kl. 08.24 skrev Davy Chan: **I have one Asterisk box registering to another via SIP and on the registar **console I keep getting: ** **-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx ** **Anyone know how to turn off this feature? Look at: http://lists.digium.com/pipermail/asterisk-users/2007-February/ 179168.html The message is popping up because Asterisk's new behavior to SIP NOTIFY messages carrying Message Waiting Indication (MWI) info. See ya... Why enable MWI notification when you don't need it? /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels hanging when SIP phone gets reset during call
21 feb 2007 kl. 12.54 skrev Steve Langstaff: Hi All. This is on Asterisk 1.2.13 I place a call between 2 SIP phones (with canreinvite=yes, qualify=yes). I reset the phones (so they don't have time to say BYE). Asterisk seems to think that the call is still ongoing. This persists until I do a 'restart now'. Check the RTP timers in sip.conf. They will hangup the call if there's no audio. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
Agreed, but your response to the OP said to lower the verbosity, and I commented that it might not be possible, due to then seeing no dialplan execution... :) How about the seeding messages then? Will you move these to a debug level? Or do a bug need to be filed in Mantis? // Tobbe Olle E Johansson wrote: 22 feb 2007 kl. 11.19 skrev Torbjörn Abrahamsson: Well, but isn't lines that begin with -- on the same verbosity level? So lowering the verbosity would in this case mean that you also stop displaying the dialplan execution steps. I have a similar problem regarding the -- SIP Seeding peer from astdb messages. I get a lot of these, so I tried to lower the verbosity, but to stop seeing these messages I had to go to verbosity level 2, and therefore got no dialplan statements. So shouldn't these informational statements be on a higher level? The SIP Seeding peer from Astdb I think has no value at all, should propably be a debug message. I would still like to see SIP errors at verbosity 3. /O___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Channels hanging when SIP phone gets resetduring call
Are the RTP timers applicable with canreinvite=yes ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson Sent: 22 February 2007 10:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Channels hanging when SIP phone gets resetduring call 21 feb 2007 kl. 12.54 skrev Steve Langstaff: Hi All. This is on Asterisk 1.2.13 I place a call between 2 SIP phones (with canreinvite=yes, qualify=yes). I reset the phones (so they don't have time to say BYE). Asterisk seems to think that the call is still ongoing. This persists until I do a 'restart now'. Check the RTP timers in sip.conf. They will hangup the call if there's no audio. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] An ISDN ISPBX to Voip Gateway??
Hello, I have 2 ISDN BRI connections, configured by my telephony provider as a ISPBX calling group. This allows me to have 4 concurrent calls. I want to use this connection with my VoIP network, with an Asterisk PBX, so I need a ISPBX to Voip (SIP) gateway. The problem is that I can't find any valid solution. The most of the gateways and cards use S0 simple BRI, and can't work with ISPBX. This is the case of the FritzBox. Does anybody known any gateway/card that I can use with this configuration and with asterisk? I rather prefer gateways than cards, but a card is ok if there is not any better solution. Thank you! -- Atentamente, LambdaStream Héctor Rivas www.lambdastream.com +34 981 17 33 44 begin:vcard fn;quoted-printable:H=C3=A9ctor Rivas G=C3=A1ndara n;quoted-printable;quoted-printable:Rivas G=C3=A1ndara;H=C3=A9ctor org:LambdaStream | www.lambdastream.com;Sistemas adr;quoted-printable;quoted-printable;quoted-printable;quoted-printable:Campus de Elvi=C3=B1a;;Edificio de Servicios de Investigaci=C3=B3n;A Coru=C3=B1a;A Coru=C3=B1a;15071;Spain email;internet:[EMAIL PROTECTED] title;quoted-printable:H=C3=A9ctor Rivas G=C3=A1ndara tel;work:+34 981173344 x-mozilla-html:FALSE url:http://www.lambdastream.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP Internet Server
Hi, This is my first post to the list so please be gentle ;) Okay, I have successfully configured Asterisk with a X100P clone card (soon to be replaced with a 1xFXO,1xFXS TDM card), and it quite happily answers the PSTN line and routes it to either a extension or voicemail. What I would like to be able to do next is have the extension accessible from across the internet. I run my own server, domain name, DNS etc so can update this all easily. Would somebody please point me to a wiki/document that shows me how to achieve this. I am more than likely being very dumb, but your help would be appreciated. Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GotoIf DURATION
Hi,
I am trying to branch a call based on it's duration but
${CDR(duration)} is always 0.
(The idea is to keep ringing the operator until a certain amount of
time has lapsed)
This does not work:
exten = s,4,Background(local/script8)
exten = s,5,Dial(${OPERATOR},30,tr)
exten = s,6,Noop(${CDR(duration)})
exten = s,7,GotoIf($[${CDR(duration)} 80]?4)
exten = s,8,Playback(local/script7)
CLI output:
-- Nobody picked up in 3 ms
-- Executing NoOp("SIP/mvn-f877", "0") in new stack
-- Executing GotoIf("SIP/mvn-f877", "1?4") in new stack
-- Goto (ivr,s,4)
-- Executing BackGround("SIP/mvn-f877", "local/script8") in new
stack
Any suggestions would be appreciated.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb, phonogram and motion pictures.
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Re: [asterisk-users] GotoIf DURATION
Marnus van Niekerk wrote:
Hi,
I am trying to branch a call based on it's duration but
${CDR(duration)} is always 0.
(The idea is to keep ringing the operator until a certain amount of
time has lapsed)
This does not work:
exten = s,5,Dial(${OPERATOR},30,tr)
Change the 30 on your dial statement to 80.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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[asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Thu, Feb 22, 2007 at 09:29:47AM +0200, Tzafrir Cohen wrote: On Thu, Feb 22, 2007 at 07:47:18AM +0100, Axel Thimm wrote: bristuff is the only patch in functionality, and for 1.2.15 I need to drop it again, because it does not apply Gee, it shows you're not on the bristuff list. Up-to-date bristuff patch for Asterisk: Not only am I not on this list, I didn't know of its existance until now, and I seem to be too dump to google it up. Can you provide a URL for the list? Thanks! -- Axel.Thimm at ATrpms.net pgpHttOVJg7FV.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie: registration failure (fwd)
Hi Sorry if this comes twice; i sendt first version from non-member address. I'm learning use Asterisk but cannot solve following problem: i have Asterisk v1.0.7 (DEbian) and Linphonec v1.2 (Debian). Every time i try to register within LAN i got 'Forbidden' message from Linphonec. Where to start searching for reason for this failure, is there more debuggin options available? Here is relevant part of sip.conf: [fujitsu] type=friend username=arimo context=siptest secret=n*i host=dynamic sip debug output: Sip read: REGISTER sip:arimo.iki.fi SIP/2.0 Via: SIP/2.0/UDP 192.168.1.67:5060;rport;branch=z9hG4bK769868031 From: sip:[EMAIL PROTECTED];tag=529821486 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 5 User-Agent: Linphone-1.2.0/eXosip Expires: 600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.1.67 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.67:5060;branch=z9hG4bK769868031 From: sip:[EMAIL PROTECTED];tag=529821486 To: sip:[EMAIL PROTECTED];tag=as0875a823 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.1.67:5060 Feb 22 12:57:12 NOTICE[14006]: chan_sip.c:7708 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.67' Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]' and sip sho peer fujitsu results: * Name : fujitsu Secret : Set MD5Secret: Not set Context : siptest Language : FromUser : FromDomain : Callgroup: (0) Pickupgroup : (0) Mailbox : LastMsgsSent : -1 Dynamic : Yes Expire : -1 Expiry : 900 Insecure : Yes Nat : No ACL : No CanReinvite : Yes PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : (Unspecified) Port 0 Defaddr-IP : 192.168.1.67 Port 5060 Username : arimo Codecs : 0x6 (gsm|ulaw) Codec Order : (ulaw|gsm) Status : UNKNOWN Useragent: Full Contact : You can still escape from the Gates of hell: Use Linux! -- arimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
22 feb 2007 kl. 11.36 skrev Eric Bishop: I do need MWI notifcation, just not on this particulary trunk. Is there anyway to to turn off MWI on a particular trunk or can it only be done globally? You enable it per device in sip.conf - that's the only way. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
22 feb 2007 kl. 11.54 skrev Torbjörn Abrahamsson: Agreed, but your response to the OP said to lower the verbosity, and I commented that it might not be possible, due to then seeing no dialplan execution... :) Well if you want that level of detail during the execution, these error messages won't really be adding a lot to the massive amount of text scrolling by anyway. How about the seeding messages then? Will you move these to a debug level? Or do a bug need to be filed in Mantis? The seeding message is already fixed in 1.4 and svn trunk. /O :-)___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels hanging when SIP phone gets resetduring call
22 feb 2007 kl. 12.20 skrev Steve Langstaff: Are the RTP timers applicable with canreinvite=yes ? how could we possibly check RTP if the RTP doesn't touch or network card at all? The timers are only used when we have RTP streams going to us. If the RTP stream is redirected, it's up to the end points to hangup due to media failure. The way to solve this is to implement the SIP timer extension. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What means: Request to schedule in the past?!?!
Hi guys, My asterisk is show me some errors on line registration. This message appear on console: Request to schedule in the past?!?! What it mean ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
On Thu, Feb 22, 2007 at 01:11:03PM +0100, Axel Thimm wrote: On Thu, Feb 22, 2007 at 09:29:47AM +0200, Tzafrir Cohen wrote: On Thu, Feb 22, 2007 at 07:47:18AM +0100, Axel Thimm wrote: bristuff is the only patch in functionality, and for 1.2.15 I need to drop it again, because it does not apply Gee, it shows you're not on the bristuff list. Up-to-date bristuff patch for Asterisk: Not only am I not on this list, I didn't know of its existance until now, and I seem to be too dump to google it up. Can you provide a URL for the list? Thanks! http://lists.three-dimensional.net/mailman/listinfo/bristuff-users And it is also availble through gmane: http://dir.gmane.org/gmane.comp.telephony.pbx.asterisk.bristuff.user And see also the voip-info page: http://www.voip-info.org/wiki/view/Bristuff -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[RESOLVED] Re: [asterisk-users] VoIP Internet Server
Apologies On Thu, 22 Feb 2007 11:49:52 +, uxbod [EMAIL PROTECTED] wrote: Hi, This is my first post to the list so please be gentle ;) Okay, I have successfully configured Asterisk with a X100P clone card (soon to be replaced with a 1xFXO,1xFXS TDM card), and it quite happily answers the PSTN line and routes it to either a extension or voicemail. What I would like to be able to do next is have the extension accessible from across the internet. I run my own server, domain name, DNS etc so can update this all easily. Would somebody please point me to a wiki/document that shows me how to achieve this. I am more than likely being very dumb, but your help would be appreciated. Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 603 driving me nuts
OK, nice. Any chance of it finding its way into 1.2-branch? I agree in some extent that one get a lot of information when looking at the dialplan execution, but the difference is that this is usefull information. Looking at the dialplan pass by is not made easier by having the seeding messages there too. It can be quite a lot of them, even when having only about 20 users, all subscribing to hints. But I understand this has already been realized, as it has been fixed in 1.4, so no need for further whining... :) // Tobbe Olle E Johansson wrote: 22 feb 2007 kl. 11.54 skrev Torbjörn Abrahamsson: Agreed, but your response to the OP said to lower the verbosity, and I commented that it might not be possible, due to then seeing no dialplan execution... :) Well if you want that level of detail during the execution, these error messages won't really be adding a lot to the massive amount of text scrolling by anyway. How about the seeding messages then? Will you move these to a debug level? Or do a bug need to be filed in Mantis? The seeding message is already fixed in 1.4 and svn trunk. /O :-)___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Cisco PRI gateway config
Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined to send and receive names via Q.sig. The definition fragments on Cisco are: isdn switch-type primary-qsig (so it will use Q.sig signalling). ... voice service voip qsig decode(This sends names out via Q.sig) fax protocol pass-through g711alaw sip controller E1 0/0/0 pri-group timeslots 1-31 ... interface Serial0/0/0:15 (This is for E1 PRI). no ip address encapsulation hdlc isdn switch-type primary-qsig isdn overlap-receiving isdn not-end-to-end 64 isdn incoming-voice voice isdn supp-service name calling (This receives names via Q.sig) isdn negotiate-bchan isdn outgoing ie facility isdn outgoing ie caller-number isdn outgoing ie called-number no cdp enable Anc the rest is quite standard. Regards, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Channels hanging when SIP phone gets resetduringcall
22 February 2007 12:22, Olle E Johansson wrote: 22 feb 2007 kl. 12.20 skrev Steve Langstaff: Are the RTP timers applicable with canreinvite=yes ? how could we possibly check RTP if the RTP doesn't touch or network card at all? You can't. I realise. The timers are only used when we have RTP streams going to us. If the RTP stream is redirected, it's up to the end points to hangup due to media failure. The endpoints have been rebooted, so they can't detect media failure (unless they have some persistent store of call state over a reboot!). The way to solve this is to implement the SIP timer extension. I see there is a discussion of this on the bug tracker... http://bugs.digium.com/bug_view_page.php?bug_id=207 Looks like I'm going to be pushing the media through the server after all... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Asterisk to Cisco's Rescue...again...AuthenticateLD Calls
From: Jason Aarons \(US\) [EMAIL PROTECTED] Glad to hear you had a workaround. I would suggest re-queing your TAC case, perhaps you got a outsourced or less experienced engineer at Cisco. Their support has varied depending on which city/group you get. Some have more experience then others. While your 2600 from 2001 timeframe it should work, you can't run any of 12.4T images over the last 3 years without maxing the DRAM/Flash. I've got 1200+ Forced Authorization Codes with 4.1(3)SR1 using 2811ISRs/VWIC2-1MFT-T1s running 12.4T with both MGCP and H323 gateways across 20 sites with no issues. Could be the old 2600s IOS as you mentioned. Thanks for the feedback Jason. We figured putting in a new router would make it work. But this was the customer's router, and they didn't want to spend the money on a new one. We didn't do our homework to check compatibility with the LD code feature. We assumed it would work, so we got caught in a pickle when it didn't. A fairly easy fix when running multiple platforms, Cisco and Asterisk, if one can't do something; usually the other one can pull the slack. Regards, JR ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP 601 help needed
I have been given the task of getting a Polycom IP 601 running SIP v2.0.1.0291 to register with our SER proxy and be able to interact with our Asterisk server for voice mail. The Asterisk server currently sends unsolicited NOTIFY messages to turn on/off the message waiting light. Most of the configuration is working however I cannot get the phone to register with SER if I set the voIpProt.server.1.address to the SRV name of our SIP domain. The only way the phone will register is if I set this parameter to the IP address of our SER server which is something we do not want to do. Is there any way to make this phone perform a SRV lookup for the server address? Thanks,Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] An ISDN ISPBX to Voip Gateway??
Try to see maybe it could be done with Patton Smartnode series gateway. Hector Rivas Gandara wrote: Hello, I have 2 ISDN BRI connections, configured by my telephony provider as a ISPBX calling group. This allows me to have 4 concurrent calls. I want to use this connection with my VoIP network, with an Asterisk PBX, so I need a ISPBX to Voip (SIP) gateway. The problem is that I can't find any valid solution. The most of the gateways and cards use S0 simple BRI, and can't work with ISPBX. This is the case of the FritzBox. Does anybody known any gateway/card that I can use with this configuration and with asterisk? I rather prefer gateways than cards, but a card is ok if there is not any better solution. Thank you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP 406 error - cause?
On 02/22/07 06:04 Michelle Dupuis said the following: I'm working on calls coming in to an asterisk box as H.323, and going out as SIP to a remote device (a VoiceMaster). The remote device is refusing the calls with SIP error 406 (Not Acceptable). I have attached the SIP debug output below. It looks like codecs overlaps - can anyone see why the call is being refused? 406s are usually returned because there're no common codecs for the call. check the codecs available on the voicemaster. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP RE-INVITE after an Answer()
Hi, I managed to get SIP re-invite working. If a call comes into my * box from my ITSP on a DiD, I can handle the call by calling Dial() in my dial plan and the call will get transferred and the media does not pass through my * box after the call is bridged. However, if I Answer() the call before calling the Dial() command, the call gets bridged OK but the media continues to go through my server. This does not happen with IAX. Is there any way to resolve this issue? The problem for me is that I need to answer in order to play an IVR recording. My setup: Asterisk 1.2.14 Redhat 9 I also have OpenSER running on a WRT54G, if that can help with a workaorond. Thanks, Hugh Powered by Execulink Webmail http://www.execulink.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] An ISDN ISPBX to Voip Gateway??
Try http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways - Original Message - From: Mindaugas Kuprys [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 22, 2007 2:49 PM Subject: Re: [asterisk-users] An ISDN ISPBX to Voip Gateway?? Try to see maybe it could be done with Patton Smartnode series gateway. Hector Rivas Gandara wrote: Hello, I have 2 ISDN BRI connections, configured by my telephony provider as a ISPBX calling group. This allows me to have 4 concurrent calls. I want to use this connection with my VoIP network, with an Asterisk PBX, so I need a ISPBX to Voip (SIP) gateway. The problem is that I can't find any valid solution. The most of the gateways and cards use S0 simple BRI, and can't work with ISPBX. This is the case of the FritzBox. Does anybody known any gateway/card that I can use with this configuration and with asterisk? I rather prefer gateways than cards, but a card is ok if there is not any better solution. Thank you! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.441 / Virus Database: 268.18.3/696 - Release Date: 2/21/2007 3:19 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Asterisk.
Hello, I am using the last version on subversion, I already configured the file sip.conf: [6614] username=6614 type=friend secret=* qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 context=meuvoip callerid=6614 [EMAIL PROTECTED] [6617] username=6617 type=friend secret=* qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 context=meuvoip callerid=6617 [EMAIL PROTECTED] and the file extensions.conf: [meuvoip] exten = 6614,1,Dial(SIP,6614,20) exten = 6614,2,Hangup exten = 6617,Dial,(SIP,6617,20) exten = 6617,2,Hangup so when I go to start asterisk... this message is showed: [EMAIL PROTECTED] ~]# asterisk -rd Asterisk SVN-trunk-r56126, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = NOTE: This is a development version of Asterisk, and should not be used in production installations. Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) I am using Mandriva Linux 2007 Plus. Please can someone help me ? Mhayk Whandson da Silva Lima www.mhayk.com.br skype: mhaykwhandson [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] fax support
But, as shipped, asterisk doesn't have native fax support, but it can be patched in via the spandsp code, giving you 2 new applications: RxFax and TxFax. You can plumb an incoming answered call to RxFax and it will decode the incoming fax stream into a TIFF file which you can then process as required. You can also use a combination of iaxmodem and hylafax to add fax capabillities to asterisk. Although this is harder to configure it was more reliable on our systems. Kind regards, Ardjan Zwartjes, Telecats. This is the route I use for my office and for my clients. It Just Works. And the configuration isn't that bad after a cursory read through the docs. --Michel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asternic Flash Panel
Has anyone gotten this configured to show all extensions vertically instead of filling up the window. If so would you mind sharing your configuration Yes I have tried searching terms like +asternic +op_panel +vertical and a slew of others. Unsucessful though. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g' http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 How a man plays the game shows something of his character - how he loses shows all - Mr. Luckey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP interface status and calllimit
I've reviewed the bugs reports. I didn't see anything that applied to this. Have you? Could you point it out to me? Olle E Johansson wrote: 21 feb 2007 kl. 15.50 skrev James Fromm: Anybody seen this behavior? To determine if it's my config or a bug, could I trouble someone running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP interface as a test? After a few hours a 'sip show inuse' should indicate the interface is on calls that it isn't. The incorrect count can be cleared up by ringing the interface for how ever many calls are incorrect. Beware, removing call-limit will require a restart to take effect. Thanks in advance for any help. A good way to check is to visit the bug tracker at bugs.digium.com If you do, you will find a few bug reports and also notice a few that has been resolved in Asterisk 1.4 svn, which is the base for the coming 1.4.1 release. Please try with latest 1.4 from subversion to test if the behaviour is fixed. Thanks, /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Configuring Asterisk.
so when I go to start asterisk... this message is showed: [EMAIL PROTECTED] ~]# asterisk -rd asterisk -h Asterisk 1.2.14, Copyright (C) 1999 - 2005, Digium, Inc. and others. Usage: asterisk [OPTIONS] Valid Options: -V Display version number and exit -C configfile Use an alternate configuration file -G group Run as a group other than the caller -U user Run as a user other than the caller -c Provide console CLI -dEnable extra debugging -f Do not fork -g Dump core in case of a crash -h This help screen -i Initialize crypto keys at startup -n Disable console colorization -p Run as pseudo-realtime thread -q Quiet mode (suppress output) -rConnect to Asterisk on this machine -R Connect to Asterisk, and attempt to reconnect if disconnected -t Record soundfiles in /var/tmp and move them where they belong after they are done. -T Display the time in [Mmm dd hh:mm:ss] format for each line of output to the CLI. -v Increase verbosity (multiple v's = more verbose) -x cmdExecute command cmd (only valid with -r) ... Try asterisk -pT first and THEN asterisk -rd --Michel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Asterisk.
You have to actually start asterisk with a command like safe-asterisk before you can connect to the console with the command asterisk -r Mhayk Whandson wrote: Hello, I am using the last version on subversion, I already configured the file sip.conf: [6614] username=6614 type=friend secret=* qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 context=meuvoip callerid=6614 [EMAIL PROTECTED] [6617] username=6617 type=friend secret=* qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 context=meuvoip callerid=6617 [EMAIL PROTECTED] and the file extensions.conf: [meuvoip] exten = 6614,1,Dial(SIP,6614,20) exten = 6614,2,Hangup exten = 6617,Dial,(SIP,6617,20) exten = 6617,2,Hangup so when I go to start asterisk... this message is showed: [EMAIL PROTECTED] ~]# asterisk -rd Asterisk SVN-trunk-r56126, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = NOTE: This is a development version of Asterisk, and should not be used in production installations. Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) I am using Mandriva Linux 2007 Plus. Please can someone help me ? Mhayk Whandson da Silva Lima www.mhayk.com.br skype: mhaykwhandson [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP interface status and calllimit
Nevermind, I found it. I'll put up an SVN version in my dev environment today. Thanks. James Fromm wrote: I've reviewed the bugs reports. I didn't see anything that applied to this. Have you? Could you point it out to me? Olle E Johansson wrote: 21 feb 2007 kl. 15.50 skrev James Fromm: Anybody seen this behavior? To determine if it's my config or a bug, could I trouble someone running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP interface as a test? After a few hours a 'sip show inuse' should indicate the interface is on calls that it isn't. The incorrect count can be cleared up by ringing the interface for how ever many calls are incorrect. Beware, removing call-limit will require a restart to take effect. Thanks in advance for any help. A good way to check is to visit the bug tracker at bugs.digium.com If you do, you will find a few bug reports and also notice a few that has been resolved in Asterisk 1.4 svn, which is the base for the coming 1.4.1 release. Please try with latest 1.4 from subversion to test if the behaviour is fixed. Thanks, /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco PRI gateway config
interesting! so it means, that you can now see caller id names between sip phones connected to asterisk and phones connected to pbx? PJ Yehavi Bourvine +972-8-9489444 wrote: Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined to send and receive names via Q.sig. The definition fragments on Cisco are: isdn switch-type primary-qsig (so it will use Q.sig signalling). ... voice service voip qsig decode(This sends names out via Q.sig) fax protocol pass-through g711alaw sip controller E1 0/0/0 pri-group timeslots 1-31 ... interface Serial0/0/0:15 (This is for E1 PRI). no ip address encapsulation hdlc isdn switch-type primary-qsig isdn overlap-receiving isdn not-end-to-end 64 isdn incoming-voice voice isdn supp-service name calling (This receives names via Q.sig) isdn negotiate-bchan isdn outgoing ie facility isdn outgoing ie caller-number isdn outgoing ie called-number no cdp enable Anc the rest is quite standard. Regards, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco PRI gateway config
interesting! so it means, that you can now see caller id names between sip phones connected to asterisk and phones connected to pbx? Yes! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: FXS - Init Indirect Registers UNSUCCESSFULLY.
ahh! I am having this same problem all of a sudden I've installed many TDM cards before ..never had this problem what gives? Trying to load zaptel 1.0.10 ... Rev. G card ... tried uncommenting the revH fix in zconfig.h ...but no go ideas?! -- Chris Michael C. Cambria [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I'm having problems with a TDM22B. The FXO modules work fine. Both FXS modules fail to initialized. The error messages I seeR when the module loads: Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. I already RMA'ed a TDM22B because of this problem. Now that the replacement shows a similar issue, I want to see if anyone else can think of something to try; at least until Monday when I can get an RMA number for this card. If it helps, I have a TDM20B (just FXS modules) that does not see this problem when I place it in the same slot. Here is what dmesg shows for the TDM22B when the system boots, or when I issues modprobe wctdm: Freshmaker version: 73 Freshmaker passed register test !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. Module 0: FAILED FXS (FCC) !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. Module 1: FAILED FXS (FCC) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed
[asterisk-users] Lastest SVN (1.4) and realtime call limit
Hello, I am running version 1.4 with realtime support. I've set (for Snom phones 300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the database). - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem was that when such a phone received a call and did attended transfer it was left in use and could not receive new calls. - After seeing reference to similar problem on this list I;ve downloaded today the latest SVN source code and installed it. The problem is that it shows the call limit as 0 and not as 1. Any idea? Thanks, __yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Asterisk.
On Thu, Feb 22, 2007 at 09:01:47AM -0600, Joe Dennick wrote: You have to actually start asterisk with a command like safe-asterisk like asterisk . safe_asterisk only adds noise and complication and doesn't help you with anything. BTW: it may also not be recommended to use -p at start, is it makes it easier to hang the system. before you can connect to the console with the command asterisk -r Right. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) BUT, when callprogress=yes, asterisk doesn't bridge the call and just ring for the caller and noise for called!! is it a bug or it's normal? The noise may indicate other problems. Yuan Liu Paradise Dove wrote: hi, is there anyway to Answer() the caller channel after the called number pickedup the phone. when an outside caller calls * system just continue ringing and not pick up the line and just dial an extension and then answer the caller channel after the called extension picked up the phone. is this possible in *? something like this: [incoming] exten = s,1,NoOp() exten = s,n,Dial(SIP/120) i've done this but when 120 extension picks up the phone just a noise will be heard and call won't be bridged to caller channel. i have also used Dial(SIP/120|M(answerme)) which runs a macro with Answer() command when called party picksup but it just re-answers the called extension!! thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CheckPoint (DMZ) + Asterisk (SIP)
Hi! Everyone, Has anybody, already experienced any issue with port forward and NAT translation into the CheckPoint Firewall? I did setup a DMZ for asterisk be accessible from the LAN and WAN as well. The needed ports were properly opened but even though I am not able to authenticate into it from both ways LAN and WAN... It looks like to be a SIP translation issue, but I am really not sure about that... Does anybody have anything to say to help me? Thanks, Alcides ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - VoiceGenie IVR
Hi,
I'm currently working on a setup between Asterisk and VoiceGenie (which
is a IVR system).
The way my setup is done, is that I have a PRI line coming in my
Asterisk server, and then VoiceGenie is connected to Asterisk via SIP,
like any other softphone basically. I'm able to receive calls in
Asterisk and then link them with VoiceGenie. But one of my issues is
that when I get an outside call, transfer the call to VoiceGenie, then
for that specific calls VoiceGenie would decide that this call has to be
transfered to an outside party, so then VoiceGenie calls up that number,
it goes through Asterisk and it reached the other person. But the link
doesn't stay up very long, max 15 seconds.
That's one of the errors that I see in Asterisk(for obvious reasons I've
replaced some numbers with *):
-- Hungup 'Zap/8-1'
Feb 15 14:10:19 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 1
(Critical Response)
Feb 15 14:10:28 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 1
(Critical Response)
-- Hungup 'Zap/1-1'
Here's a part of my dialplan for outside calls:
exten = _9XX,1,Set(CALLERID(all)=450-655-)
exten = _9XX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
And here's a Macro that I use for incoming call for VoiceGenie:
[macro-voicegenie]
exten = s,1,Answer
exten = s,2,SIPAddHeader(X-Asterisk-DID: ${ARG1})
exten = s,3,SIPAddHeader(X-Asterisk-CallerName: ${ARG2})
exten = s,4,Dial(SIP/108)
exten = 514380,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514380,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
Here's the config in sip.conf:
[108]
type=friend
context=internal
host=10.1.1.40
callerid=VoiceGenie 108
progressinband=never
disallow=all
allow=ulaw
Also, the support team at Voicegenie they asked me if I stop sending
183 Session Progress before 180 Ringing.
It seems that this could be part of my issue.
Thanks,
--
Eric Rousse
System Administrator
514.380.2992
450.655.1001
1.888.641.5800
Telmatik inc.
204 Montarville, suite 250
Boucherville, QC, Canada
J4B 6S2
www.telmatik.com
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Re: [asterisk-users] CheckPoint (DMZ) + Asterisk (SIP)
On Thu, 22 Feb 2007 13:46:07 -0200, Alcides [EMAIL PROTECTED] wrote: Hi! Everyone, Has anybody, already experienced any issue with port forward and NAT translation into the CheckPoint Firewall? I did setup a DMZ for asterisk be accessible from the LAN and WAN as well. The needed ports were properly opened but even though I am not able to authenticate into it from both ways LAN and WAN... It looks like to be a SIP translation issue, but I am really not sure about that... Does anybody have anything to say to help me? Thanks, Alcides ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Is this any help ? http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problem with busydetect and cell phones
I just wanted to share the solution for this problem. The busydetect feature is working with all cell phone carriers now as well. I added the following to my Zapata.conf. rxgain=4. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor G. Hammonds Sent: Monday, February 19, 2007 12:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Problem with busydetect and cell phones Ryan McDaniel wrote: I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal correctly and clears the line. If I call from an internal extension to a cell phone and then hangup the cell phone Asterisk will never detect the busy signal though it is clearly there. Asterisk will happily sit there listening to the busy signal. I suspect that the busy signal styles are slightly different though it is undetectable to me. How can I fix this??? It causes severe issues when a call is forwarded to a cell phone via the Zap interfaces as once you hangup the cell phone Asterisk never releases the channel. The landlines are with ATT. The cell phones I'm testing with are Cingular (ATT subsidiary). There must be a subtle difference in the busy signals. How can I make it catch busy signals from both carriers? Have you tried calling ATT and asking for call disconnect supervision? I realise that this can be a thankless and tedious endeavour, but it IS worth trying. There are almost no commercial switches that don't support this; it's a matter of activating it for the specific circuit in software. Particularly if you have a business line -- you can demand it. All PBXs need it if they use analog lines (and plenty still do) so I'm sure this is not an alien concept to ATT. It's just a matter of finding the right Earthling there who can help you. This might be one of those times where a beer with the technician will get you some joy, if calling Repair doesn't give you any joy. -Stephen- Unfortunately I tried that. Apparently my lines are on one of the last really ancient junction boxes in Southern California. When using busydetect is it looking for any on / off repetitive sound to identify the busy signal, or for a specific length sound as defined in the indications.conf region? I'd really like to avoid using callprogress if possible. Is there a way to tweak it so it will accept a wider variety of busy patterns? - Ryan Ryan, Even 1AESS switches offer disconnect supervision -- and I am not aware of any of those still in primary service in Southern California. By early 2000, Pacific Bell (then SBC, now ATT) replaced all the analogue 1As with DMS-100s. If you care to contact me off list, I may be able to help get you in touch with the right department to assist you. Sincerely, Trevor Hammonds ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
Paradise Dove wrote: On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) BUT, when callprogress=yes, asterisk doesn't bridge the call and just ring for the caller and noise for called!! is it a bug or it's normal? Don't use callprogress. It doesn't work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lastest SVN (1.4) and realtime call limit
22 feb 2007 kl. 16.38 skrev Yehavi Bourvine +972-8-9489444: Hello, I am running version 1.4 with realtime support. I've set (for Snom phones 300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the database). - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem was that when such a phone received a call and did attended transfer it was left in use and could not receive new calls. - After seeing reference to similar problem on this list I;ve downloaded today the latest SVN source code and installed it. The problem is that it shows the call limit as 0 and not as 1. Any idea? Call limits are in memory flags that we don't keep in the database. Realtime peers are *not* by default kept in memory and not guaranteed to stay in memory. Using call limits on them might work, but is not guaranteed to work. Realtime peers/users are made to be optimal for large installations, but lack a lot of the features in regards to call limits, subscriptions, message waiting indications. The bug where the transferer was kept in use after the transfer was fixed a few days ago in 1.4 svn. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox -- ACPI and IO-APIC?
On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote: My point is that if it's going to involve rebuilding a kernel to support IO-APIC, then I'd just as soon build from the ground up. And my point is that this is the Asterisk Users mail list, not the Trixbox list. Either ask other there or ask on a CentOS list. Once you decide to build from the ground up, your Asterisk questions can be reliably answered here. Most of us don't have any idea what all kinds of weird stuff they put in Trixbox these days, which is why I saw reliably answered. The people on here could give you a solution to something that would break a Trixbox install. Your question though, sounds like it needs to be directed to a CentOS, or as Kodak said, a RHEL list or forum. I personally don't have any idea what you are asking, I'm pretty sure it's not an Asterisk config question, though. I don't mean to be rude, just trying to point you in the direction to get the best answers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] An ISDN ISPBX to Voip Gateway??
Mindaugas Kuprys wrote: Try to see maybe it could be done with Patton Smartnode series gateway. Thank you. I take a look at the SmartNode 1200 [1]. It's great, but I don't known too much about ISDN technologies and I'm not sure if this device will work with a ISPBX line. As far as I known, the ISPBX (aka ISDN PABX???) offers ISDN T interface. My fritzbox doesn't work because it needs a ISDN S interface. Is this correct? If it's correct, Patton Electronics SmartNode 1200 should work, since it can be connected to a ISDN NT BRI S0 (S/T) interface. Can somebody tell me if it really could work with this configuration? [1] http://www.dceexpress.com/SmartNode1200.htm -- Atentamente, LambdaStream Héctor Rivas www.lambdastream.com +34 981 17 33 44 begin:vcard fn;quoted-printable:H=C3=A9ctor Rivas G=C3=A1ndara n;quoted-printable;quoted-printable:Rivas G=C3=A1ndara;H=C3=A9ctor org:LambdaStream | www.lambdastream.com;Sistemas adr;quoted-printable;quoted-printable;quoted-printable;quoted-printable:Campus de Elvi=C3=B1a;;Edificio de Servicios de Investigaci=C3=B3n;A Coru=C3=B1a;A Coru=C3=B1a;15071;Spain email;internet:[EMAIL PROTECTED] title;quoted-printable:H=C3=A9ctor Rivas G=C3=A1ndara tel;work:+34 981173344 x-mozilla-html:FALSE url:http://www.lambdastream.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] An ISDN ISPBX to Voip Gateway??
George Camilleri wrote: Try http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways Thanks. I visited this reference before, but I don't known if this gateways can work with my current configuration. I further describe my problem in other post. -- Atentamente, LambdaStream Héctor Rivas www.lambdastream.com +34 981 17 33 44 begin:vcard fn;quoted-printable:H=C3=A9ctor Rivas G=C3=A1ndara n;quoted-printable;quoted-printable:Rivas G=C3=A1ndara;H=C3=A9ctor org:LambdaStream | www.lambdastream.com;Sistemas adr;quoted-printable;quoted-printable;quoted-printable;quoted-printable:Campus de Elvi=C3=B1a;;Edificio de Servicios de Investigaci=C3=B3n;A Coru=C3=B1a;A Coru=C3=B1a;15071;Spain email;internet:[EMAIL PROTECTED] title;quoted-printable:H=C3=A9ctor Rivas G=C3=A1ndara tel;work:+34 981173344 x-mozilla-html:FALSE url:http://www.lambdastream.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What means: Request to schedule in the past?!?!
On 2/22/07, Frederico Madeira [EMAIL PROTECTED] wrote: My asterisk is show me some errors on line registration. This message appear on console: Request to schedule in the past?!?! I could be wrong here, but I think one of the symptoms of that could be not have any zaptel devices and not having ztdummy loaded. I've had that a few times on new systems, and I think that was what I narrowed it down to. Or, if you have a PRI card, the timing being incorrect. I think it was one of those two. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New tutorial: DTMF tone detection
Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue information into db
Not sure about * 1.4, but you can definitely use our Qloaderd script to do that - see http://queuemetrics.com/download.jsp . That script is pretty smart (to be a loader script...) and is able to handle restarts and database disconnections. l. In data Thu, 22 Feb 2007 09:20:59 +0100, nik600 [EMAIL PROTECTED] ha scritto: Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What means: Request to schedule in the past?!?!
Lacy Moore - Aspendora wrote: On 2/22/07, Frederico Madeira [EMAIL PROTECTED] wrote: My asterisk is show me some errors on line registration. This message appear on console: Request to schedule in the past?!?! I could be wrong here, but I think one of the symptoms of that could be not have any zaptel devices and not having ztdummy loaded. I've had that a few times on new systems, and I think that was what I narrowed it down to. Or, if you have a PRI card, the timing being incorrect. I think it was one of those two. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users check the date on the machine? signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue information into db
I am planning to develop an open source (GPL) queue statistic/analyzer. Can i use that to store data into the db? Or shall i wrote some php code to do that? On 2/22/07, lenz [EMAIL PROTECTED] wrote: Not sure about * 1.4, but you can definitely use our Qloaderd script to do that - see http://queuemetrics.com/download.jsp . That script is pretty smart (to be a loader script...) and is able to handle restarts and database disconnections. l. In data Thu, 22 Feb 2007 09:20:59 +0100, nik600 [EMAIL PROTECTED] ha scritto: Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Asterisk use SIP info command
From: Olle E Johansson [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 10:36:45 +0100 21 feb 2007 kl. 21.58 skrev Yuan LIU: What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. None. What do you want to do with SIP INFO? /O I was watching the send variable thread and thought INFO would be a handy tool for that in the middle of a session. SIP headers are only sent along with INVITE, it seems. And if the situation requires the recepient to send back a value, INVITE would be impossible. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 10:09:07 -0600 Paradise Dove wrote: On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) BUT, when callprogress=yes, asterisk doesn't bridge the call and just ring for the caller and noise for called!! is it a bug or it's normal? That zap channel happens to use usecallprogress=yes, and it did not have this problem. I'm very confused about all these feature names like why usecallprogress and callprogress (some examples use one, others use the other), version compatibility, etc. But this particular setting does not affect SIP/RTP connection. Come to think about it, callprogress only affects Zap channel and should not affect RTP. There must be other things that prevent RTP from streaming. Don't use callprogress. It doesn't work. Until you are desperate and callprogress is the last straw in sight:-) Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New tutorial: DTMF tone detection
From: lenz [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 17:30:44 +0100 Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 Would this prevent POTS phones from interacting with Asterisk? Yuan Liu I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
On 2/22/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Paradise Dove wrote: On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) BUT, when callprogress=yes, asterisk doesn't bridge the call and just ring for the caller and noise for called!! is it a bug or it's normal? Don't use callprogress. It doesn't work. GOOD NEWS: Problem Fixed! i wrote a patch for dsp.c and chan_zap.c now both callprogress and answer problem work fine together. i also add a config option in zapata.conf to tune callprogress now it works with over 95 percent accuracy. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macros, Background(), Return Values...
I am programming a very large dialplan right now (Asterisk 1.4), and a couple of things are annoying the heck out of me. 1. When in a macro, background() does not work properly. If you use the background() app inside a macro, and then press a key, execution returns back to the calling context where it tries to match that extension. I believe this is a known bug. 2. Is there any way to have macros return a value? I can pass arguments to macro's with ARG1..ARGN, but the only way to set a return variable is to set a channel variable. Essentially, I have a large number of global variables which is never good. 3. If you use Gosub to and Return to jump into and out of contexts, and use them like macros to get around the background() problem, the global variable issue becomes worse as there's no way to explicitly pass variables to the contexts when you do this. 4.Every time you make a decision, you have to use GotoIf, which means more code to do simple things like set variables, or do things based on a decision. It would be great if there was SetIf(), MacroIf(), or even doIf() applications. Has anyone tried to program large complex dialplans before and come across some of these issues? How did you resolve them? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Jabber/Asterisk Integration
Chris Earle wrote: agent monitoring screen? curious, which app are you using for that? Unfortunately a proprietary app written in a closed language called Progress. However, the basics are that we embedded the Ipworks ActiveX xmpp control, created a text box for each agent of a queue, and changed the colours / labels / tooltips according to the presence message of each jabber client. Julian. -- Chris Earle Julian Lyndon-Smith [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Kyle Sexton wrote: Started playing with 1.4 and I'm curious what uses people have come up with for the Jabber integration? So far I can think of presence based call routing, but I'm sure there are other ideas. How are YOU using the new Jabber features in 1.4? :) We've been using it since July last year (brave / stupid - make your choice) for integrating our custom application with the asterisk system. The phone system sends all sorts of call information to the agent about to receive the call, whilst the agent monitoring screen is used to monitor the presence of the agents and their dialplan status (dialling / calling / etc etc) Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Asterisk use SIP info command
Yuan LIU wrote: From: Olle E Johansson [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 10:36:45 +0100 21 feb 2007 kl. 21.58 skrev Yuan LIU: What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. None. What do you want to do with SIP INFO? /O I was watching the send variable thread and thought INFO would be a handy tool for that in the middle of a session. SIP headers are only sent along with INVITE, it seems. And if the situation requires the recepient to send back a value, INVITE would be impossible. I thought it might be useful to be able to ask Asterisk for the current SIP CSeq through the Manager API in order to send your own SIP messages during a call outside of Asterisk (for AOC, whatever). Each time you ask for the CSeq Asterisk should increment the value so it does not get out of sync. Anyone sharing my opinion? We might open a feature request. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New tutorial: DTMF tone detection
Well, kind of - it is meant for weird situations where mostly you do not have regular POTS phones. Of course all DTMF detection would be disrupted. l. In data Thu, 22 Feb 2007 18:59:50 +0100, Yuan LIU [EMAIL PROTECTED] ha scritto: From: lenz [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 17:30:44 +0100 Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 Would this prevent POTS phones from interacting with Asterisk? Yuan Liu -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Macros, Background(), Return Values...
From: Doug Garstang [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 10:17:20 -0800
I am programming a very large dialplan right now (Asterisk 1.4), and a
couple of things are annoying the heck out of me.
I have not programmed large dial plans, but have encountered some of the
nuances.
1. When in a macro, background() does not work properly. If you use the
background() app inside a macro, and then press a key, execution returns
back to the calling context where it tries to match that extension. I
believe this is a known bug.
Or feature? I posted this question before and haven't got an answer. It
might be a feature on the premise that a macro is not a context. (Even
though I can use Goto to wander inside.)
2. Is there any way to have macros return a value? I can pass arguments to
macro's with ARG1..ARGN, but the only way to set a return variable is to
set a channel variable. Essentially, I have a large number of global
variables which is never good.
In my wild experimentation, I tried to
Goto(${MACRO_CONTEXT},${return_value},1). But another way to do this is to
abandon macro and use Local channel. Expect to set all necessary _variables
yourself before dialing into a local channel.
3. If you use Gosub to and Return to jump into and out of contexts, and use
them like macros to get around the background() problem, the global
variable issue becomes worse as there's no way to explicitly pass variables
to the contexts when you do this.
Local channel may be your friend.
4.Every time you make a decision, you have to use GotoIf, which means more
code to do simple things like set variables, or do things based on a
decision. It would be great if there was SetIf(), MacroIf(), or even doIf()
applications.
Have you looked at AEL?
Yuan Liu
Has anyone tried to program large complex dialplans before and come across
some of these issues? How did you resolve them?
Doug.
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[asterisk-users] AG-188
Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (haven't tried SIP), there is no ring. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. So my call flow will be PRI - Asterisk 1.2.x Out the 2nd PRI to the 3660 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA. I have the 3660 there to take the call via TDM and convert to T38. I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned. That's how I plan to set it up. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson Sent: Wednesday, February 21, 2007 10:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax with T.38 Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, *Thomas Deillon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] De la part de Thomas Deillon Envoyé: jeudi, 15. février 2007 11:26 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30
Re: [asterisk-users] How does Asterisk use SIP info command
From: Philipp Kempgen [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 19:34:37 +0100 Yuan LIU wrote: From: Olle E Johansson [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 10:36:45 +0100 21 feb 2007 kl. 21.58 skrev Yuan LIU: What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. None. What do you want to do with SIP INFO? /O I was watching the send variable thread and thought INFO would be a handy tool for that in the middle of a session. SIP headers are only sent along with INVITE, it seems. And if the situation requires the recepient to send back a value, INVITE would be impossible. I thought it might be useful to be able to ask Asterisk for the current SIP CSeq through the Manager API in order to send your own SIP messages during a call outside of Asterisk (for AOC, whatever). Each time you ask for the CSeq Asterisk should increment the value so it does not get out of sync. A little reading led to this interesting discussion: http://www.voip-info.org/wiki/view/SIP+method+invite. Yuan Liu Anyone sharing my opinion? We might open a feature request. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
On Thu, Feb 22, 2007 at 09:40:54PM +0330, Paradise Dove wrote: On 2/22/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Paradise Dove wrote: On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) BUT, when callprogress=yes, asterisk doesn't bridge the call and just ring for the caller and noise for called!! is it a bug or it's normal? Don't use callprogress. It doesn't work. GOOD NEWS: Problem Fixed! i wrote a patch for dsp.c and chan_zap.c now both callprogress and answer problem work fine together. i also add a config option in zapata.conf to tune callprogress now it works with over 95 percent accuracy. Great! Mind posting your patch on http://bugs.digium.com ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing call status/progress between protocols
We have a * box with sip in, and h.323 out. When the H.323 call setup is underway, will Asterisk translate the progress/status/result codes to SIP automatically? Ordo we have create our own result codes in SIP headers? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] upgrading from A101 to....A102
Any benefit on getting the PCI Express version? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Possible to light up a LED on Snom phones?
Hi everybody! I've setup my dialplan so that if an extension dials *21*, that extension is added/removed as a queue member to a queue. (State toggled). But it would be great to get an optical feedback of that phone's state regarding the queue membership. Does someone know if it is possible to light up a LED under this szenario? Many thanks! Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible to light up a LED on Snom phones?
On Thursday 22 February 2007 22:24, Norbert Zawodsky wrote: Hi everybody! I've setup my dialplan so that if an extension dials *21*, that extension is added/removed as a queue member to a queue. (State toggled). But it would be great to get an optical feedback of that phone's state regarding the queue membership. Does someone know if it is possible to light up a LED under this szenario? AFAIR if you use BRIstuff it is possible with the devstate application. If you want an example I might be able to dig it up. HTH -- Sune Kloppenborg Jeppesen (Jaervosz) pgpXtt2rddNxB.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What means: Request to schedule in the past?!?!
On 2/22/07, Derek Whitten [EMAIL PROTECTED] wrote: My asterisk is show me some errors on line registration. This message appear on console: Request to schedule in the past?!?! check the date on the machine? Also check if you are running a NTP client, either ntpd or a periodic call to ntpdate from cron. It's not uncommong of for system clocks to drift over time, and a run of ntpdate can cause your clock to jump a considerable distance, causing * (and other programs) to get confused. Running ntpd should mitigate this as it tries to move the clock in a series of small steps instead of one big kick, but many OSs use ntpd startup scripts that first call ntpdate, so a 'service ntpd restart' or the equivalent for your OS could also cause this. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Asterisk use SIP info command
22 feb 2007 kl. 19.34 skrev Philipp Kempgen: Yuan LIU wrote: From: Olle E Johansson [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 10:36:45 +0100 21 feb 2007 kl. 21.58 skrev Yuan LIU: What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. None. What do you want to do with SIP INFO? /O I was watching the send variable thread and thought INFO would be a handy tool for that in the middle of a session. SIP headers are only sent along with INVITE, it seems. And if the situation requires the recepient to send back a value, INVITE would be impossible. I thought it might be useful to be able to ask Asterisk for the current SIP CSeq through the Manager API in order to send your own SIP messages during a call outside of Asterisk (for AOC, whatever). Each time you ask for the CSeq Asterisk should increment the value so it does not get out of sync. Anyone sharing my opinion? We might open a feature request. We're trying to keep the Asterisk architecture multiprotocol and do things in a uniform way from the dialplan. Things like this would certainly break that, since it is very SIP- specific. Better to implement needed functionality in Asterisk. And, you need much more than the Cseq and you also assume this is not NAT and not secure. We do support sending MESSAGE with sendtext() during a session. How would a multiprotocol interface from the dialplan to something like INFO look like? /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible to light up a LED on Snom phones?
On 2/22/07, Norbert Zawodsky [EMAIL PROTECTED] wrote: Does someone know if it is possible to light up a LED under this szenario? 1.2 or 1.4? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk with TCP transport
How does one configure asterisk for using TCP transport for SIP and not UDP? Thanks, Jerry --- SIP has the following features: ·Lightweight, in that SIP has only six methods, reducing complexity. ·Transport-independent, because SIP can be used with UDP, TCP, ATM so on. · ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with T.38
22 feb 2007 kl. 21.02 skrev Bill Gibbs: Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. As I've stated a few times, T.38 passthrough is broken in 1.4.0. Either use 1.4 from subversion or wait for 1.4.1. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible to light up a LED on Snom phones?
22 feb 2007 kl. 22.30 skrev Sune Kloppenborg Jeppesen: On Thursday 22 February 2007 22:24, Norbert Zawodsky wrote: Hi everybody! I've setup my dialplan so that if an extension dials *21*, that extension is added/removed as a queue member to a queue. (State toggled). But it would be great to get an optical feedback of that phone's state regarding the queue membership. Does someone know if it is possible to light up a LED under this szenario? AFAIR if you use BRIstuff it is possible with the devstate application. If you want an example I might be able to dig it up. And we have a new dialplan function in Asterisk trunk for this too, so it will be a standard feature in the next release of Asterisk, after 1.4 /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
