[asterisk-users] Re: registration not timing out?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... CLI sip show registry HostUsername Refresh State iinettrunk:5060 [EMAIL PROTECTED] 3584 Request Sent sip.pennytel.com:5060 N 280 Registered Yes, I have same problem. Have you find the solution? -- Tomislav Parcina [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
[asterisk-users] Re: Billing pulses
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You then ask the telco to include Advice of Charge (AOC) in your ISDN setup. The AOC then is included somewhere in the Asterisk CDR, but I don't have direct experience of this. You can then get appropriate software to issue bills to telephone users. Unfortunately, as far as I know, Asterisk can't store AOC messages in database. So, provider sends perfectly usable messages, and Asterisk detects them (they are shown on CLI) but it can't store them anywhere. Said. -- Tomislav Parcina [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
[asterisk-users] Re: Pickup() ringing extension and call waiting
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
What do you mean by mapping the 200 ?
In this example I can pickup any ringing extension:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
If phone with number 42 rings you can catch the call by dialing 742. You
don't need to use the context
exten = _7.,1,Pickup(${EXTEN:1}) works for all contexts.
Regarding call waiting, internally when I'm having a conversation and someone
calls me, then my second line button blinks and I can pickup a second call
putting first one on hold. Problem just with real call waiting from PSTN.
Hi Dominik!
Information's on that page are wrong. Read this:
pbx*CLI show application Pickup
pbx*CLI
-= Info about application 'Pickup' =-
[Synopsis]
Directed Call Pickup
[Description]
Pickup([EMAIL PROTECTED]): This application can pickup any ringing channel
that is calling the specified extension. If no context is specified, the current
context will be used.
So, if application Pickup isn't in same context with Dial which you are trying
to pickup, then you have to specify context.
Hope this helps.
--
Tomislav Parcina
[EMAIL PROTECTED]
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[asterisk-users] Re: Comments on Billing reconcillation with providers
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I just want out find out how to do bill recon's when you send calls to a provider. They send me their CDR's, and when I compare it to my * CDR's, some calls are 1 second off, either way. How in general is it done by others? Most providers send advice of charge messages (AOC). Unfortunately, asterisk can't store them in database or manipulate with them at any way. -- Tomislav Parcina [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
[asterisk-users] Re: Asterisk Faxing Support
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Asterisk 1.2 has no support of t.38 whatsoever, the call will drop before t.38 is ever utilised, not even pass-thru. 1.4 Adds support for T.38 pass through only and no other sort of faxing, the endpoint must support T.38 and you must send your call to a T.38 gateway and you must not use NAT anywhere in your network and you must enable re-invites which could cause CDRs not to reflect the true details of the call. Asterisk/Digium also has no interest in any further interest in expanding T.38 or faxing support in Asterisk. Steve Underwood and the other fine persons that have helped to develop the software DSPs and other stuff required for FoIP support also have no interest in writing any further faxing support for Asterisk (RxFax, TxFax + the newest span_dsp wont even compile, much less work under Asterisk any more) probably because they know it will never be included into the Asterisk code. Someone please tell me this isn't truth. -- Tomislav Parcina [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
[asterisk-users] Digium cards on Vmware
Is it possible to use Digium (or Sagnoma, or Beronet) cards with Asterisk on Vmware? Has anyone done it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Pickup
On one installation (* 1.2.13) Pickup doesn't work. This is what I have in
extensions.conf
exten = _**2X,1,Pickup(${EXTEN:2}8${EXTEN:3}tuevents)
exten = _**2X,n,Hangup
This is what I get on CLI
-- Executing NoOp(mISDN/3-1, incoming-beronet 80 - dolazni poziv s broja
270248) in new stack
-- Executing LookupCIDName(mISDN/3-1, ) in new stack
-- Executing Dial(mISDN/3-1, SIP/20|30|t) in new stack
-- Called 20
-- SIP/20-08cdad80 is ringing
Extension Changed 20 new state Ringing for Notify User 27
Extension Changed 20 new state Ringing for Notify User 21
Extension Changed 20 new state Ringing for Notify User 28
-- Incoming call: Got SIP response 415 Unacceptable Content-Type back from
192.168.2.107
Extension Changed 27 new state InUse for Notify User 21
Extension Changed 27 new state InUse for Notify User 20
Extension Changed 27 new state InUse for Notify User 28
-- Executing Pickup(SIP/27-b65a1100, 2080tuevents) in new stack
== Spawn extension (sip2, **20, 1) exited non-zero on 'SIP/27-b65a1100'
Extension Changed 27 new state Idle for Notify User 21
Extension Changed 27 new state Idle for Notify User 20
Extension Changed 27 new state Idle for Notify User 28
Why do I get == Spawn extension (sip2, **20, 1) exited non-zero on
'SIP/27-b65a1100'
I have to pickup either 2X, 8X, t or uevents extension (phone will ring on any
of those).
Have I done something wrong?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Re: Mabe OT? What managed switch is best for VoIP application?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I worked with Cisco and HP and they should do what you are looking for. I even worked with cheap unmanaged switches ~20 Euro and they work with VoIP. Do you know for switch that can tell me that on port 7 there are two active SIP calls. One of them goes to x.x.x.x IP address and another to sip.mydomain.com. First lasts for 34 and another 51 seconds. -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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: Cordless SIP Phones
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Siemens Gigaset IP phones (C450-IP, S450-IP) are not that bad (gigaset.siemens.com). C450IP costs less than 100 USD (in Italy at least), S450 is slightly more expensive. I have Siemens C450 IP for two days and it seams weary good. I'm looking for S450 IP, but I can't buy it in Croatia :( -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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] mISDN
Hi list! How to make outgoing call thru other mISDN channel group of all channels on first group are busy? I believe I'll need to - Check of there is free channel on group1 - if there is free channel call thru group1 - if there are no free channels call thru group2 -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr winmail.dat___ --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: mISDN
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Iirc you can put more than 1 interface in a group and it should just use any free channel of whichever interface that has a free channel. Check the sample config. Hi Patrick! Yes, I know that and I'm using that. But then I need to change my CID number, because I can't use same numbers on both ports. -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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] CDR - uniqueid
Is uniqueid globally unique? I have three Asterisk installations and I need to store data from all of them in same database, in same table. Will this uniqueid field be unique? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr winmail.dat___ --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: Disconnected Calls
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I upgraded to the newest 1.2 Zaptel release and this is still occurring. I checked and the digium card is not sharing an IRQ with any other devices. I also changed busycount=8, and set callprogress=no. The call drops are still occurring. Mid-conversation ` in 10 calls will be disconnected. Any other suggestions? This is a relatively low volume system. Usually running less than 1 or 2 concurrent calls. Would turning on debugging logs to a file cause a problem? Many thanks, Ejay Hire Hi Ejay! Why have you excluded possibility that the problem is on telco side? -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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: How to exit from console?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... E.g: because you have a valid PID file of the controlling process. If you actually want to kill it, you can. And you don't need physical access to the system to get to the one and only real console. OTOH, if you do have physical access, you have full control of Asterisk, as you may inject custom dialplan. And if, for some reason Asterisk dies, you have to start it manually? -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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.conf
Why there is no asterisk.conf.sample file? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr winmail.dat___ --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: How to exit from console?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Try safe_asterisk , for an easy way to start asterisk in background, a plain 'asterisk' is even better and safer. asterisk -U asterisk . is better. /etc/init.d/asterisk start is similar. Why is this better than safe_asterisk? -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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: AOC on misdn?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, i can see AOC messages on the asterisk console. Can i sendtext() them to the caller or put them in cdr? Regards, Andreas. I'm also interested in this. If you find solution, please mail it to the list. -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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: Answer a call that is not ringing on yourextension
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Another solution is to use the Pickup() command. It will pick up a call on a specific extension that is in the ringing state: [Description] Pickup([EMAIL PROTECTED]): This application can pickup any ringing channel that is calling the specified extension. If no context is specified, the current context will be used. For example, my co-workers extension is 203. I hear his phone ringing, and I dial my pre-defined pickup extension (**203) to pickup his call. Dialplan example: Exten = **203,1,Pickup(203) Exten = **203,2,Hangup() Note: the read I use ** is GXP-2000 phones will dial **exten while the BLF light is in ringing state. You can use whatever you want. Wes Baehr Wes, thank you for this information! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: centos 4.4 + asterisk
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... That kernel-devel fix is just for ZAPTEL. The bug has been solved in 4.4 To make it more understandable - Cent OS 4.4 doesn't have problems with Zaptel installation. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: SetCallingPres propagation
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello, We have several regional asterisk's connected to a central one making the the PRI calls through a TE410P card. When using SetCallingPres(prohibited) on a call at the regional level, that setting it not forwarded to the central asterisk and the call is made as if no callerid had been sent: the telco substitutes the network number. Using SetCallingPres(prohibited) on the central asterisk works though: the call is received with no callerid at all. How can I suppress callerid presentation at the regional level and keep that setting when trunking the call from regional to central asterisk's? Hi Louis! You shouldn't prohibit CallerID on regional level. Instead you should send *79 (or something else) to central Asterisk. When central Asterisk receives *79SOME_EXTEN he should cut of *79 and execute SetCallingPres(prohibited). Hope this helps. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Switching from FreeBSD to Linux - which distro?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs great on it. Debian is good too. They have Asterisk packages, but they're generally a little bit old. Source installations work fine. Both have large, active developer and user communities. Hi Carla! Can you tell me from where do you download rpm's for Cent OS 4? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Distinctive ring
Hi list!
I need help with distinctive ring on Cisco 7940 phone. I'm using Asterisk 1.2.5
(I know, I should upgrade) and in dial plan I have:
exten = _64X,n,Set(_ALERT_INFO=Chirp2)
exten = _64X,n,Dial(SIP/${EXTEN},30,wWtT)
On Cisco in Settings = Ring type I have Chirp1 and Chirp2. By default
phone is ringing sound Chirp1. For internal calls I'm using dial plan I have
sent you above. Problem is that Cisco doesn't ring with Chirp2, but with
slightly different Chirp1 (instead of ring, pause, ring he sounds ring ring
pause).
Is there any way that my Cisco 7940, thru dial plan, can ring Chirp2 instead of
Chirp1?
Thank you for your time!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Re: Cisco 7940 Firmware 8.2
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Greetings, I am cutting my teeth with SIP phones and my first issue is getting a Cisco 7940 to Authenticate with my VoIP provider (BBTelsys). I did read some notes on the vo-ip website about 7.5 being the better firmware version. Has anyone had trouble with 8.2 and SIP registering? Should I just downgrade to 7.5 and give it a go? I think SIP uses UDP 5060 correct? The phone is behind a firewall(NAT) I figure this might be an issue as well. Thoughts? Thank you for your response. I'm using 7.4 firmware. I didn't noticed any problems. I'm not familiar that and further firmware brings anything that will make me change firmware. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: unable to get channel lock BAD BAD BAD
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Deadlocks are not a config or Trixbox issue. I'm confirming this one! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: T.38 - make conclusion
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I want to guess that it's your SIP provider. Faxing via VoIP (SIP) is not reliable unless you are using T38, my guess is your sip provider is providing this feature. Hi Doug! Have I understand it right. You are saying that my provider, when it detects FAX stream, he is trying to use T.38. And since my Asterisk doesn't support it I get the error message? So, it does nothing to do with FAX machine that is (over ATA) plugged to my Asterisk, or with FAX that is on the other side? Please confirm or deny this. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] T.38 - make conclusion
This is one long letter about T.38 and Asterisk. I hope it will help me, and lots of other Asterisk users to understand some T.38 problems with Asterisk. This is my situation: I have Panasonic DX600 FAX machine. It's connected to Asterisk 1.2.13 thru ATA adapter (I have used both, Cisco 186 and Grandstream HandyTone 386). Asterisk is connected with my SIP provider. That link that my provider gives me is dedicated for VoIP. QOS is up and it doesn't go thru Internet, it goes thru dedicated line. That Panasonic DX600 can receive FAX from everybody, but it can send FAX to 70% of numbers. The numbers to who I can send FAX works all the time (none fax has failed). The numbers I can't send fax newer work (all fax fails). Every time I try to send fax to some number that fails, I get following warning: WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 As far as I understand, T.38 is VoIP protocol, so Panasonic DX600 on its analog port shouldn't use T.38. My HT386 is in pass thru mode, so he shouldn't use T.38 also. Cisco 186 ATA doesn't support T.38. Big question is, who tries to establish T.38 connection? - Panasonic DX600 - because it's connected on analog port, I believe he doesn't - ATA - Cisco doesn't support it and on HT386 it's turned off - Asterisk - doesn't support it So who tries to establish T.38 connection? I'm not planning to use T.38. I'm only trying to make FAX work. To do that I need to find out who is causing the problem that is giving me this warning WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 I'll appreciate any info on this topic. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] T38 problem
I have problem with fax machine Panasonic DX600. It's connected to Grandstream Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP provider. To some numbers I can't send FAX, and I get following error on CLI. WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 I believe that Panasonic DX600 machine supports T38. And when I have another T38 fax machine on other end they try to send FAX using T38 protocol. And than I believe I get above error and sending FAX fails. Is there any way to solve this? I hear that there is T38 support in Asterisk 1.4, but I can't wait for version 1.4. In manual for Panasonic DX600 I didn't find any instructions how to turn T38 off. Please suggest something. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: unable to get channel lock BAD BAD BAD
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I am seeing the following in my log file (standard trixbox install). One seems to be complaining about an error in the dialplan but it won't tell me what file or what line. The other (maybe related) is complaining about a channel lock. How to do go about trying to figure out what the problem is and how to solve it? Nov 14 07:20:44 ERROR[24091] chan_sip.c: BAD! BAD! BAD! Yes, I get same error message in my log. Anybody has any info on this one? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: ATA with reliable FAX?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... (a) If you are not running a version of Asterisk that has working SIP jitter buffering (is there such a thing?), then abandon all hope now. (b) We have no experience with the Cisco ATAs, but the Linksys (nee Sipura) SPA-210x is markedly better than the SPA-100x and SPA-200x, probably because they have better jitter buffering (it goes without saying we do not pass our fax traffic through Asterisk). (c) T.38 is the way to go, G.711 a poor and distant second choice (again, Asterisk's T.38 pass-through is far from ready for prime time). Hi George! You said that T.38 is the way to go. I have problems with T.38 and I don't know how to solve them. Maybe you can help me. I often get this message on CLI: Nov 15 14:56:03 WARNING[2237]: chan_sip.c:3602 process_sdp: Unknown SDP media ty pe in offer: image 31512 udptl t38 What could be the reason and how to solve it? I have Fax machine - Grandstream Handy Tone 386 - Asterisk - my SIP provider Thank you for your time! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: unable to get channel lock BAD BAD BAD
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Are you using trixbox? It would be nice to try and isolate this problem by ruling out a bad config in trixbox. No, I'm running Asterisk. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: T38 problem
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi Tomislav, It sounds to me that you have t.38 enabled on your Grandstream Handytone 386. You should disable this on the Handytone. I have a handytone 286 which has an option to disable t.38 and use fax passthrough. This should get rid of your t.38 messages on the cli. Hi Craig! Thank you for your mail. Yes I thought the same. Unfortunately I'm not on the location where ATA is, and I have told one guy on that location to check that option. He checked, few times, and he's positive that option pass-through is on (and T.38 is off). -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: State of a public number
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi guys, i would check the state of a number on a Zap channel, i suppose that i cannot use ExtensionState that works only for SIP and IAX. Anyone has any ides ? Could i check the state of a pubblic number before transfer it a internal call? Thanks in advance Hi Giordano! If I have understand you correctly, you can use ChanIsAvail(Zap/4Zap/3) - it will return Zap/4 or Zap/3, the first one that is available. Hope this helps. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Cisco 7960 - Fast dial
Cisco 7960 has six buttons/lines. Can some of them be configured for fast dialing? If it can't be configured on the phone, how can I configure it on Asterisk? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Can some moderator kick this person out of the list
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I myself and I am sure hundreds of other users on this mailing list are getting very much annoyed on receiving follwoing autogenerted message several times a day from [EMAIL PROTECTED] Is there any moderator on the list who can take care of this. It comes replied to every post and almost to every answer to it. I definitely support your request! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: VM Language
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... What is the best way to have the voicemail system and system do more than one language I know I have to have all wav, gsm files on the system. Look for CID and than change language in dial plan. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Grandstream HandyTone-488 with Asterisk ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It's possible, but the FXO port on that device is more of a power out fail over to PSTN in my opinion. The FXO can be made to work, but it always had issues with my setup. 1) Echo problems. I have a long loop. 2) If calls are picked up on the first ring, they go into la la land. 3) General reliability and stability issues. I confirm all what Martin has said. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 Call Statistics
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... of course you can always use http://cacti.net/download_cacti.php Hi Moisies! I heard that Munin can (or they are working on that) log how many simultaneous call on each interface Asterisk has. Can Cacti do the same? I have tried Cacti once and I liked it weary much. It's easy to configure and has nice interface. I definitely need to install it again! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 Call Statistics
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... http://www.areski.net/asterisk-stat-v2/about.php Hi Doug! I don't recommend anybody using Asterisk stat. Last version is V2.0.1 (07 March 2005). It's obsolete. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: IAX2 show peers - description
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I think the (T) is for Trunk. Regards Fred Hi Fred! I believe that T is for trunk. Thank you. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Forwarding recorded calls to Voicemail
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I was wondering if anyone has implemented a feature that would allow a user to record a phone call and once the call has ended, the call is forwarded to his voicemail? Hi Tom! I was looking for something like this, but I was unable to find anything useful. Hopefully someone will answer your mail. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Intel S3000AHLX - Digium TE110P
Does anybody use Intel S3000AHLX board with Digium TE110P E1 card? Have you experienced any problems? I'm planning following configuration, so I would appreciate any experience both positive and negative. Best regards, -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: IAX2 show peers - description
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi people, pls does anybody know what (T) and (D) letter means? server3*CLI iax2 show peers Name/UsernameHost Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0 unmonitored] Hi Marian, Near host you can have D (dynamic) or S (static). Near port you can have T, but I don't know what it means. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For my home Asterisk setup I have a single PSTN line, and then I use a variety of different voip providers. I use two different providers for my DID's (one toll free, and one normal). I use yet a different provider for terminating outgoing calls. So, when making an outgoing call via voip, what number should I use to identify myself? I currently use the number of my PSTN line, since that is our public inbound number. Hi John! I have same situation, and I certainly agree about everything you said. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Cisco 7971G-GE SEP{MAC}.cnf.xml
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Would anyone happen to have a working configuration for the 7971G-GE (running SIP70.8-0-4SR1S) they would care to share, or allow me to purchase. Hi Kelvin! I have Cisco 7970 and firmware SIP70.8-0-2SR1S and I use Another SEPmac.xml.cnf example from this page http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP I suggest you to try the same firmware and same conf file. Best regards, -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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.conf - srvlookup
I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues (Asterisk stops responding). I use VoIP Buster and in sip.conf I use sip1.voipbuster.com. When I do sip show peers in CLI I get voipbuster/tomo 194.221.62.207 5060 OK (27 ms) And when I ping sip1.voipbuster.com [EMAIL PROTECTED] ~]# ping sip1.voipbuster.com PING sip1.voipbuster.com (194.221.62.206) 56(84) bytes of data. So, Asterisk is registered at 194.221.62.207 and DNS lookup gives me 194.221.62.206 IP address. Question is, which IP address should I use, 206 or 207? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: echotraining=yes in misdn.conf is invalid or out of range.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi Jarkko, I had the same problem..It worked with an old version of misdn-install (taken from beronet site) but not with actual mqueue-misdn-install. I tried to put it in every misdn.conf section I have without success. The updated beronet install manual doesn't mention that parameter anymore so I removed it from misdn.conf. I have also BeroNet card but I'm unable to start Asterisk with chan_misdn. This is the error that I get on CLI. Oct 18 15:10:21 ERROR[5860] chan_misdn.c: Unable to initialize mISDN Oct 18 15:10:21 WARNING[5860] loader.c: chan_misdn.so: load_module failed, returning -1 Oct 18 15:10:21 VERBOSE[5860] chan_misdn.c: -- Unregistering mISDN Channel Driver -- Oct 18 15:10:21 WARNING[5860] loader.c: Loading module chan_misdn.so failed! And I have started misdn-init start [EMAIL PROTECTED] ~]# /etc/init.d/misdn-init start which: no lsusb in (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/lo cal/bin:/sbin:/bin:/usr/sbin:/usr/bin:/usr/X11R6/bin:/root/bin) [!!] FATAL: lsusb not in path, please install. - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type=0x4 protocol=0x2,0x2,0x2,0x2 layermask=0 xf,0xf,0xf,0xf poll=128 debug=0 modprobe mISDN_dsp debug=0x0 options=0 poll=128 dtmftreshold=100 [i] creating device node: /dev/mISDN And I believe I have all modules loaded: [EMAIL PROTECTED] ~]# lsmod Module Size Used by mISDN_dsp 202764 0 mISDN_capi103180 0 l3udss145020 0 mISDN_l2 41812 0 mISDN_l1 12732 0 capi 18049 0 capifs 5961 2 capi kernelcapi 46689 2 mISDN_capi,capi md5 4033 1 ipv6 266433 10 parport_pc 28805 0 lp 13001 0 parport39689 2 parport_pc,lp autofs427333 2 rfcomm 42589 0 l2cap 30021 5 rfcomm bluetooth 55109 4 rfcomm,l2cap sunrpc162821 1 ztdummy 3924 0 wcusb 19488 0 wctdm 35392 0 wcfxo 13216 0 wctdm24xxp120384 0 wcte11xp 36384 0 wct1xxp2 0 wct4xxp 312000 0 tor2 92704 0 zaptel206468 9 ztdummy,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wc t1xxp,wct4xxp,tor2 crc_ccitt 2113 1 zaptel video 15941 0 button 6609 0 battery 9413 0 ac 4805 0 ohci1394 39817 0 ieee1394 304057 1 ohci1394 uhci_hcd 34897 0 ehci_hcd 39757 0 shpchp 91205 0 i2c_viapro 8145 0 i2c_core 21825 1 i2c_viapro hfcmulti 79144 0 mISDN_core 79840 6 mISDN_dsp,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1,h fcmulti via_rhine 27465 0 mii 5569 1 via_rhine dm_snapshot17669 0 dm_zero 2113 0 dm_mirror 25261 0 ext3 132297 2 jbd79449 1 ext3 dm_mod 58997 6 dm_snapshot,dm_zero,dm_mirror Do you know what could be the problem? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Cisco 7970 - versionStamp
If I put versionStamp in cnf.xml file, how do I check it on the phone? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Digium on Dell PowerEdge 1850
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... We're running 2 TE412P's in a Dell 1850 just fine, been running like this for well around 6 months to a year now without any problems. They're not exactly 212P's but I imagine it won't be much different. On Wed, 2006-10-18 at 10:54 +0200, Tomislav ParÄŤina wrote: Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Hi Aaron! Thank you for your mail. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Digium on Dell PowerEdge 1850
Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Dial - i parametar
There was patch for 1.0.x version of Asterisk that is quite useful. Is there patch for 1.2.x version and will this i parameter be in 1.4.x version of Asterisk? Have a nice day! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: unauthenticated calls
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... is it possible on asterisk to reject unauthenticated calls or not registered phones to call? You can send them to [default] context that has only extensions like this: exten = i,1,Hangup exten = s,1,Hangup -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Cisco 7970 SIP won't update?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. You setup versionStamp in cnf.xml file, but how do you check it on the phone? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Call Asterisk : It calls me backup with a dial tone
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Look into DeadAGI, should be easy enough that illl implement tomorow ;) If you implement it, please send us your configuration. Best regards, -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Beronet BN4S0 instalation
I'm having trouble installing Beronet BN4S0 card. I have downloaded instructions from here http://www.beronet.com/download/card_installation_guide.pdf And when I download install-misdn-mqueue[1].tar.gz I untar it and execute make and make install. This is the output that I get. [EMAIL PROTECTED] install-misdn-mqueue]# make install make -C app_bundle make[1]: Entering directory `/usr/src/install-misdn-mqueue/app_bundle' make[1]: Nothing to be done for `all'. make[1]: Leaving directory `/usr/src/install-misdn-mqueue/app_bundle' cd mqueue-misdn; cvs -d:pserver:anonymous:[EMAIL PROTECTED]:/i4ldev co mISDN mISDNuser ; cvs [checkout aborted]: connect to cvs.isdn4linux.de(217.160.76.191):2401 failed : No route to host make: *** [misdn] Error 1 But I'm able to connect to cvs.isdn4linux.de ftp server [EMAIL PROTECTED] install-misdn-mqueue]# ftp cvs.isdn4linux.de Connected to listserv.isdn4linux.de. 220-*** 220-Welcome to ftp.isdn4linux.de 220- 220-All transfers are logged with your username and hostname (as printed in 220-the first line of this message). If you don't like this policy, 220-disconnect NOW! 220- 220-If you have problems accessing this server, please drop a note to [EMAIL PROTECTED] 220-*** 220 530 Please login with USER and PASS. 530 Please login with USER and PASS. KERBEROS_V4 rejected as an authentication type Name (cvs.isdn4linux.de:root): anonymous 331 Please specify the password. Password: 230 Login successful. Remote system type is UNIX. Using binary mode to transfer files. ftp Can someone please help me on this? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: CDR stats to one mysql database, multiple webstats packages
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Have a number of asterisk servers and want to get some good stats tracking going (with Asterisk-Stat) -- but this requires cdr logging to mysql, apache and the stats software running on each server. Or does it? Of course, I can either run the stats package on the webserver and direct it to each individual server's local mysql db --- or have each asterisk server logging to an external mysql db somewhere.(on the webserver I suppose) Thoughts on this? Good idea/Bad idea to log to an external source? One thing that might be an issue is if for some reason the external source becomes unreachable or goes offline ...then what happens to the CDR data for that time period? Suggestions appreciated Hi Chris! I have three Asterisk and every one of them is logging CDR's to MSSQL database that is on same location (same room) as Asterisk. So, there is only switch between them. Two of three MSSQL servers are doing log shipping on third MSSQL server on new database. That way every * logs to database which is close to him - should be stable enough. Because of secure log shipping I have all data from every Asterisk in one database. I calculate everything from that one database. Hope this helps. P.S. If any MSSQL fails, then I import data from Master.scv -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Voip Buster - CID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I use SellVOIP and Voxee which both seem to allow that. Hi Ira. Thank you for this information! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Voip Buster - CID
You can try VoipJet (http://www.voipjet.com)
A simple configuration in you extensions.conf as below will solve your
problem.
exten = _X.,1,SetCIDNum(1341212)
exten = _X.,n,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Thank you!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Re: RPID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Tomislav, RPID is short for Remote-Party-ID. Basically, Remote-Party-ID is a way, using a header (Remote-Party-ID) to completely separate caller id presentation from authentication information with SIP. I should point out that in standards tracks, Remote-Party-ID has been replaced by PAI (P-Asserted-Identity). Gotta love those standards :). Hi Kristian! Thank you for informations. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: max number of devices in hint
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm glad you asked :-) If we had Shared Line Appearances, I would not have to do this. However, I could be at any of about 6 different phones, and on any of about 4 lines per phone. Therefore, to monitor whether or not I am on the phone would take a 24 BLF buttons or just one, if hinting allowed that many. How many hands/ears you have? ;)) -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Voip Buster - CID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... There are not many that will allow you to set your own CID even then they normally ask for proof of the numbers you wish to use. Hi Chris! So, you are saying that I can't set outgoing CID number on Voip Buster? Do you know for any VoIP provider that allows that? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: ASTTAPI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI. Hi Mike! I have been using ASTTAPI, but it takes time to configure it and I'm not sure it's developing any more. Now I'm using SNAP for several days but it seams that it has some bugs. I'm using Snap's forum to check with developer about this, but it's going slowly. I don't think that Snap is for business production yet. If developer doesn't solve those problems with Snap, I'll try Etelescript. Is Etelescript free? Is it open source? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: RPID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Has anyone successfully gotten rpid working between two phones through asterisk? Hi Aaron! Can you please tell me what is RPID? Wikipedia and Google - define: RPID didn't help me. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 = E1 = Alcatel OXO
Hi list! Has anyone of you connected Asterisk (Digium TE205) with Alcatel OXO thru E1 lines? I need to configure Asterisk so that every call from Alcatel OXO passes thru it. Asterisk will be between my provider (T-com in Croatia) and Alcatel. Thing is that, probably next week, I'll go on site to install Asterisk. And I need to prepare as best I can to make it work. And as far as I'm concern, best preparation would be working configuration. So, if anyone of you has done it, please send me your zapata.conf, zaptel.conf and extensions.conf files. Thank you. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 Buster - CID
Hi List! Is there any way to set outgoing CID number when making VoIP calls using VoIP Buster? I have search on their forum and I couldn't find anything useful. There is no support mail on their web pages :(( P.S. I use them because they are cheep and sound quality is satisfying -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: max number of devices in hint
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have one extension that rings in many places. It has just come to my attention that I can only monitor 4 devices within a hint. Ex: exten = 132,hint,SIP/DEVASIP/DEVBSIP/DEVCSIP/DEVD if I add SIP/DEVF, DEVF is not monitored. I'm interested, why do you monitor multiple devices within a hint? If one device is in use (and three are free), how does it show - in use or as free? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Advice of charge
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... No. I once tried to create a channel variable during hangup. Then, in the hangup extension this variable was added to the user defined CDR field. This generally works, but only if the call leg hangs up, on which the AOC is received. In other cases (e.g. sip to zap calls) when the SIP user hangs up, I had to fetch the last AOC-D value from the bridged channel, which does not work well. There should be a generic method in Asterisk for storing/retrieving AOC, thus I stoped my work. Hi Klaus! Have you provide those information's to developers? Is there any interest to make this work? Approximately, in your opinion, how much work there has to be done? P.S. There are few programmers in company I work for. Can you please send me all relevant code and maybe I can persuade them to look at it. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Dual core
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Well, it would seem to me that with a little attention to processor affinity, you could run your Asterisk and DBMS code on one processor, and let the other one handle the device interrupts; ie: that sounds to me like a feature, rather than a bug... Ok, and if I have two dual core processors? It doesn't sound like very useful feature to me (in some situations). -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Dual core
For what we do with Asterisk(lots of meetme and Zap - IAX2) It does spread the load across both cores. In our initial comparisons for equal call traffic, the P4-D had half or the average loadavg for a 6 hour time period of the P4 of the same speed. MATT--- Hi Matt! Thank you for information's. Can you please tell me have you made any special adjustments or steps in Asterisk install or configuration to achieve this? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 cell phone network
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 1/2/4 simslot pci card: http://www.junghanns.net/en/GSM-PCI_produkt.html If they are as stable as the quad/octo BRI cards they have it's a real winner. Where can I see the prices of this cards? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Dual core
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have a few dual core that I have installed Asterisk on without any issues. Hi Bill! Sure you don't have any issues, but do you take any advantage of dual core processor? Why would I pay for something if I can't profit from it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Dual core
Asterisk is very happy on dual core. It greatly reduces load. We just put a Pentium-D in poduction last week and it is working verry well. We have a Core 2 Duo on order that we should be putting in production next week. MATT--- Hi Matt! Thank you for this information. Can you please tell me if you weight Asterisk, does it divide that job on both processors or it's only one that does the job? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Dual core
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... My home Asterisk server is running dual proc dual core zeon 3ghz, seems happy, no crashes that I didn't bring about myself. ;) mpg123 does occasionally hang a pid at 100% now and then, but it does that on single proc/single core systems too. Hi Nick! You should use native MOH. Than you won't have that problems with mpg123. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Very high ping times from 7960 phones
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have two 7960 phones with 7.4 firmware and sip show peers tells me that response time is 70 and 72 ms. Hope this helps. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Cisco 7970 - DTMF
In sip.conf for one friend (Cisco 7970 phone) I have define this dtmfmode=inband And in xml.conf of that phone I have preferredCodecnone/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandnone/dtmfOutofBand But DTMF doesn't work for that phone. Phone establishes call using g711 alaw codec. How should I configure phone and sip.conf to make DTMF work? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Cisco 7970 - DTMF
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In asterisk sip.conf, use dtmfmode=rfc2833 for that extension, and in the SIPDefault.cnf boot file for the cisco, include: dtmf_inband: 1 dtmf_outofband: avt dtmf_db_level: 3 (you'll need to translate the above 7960 parameters into the 7970 xml parameters since I don't have a 7970 to play with.) Taking a wild-ass guess, you might be able to get by simply using the dtmfmode=rfc2833 parameter in asterisk without touching the phone. Try it. Hi Rich! dtmfmode=rfc2833 in sip.conf with dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandavt/dtmfOutofBand In sepmac.cnf.xml works well. Thank you! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Can you explain why multiple registration isan important (missing) feature ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Perhaps you are tying to use wildcard destinations in your setup. This
does not scale.
Wildcard:
exten = 1234567,1,Dial(SIP/${EXTEN})
This does not scale.
Each extension should have it's own exten = line and Dial(... line.
exten = 1234567,1,Dial(SIP/[0004f201e443-a) because 0004f201e443-a is
the userid of the phone that you want to send the call to.
As far as I'm concern that isn't acceptable. I would newer make such
configuration. Imagine 1000 extensions and for every one of them you have to
create line like above in extensons.conf.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Dual core
Hi list. I have one quick question. Does Asterisk work with dual core processors in version 1.2? Will it work with dual core processors in 1.4? I'm planning to buy new machine for one installation and I have to decide will I buy single or dual core processor. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] E1 - PCI-Express
Is there any (I prefer one port, but I could also buy two port) E1 PCI-Express card? As far as I can see, all Digim cards are PCI. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Can you explain why multiple registration isan important (missing) feature ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Your definition in the sip.conf would be defining devices according to their MAC addresses. Your dial plan would call these devices based on extensions. exten = 100,1,Dial(SIP/MAC) ; where MAC is the MAC address of the phone All right. Then I give to my girlfriend my number 1234567 and she calls me in, how will I know to which MAC address I need to pass call? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Playtones
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It looked promising so I tried it. Unfortunately it didn't help. Calling person doesn't hear ringing. I don't know why this application didn't work as it should. I have tried with and without wait command. -- Executing Playback(SIP/198-d5e2, lama/dobro-jutro|skip) in new stack -- Playing 'lama/dobro-jutro' (language 'hr') -- Executing Goto(SIP/198-d5e2, s|11) in new stack -- Goto (aahrvatski,s,11) -- Executing BackGround(SIP/198-d5e2, lama/odjeli) in new stack -- Playing 'lama/odjeli' (language 'hr') == CDR updated on SIP/198-d5e2 -- Executing Ringing(SIP/198-d5e2, ) in new stack -- Executing Wait(SIP/198-d5e2, 5) in new stack -- Executing Goto(SIP/198-d5e2, sip_queue|148|1) in new stack -- Goto (sip_queue,148,1) -- Executing Dial(SIP/198-d5e2, SIP/148|30|wtr) in new stack -- Called 148 I have test it by calling from SIP phone to AA menu, and it doesn't work. Then I tried from ZAP interface and the phone rings. Since this AA will be for incoming calls from ZAP interface I can take this one as solved. But there is another thing. Is this not ringing on Sip interface u a bug? I'm using Asterisk 1.2.5. Can somebody check this on Asterisk 1.2.12.1? I don't want to report u BUG if it's already fixed. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: mpg123
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi all, I'm using * 1.0.9 which use mpg123 for music on hold. But sometimes starts eating up a lot of CPU. I sthere any alternative method to use moh without use mpg123? I tryied this http://astrecipes.net/?n=152 but i doesn't wotks for me. Anyone can help me pls ? Upgrade to Asterisk 1.2 and use native sounds. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: unable to change the emailbody for email notification
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi all, the default message for email notification looks like: Is there something wrong with my config? thx in advance This should work. Have you reloaded Asterisk? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Playtones
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
what about this?
show app ringing?
exten = _7XX,1,Ringing
exten = _7XX,2,Goto(local,${EXTEN},1)
It looked promising so I tried it. Unfortunately it didn't help. Calling person
doesn't hear ringing. I don't know why this application didn't work as it
should. I have tried with and without wait command.
-- Executing Playback(SIP/198-d5e2, lama/dobro-jutro|skip) in new stack
-- Playing 'lama/dobro-jutro' (language 'hr')
-- Executing Goto(SIP/198-d5e2, s|11) in new stack
-- Goto (aahrvatski,s,11)
-- Executing BackGround(SIP/198-d5e2, lama/odjeli) in new stack
-- Playing 'lama/odjeli' (language 'hr')
== CDR updated on SIP/198-d5e2
-- Executing Ringing(SIP/198-d5e2, ) in new stack
-- Executing Wait(SIP/198-d5e2, 5) in new stack
-- Executing Goto(SIP/198-d5e2, sip_queue|148|1) in new stack
-- Goto (sip_queue,148,1)
-- Executing Dial(SIP/198-d5e2, SIP/148|30|wtr) in new stack
-- Called 148
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] RE: Asterisk 1.4 Docs
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... One of the providers that I use already offers this feature via a macro in the dail plan http://connect.voicepulse.com/FlexRate.aspx Hi Jason! This is interested, although it's not related to AOC messages. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Can you explain why multiple registration is an important (missing) feature ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... And there is your problem. Using the extension as the SIP User ID does not scale, is confusing, and limits your thinking about devices and extensions. There are several reasons this is a bad idea. Multiple extension numbers ringing on the same device / line appearance is the most common. We use the MAC address of the device as the SIP User ID. We append a -a, -b, -c, etc to the MAC address for each line appearance. This does not work well for Softphone, but since All Softphones Suck(TM), we don't really care about this limitation. Users seldom need to know their SIP User ID. Can you please tell me more about this. I don't follow you weary well. I understand that we need to treat phone and users different, but I don't thing that is easy to do with Asterisk 1.2. Maybe something will change, but till then... -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 - Agent language
I have Queue with static members (without agents.conf file). When someone calls queue I can set his language, but how to set agent's language? I would like to Hold time less than two minutes to be read in Croatia (hr) language. -- Playing 'lama/najava-programeri' (language 'en') -- Playing 'queue-reporthold' (language 'en') -- Playing 'queue-less-than' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'queue-minutes' (language 'en') In queues.conf I have: member = SIP/888,1 And in sip.conf, in general section I have: language=hr But it doesn't help. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Log out an Agent on RNA
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello all, Is it possible to automatically log off an agent on RNA (Ring No Answer) when the agent is logged in with AgentCallbackLogin? By default it logs off agent. Check agents.conf and queues.conf. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Playtones
I have auto attendant menu. When calling person dials one number one extension
rings. Problem is that while extension rings caller doesn't hear ringing. I
understand that caller doesn't hear ringing because phone call is already
established, but I need to tell to caller that extension is ringing. How to
do that?
My extensions.conf
[incoming]
exten = s,1,Answer
exten = s,n,ResponseTimeout(5)
exten = s,n,Playback(mymessage,skip)
exten = s,n,Background(mymessage2)
exten = s,n,Background(silence/3)
exten = _7XX,1,Goto(local,${EXTEN},1)
[local]
exten = _7XX,1,Dial(SIP/${EXTEN},30,wtr)
exten = _7XX,n,VoiceMail,u${EXTEN}
exten = _7XX,n,Hangup
exten = _7XX,102,VoiceMail,b${EXTEN}
exten = _7XX,n,Hangup
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
___
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[asterisk-users] Re: PRI: sometimes Asterisk drop calls
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I do not use queues but I have a lot of messages like that. I talked a lot with Steve about this It seems like Asterisk cannot agree with telco about which channels are busy and which are not. Maybe a bug? I do not know...it seems too strange Asterisk has a so big problem. There must be something we do not knowBy the way, the solution seems to be using the higher channels of the span, in other words to make calls using G instead of g inside Dial command (thans to Steve and others!!) I don't think that could be the problem. Because Asterisk has already established connection with provider on certain channel. So why would they negotiate another channel? When I transfer phone call to another extension, incoming channel doesn't change. I think something else is the problem, but I do encourage to use G in dialplan's Dial command. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Maximum retries exceeded on transmission
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 12 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1620 (Critical Response) Sep 12 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call [EMAIL PROTECTED] no reply to our critical packet. What is the critical packet that is not being responded to? Please help. I head this problem with SJ phone softphone on one installation. I have uninstalled soft phone's and now I use hard phones (Grandsteram GXP 2000) and I don't get that errors anymore. Hope this helps. If you find what exactly is the problem, please let me know. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: BLF across asterisk trunks
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I second this wish. I third this wish :)) -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 1.4 Docs
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... No mention of Shared Line Appearance in the v1.4 new release. Anyone know if they still plan to include it or not? Digium has been kind of quiet on their work on that feature. With their new Asterisk appliance running v1.4 I certainly hope they have SLA as all other traditional/proprietary PBX's in that market segment do. Yes, and I'm interested in AOC messages. If I'm only able to manipulate with them, store them somewhere. I believe every Asterisk user will benefit with this, it just that people are not familiar what AOC does. AOC messages (Advice of charge) are messages that your provider sends you at the end of call. They tell you how much units jour provider will charge you for that call. And if you would like to know how much money is that, you simply multiply it with price of every unit. It will solve charging problems with Asterisk! We wouldn't have to keep up to date our databases with prices. Provider will directly tell us how much he will charge every our call. How to help/motivate developers to work on AOC in Asterisk? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] AOC - advice of charge
I'm interested in AOC (Advice of Charge) messages in Asterisk. As far as I know, * does get AOC messages, but it's unable to do anything with them. What I would like to know is: - what is current status of AOC in Asterisk? - is there any work going on AOC in Asterisk? - is there anything I could do to make thing go faster in developing AOC in Asterisk? (unfortunately I'm not programmer) What I would like to be able to do with AOC messages is to manipulate with them and to store them in CDR or in some other database so that I could do billing. I believe every Asterisk user will benefit with this, it just that people are not familiar what AOC does. AOC messages (Advice of charge) are messages that your provider sends you during or at the end of call. Provider can send you charging Info in currency or charging Info pulse. Come on guy, lets make our life's a little bit easier! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Cisco 79xx and vlan
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I know that this has been asked before, but I couldn't find an answer .. In the office, my 79XX phones (connected to dell / hp switches) are all on their own separate network (i.e. we have data going through separate switches). When they boot, they take ages on the configuring VLAN screen. However, I also have a 7960 at home, connected to work through a vpn. This one boots very quickly indeed. It's not the phone settings, as I took this phone into the office and it then had the same symptoms. Has anyone got any idea on how to speed this process up ? On a side note, does anyone know how to send a reload config command to the 7940 without having to reboot it ? Hi Julian! I have several Cisco phones and I'm interested to get answers to your questions. If you find solution, please send mail to the list. P.S. Are Cisco phones able to do paging/intercom? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Can you explain why multiple registration is an important (missing) feature ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In some cases : Yes. But we have the following situation : We re using cisco 7960 phones in each office (about 150 of them) Do Cisco phones support paging/intercom? If yes, please send me link to some useful pages. Now we want to give the user's the ability to take their number with them. So when you change places you can call a defined number which will write you a config file for your new phone. To much work. Is it working right? Now, if I have extension 1234 and go to a different office, or to a meeting room, etc and log into that phone using my extension, if i did not log out my normal phone we have a problem because we have to SIP/1234. I haven't found a good solution for that yet, but if I could register two SIP/1234 phones the problem would be solved. I would like that Asterisk supports multiple registers, but till then you could use dynamic agents. Agent can log in from every phone. And you send incoming phone call to agent instead to extension. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: PRI: sometimes Asterisk drop calls
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Do you have queues/agents configured? No, I don't have queues nor agents configured. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: voicemailmain errors on CLI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You have to leave a message in the voicemail, then listen it and the error will not apear again. That's bad procedure. Because, all of my clients receive voicemails on e-mail with delete option. So, they will newer listen voicemail and Old directory won't be created. And I'm always getting this error. This should be changed! What is the procedure to change this in code? Is the bug tracker right way to do it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Queue - static members
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You probably already figured this out, but you use either Agent or SIP, not both. Use Agent if they login through AgentLogin or SIP if it is calling the SIP phone directly. Yes, I have figured it out later. It should go like this: member = SIP/148,1 member = SIP/143,2 -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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: Features.. phone vs. asterisk?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I tried a lot of SIP and IAX softphones looking for ones I liked, noticing some have certain features and others did not. For things like call transfer, call park, group pick-up, line presence, and all those kinds of extras I have a bit of confusion on where it is implemented? Are these functions that Asterisk handles and the phone just triggers them with some out-of-band signal or DTMF sequence? Or does some of this rest on the phone itself? (Here is where I would love TFM to R. :) Just having a hard time finding what to read.) Hi Nick! As far as I know, most of them are telephone features. And just like you have said, Asterisk features you use with DTMF. I'm looking for solution how to use phone buttons to trigger Asterisk features. So when I press Transfer button on phone, that he sends #1 to my Asterisk so that transfer can be completed on *. Still looking... :)) -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 - persistent members
Hi list! I have few questions about queue and persistent queue members. If there is queue with only one persistent member, what happens if it doesn't answer the phone for timeout = 10 seconds? Calling person still waits in queue and what happens with agent? Will his phone ring after retry = 20 seconds? When phone call from queue comes to only queue member, can somebody from pickup group pick up phone call that is from queue? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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
