[Asterisk-Users] agents.conf
One simple question. I'm using asterisk 1.2.1, can one agent be defined in more than one group? Example: group=1 ; queue1 agent = 401,401,Tomislav Parcina agent = 402,402,Katarina Ivanisevic agent = 403,403,Sasa Juginovic group=2 ; queue2 agent = 401,401,Tomislav Parcina agent = 402,402,Katarina Ivanisevic agent = 404,404,Marija Bilic agent = 405,405,Ana Kaliterna Will this work? Will agents 401 and 402 be in both groups? If I join every group to another queue, will one agent be in both queue's? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 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] What ATA should I buy?
I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of those ATA-s? Sipura SPA-2100 SIP-ATA 160$ Sipura SPA-1001 SIP-ATA 125$ ALL7902 IP SIP ATA Adapter / Router 106$ Grandstream HandyTone ATA486142$ Thank you for any suggestions. P.S. If this is second time you see this message, then sorry for resending, but I didn't see it on list... -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 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 - joinempty
Hi everybody! In queue.conf I have joinempty = no What happens with call of calling person tries to join that queue? Does it goes to next priority? This is my extensions.conf [callcentre] exten = 311,1,Answer exten = 311,n,Playback(callcentar/anonuce,skip) exten = 311,n,Queue(queue|th|||3600) Do I have to add one more line (311,n) which will define what will happen with call if no agents are logged in that queue? Can I do that on any other way? Thank you for your time! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 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 - check agent
I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's? What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call. Thank you for your time. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 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 transfer
When I try to make att transfer (*2) of call that was in queue the call get's disconnected. Blind transfer (#1) works fine. In dial plan I don't have any h or H (hangup call with *). In features.conf I have this line disconnect = *0. What could be the reason why call hang's up? -- Tomislav Parcina tparcina#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] Voicemail - direct call
Hi list! How to send a call directly to voicemail recording? When I put this exten = 313,n,VoiceMail,u221 Or this exten = 313,n,VoiceMail,b221 In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? -- Tomislav Parcina tparcina#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] Call centre - * hang's up
When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help would be appreciate. This is debug from console. Feb 14 08:27:08 DEBUG[13349]: chan_sip.c:2969 sip_rtp_read: * Detected inband DTMF '*' Feb 14 08:27:08 DEBUG[13349]: channel.c:3253 ast_generic_bridge: Didn't get a frame from channel: Agent/401 Feb 14 08:27:08 DEBUG[13349]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels SIP/211-5396 and Agent/401 Feb 14 08:27:08 DEBUG[13349]: chan_agent.c:760 agent_hangup: Hangup called for state Up -- Tomislav Parcina tparcina#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 Register
I'm having trouble making calls over my VoIP provider. I do successfully
register, and when I try to establish a phone call Asterisk sends wrong
username and password. Instead of sending username and pass that I have
provided, he send username and pass of the SIP phone that is registered to *
(the phone from which I try to make a call).
What have I done wrong?
This is my sip.conf
[general]
context=sip
port=5060
bindaddr=0.0.0.0
srvlookup=no
tos=184
maxexpirey=3600
defaultexpirey=120
disallow=all
allow=ulaw
allow=alaw
allow=gsm
musicclass=default
useragent=PBX Lama
nat=no
externip = 200.200.200.200 ; my external IP
localnet = 10.0.0.0/255.255.255.0
realm=lama.hr
register = myusername:[EMAIL PROTECTED]
canreinvite=no
[iskon1]
type=friend
username=myusername
secret=mypass
host=sip.iskon.hr
nat=yes
canreinvite=no
[214]
callerid=Vice Lacmanovic 214
type=friend
username=214
secret=vice
host=dynamic
mailbox=214
canreinvite=no
dtmfmode=inband
And this is part of my extensions.conf - the line I use for calling out.
exten = _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Again, problem is that Asterisk to my VoIP provider sends username 214 and pass
vice (data of my SIP phone) and not the data that I have provide to it
(myusername and mypassword for that VoIP provider).
Thank you for your time.
--
Tomislav Parcina
tparcina#lama.hr
___
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[Asterisk-Users] RE: SIP Register
Subject: RE: SIP Register From: Tomislav Parcina [EMAIL PROTECTED] In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... First impressions telling me you want to check your phone settings. What phone are you using and what are the config settings? Hi Mark, thank you for your reply. I'm using Cisco 7905 with SIP version 1.3.1(050608A). This phone has tone of settings (few pages). What exactly would you need? Why do you think it's phone problem and not Asterisk? Asterisk is the one that contents my provider. * is the one who should decide what information's to send to my VoIP provider... Anyway, I'm inexperienced with this and I'm just trying to understand what is happening and where could be the problem. -- 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: Queue - check agent
Subject: RE: Queue - check agent From: Tomislav Parcina [EMAIL PROTECTED] In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello, I might be wrong here, but I thought that in Queues.conf, if you defined a queue with joinempty=no, or joinempty=strict then no calls will be placed in the queue, and asterisk will go onto the next extension in the dial plan. This is fine if it goes to next extension. ; If you wish to remove callers from the queue when new callers cannot join, ; set this setting to one of the same choices for 'joinempty' ; ; leavewhenempty = yes Where the caller goes if last agent exits queue? -- 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: SIP Register
Why do you think it's phone problem and not Asterisk? Asterisk is the one that contents my provider. * is the one who should decide what information's to send to my VoIP provider... Anyway, I'm inexperienced with this and I'm just trying to understand what is happening and where could be the problem. One more thing. Now I have tried with softphone. I have the same problem. Asterisk sends user and password of SIP account (SIP phone) that is making a call but not the account information's that I have received from my service provider. Question: How to configure Asterisk so he sends right user information's? -- 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: virtual extension per user ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... This can easily be accomplished with AMP using the Users and Devices mode. http://voipspeak.net/index.php?/content/view/49/28/ How can this be done without AMP? Using personal queue's and agents? I need information's to get better picture about this one. -- 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: RE: virtual extension per user ?
You can do this with agents, no need for a queue. Define agents in agents.conf In your dialplan, instead of Dial(SIP/bedroom) use Dial(Agent/200) Let the phones login as agent :) OK, I know I have to Dial(Agent/200), but how will I login agents if I don't use queue? If phone log's in as agent, then I didn't do anything, because that agent will always be on that phone (and that is something I would like to avoid - because of that I started to use agents in first place). Maybe I didn't understand something right. -- 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: RE: virtual extension per user ?
AMP doesn't do miracles! Look at its dialplan. I believe he doesn't, but I don't have AMP installed. Next week I think I'll have enough free time to try it. Will [EMAIL PROTECTED] do the trick? -- 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: What ATA should I buy?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... We have got some ATA for only $55 if you are interested? Sam Yes Sam, I'm interested. If they work with FAX I'll definitely buy one of them for testing. -- Tomislav [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: What ATA should I buy?
Since you have no Digium hardware (and thus no connection to POTS or PRI)... are you routing your phone calls via VoIP? If so, it is not recommended to run FAX via VoIP. The two don't mix. FAX is not able to handle packet loss like VoIP. Also, any codec other than uLaw will not even come close to working, as the codecs are designed to compress voice. Hi Ron! Thank you for your mail. I know there could be some issues, but if I use ulaw, most of FAX should pass true. In few years people won't send faxes anymore, but till then I need something that will work with 90% success. -- 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: What ATA should I buy?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... AFIK, fax is supported and installed with with app_txfax app_rxfax If this proves to be true why would you need the ATA? I'm working on this one. I have to install app_rxfax but I have failed. Soon, I'll try again (hopefully next week). Anyway, I'll need ATA even then. Because it isn't just receiving FAX, but sending it. It is problem to scan paper then send it by mail or app_txfax. -- 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: Re: asterisk logger - urgent!!!
Why don't you simply rotate the logs with logrotate ? How to do that? -- 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: SIP groups
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You can not define groups in sip.conf But there are, as you hint, other ways to solve the problem, like using queues or implementing it in dialplan logic. Do you have any example how to do that? -- 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: Voicemail - direct call
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... aHR0cDovL3d3dy52b2lwLWluZm8ub3JnL3Rpa2ktaW5kZXgucGhwP3BhZ2U9QXN0ZXJpc2srY21k K1ZvaWNlTWFpbAoKSXQncyBhbGwgaW4gdGhlcmUKCk9uIDIvMTMvMDYsIFRvbWlzbGF2IFBhcuhp bmEgPHRwYXJjaW5hQGxhbWEuaHI+IHdyb3RlOgo+Cj4gSGkgbGlzdCEKPgo+IEhvdyB0byBzZW5k IGEgY2FsbCBkaXJlY3RseSB0byB2b2ljZW1haWwgcmVjb3JkaW5nPwo+Cj4gV2hlbiBJIHB1dCB0 aGlzCj4gZXh0ZW4gPT4gMzEzLG4sVm9pY2VNYWlsLHUyMjEKPiBPciB0aGlzCj4gZXh0ZW4gPT4g MzEzLG4sVm9pY2VNYWlsLGIyMjEKPiBJbiBteSBkaWFsIHBsYW4sIGNhbGxpbmcgcGVyc29uIGZp cnN0IGhlYXJzIGdyZWV0aW5nIG1lc3NhZ2UgKGJ1c3kgb3IKPiB1bnZpYWJsZSkuIEkgd291bGQg bGlrZSB0byBhdm9pZCBncmVldGluZyBtZXNzYWdlIChJIHdvdWxkIHBsYXkgc29tZXRoaW5nCj4g d2l0aCBQbGF5YmFjayBhcHBsaWNhdGlvbikuIElzIGl0IHBvc3NpYmxlPwo+Cj4KPiAtLQo+IFRv bWlzbGF2IFBhcmNpbmEKPiB0cGFyY2luYSNsYW1hLmhyCj4gX19fX19fX19fX19fX19fX19fX19f X19fX19fX19fX19fX19fX19fX19fX19fX18KPiAtLUJhbmR3aWR0aCBhbmQgQ29sb2NhdGlvbiBw cm92aWRlZCBieSBFYXN5bmV3cy5jb20gLS0KPgo+IEFzdGVyaXNrLVVzZXJzIG1haWxpbmcgbGlz dAo+IFRvIFVOU1VCU0NSSUJFIG9yIHVwZGF0ZSBvcHRpb25zIHZpc2l0Ogo+ICAgIGh0dHA6Ly9s aXN0cy5kaWdpdW0uY29tL21haWxtYW4vbGlzdGluZm8vYXN0ZXJpc2stdXNlcnMKPgo=___ --Bandwidth and Colocation provided by Easynews.com -- Thank you, but this is how I see your mail. How can I see it right? -- 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: segmentation fault
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi
Asterisk died this morning with this message
safe_asterisk: line 83: 6828 Segmentation fault (core dumped)
asterisk ${CLIARGS} ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Hi Patrick,
I'm new to Linux, so can you please tell me how do you check how did Asterisk
died?
Thank you.
--
Tomislav Parcina
[EMAIL PROTECTED]
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[Asterisk-Users] Re: Call centre - * hang's up
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I think it's a bit of a known fault - the attended transfer function does not work from the queue system. It would be nice if it did, though. Hi Paul! Is there any explanation about this? Is that something that will change? -- 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: Call centre - * hang's up
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You'll have to use uattended transfers for CCs. l. I have read Paul's mail. Is this bug or feature? -- 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: Re: RE: virtual extension per user ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You have to use AgentCallbackLogin for that. If a phone logs in that way, it's reachable as Agent/200 You can also use AgentCallbackLogin to logout the agent. You don't have to worry about an agent that forgets to logout on phone X when they walk to phone Y, cause AgentCallbackLoging will overwrite asterisk database entry for that agent so it's only reachable on the phone where they last login (asuming they didn't logout there) This is cool. Another thing, how can I limit outgoing phone calls form IP phone, if no agent isn't logged on that phone? And, in CDR, does it say which agent has made specific phone call? When I get home later today I will put an example in my system and post it here. Now I understand, but (as you can see) now I have new questions :)) Thank you for your time! -- 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: Outbound ZAP Dialing
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have server with a total of 6 Analog ports...using TDM04B and TDM02B. I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have worked through getting the DIDs to work and route to the extensions...now what I need to do is when Extension picks up the phone to dial, I would like them to use their DID analog line first, unless someone has called in on it and they are trying to call out to conf someone else, then roll to one of the other 3 rollover lines. I have come up with one option of using different prefixes...which is tied to each DID...7 for did , 8 for and 9 for ..but as you can see this pretty silly...and very limiting... Is there a way...please help... Why don't you group all six lines? And call out over group. -- 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: Re: Voicemail - direct call
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Thank you, but this is how I see your mail. How can I see it right? http://lists.digium.com/pipermail/asterisk-users/2006-February/146742.html Thank you! -- 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: Re: Call centre - * hang's up
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... But using the native transfer on the phone causes the system to think the agent is still on the call Yes, and I have desabled that options on my phones. Sometimes I have delay if I use transfer or three way calling on Cisco phones. Anyway, that is why I have PBX, to make all this options avaible on it, not on the phone. -- 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: Linear Queues Strategies for 3rd Party Application
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does anyone know how to setup a linear type of queue strategy? By that I mean that agents will be tried in a particular order and the call will be routed to them unless they are on the phone or not logged in. I want a 3rd party app to be able to re-arrange this order on the fly based on sales and other metrics. Anybody setup something similar? Any pointers or products already out there open source or not? Thanks, Steve Totaro Hi Steve! Why don't you use weight=10 from queues.conf? -- 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: Fromstring when sending e-mail on recieved voicemail
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi. I'm having trouble controlling the user info when sending e-mails from asterisk via sendmail to a Microsoft exchange server. When I receive the email the sender is always [EMAIL PROTECTED] and the name of the sender is always Added by portage for asterisk. I want to change both sender-address and the name of the sender. In voicemail.conf you have [general] [EMAIL PROTECTED] fromstring=My name -- 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: Call queue design issues and suggestions
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I don't know if this works for you, but I use the following mechanism. I don't use the agent call back stuff, just the (Add|Remove)QueueMember stuff. For each queue, dialing the extension (), puts the caller into the queue (ie, a customer calling for reservations). I use ** to sign a phone into the queue and * to sign out of a queue. Good idea, maybe sometimes I'll need it. You can use the manager to see who is currently logged into a port. It doesn't take much to write a cgi script that outputs the Cisco XML for the phones. I've built a few apps that do interesting things. It would be quite easy to write an app that: It could be easy for someone with experience, but if you have never done it before (like me) it isn't like that. Can you send us what you have done? -- 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] Cisco 79xx firmware
I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me technical support agreement which cost around 75$ for every phone! -- Tomislav Parcina tparcina#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 79xx = Asterisk - SIP or SCCP?
One easy question for experienced users. Should I use Cisco VoIP phones with SIP or SCCP? What are the (dis)advantages of one or another? Please tell me your stories. -- 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: Cisco 79xx firmware
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... CDW and other large resellers like them have a difficult time selling service contracts. The issue is they _must_ provide Cisco with a serial number (of the phone) which is checked by Cisco to see if the company ... First they are expensive, and than they have dealers that don't know how to sell their products. It's really frustrating! -- 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: FC4 and yum install; how to configure questions
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I installed FC4, ran command, # yum install asterisk. A bunch of stuff happened, but can't locate .conf files. I have a list of files: /usr/share/doc/asterisk-1.2.4/configs/features.conf.sample /usr/share/doc/asterisk-1.2.4/configs/rtp.conf.sample /usr/share/doc/asterisk-1.2.4/configs/extensions.conf.sample Hi Tom! Read the book! Those files are sample configuration files. You can copy them to /etc/asterisk dir without .sample at end (example - sip.conf). Also, you can do cd /usr/src/asterisk-1.2.4/ and make sample. That way Asterisk will copy some example files to your /etc/asterisk/ dir. Read the book! -- 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: Keep getting message in logs that pbx.c cannot find extension context 'default'
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I am getting repeated messages in my logs with the following: * Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Feb 23 07:56:12 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:12 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Feb 23 07:56:14 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:14 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] * I do not have a default context used in my extensions.conf - I use other names. Am I required to have a context named 'default'?? Thanks Hi Chuck! In sip.conf you have defined context=default in general or some of user/peer/friend section. So, Asterisk tries to send call to default context which you don't have in extensions.conf. You can 1. in sip.conf change from context=default to some other context that exist in extensions.conf or 2. create default context in extensions.conf -- 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: Important: Application DIALPLAN STANDARD/GUIDELINES needs to be established.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello Asterisk community. We have a small User-group in Melbourne Australia. Recently I brought up the issue of STANDARDS for dialing Applications on a PBX. This generated some interest but also the fact little has been done on this topic. Below is a rundown of our THREAD. (start from bottom and go up) I myself, feel this to be an important issue. With Asterisk being so programmable, anything can be done. But should it. I would like to see some type of guide line or standard for extension layouts. We have not been able to find any reference to this. However, I hope the greater Asterisk community has, and if so, please share. Thanks, James Hi James! I must say that I like your idea. It would be great that there is some recommendation for standard options of PBX. Maybe there is some RFC that is already dealing with this but I'm not familiar with that. Any further information's about this are more than welcome. -- 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] Asttapi - what's wrong?
When I try to call from asttapi one number, I get message No one is available to answer at this time (1:0/0/0). Immediately after that I try to call the same number from SIP phone (the same one that is used with asttapi) and call goes true. What have I done wrong? This is how it looks on CLI. == Parsing '/etc/asterisk/manager.conf': Found == Manager 'tomo' logged on from 10.0.0.203 Channel SIP/341-062e was answered. -- Executing Dial(SIP/341-062e, OOH323/[EMAIL PROTECTED]) in new s tack -- Called [EMAIL PROTECTED] == No one is available to answer at this time (1:0/0/0) -- Executing Hangup(SIP/341-062e, ) in new stack == Spawn extension (sip, 00989970434, 2) exited non-zero on 'SIP/341-062e' == Manager 'tomo' logged off from 10.0.0.203 -- Executing Dial(SIP/341-9e85, OOH323/[EMAIL PROTECTED]) in new s tack -- Called [EMAIL PROTECTED] -- OOH323/85.114.35.42-b1b4 is ringing == Spawn extension (sip, 00989970434, 1) exited non-zero on 'SIP/341-9e85' ___ --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 can I debug spandsp?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Asterisk's debug facilities need to be enabled before you'll get debugging information. And how do you turn on Asterisk's debug facilities? -- 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: res_features pickupexten
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... the callgroup/pickupgroup settings are correct... dialing *8 or *8# on any client (zap/sip/sccp) results in unknown extension... To pick-up with SIP phone, it has to be defined in sip.conf. Same goes for zap and iax2. -- 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] My or provider error?
Situation. I call out from SIP phone over h323 trunk and called person decides
not to pick up (on mobile phone they press red button - NO - hang-up). Until
the called person press the NO button, I can hear ringing. When called person
press the button, I don't hear anything. Asterisk waits until timeout and than
ends the call.
How can I get busy or some other appropriate signal on SIP phone headset?
This is what I have in extensions.conf. I use Asterisk 1.2.1 (soon I'll use
1.2.4)
exten = _0.,1,Dial,OOH323/${EXTEN:[EMAIL PROTECTED]
exten = _0.,n,Hangup
--
Tomislav Parcina
tparcina#lama.hr
___
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[Asterisk-Users] Re: Re: How can I debug spandsp?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Edit logger.conf and uncomment full. Start Asterisk with the the -d option. View debugging information in the /var/log/asterisk/full Is -d option necessary? Anyway, done that. Just thought that you think about something else. Thank you! -- Tomislav Parcina tparcina#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: Re: res_features pickupexten
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... callgroup and pickupgoup is configured in the config-files (zap/sip/sccp) - is anything else needed ? Sorry, I'm not up to this. -- Tomislav Parcina tparcina#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 - what's wrong?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... When I try to call from asttapi one number, I get message No one is available to answer at this time (1:0/0/0). Immediately after that I try to call the same number from SIP phone (the same one that is used with asttapi) and call goes true. What have I done wrong? Solved! Problem vas that manager adds default caller ID (not the one that was defined in sip.conf for the phone from which I'll will speak). And I need to sent to provider specific caller ID. Now, I have question. In agents conf, can I define Caller ID for every user (manager)? If not, that is something that defiantly should be implemented. -- Tomislav Parcina tparcina#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 hangs up - h323
This is third time today that my Asterisk hangs up. It seams that I have problems with h323. I'm using ooh323 from Asterisk add-ons. I have the following configuration Asterisk 1.2.1 Asterisk-addons 1.2.1 Fedora Core 4 I'm using SIP phones and h323 trunk to my VoIP provider Like I said this is third time today that he hang's up. First time, I came at work and Asterisk was down. Second time I tried to call, and Asterisk was down (not sure at that wary moment or before I tried to call). So, I decide to start logging and this is what I received just before Asterisk died. Anyway, I tried to reload from CLI and that is when he died. What can I do to check why it's happening? I have plenty of disk space, lots of free ram and processor is idle for more than 80%. I think it could be because of alaw codec that I use (my provider requires it) and this is what is in ooh323.conf file (ONLY ulaw, gsm, g729 and g7231 supported as of now). But Like I said, it works for several hours and then it dies... So I don't think that is it. ooh323.conf [general] bindaddr=xxx.xxx.xxx.xxx h323id=ObjSysAsterisk e164=100 callerid=asterisk gatekeeper = DISABLE context=incomingh323 disallow=all allow=alaw allow=ulaw allow=gsm dtmfmode=rfc2833 full.pbx Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/manager.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Feb 28 14:04:15 NOTICE[5018] cdr.c: CDR simple logging enabled. Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/rtp.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/rtp.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: == RTP Allocating from port range 1 - 2 Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/musiconhold.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/musiconhold.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_indications.so' (Indications Configuration) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_adsi.so' (ADSI Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_features.so' (Call Features Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/features.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/features.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Blind Transfer (blindxfer) to sequence '#1' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Attended Transfer (atxfer) to sequence '#2' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature One Touch Monitor (automon) to sequence '#3' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Disconnect Call (disconnect) to sequence '#0' Feb 28 14:04:15 DEBUG[5018] res_features.c: Removed old parking extension [EMAIL PROTECTED] Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Added extension '700' priority 1 to parkedcalls Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_config_mysql.so' (MySQL RealTime Configuration Driver) Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Host: Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Port: 0 Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime User: Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Password: Feb 28 14:04:15 ERROR[5018] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '' (111) Feb 28 14:04:15 WARNING[5018] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '' (111) Feb 28 14:04:15 VERBOSE[5018] logger.c: == MySQL RealTime reloaded. Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2)) Feb 28 14:04:15 ERROR[5018] chan_iax2.c: Unable to load config iax.conf Feb 28 14:04:15 VERBOSE[5018] logger.c: == Loaded firmware 'iaxy.bin' Feb 28 14:04:15 NOTICE[5018] iax2-provision.c: No IAX provisioning configuration found, IAX provisioning disabled. Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny)) Feb 28 14:04:15 NOTICE[5018] chan_skinny.c: Unable to load config skinny.conf, Skinny disabled Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_local.so' (Local Proxy Channel) Feb 28 14:04:15
[Asterisk-Users] ooh323 codec's - alaw
Does ooh323 from asterisk-addons 1.2.1 support alaw codec? This is what is written in h323.conf.sample that can be found in asterisk-addons dir. The codecs to be used for all clients.Only ulaw and gsm supported as of now. Default - ulaw ONLY ulaw, gsm, g729 and g7231 supported as of now disallow=all allow=gsm allow=ulaw So, it shouldn't support alaw, but I manage to establish calls with alaw codec. The problem is that sometimes, because of h323, my asterisk dies. Now I just would like to check is it because of codec. -- Tomislav Parcina tparcina#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: How to check if transcoding is setup to work properly
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... How can you check if transcoding is configured to work properly on a system? Is there a way of knowing that transcoding is configured properly and is giving some output to indicate so? CLI show translation -- Tomislav Parcina tparcina#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 7905 - vad, cng
How to disable silence suppression (or Voice activity detection - VAD) on Cisco 7905 phone? On Cisco 7940 I use enable_vad: 0, but I can't find anything similar for 7905. -- Tomislav Parcina tparcina#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] MOH native files
Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#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] Info about F1000G
Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- Tomislav Parcina tparcina#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: MOH native files
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You need to use mpg123 to convert the mp3 files to wav files first. mpg123 -w out.wav in.mp3 This one works. Thank you! sox out.wav -r 8000 out.gsm I have problem with this command. It runs fine, but when I play that file it is twice long as it should be and double slow as it should be. So wav file that was 2 min long becomes 4 min long gsm file. How can I fix that? -- Tomislav Parcina tparcina#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: MOH native files
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... sox -V foo.mp3 -t au -r 8000 -U -b -c 1 foo.ulaw resample -ql Chris This is what happens. [EMAIL PROTECTED] mohmp3]# ls fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 [EMAIL PROTECTED] mohmp3]# sox -V fpm-calm-river.mp3 -t au -r 8000 -U -b -c 1 fpm-calm- river.ulaw resample -ql sox: resample opts: Kaiser window, cutoff 0.94, beta 16.00 sox: Failed reading fpm-calm-river.mp3: Do not understand format type: mp3 Have I done anything wrong? -- Tomislav Parcina tparcina#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] Native music on hold - Error
I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial(SIP/341-5931, SIP/344|20|wWtT) in new stack -- Called 344 -- SIP/344-5e4e is ringing -- SIP/344-5e4e answered SIP/341-5931 -- Attempting native bridge of SIP/341-5931 and SIP/344-5e4e -- Started music on hold, class 'native', on SIP/344-5e4e Mar 2 11:17:50 WARNING[7717]: format_wav.c:161 check_header: ot in mono 2 ar 2 11:17:50 WARNING[7717]: file.c:432 ast_filehelper: nable to open file on / var/lib/asterisk/moh-native/fpm-sunshine.wav ar 2 11:17:50 WARNING[7717]: res_musiconhold.c:225 ast_moh_files_next: nable to open file '/var/lib/asterisk/moh-native/fpm-sunshine': No such file or director y -- Stopped music on hold on SIP/344-5e4e == Spawn extension (sip, 344, 1) exited non-zero on 'SIP/341-5931' What have I done wrong? That file IS in that directory. When this starts to work I'll put more files in gsm and g729 format, but till then asterisk should encode this files. For this call I have use alaw codec. -- Tomislav Parcina tparcina#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: Agents, queues and Pentalties
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... But when a call enters queue_1 or queue_2 it allways rings everyone directly without checking if Agent1 is available or not. It should distribute the calls from queue_1 to the other agents only when agent/1 is unavailable and agent/1 should only get calls from queue_2 when all other agents of queue_2 are unavailable Ringall does exactly what you said - it rings all. You should use some other ring strategy. -- Tomislav Parcina tparcina#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: Info about F1000G
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello Tomislav, I borrowed F1000 from my friend for testing. I am not sure if that is different from F1000G, but I am experiencing the following issues: 1. As a user, it is not easy to get a firmware update as I need to have a service contract. 2. Even with the latest firmware I got from sipgate.de (version 3.80st), I can only have WPA-PSK with TKIP encryption, while I prefer AES. 3. The voice quality is sometimes really bad when using codecs with compression (G729 and G726). No problem with G.711. 4. The battery does not last long, just around 22 hours. I don't have any other issues a part from those. Cheers, Anto Hi everybody! Thank you all for information's that you have provide to me. Now I have pretty clear picture what to expect from this phone. Have a nice day! -- Tomislav Parcina tparcina#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: res_features pickupexten
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... i can confirm that this bug exists in 1.2.4 as well. we've managed to fudge it by dialplan tricks and Pickup(). Please report the bug. In 1.2.1 it works fine. -- Tomislav Parcina tparcina#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: Re: MOH native files
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You need to install either libmad or libmp3lame. Sox checks for this on startup. This is what I get when I enter yum install libmp3lame or libmad Parsing package install arguments No Match for argument: libmp3lame Nothing to do Parsing package install arguments No Match for argument: libmad Nothing to do Is something wrong with my repository or you have provide me wrong packet names? -- Tomislav Parcina tparcina#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: Native music on hold - Error
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... what are the file permissions/ownership and are they readable by the asterisk process ? Asterisk runs like root and permissions are 755. So, as far as I know, that should be fine. -- Tomislav Parcina tparcina#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: Get no busy signal on my analog line
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does this belong to my dialplan or my sip registration settings? To your SIP registration settings. You should limit that user/peer/friend to only one line. -- Tomislav Parcina tparcina#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: Native music on hold - Error
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... what are the file permissions/ownership and are they readable by the asterisk process ? The problem was that wav files where in stereo mode. I have encode them and now it works fine. -- 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] Set(LANGUAGE()=language) - for queue
Hi group! How to set language for queue? I have several queue's. In every queue, agents speaks different language. I need to announce queue-youarenext and similar on different languages. This is what I have in my extensions.conf and it does set language, but when calls enters queue, it doesn't use that language. exten = 313,1,Answer exten = 313,n,Set(LANGUAGE()=de) exten = 313,n,Playback(callcentar/qnjemacki,skip) exten = 313,n,Queue(njemacki|t|||3600) exten = 313,n,GotoIfTime(8:00-16:00|mon-fri|*|*?313,8) exten = 313,n,Playback(callcentar/rvnjemacki,skip) exten = 313,n,VoiceMail,u221 exten = 313,n,Hangup exten = 313,n,VoiceMail,b221 exten = 313,n,Hangup And this is how it looks on CLI. -- Executing Goto(SIP/211-793f, callcentre|313|1) in new stack -- Goto (callcentre,313,1) -- Executing Answer(SIP/211-793f, ) in new stack -- Executing Set(SIP/211-793f, LANGUAGE()=de) in new stack -- Executing Playback(SIP/211-793f, callcentar/qnjemacki|skip) in new stack -- Executing Queue(SIP/211-793f, njemacki|t|||3600) in new stack -- outgoing agentcall, to agent '401', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/401 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/211|20|wWtT) in new stack -- Called 211 -- SIP/211-5996 is ringing -- Agent/401 is ringing -- SIP/211-5996 answered Local/[EMAIL PROTECTED],2 -- Agent/401 answered SIP/211-793f -- Playing 'callcentar/gpnjemacki' (language 'en') == Spawn extension (internal, 211, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- Playing 'queue-reporthold' (language 'en') -- Playing 'queue-less-than' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'queue-minutes' (language 'en') -- Tomislav Parcina tparcina#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 add-ons - H323
How to upgrade h323 from Asterisk add-ons (from version 1.2.1 to 1.2.2)? In INSTALL they don't say anything about upgrade... Thank you for your time! -- Tomislav Parcina tparcina#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] Gmane - Asterisk Users Mailing List
Hi group! Does anybody knows about any news server that works the same way that Gmane www.gmane.com/ does it? I was satisfied with Gmane for few months, but now it seams that it doesn't work any more (no new posts in past few days). Now I'm looking for alternative. -- Tomislav Parcina tparcina#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] Send One Touch Record to mail
How can I send recordings, that I have recorded with One Touch Record, to e-mail address that is defined in voicemail.conf? Thank you for your ideas. -- Tomislav Parcina tparcina#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
RE: [Asterisk-Users] Send One Touch Record to mail
Hi Joe! Thank you for your mail. The thing is that I have never program anything so it will take a lot of my time, which I don't have right now. Hopefully, when I finish started projects I'll be able to play with this stuff. In the meantime if anybody solves this problem, please let the group know. --Tomislav Parcinatparcina#lama.hr From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe PukepailSent: 7. ožujak 2006 20:41To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Send One Touch Record to mail As far as I know, you will need to do this yourself with some creative scripting. There was some talk on the list awhile ago to move the recording tovoicemail, but I dont' know if anyone has made a patch to do it yet. ___ --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] Can't hear busy tone
HI Group! I have strange problem. Since I started to use H323 with my VoIP provider when I dial the person that is currently busy, I can't hear busy tone on my handset. What could be the problem? What should I look for? How is this exactly called (because I even don't know what to look for). Hopefully someone will be able at least to give me some starting point. -- Tomislav Parcina tparcina#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: 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: 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: 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] 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] 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] 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] 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] 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] H264
As far as I can see on this web page http://www.voip-info.org/wiki-Asterisk+video Asterisk doesn't support h264 codec. I can see the same on this pages http://www.asterisk.org/features Question is, can I somehow enable H264 codec support in Asterisk? I have Grandstream GXV-3000 video IP phone which supports only h264 codec. Right now I can make only direct IP video phone calls, and I would like to make calls true 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: GSM gateway and FXO ATA
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. Hi Marty! Can you tell me more about this? You mean when call from SIP goes to FXO port, if phone attached on FXO port answers after the first ring (before second) ATA will always stop to 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: Nokia E60/61/70 and SIP
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 1. you need qualify set as the wifi radio on the phone sucks big oranges What is qualify set? -- 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: DNS
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Have you tried setting timeout, attempts and rotate in resolv.conf? Can you please tell me more about this? How to do it and what would I achieve with 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: H264
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Actually you need to use the SVN version of Asterisk to support H264 video. It should be part of the planned 1.4 release. When can I expect 1.4 release? Will it be this year? First quarter of 2008? -- 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
Is 8.0.2.SR1 still the latest firmware? I still haven't managed to do anything useful with that weary expensive phone. It still only receives and places calls, nothing else. Is there any exciting feature that can work with asterisk and SIP firmware? Has anybody managed to do anything of the following: - my screensaver - picture of calling person - External directory - dialplan.xml - How to setup hinting (Multiple Call Appearance) - How to login true ssh? -- 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 Development and Release Cycle
Do Asterisk team care about this anymore? Whole text can be found here: http://www.asterisk.org/developers/releasecycle -- 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: Nokia E60/61/70 and SIP
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... qualify=yes Put in in the sip.conf file in the configuration section for the specific phones. I don't think he thought on 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: Asterisk Development and Release Cycle
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Of course we care. Turns out that schedule was unrealistic, and when we start the next cycle we will regroup and decide if we either stretch out the cycle or reduce the amount of new functionality that gets added during the cycle. OK, thank you for info. -- 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 Development and Release Cycle
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I don't know. Do you use Asterisk? That makes you part of the team. Have you tested the trunk version? Provided assistance testing out patches waiting for completion? Really, once all the new features have been completed, it will be released. If you would prefer it to be released now (I.E. before everything has been tested and possibly fixed), just download SVN trunk. Hi Matt, Yes, I have downloaded SVN trunk. I'm using H264 codec from it. There is one question I need to ask. How can I find out what are new options in SVN trunk? Right now I know only for H264, where can I find the list of others? -- 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 Development and Release Cycle
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Err, wasn't the patch for H.264 just changing one digit for another? Hi Thomas, I don't know. I should check BUG page for 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: Asterisk Development and Release Cycle
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Other cool things: make menuconfig Jingle/jabber support IAX2 media transfers new sound files New answer machine detection (AMD) and much much more! Hi Matt, thank you for info! Bye. -- 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 7960G SIP firmware 8.4
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Seems to be working ok on my handset for the past couple of weeks. No major bugs, registration, xml services and MWI works etc..etc.. Have not given it a thorough testing though. Hi Nathan, Does it have any new options? I would like to see hinting on 7940/7960. Can you send me your's Phone Directory xml files? I can't manage to add second page so I have only 32 numbers :(( Also, I can't manage to enable search thru directory. Other thing, can personal directory be in xml file? -- 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 7960G SIP firmware 8.4
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any info? What version should I download? Is this one all right? cmterm-7940-7960-8.4.00-sip.cop.sgn Signed SIP Firmware for CCM versions 5.0(4) and later -- 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 8.0.4 SIP firmware
Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? -- 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 8.0.4 SIP firmware
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does anybody use 8.0.4 SIP firmware for Cisco 7970 IP phone? I have upgrade my phone and now it doesn't register with Asterisk. In full.log file I don't see any reason why phone doesn't register. Has anybody head problems like this one? Now I have downgrade to 8.0.2 version and phone has registered fine. Does anybody know what is the problem with SIP 8.0.4 firmware and how to solve 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: GSM gateway and FXO ATA
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Actually it's kind of the opposite... When a call comes in to the FXO, and it rings the FXS, if the FXS answers on the first ring, the call goes somewhere but who knows where. The picking up party hears a dial tone, and the caller hears dead air. Hi Marty, I can live with that. I don't have anything connected to FXS port :) -- 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 8.0.4 SIP firmware
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I tried that image for about 5 minutes. Kept getting errors in asterisk from the phone and it wouldn't stay registered. Rolled back to 8.0.2 and that works fine for us for now. Hi Aaron! When you define all 8 line buttons (on the right side of display) does screen go brown for you? I mean a little bit darker on lines where you have defined something. Is it possible to avoid that darkness on whole screen? It makes my picture look bad :( Does services and directories button work for you? I always get HTTP 404: Page Not Found and I have that pages on http server and in log I see that phone asks for them... Have you done anything in Java for this phone? -- 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
