[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] Xorcom Astribank
Hi! Does some one of you have practical experience with the Astribank channel bank (I've tried to contact xorcom directly but didn't received any answer)? We plan to use an Asteribank-16 channel bank to connect analog fax devices to Asterisk (to get rid of buggy Fax over G.711 with SIP terminal adapters) Does Astribank support fax devices (as fax is sensitive against jitter, delay, synch ...)? Are there special configuration issues when dealing with fax devices? regards Klaus ___ --Bandwidth 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] amr codec
On 17 Sep 2006, at 23:04, Net Nut wrote: Well this would not be for comercial use.. I just want it for my own cell phone to talk on my own asterisk system. is that ok? Probably not. Unless you live in a country where the patent isn't valid. You'd have to read the license terms from the patent holder(s) to be sure. On the other hand I doubt that they would go after you for a single channel. Conversely, you should consult the licensing folks at Digium to see if adding a module that contains a patented codec is ok with them. You certainly can't distribute the result in any way without breaching the GPL/Digium's license. Likewise no one can help you building it since you may not distribute the sourcecode either. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: PRI: sometimes Asterisk drop calls
Hi, you are right. We have an outbound calls problem too. We tried not to use groups but to ask Asterisk for an available channel using chanisavail command. Asterisk gives us back the channel but when we try to call using that channel Asterisk says the channel is not availableand nobody is using it Seems strange nobody noticed it before. This risks to be a dangerous problem because this means Asterisk does not know which channels are really available and which are not. In other words if nobody explains why Asterisk behaves that way, it surely be possible to consider this as a bug. Is there anybody trying to use chanisavail with success?? TIA Giorgio Incantalupo Tomislav Parčina wrote: 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 ___ --Bandwidth 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] amr codec
On Sun, Sep 17, 2006 at 03:04:35PM -0700, Net Nut wrote: Well this would not be for comercial use.. I just want it for my own cell phone to talk on my own asterisk system. is that ok? Voiceage are quite agressive in terms of licensing. However as an individual it's probably not worth their efforts to do anything as the results wouldn't be worth it. If you run a business and the business has assets, then it's a different matter. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is chanisavail command reliable?
Hi,
I have an Asterisk box on a Debian Sarge distro with a Sangoma a102 PRI
card and I have a big problem: Asterisk drops a lot of outbound calls.
I do not use groups, I want to use a free channel given from Asterisk.
What I do is:
1) use chanisavail command to ask Asterisk for a free channel to use
2) use that channel to dial outbound calls
Asterisk gives me the free channel but when I make a call Asterisk tells
me the channel is not available (I check variable ${AVAILSTATUS}). I
checked but that channel is not busy on another call.
How can it be possible? Is there anybody who can help me?
TIA
Giorgio Incantalupo
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[asterisk-users] Log out an Agent on RNA
Hello all, Is it possible to automatically log off an agent on RNA (Ring No Answer) when the agent is logged in with AgentCallbackLogin? Thanks and Regards Markus ___ --Bandwidth 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] Why not g726-32?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: 17 September 2006 17:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why not g726-32? RR wrote: On 9/16/06, Rich Adamson [EMAIL PROTECTED] wrote: RR wrote: All, is there anyone who uses g726-32 ? If not, then does anyone know why don't people use it? I use g726 on iax links between systems and to teliax.com for LD calls. Have no idea if its -32 or what though. What ships with asterisk (in terms of g726) has been working very well for us with the exception of a period of time where all g726 calls via teliax were not usable. Teliax had to have had a problem or was playing around as that was the only iax link that had bad audio. Thanks Rich for the positive email about g726. Just FYI, (*) supports only g726-32 AFAIK so that's probably what you've been using. I don't have the worry of Teliax as I'd probably never be using them or at least not in the immediate/near future. Like I said, all I want to do is avoid usage of license fees, save bandwidth, and not stress out my systems with cpu intensive codecs like g729 and maybe find something that can still deliver comparable quality. I'm still confused as to why more people and carriers don't use g726 however. I can only guess that many itsp's actually support it, but don't advertise its availability, just like they don't advertise ilbc, etc. I'd also have to guess that phone manufacturers haven't implemented it (in the past) due to limits on memory, etc. There are actually two conflicting methods of packing G.726-32 samples into bytes. RFC 3551 has this to say: Note that the little-endian direction in which samples are packed into octets in the G726-16, -24, -32 and -40 payload formats specified here is consistent with ITU-T Recommendation X.420, but is the opposite of what is specified in ITU-T Recommendation I.366.2 Annex E for ATM AAL2 transport. A second set of RTP payload formats matching the packetization of I.366.2 Annex E and identified by MIME subtypes AAL2-G726-16, -24, -32 and -40 will be specified in a separate document. That's all well and good, but there are some phones out there that pack samples into RTP payloads using the AAL2 direction. This causes interop nightmares (i.e. your phones talk G.726-32, someone elses phones talk G.726-32, but it sounds rubbish when you attempt a conversation). I would guess that this might be why people avoid the G.726 codec. ___ --Bandwidth 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] Xorcom Astribank
On Mon, Sep 18, 2006 at 09:15:11AM +0200, Klaus Darilion wrote: Hi! Does some one of you have practical experience with the Astribank channel bank I'm well familiar with it, sure. ;-) (I've tried to contact xorcom directly but didn't received any answer)? It seems that indeed we have not recieved your mail. Anyway, you're always more than welcome to contact [EMAIL PROTECTED] . We plan to use an Asteribank-16 channel bank to connect analog fax devices to Asterisk (to get rid of buggy Fax over G.711 with SIP terminal adapters) Does Astribank support fax devices (as fax is sensitive against jitter, delay, synch ...)? Yes. Are there special configuration issues when dealing with fax devices? First off, be sure to disable echo cancelling. (echocancel=no for the channel of the fax in zapata.conf). That depends on where the fax is coming from (or going to). If it is coming from an FXO extension on a Astribank FXS/FXO device, you have no problems. If it's from a different zaptel device: there may be some sync problems. For a normal call those may be clicks. For a fax those will basically terminate the fax. I'm working on a patch to Zaptel and to our driver to solve this. If it is from a SIP/IAX peer, then basically see http://www.voip-info.org/wiki/view/FoIP and the many messages on this list about this subject. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disconnect code in featuremap doesn't work on unanswered calls
I have the following in features.conf [featuremap] disconnect = ### ; Disconnect yet when a call is Dial()ed with the H option, ### doesn't disconnect the call when the far end hasn't yet answered, but * does. (### disconnects calls properly after the call is answered.) I even tried changing the default disconnect code in res_features.c and recompiling and it didn't change a thing. :( Any other ideas on how to get it to work consistently? -SC -- Stanley Cline // aka roamer1 // sc1 at roamer1 dot org // www.roamer1.org it seems like all you ever buy is Abercrombie and cell phones --a friend ___ --Bandwidth 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] rxfax, spandsp and lack of ecm
I'm using snapshot 20060915 for days and it's much better than before. Still have some missing lines might related to bad quality line. Thanks again! bye, Zsolt On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote: Hi Bruce, Looks like your typing is as bad as mine :-) Try http://www.soft-switch.org/downloads/snapshots/spandsp Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Xorcom Astribank
On Mon, 18 Sep 2006, Klaus Darilion wrote: Does some one of you have practical experience with the Astribank channel bank (I've tried to contact xorcom directly but didn't received any answer)? We plan to use an Asteribank-16 channel bank to connect analog fax devices to Asterisk (to get rid of buggy Fax over G.711 with SIP terminal adapters) We have two Astribank-8 units. We couldn't get our fax machine to work with it (it didn't like the echo), and the echo on analogue calls can be a bit bad at times (especially far end echo). Xorcom are supposed to be working on the echo issue, but the last drivers they supplied us didn't fix it completely, and we haven't heard from them in months about versions with better echo cancel. If you can get one on a trial, I'd say give it a whirl and see if the echo problems affect your calls or not. You might get lucky, otherwise you might have more luck getting working echo cancel drivers out of Xorcom... Nick ___ --Bandwidth 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
Re: [asterisk-users] Why not g726-32?
That's all well and good, but there are some phones out there that pack samples into RTP payloads using the AAL2 direction. This causes interop nightmares (i.e. your phones talk G.726-32, someone elses phones talk G.726-32, but it sounds rubbish when you attempt a conversation). I would guess that this might be why people avoid the G.726 codec. Interesting, maybe the reasons you and Rich stated might be some of the reasons I suppose. Thankfully neither of these will affect us since all the voip gateways/IADs and phones will be distributed and certified by us and BYOD type of a scenario will be highly discouraged PLUS I'm thinking of using g726 only when people want to interact with *. Every other time they'll be using g711 or g729 for off-net calls. This topic is still open, if anyone else has some interesting comments about it :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pickup call little complicated
Hi allI have a rather complex problemI would like to steal a call on asterisk but on like pickup application does or steal application.exampleI have 2 SIP extensions 100 and 101100 is calling na number like XX and channel is ringing or picked up now i want to be able to steal the call from extension 100 with extension 101 and talk to number XXX. Is that possible?greetingsmk ___ --Bandwidth 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] Variable that gives the SIP channel
Hi, I have a dialplan code to flash hook from a SIP phone. Everything works great except that it requires the SIP phone to have 2 lines since when the call comes back after the dialplan flash hook, the 1st line instance on the SIP (softphone) is still active. What I would like to do is in my flash hook dialplan code to ass something like Hangup(SIP/100-fe65), but where can I get that SIP/100-fe65 ? Is there a variable set with this information available in the dialplan ? Andre Courchesne ___ --Bandwidth 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] Problem with Asterisk Realtime (MySQL)
I have installed and configured Asterisk and that's working without problems in static mode. To use Realtime, i have created a table (sipuser) and put this line in extconfig.conf sipusers = mysql,asterisk,sipuser The connection to the database is alive. bechtoldsheim*CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username asterisk for 10 minutes, 34 seconds. 2006-09-18 14:36:25 DEBUG[23680]: res_config_mysql.c:674 mysql_reconnect: MySQL RealTime: Everything is fine. There is one record in the table sipuser, with the name 'edwin'. nameedwin contextedwin hostdynamic typefriend deny0.0.0.0/0.0.0.0 permit10.0.0.37/255.255.255.255 But when i try to register the sipuser 'edwin' with kphone (KDE SIP phone), the following error is shown in the Asterisk log. 2006-09-18 14:34:43 NOTICE[23672]: chan_sip.c:11084 handle_request_register: Registration from 'Edwin Pauli sip:[EMAIL PROTECTED]' failed for '10.0.0.37' - Username/auth name mismatch This information is in sip.conf (the same as what i have insert into the database). [edwin] type=friend host=dynamic context=edwin deny=0.0.0.0/0.0.0.0 permit=10.0.0.37/255.255.255.255 What is the problem? -- Edwin ___ --Bandwidth 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] pickup call little complicated
there 3 things you can do:1) transfer the call from 100 to 111 once the called to XXX is answered.2) park the call from 100, and pick up from 1113) with * version 1.4 you can configure 100 and 111 as a sla group and pickup the call at either location using the hold button mikeOn 9/18/06, Miloš Kocbek [EMAIL PROTECTED] wrote: Hi allI have a rather complex problemI would like to steal a call on asterisk but on like pickup application does or steal application.exampleI have 2 SIP extensions 100 and 101100 is calling na number like XX and channel is ringing or picked up now i want to be able to steal the call from extension 100 with extension 101 and talk to number XXX. Is that possible?greetingsmk ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable that gives the SIP channel
What I would like to do is in my flash hook dialplan code to ass
something like Hangup(SIP/100-fe65), but where can I get that
SIP/100-fe65 ? Is there a variable set with this information available
in the dialplan ?
${CHANNEL}
have a look here : http://www.voip-info.org/wiki-Asterisk+variables
hth
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Re: [asterisk-users] Polycom Expansion Module
Poly 2.0.1 says it can do 48 On Sep 17, 2006, at 8:06 PM, Douglas Garstang wrote: As far as I know, it's 12. -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Sun 9/17/2006 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Polycom Expansion Module Hi Kevin - Has anyone used the Polycom expansion module with multiple lines? My application is for 20 lines and read there was a limit of 7 at one point. I heard rumors that the newest version of the polycom sip firmware (2.01) would lift the limit of 7. It just came out, and I haven't had time to test it yet, but you can give it a try. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playtones
I have auto attendant menu. When calling person dials one number one extension
rings. Problem is that while extension rings caller doesn't hear ringing. I
understand that caller doesn't hear ringing because phone call is already
established, but I need to tell to caller that extension is ringing. How to
do that?
My extensions.conf
[incoming]
exten = s,1,Answer
exten = s,n,ResponseTimeout(5)
exten = s,n,Playback(mymessage,skip)
exten = s,n,Background(mymessage2)
exten = s,n,Background(silence/3)
exten = _7XX,1,Goto(local,${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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Re: [asterisk-users] pickup call little complicated
2006/9/18, mike pham [EMAIL PROTECTED]: 3) with * version 1.4 you can configure 100 and 111 as a sla group and pickup the call at either location using the hold button Hi Mike,Have you tried yourself SLA already ?Would you advise someone to use it on a production server now ?Regards ___ --Bandwidth 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] User authentication
How does one configure user authentication on asterisk . ___ --Bandwidth 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 Design Question
Hi, Right now I am in the process of setting up an asterisk box. I was thinking of having two asterisk box, one that is hooked up to the PSTN using a digium TE405P card and the other asterisk box will be used to store all the sip user features and routing information. Do you think this a good design? Or do you think I should just stick with having one asterisk box that does everything. I plan on having a lot of users hooked up to it in the future. The system specs are 3.0 GHz Pentium 4, 1 GB RAM, and a 40 GB hard drive. Thanks, Remi ___ --Bandwidth 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] User authentication
go here http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate and read ;) Siqhamo Sifo schrieb: How does one configure user authentication on asterisk . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playtones
what about this?
show app ringing?
[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,Ringing
exten = _7XX,2,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
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[asterisk-users] Chanspy crashing the server, again
I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was once or twice per week, now EVERY TIME someone uses chanspy it crashes the machine. Anyone with any ideas? This is killing me. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth 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] Chanspy crashing the server, again
Are you sure you deleted all the old asterisk modules when upgrading? On 9/18/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was once or twice per week, now EVERY TIME someone uses chanspy it crashes the machine. Anyone with any ideas? This is killing me. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth 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] Chanspy crashing the server, again
Iam experiencing the same problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Monday, September 18, 2006 10:03 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Chanspy crashing the server, again I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was once or twice per week, now EVERY TIME someone uses chanspy it crashes the machine. Anyone with any ideas? This is killing me. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth 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] Chanspy crashing the server, again
Title: Re: [asterisk-users] Chanspy crashing the server, again Anybody have a backtrace? On 9/18/06 9:23 AM, Richard [EMAIL PROTECTED] wrote: I am experiencing the same problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, September 18, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Chanspy crashing the server, again I upgraded to 1.2.12.1 - the problem is much, MUCH worse. Before, it was once or twice per week, now EVERY TIME someone uses chanspy it crashes the machine. Anyone with any ideas? This is killing me. Check Out the new free AIM(R) Mail http://pr.atwola.com/promoclk/100122638x1081283466x1074645346/aol?redir=http%3A%2F%2Fwww%2Eaim%2Ecom%2Ffun%2Fmail%2F -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Texter ___ --Bandwidth 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: Mediatrix 1204 trix
Thank you, C F and Florian. Now I must expose my ignorance about SIP and Mediatrix... I've adapted my sip.conf to essentially conform with what you've posted. So when I restart the Asterisk server, ethereal indicates that a NOTIFY goes to the Mediatrix (at 192.168.20.188), which responds with a 481, resulting in this message: -- Got SIP response 481 Subscription does not exist back from 192.168.20.188 My guess is that I'm missing a piece of the puzzle on the Mediatrix side of the configuration. Similarly, I've configured the Mediatrix via snmpset commands such that: telephonyAttributesAutomaticCallEnable[*] = 1 and telephonyAttributesAutomaticCallTargetAddress[*] = my desired extension(s) When I call the Mediatrix from POTS, it sends INVITE to Asterisk with the appropriate extension, but Asterisk responds with 404. I think I'm missing something involving REGISTER, but I'm foggy... would somebody clear the haze, please? In my floundering, I tried putting this into sip.conf: register = [EMAIL PROTECTED]/441 But the Mediatrix was unimpressed, rebuffing my entreaty with a: 405 Method Not Allowed I don't take rejection well, and so I'm loathe to speak with the Mediatrix again. I really need someone wiser to advise me... Message: 15 Date: Sat, 16 Sep 2006 21:59:34 -0400 From: C F [EMAIL PROTECTED] Subject: Re: [asterisk-users] Mediatrix 1204 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I have the same setup as Florian, however I have dtmfmode set to rfc instead of inband On 9/16/06, Florian Overkamp [EMAIL PROTECTED] wrote: Bill Michaelson wrote: Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204? Ours works with: [mtrix1] type=peer host=172.28.4.46 mask=255.255.255.255 context=in-mtrix1 qualify=no canreinvite=no dtmfmode=inband disallow=all allow=ulaw Best regards, Florian ___ --Bandwidth 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] FOP Installation help
I had FOP installed previously on [EMAIL PROTECTED] which was working perfectly fine until today when I decided to upgrade it to its ver. 2.6 I ended up losing my old FOP and new one no success. First I backed up /var/www/html/panel to /var/www/html/panel_old. But after no sucess with new FOP, moved the files back to original location. But now the errorI get is that it doesn't show all the extensions, and no trunks and queues at all. Please help me how to fix this problem. I didn't change anything in these files at all, then why all of a sudden they've stopped working.-- Zeeshan A Zakaria ___ --Bandwidth 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-dev] open letter
Hi all, Frank, this were my first thougts, why the hell does he say something like this on asterisk and doues not say whats wrong! I've spent some time to have both asterisk and ser working together. In my last post I asked how asterisk could send invite to a outboundproxy (SER) because of a 491 request pending was sent back to the caller . I use asterisk svn-trunk so i posted the mail to users and dev . I'm not convinced, that asterisk is perfect, as Mr Gaillac is as well. but i try to make it better day by day, doing bug reports and even writing own code (func_math). I'm not a member of the asterisk community that long (5 month or sth like this) but I enjoy asterisk as much as i hate it ;) The Asterisk project is good so Dear Mr Harry Gaillac, don't let us die without knowing why you left us ;) plz tell us whats wrong with asterisk. Kind regards Kai Ober Harry ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unable to change the emailbody for email notification
Hi all,
the default message for email notification looks like:
New 0:09 long msg in box 2001
from XliteUser2002, on Monday, September 18, 2006 at
04:24:11 PM
i try to change it with emailbody= but i always get
the default message body.
my voicemail.conf looks like
[general]
format=wav49|gsm|wav
attach=no
maxmessage=180
maxgreet=60
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emailbody=Dear ${VM_NAME}:\n\n\t you were just left a
${VM_DUR} long message (number ${VM_MSGNUM})\nin
mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on
${VM_DATE} \n
[default]
2001 = 2001,2001,Coco Richard,[EMAIL PROTECTED]
Is there something wrong with my config?
thx in advance
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[asterisk-users] Chanspy crashing the server, again
Yes, I deleted the old modules. All the modules come up listed as the date of the upgrade. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth 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] A1200+fxo, anyone using this?
Nick, I use one and it works just fine for me with 2 FXO and 2 FXS at the moment. I would say it is a great board to have and experiment with and as you say not too big or too small. Peter Nick Ellson wrote: I know it's not a digium product, but the 12 port A1200P card with a single FXO module at pbxeq.com at first glance would seem to be the way to get started for me with an in-system controller card. 4 ports seems too small for expansion, the huge 24 port card a tad too big (and spendy). So has anyone used this card with Asterisk? I googled for reviews and have not found anything, and I am tryingto find a way to search the archives without looking at each month one at a time. Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom SoundPoint 2.0.1 SIP firmware?
Hi, all -- since the new 2.x firmware seems to support NAT -- and, since I'm not an authorized dealer -- I'm kind of wondering if anyone knows where I can get it. freedomphones.net/polycom/files/ only goes up to 1.6.7. If anyone can either mail it to me, or mail me a link, I'd certainly be appreciative. Thanks! -Ken ___ --Bandwidth 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] Chanspy crashing the server, again
Do you have a backtrace so we can see where it crashed and have you reported a bug with the backtrace?JoshuaColpSoftwareDeveloperDigium,Inc.- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Monday, September 18, 2006 11:45:14 AM GMT-0800Subject: [asterisk-users] Chanspy crashing the server, again Yes, I deleted the old modules. All the modules come up listed as the date of the upgrade. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth 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] Chanspy crashing server, again
Here is the log excerpt from rright before the crash: Sep 18 09:58:58 NOTICE[23492] app_chanspy.c: Attaching SIP/403811-0a271260 to SIP/403415-0a2d8020 Sep 18 09:59:17 NOTICE[23492] app_chanspy.c: Attaching SIP/403811-0a271260 to SIP/403425-0a2f0cf8 Sep 18 09:59:53 WARNING[23698] pbx.c: Timeout, but no rule 't' in context 'qa' Sep 18 09:59:55 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403429-0a18 Sep 18 10:00:03 WARNING[23689] pbx.c: Timeout, but no rule 't' in context 'custcare' Sep 18 10:01:57 WARNING[23977] pbx.c: Timeout, but no rule 't' in context 'custcare' Sep 18 10:02:19 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403423-0a37c0e0 Sep 18 10:02:48 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403416-0a384540 Sep 18 10:05:40 NOTICE[23492] app_chanspy.c: Attaching SIP/403811-0a271260 to SIP/403404-0a1e6208 Sep 18 10:06:36 WARNING[23492] file.c: Failed to write frame Sep 18 10:10:09 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403403-0a1e6208 Sep 18 10:11:12 WARNING[23339] channel.c: Avoided initial deadlock for '0xb7a1c4f8', 10 retries! Sep 18 10:11:57 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403423-b7ad3d58 Sep 18 10:12:33 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403414-b7a9d748 Sep 18 10:13:39 NOTICE[23727] app_chanspy.c: Attaching SIP/403419-0a319718 to SIP/403426-0a49d028 Sep 18 10:14:49 WARNING[26819] pbx.c: Timeout, but no rule 't' in context 'CancelSave' Sep 18 10:15:05 ERROR[23351] chan_sip.c: We could NOT get the channel lock for SIP/403101-b7844c38! Sep 18 10:15:05 ERROR[23351] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK Sep 18 10:15:05 ERROR[23351] chan_sip.c: BAD! BAD! BAD! The problem I have withthis though, is that 403101 was not being monitored or monitoring a call. There must be something else. I downgraded back to 1.2.7.1, everything seems to be stable again. I am going to test chanspy later and see if it works at all. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth 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 7940 Problem (Mess)
I know this isn't directly Asterisk related. - But I do appreciate the responses I get from you folks. - At least I don't get flamed like I've seen on the Java Perl-Mod lists (geesh!). Okay - Here's what I've got. A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll work on my Asterisk box (outside of the FXO FXS modules on the TDM card in the Asterisk server, I only run SIP on the hardphones). I don't know the phone's password (sound familiar?). - Have tried everything, cisco, *##, etc Nothing works. So I figured I would set up a .cnf or .tlv (or whatever file MGCP looks for) with a one-line entry something like 'phone_password:cisco' then let the phone TFTP the file and reset it's password. - Then I can get into the 'innards' and set up my local stuff so I can upgrade/reload the phone with SIP and get it ready for my Asterisk box. - Sound do-able? This is going to be an exercise in 'Networking' for sure... The only catch is that per the phone's network settings: The phone uses a static IP of something like 192.168.0.220 with a Gateway of 192.168.0.1. - Standard class 'C' netmask (255.255.255.0). The phone has DHCP DISABLED. The phone has it's TFTP server set to something like 62.120.xxx.xxx (something completely outside of the local network). My home LAN uses 10.0.0.xxx on the local side. But I can reset my XP-Box to 192.168.0.99 and ping the phone with no probs. But If I set my XP-Box to a static IP of the Phone's TFTP server, my 'SolarWinds' TFTP server sees nothing. - Figured as much as the phone's IP and my TFTP server's IP are not in the same 'net. Bottom line. - I've got to figure out a way to build a 'mini-network' so the phone'll be happy but also set up a PC TFTP with the same address that's set in the phone. - Perhaps I can fake-out the phone into thinking it's hitting a TFTP box on the Internet (). I've got plenty of PC's even a spare Linksys router. Any ideas. - Or is there something simpler. - Or have I just bought an expensive paper-weight? Thanks in advance. Gary Guthary [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A1200+fxo, anyone using this?
Thanks for the feedback Peter, I am going to try one with an FXO and then one of the $30 fixed port single FXO PCI cards from pbxeq.com as well. See if there is a real difference there. Anybody try the A-100PCI card? When I do, I'll post what I find. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Mon, 18 Sep 2006, Peter Lindquist wrote: Nick, I use one and it works just fine for me with 2 FXO and 2 FXS at the moment. I would say it is a great board to have and experiment with and as you say not too big or too small. Peter Nick Ellson wrote: I know it's not a digium product, but the 12 port A1200P card with a single FXO module at pbxeq.com at first glance would seem to be the way to get started for me with an in-system controller card. 4 ports seems too small for expansion, the huge 24 port card a tad too big (and spendy). So has anyone used this card with Asterisk? I googled for reviews and have not found anything, and I am tryingto find a way to search the archives without looking at each month one at a time. Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pickup call little complicated
I'm testing it against the Aastra 9133i atm. I am trying to rely on the 9133i native BLA/SLA over Asterisk. The hard part is trying to figure out how to do it atm.MikeOn 9/18/06, Olivier [EMAIL PROTECTED] wrote: 2006/9/18, mike pham [EMAIL PROTECTED]: 3) with * version 1.4 you can configure 100 and 111 as a sla group and pickup the call at either location using the hold button Hi Mike,Have you tried yourself SLA already ?Would you advise someone to use it on a production server now ?Regards ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Mediatrix 1204 trix
Keep in mind that the Mediatrix does not support register (AFAIK, anyhow). You have to create a static entry in sip.conf that has host set to the IP address of the Mediatrix On 9/18/06, Bill Michaelson [EMAIL PROTECTED] wrote: Thank you, C F and Florian. Now I must expose my ignorance about SIP and Mediatrix... I've adapted my sip.conf to essentially conform with what you've posted. So when I restart the Asterisk server, ethereal indicates that a NOTIFY goes to the Mediatrix (at 192.168.20.188), which responds with a 481, resulting in this message: -- Got SIP response 481 Subscription does not exist back from 192.168.20.188 My guess is that I'm missing a piece of the puzzle on the Mediatrix side of the configuration. Similarly, I've configured the Mediatrix via snmpset commands such that: telephonyAttributesAutomaticCallEnable[*] = 1 and telephonyAttributesAutomaticCallTargetAddress[*] = my desired extension(s) When I call the Mediatrix from POTS, it sends INVITE to Asterisk with the appropriate extension, but Asterisk responds with 404. I think I'm missing something involving REGISTER, but I'm foggy... would somebody clear the haze, please? In my floundering, I tried putting this into sip.conf: register = [EMAIL PROTECTED]/441 But the Mediatrix was unimpressed, rebuffing my entreaty with a: 405 Method Not Allowed I don't take rejection well, and so I'm loathe to speak with the Mediatrix again. I really need someone wiser to advise me... Message: 15 Date: Sat, 16 Sep 2006 21:59:34 -0400 From: C F [EMAIL PROTECTED] Subject: Re: [asterisk-users] Mediatrix 1204 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I have the same setup as Florian, however I have dtmfmode set to rfc instead of inband On 9/16/06, Florian Overkamp [EMAIL PROTECTED] wrote: Bill Michaelson wrote: Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204? Ours works with: [mtrix1] type=peer host=172.28.4.46 mask=255.255.255.255 context=in-mtrix1 qualify=no canreinvite=no dtmfmode=inband disallow=all allow=ulaw Best regards, Florian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Design Question
Remi Quezada wrote: Hi, Right now I am in the process of setting up an asterisk box. I was thinking of having two asterisk box, one that is hooked up to the PSTN using a digium TE405P card and the other asterisk box will be used to store all the sip user features and routing information. Do you think this a good design? Or do you think I should just stick with having one asterisk box that does everything. I plan on having a lot of users hooked up to it in the future. The system specs are 3.0 GHz Pentium 4, 1 GB RAM, and a 40 GB hard drive. I attended a cisco presentation a while back and they indicated the architecture of their system was changing somewhat (away from Windows, now on Linux, etc). The presentation suggested that certain functions are dedicated to certain systems/boxes, and if one needed more of a certain function then add another box. For example, if transcoding is a requirement, then dedicated a box or two to that function. As the overall system grows and more transcoding is needed, add another box for that. Since I'm not a cisco reseller, etc, I didn't keep very many notes relative to the above. But, the approach seems to be one that can support long term growth is small increments of hardware/software. Your approach kind of follows cisco's in a way. The only issue (from a high level) that might be difficult to handle is that asterisk really wasn't designed to distribute functions to multiple boxes. E.g., if growth dictated two pstn interface boxes, how does one manage the distribution of pstn calls from a single routing box (including pstn T1 failures, overloads, etc)? ___ --Bandwidth 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] is chanisavail command reliable?
If you have a PRI, why are you telling asterisk which channel to use?
why don't you set groups in zapata.conf and just use g1 or the like?
On 9/18/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I have an Asterisk box on a Debian Sarge distro with a Sangoma a102 PRI
card and I have a big problem: Asterisk drops a lot of outbound calls.
I do not use groups, I want to use a free channel given from Asterisk.
What I do is:
1) use chanisavail command to ask Asterisk for a free channel to use
2) use that channel to dial outbound calls
Asterisk gives me the free channel but when I make a call Asterisk tells
me the channel is not available (I check variable ${AVAILSTATUS}). I
checked but that channel is not busy on another call.
How can it be possible? Is there anybody who can help me?
TIA
Giorgio Incantalupo
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[asterisk-users] Dial and Timeout
Hi, we have experienced som troubles with the timeout option of the Dial-App. It seems the Dial startts counting down the timeout imediatly, but there are great differences when the called phone actually starts ringing. If i call a landline phone in my own country it is nearly the same, but if i want to call a cell phone it could take up to 10 secs until the phone begins to ring. So, if i want to dial with a timeout of 30 secs until i want to do something else, a landline user has 30 seconds of time to answer the call, but the cell user has only 20 seconds, more or less. In use cases where i don't know what type of phone is to be called it becomes quite difficult to set an appropriate timeout. Is there someway to get Dial() to start the countdown, when the channel state changes to ringing ? Thx for any advice, Tobias Wolf ___ --Bandwidth 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] A1200+fxo, anyone using this?
On Mon, Sep 18, 2006 at 08:22:03AM -0700, Nick Ellson wrote: Thanks for the feedback Peter, I am going to try one with an FXO and then one of the $30 fixed port single FXO PCI cards from pbxeq.com as well. See if there is a real difference there. Anybody try the A-100PCI card? When I do, I'll post what I find. As for those cards: read about X100P . There are some major differences. Openvox's A100P card is basically a clone of Digium's obsolete X100P, which is basically an old winmodem card. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Rates
Im going to get a trial account, .014 to US is not bad at all!Only downside is that g729 is only codec they allow :(On 9/17/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I saw this termination company, www.BuyMin.comthe rates looks good. Has anyone any experience with this company? I use Gafachi, very reliable but expensive. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LDAP athentication
Hello, Does anyone have a solution for having SIP users to authenticate against a LDAP server?Best Regards,Andre O. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] Polycom SoundPoint 2.0.1 SIP firmware?
Hello, have a look here . ftp://nxs.yi.org Harry --- Ken D'Ambrosio [EMAIL PROTECTED] a écrit : Hi, all -- since the new 2.x firmware seems to support NAT -- and, since I'm not an authorized dealer -- I'm kind of wondering if anyone knows where I can get it. freedomphones.net/polycom/files/ only goes up to 1.6.7. If anyone can either mail it to me, or mail me a link, I'd certainly be appreciative. Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic
On Saturday 16 September 2006 20:35, Tzafrir Cohen wrote: On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote: I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I get this : laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel # modprobe zaphfc FATAL: Error inserting zaphfc (/lib/modules/2.6.13-15.11-smp/misc/zaphfc.ko): Invalid module format and this in dmesg : zaphfc: version magic '2.6.13-15.11-smp gcc-4.0' should be '2.6.13-15.11-smp SMP gcc-4.0' This means you built zaptel with the wrong kernel headers. Is there a SUSE power user in the crowd? From memory as I do not use SUSE anymore.. Suse already has the zaptel modules in its kernel under the (IIRC) the extra directory (instead of misc) . you end up with 2 sets of zaptel modules in the linux module tree and modprobe then gets confused as to which to load. Look for a directory /lib/modules/2.6.13-15.11-smp/extra You must (again from memory) delete the old modules abd rerun depmod or rebuild your asterisk Paul ___ --Bandwidth 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] ASTERFAX
Hi All Anyone have any knowledge of using the above? I have installed it and wish to send out a fax via a SIP channel on an ISDN3O. I have used the test script which says it has gone though ok but i never see any activity in asterisk. Checking the /var/spool/asterfax/tmp directory i can see all the faxes sitting there? Can anyone help of point me to some config to look at? Many Thanks SP ___ --Bandwidth 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] Polycom SoundPoint 2.0.1 SIP firmware?
It's up on http://www.freedomphones.net/polycom/files/ now with release notes. MATT--- On 9/18/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote: Hi, all -- since the new 2.x firmware seems to support NAT -- and, since I'm not an authorized dealer -- I'm kind of wondering if anyone knows where I can get it. freedomphones.net/polycom/files/ only goes up to 1.6.7. If anyone can either mail it to me, or mail me a link, I'd certainly be appreciative. Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is chanisavail command reliable?
On Mon, 2006-09-18 at 10:53 +0200, Giorgio Incantalupo wrote: Hi, I have an Asterisk box on a Debian Sarge distro with a Sangoma a102 PRI card and I have a big problem: Asterisk drops a lot of outbound calls. I do not use groups, I want to use a free channel given from Asterisk. What I do is: 1) use chanisavail command to ask Asterisk for a free channel to use 2) use that channel to dial outbound calls I think this is not a good idea since an inbound call could arrive between 1) and 2) making the channel no longer available (the app isn't named ChanIsAvailAndReserveForOutgoingCall). Use ZAP/g1 (or whatever your group is) - works for me on PRI (both as net and as cpe) - depending on the behaviour of the other side ZAP/G1 could reduce the likelihood of the same channel being used by both sides. -- Dr. Michael Neuhauser mailto:[EMAIL PROTECTED] Firmix Software GmbH sip:[EMAIL PROTECTED] Vienna/Austria/Europe tel:+43-1-7890849-30 Linux Development and Services http://www.firmix.at/ ___ --Bandwidth 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] Changes in extensions.conf handling between 1.2 1.4
I've always had ignorepat = 9 followed by _9. in my outgoing context now if I use this in 1.4 it jumps straight to BUSY. I also have _70. and _80. which make calls through other * servers, they work as expected. What is really surprising is that _90. also jumps straight to BUSY. Any ideas? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is chanisavail command reliable?
On Monday 18 September 2006 04:53, Giorgio Incantalupo wrote:
What I do is:
1) use chanisavail command to ask Asterisk for a free channel to use
2) use that channel to dial outbound calls
The problem is that a race condition exists; the channel could be available at
time (1), but something could snag it at time (2). There (currently) is no
way to lock a channel after it's been detected as free so that something
later could use it.
My suggestion isn't to use ChanIsAvail() at all; simply Dial and catch
${DIALSTATUS}, like so:
exten = foo,1,Dial(Tech/exten,,g)
exten = foo,n,Goto(foo-${DIALSTATUS},1)
exten = foo-CHANUNAVAIL,1,NoOp(Nope, wasn't available, gotta do something
else...)
exten = foo-BUSY,1,NoOp(busy...)
exten = foo-NOANSWER,1,NoOp(no answer...)
that way there is no race condition. What would maek this better is if you
could group a bunch of channels, irrespective of technology, together and
dial that group... something like this
(assume a groups.conf)
1 = SIP/500, IAX2/[EMAIL PROTECTED]/${EXTEN}, Zap/g1
2 = Zap/1,Zap/3,Zap/5
3 = SIP/[EMAIL PROTECTED]
and then do something like this:
exten = _NXXNXX,1,Dial(GROUP/1,,g)
which would dial SIP/500, IAX2/[EMAIL PROTECTED]/9165551212 and the first
available
channel in the Zaptel group 1... (assuming 9165551212 was what matched)...
variable expansion would be crucial in this kind of design. :-)
-A.
Asterisk gives me the free channel but when I make a call Asterisk tells
me the channel is not available (I check variable ${AVAILSTATUS}). I
checked but that channel is not busy on another call.
How can it be possible? Is there anybody who can help me?
TIA
Giorgio Incantalupo
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[asterisk-users] X100P and zaptel 1.2.8
Hello list. I have Asterisk installed on my small home Linux machine, and it was working with softphones. I later got a X100P card (I think it was a clone, buit Asterisk saw it as a real X100P). I then got side-tracked and the hard drive later failed in the machine. I've replace the whole machine, and installed Slackware 10.2. I've re-compiled the kernel with 2.4.33, and I'm having a problem compiling zaptel-1.2.8. During the make process I see many; warning: dereferencing type-punned pointer will break strict-aliasing rules But it doesn't actually fail. But when I attempt a make install I get depmod: *** Unresolved symbols in /lib/modules/2.4.33/misc/pciradio.o (where pciradio.o changes to tor2.0, torisa.o ... zadynamic.o) Am I missing an option in my kernel's compilation, or is there a package missing(I'm almost positve I selected everything, and it did work on my previous Slackware installation, but that might have been 10, or 10.1) Any help/pointers would appreciated. Tim ___ --Bandwidth 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] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic
On Sun, Sep 17, 2006 at 03:34:05PM +0200, Paul Hewlett wrote: On Saturday 16 September 2006 20:35, Tzafrir Cohen wrote: On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote: I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I get this : laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel # modprobe zaphfc FATAL: Error inserting zaphfc (/lib/modules/2.6.13-15.11-smp/misc/zaphfc.ko): Invalid module format and this in dmesg : zaphfc: version magic '2.6.13-15.11-smp gcc-4.0' should be '2.6.13-15.11-smp SMP gcc-4.0' This means you built zaptel with the wrong kernel headers. Is there a SUSE power user in the crowd? From memory as I do not use SUSE anymore.. Suse already has the zaptel modules in its kernel under the (IIRC) the extra directory (instead of misc) . you end up with 2 sets of zaptel modules in the linux module tree and modprobe then gets confused as to which to load. Look for a directory /lib/modules/2.6.13-15.11-smp/extra You must (again from memory) delete the old modules abd rerun depmod or rebuild your asterisk deleting files from a package is probably not smart and works agains rpm. Why not just remove the package altogether (rpm -e) ? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to make Polycom 501 go off hook when pressing any digits
I'm trying to make the Polycom 501 go off-hook (in speaker phone mode) when any digits is dialed and the handset hasnt been lifted. Isthis possible? Mike ___ --Bandwidth 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] Xorcom Astribank
Hi I just want to note some technical points here: On Mon, Sep 18, 2006 at 12:08:20PM +0100, Nick Burch wrote: On Mon, 18 Sep 2006, Klaus Darilion wrote: Does some one of you have practical experience with the Astribank channel bank (I've tried to contact xorcom directly but didn't received any answer)? We plan to use an Asteribank-16 channel bank to connect analog fax devices to Asterisk (to get rid of buggy Fax over G.711 with SIP terminal adapters) We have two Astribank-8 units. We couldn't get our fax machine to work with it (it didn't like the echo), and the echo on analogue calls can be a bit bad at times (especially far end echo). Some technical notes: the echo won't be a problem with faxes (bad echo cancelling may be). It is a problem for voice calls, indeed, and nees cancelling there. With faxes the problem is syncing between different zaptel devices. That is: a fax sent between an FXS posrt and an FXO port of a TDM400P card or between different ports of an Astribank FXS/FXO unit will get through. Faxes from a T1/E1 card to a fax on an analog port will have sync problems. Xorcom are supposed to be working on the echo issue, but the last drivers they supplied us didn't fix it completely, and we haven't heard from them in months about versions with better echo cancel. Again, please separate the faxes and the echo issues. We have done some work on the echo issue, and the result is already in our latest drivers. If you have any echo issues, please let us know so we could resove them. As for the issue of the fax: I'll keep you posted when we have something working. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fedora
Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fedora
We use asterisk on Fedora Core 3 without any issues in one of our installations. If you're after Redhat without the cost, though, you might check out CentOS as a distribution. :) Fedora can sometimes be a little bleeding edge when it comes to a production environment. N. On Mon, 18 Sep 2006 11:34:04 -0700 (PDT), bilal ghayyad wrote Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fedora
Yes. We use fedora as a test system before moving it onto RHEL. On Mon, 2006-09-18 at 11:34 -0700, bilal ghayyad wrote: Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth 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] Fedora
bilal ghayyad wrote: Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. If you want enterprise class software look at CentOS. http://www.centos.org But in answer to your question, Fedora should be supported. There are some quirks with Redhat distros so you might want to google for the fixes/issues before pulling your hair out. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fedora
I have been using asterisk on FC4, FC5 and now FC6t3 with no irritating problem. - Original Message - From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, September 18, 2006 8:34 PM Subject: [asterisk-users] Fedora Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fedora
On Mon, 2006-09-18 at 11:34 -0700, bilal ghayyad wrote: Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. check out Centos. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CSR introduces UniVox reference platform
I'm not anything to do with them, but sounds a nice design. CSR have introduced a VoWiFi reference design that costs around $20. The interesting thing is that it supports both SIP and IAX2. Maybe Digium should make a WiFi handset ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problemwith module versionmagic
- Original Message - From: Paul Hewlett [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, September 17, 2006 3:34 PM Subject: Re: [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problemwith module versionmagic On Saturday 16 September 2006 20:35, Tzafrir Cohen wrote: On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote: I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64 x86_64 x86_64 GNU/Linux and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. I get this : laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel # modprobe zaphfc FATAL: Error inserting zaphfc (/lib/modules/2.6.13-15.11-smp/misc/zaphfc.ko): Invalid module format and this in dmesg : zaphfc: version magic '2.6.13-15.11-smp gcc-4.0' should be '2.6.13-15.11-smp SMP gcc-4.0' This means you built zaptel with the wrong kernel headers. Is there a SUSE power user in the crowd? From memory as I do not use SUSE anymore.. Suse already has the zaptel modules in its kernel under the (IIRC) the extra directory (instead of misc) . you end up with 2 sets of zaptel modules in the linux module tree and modprobe then gets confused as to which to load. Look for a directory /lib/modules/2.6.13-15.11-smp/extra You must (again from memory) delete the old modules abd rerun depmod or rebuild your asterisk Thanks for the hint. I already did that. To me it seems that something is messed up with settings, so zaphfc compiles with slightly different version magic - note that the problem lies only in 'SMP' : '2.6.13-15.11-smp gcc-4.0' vs '2.6.13-15.11-smp SMP gcc-4.0' That's weird for me Any further help ? Thanks in advance, regards, Rob. ___ --Bandwidth 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] how to transfer a caller out of a queue ?
Hi, I would like to give a caller the chance to leave a queue after an agent has already accepted the call. The caller enters the queue by dialing 333: [from-sip] exten = 300,1,Answer() exten = 300,2,Queue(q1|tT) When the caller presses # and e.g. 1, asterisk is looking for this extension in the context where the call came in. In my configuration this means, that my office phone is ringing: exten = 1,1,Answer() exten = 1,2,DIAL(CAPI/@8304499:8304498,30,tTr) exten = 1,3,Hangup But in this case not the caller, but the agent has been transferred! Isn't there a chance for the caller to stop the conversation e.g. because the agent told him that he has called the wrong queue and that he should dial #1 to get to the right queue or directly to another person? If the agent does this, the caller get's transfered to the office phone, as expected. As far as I understand the documentation, the context that is assigned to a queue in queue.conf is only valid before an agent has accepted the call. I'm still running Asterisk 1.0.6, which is the current version for SuSE 9.3. Maybe Asterisk 1.2.x would help? Thanks for help hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth 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] INSTALL_PREFIX=
I am looking to setup asterisk on a DRBD distributed volume. The overall goal is to setup a redundant server that will kick in when the primary fails. So I am thinking of just setting up a directory like /drbd (or /mirror) and install the entire asterisk application (logs, var run, etc) all on /drbd by compiling with INSTALL_PREFIX=/drbd. The only thing I would put on the actual server and not get replicated is a couple init scripts and zaptel configs. I am going to try and setup hearbeat to start asterisk on the secondary server when the primary fails. I am hoping that this will help with sharing issues like run pid. Some downtime is OK. If the entire failover takes 5-10 minutes it's OK. My question is. Has anyone tried this? Does anyone for see any problems? Thanks for all the help this list has provided in the past... Forrest ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] how to transfer a caller out of a queue ?
can't the agent just transfer the caller to another extension, whether that be another queue or a person ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan-Michael. Guenther (in-put GbR) Sent: Monday, September 18, 2006 3:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to transfer a caller out of a queue ? Hi, I would like to give a caller the chance to leave a queue after an agent has already accepted the call. The caller enters the queue by dialing 333: [from-sip] exten = 300,1,Answer() exten = 300,2,Queue(q1|tT) When the caller presses # and e.g. 1, asterisk is looking for this extension in the context where the call came in. In my configuration this means, that my office phone is ringing: exten = 1,1,Answer() exten = 1,2,DIAL(CAPI/@8304499:8304498,30,tTr) exten = 1,3,Hangup But in this case not the caller, but the agent has been transferred! Isn't there a chance for the caller to stop the conversation e.g. because the agent told him that he has called the wrong queue and that he should dial #1 to get to the right queue or directly to another person? If the agent does this, the caller get's transfered to the office phone, as expected. As far as I understand the documentation, the context that is assigned to a queue in queue.conf is only valid before an agent has accepted the call. I'm still running Asterisk 1.0.6, which is the current version for SuSE 9.3. Maybe Asterisk 1.2.x would help? Thanks for help hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk / Audiocodes annoying issue - Seeking Suggestions
I have installed some AudioCodes analog gateways in conjunction with * and am having an annoying problem on the mp-118-fxs side. call quality is decent but on the mp118 you hear yourself in your ear with around 1/10 to 1/4 of a second delay. this is very annoying and makes it difficult to carry on a conversation. i have tried nearly every combination of codec including g711 a-law. i have tried every codec with and without silence suppression. even though it is now disabled, i have also increased the minimum buffer for jitter correction to 70ms and the jitter setting to 7 with no improvement in quality. i have tried setting the maximum echo buffer to the maximum amount 128ms but no improvement (i don't believe this affects anything anyways because by default if should go that far anyways). these sites are connected via an mpls vpn setup and the connection is very stable between sites. running a pingtest (ping every 50ms) i am getting consistent pings of between 71-72ms over a period of 10 minutes. that is why i have reduced the minimum jitter buffer to 0 and have set it to the very aggresive setting of 0 (remember i changed it to default of 70/7 with no improvement). it seems like the unit is trying to play back my voice in my ear like a regular phone would, but my voice is coming back to me only after it reaches the far end first. is there any way to prevent me from hearing my own voice in my ear at all or to filter that out? i do not hear an echo of the person coming from the MP114 FXO unit, i only hear my own voice delayed to myself. If anyone is using Audiocodes, and has corrected similar problems, please shoot me an email. Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / Audiocodes annoying issue - Seeking Suggestions
Cory That sounds like glare and could be caused by the AudioCodes. Check your amplification or line power as I am sure you are not powering 5 miles of copper. On 9/18/06, Cory Andrews [EMAIL PROTECTED] wrote: I have installed some AudioCodes analog gateways in conjunction with * and am having an annoying problem on the mp-118-fxs side. call quality is decent but on the mp118 you hear yourself in your ear with around 1/10 to 1/4 of a second delay. this is very annoying and makes it difficult to carry on a conversation. i have tried nearly every combination of codec including g711 a-law. i have tried every codec with and without silence suppression. even though it is now disabled, i have also increased the minimum buffer for jitter correction to 70ms and the jitter setting to 7 with no improvement in quality. i have tried setting the maximum echo buffer to the maximum amount 128ms but no improvement (i don't believe this affects anything anyways because by default if should go that far anyways). these sites are connected via an mpls vpn setup and the connection is very stable between sites. running a pingtest (ping every 50ms) i am getting consistent pings of between 71-72ms over a period of 10 minutes. that is why i have reduced the minimum jitter buffer to 0 and have set it to the very aggresive setting of 0 (remember i changed it to default of 70/7 with no improvement). it seems like the unit is trying to play back my voice in my ear like a regular phone would, but my voice is coming back to me only after it reaches the far end first. is there any way to prevent me from hearing my own voice in my ear at all or to filter that out? i do not hear an echo of the person coming from the MP114 FXO unit, i only hear my own voice delayed to myself. If anyone is using Audiocodes, and has corrected similar problems, please shoot me an email. Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / Audiocodes annoying issue - Seeking Suggestions
Try to decrease the gain value on the Audiocodes. That solved the echo issues we had. On 9/18/06, Cory Andrews [EMAIL PROTECTED] wrote: I have installed some AudioCodes analog gateways in conjunction with * and am having an annoying problem on the mp-118-fxs side. call quality is decent but on the mp118 you hear yourself in your ear with around 1/10 to 1/4 of a second delay. this is very annoying and makes it difficult to carry on a conversation. If anyone is using Audiocodes, and has corrected similar problems, please shoot me an email. -- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to transfer a caller out of a queue ?
Hi Rick, Am Montag, 18. September 2006 21:30 schrieb Rick Smith: can't the agent just transfer the caller to another extension, whether that be another queue or a person ? yes, that's the easy part. But my client wants the caller (!) to be able to transfer himself into another context. The reason for this is, that the caller must decide whether he wants to be transfered from a free support line to a support line for which he would have to pay. Stefan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan-Michael. Guenther (in-put GbR) Sent: Monday, September 18, 2006 3:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to transfer a caller out of a queue ? Hi, I would like to give a caller the chance to leave a queue after an agent has already accepted the call. The caller enters the queue by dialing 333: [from-sip] exten = 300,1,Answer() exten = 300,2,Queue(q1|tT) When the caller presses # and e.g. 1, asterisk is looking for this extension in the context where the call came in. In my configuration this means, that my office phone is ringing: exten = 1,1,Answer() exten = 1,2,DIAL(CAPI/@8304499:8304498,30,tTr) exten = 1,3,Hangup But in this case not the caller, but the agent has been transferred! Isn't there a chance for the caller to stop the conversation e.g. because the agent told him that he has called the wrong queue and that he should dial #1 to get to the right queue or directly to another person? If the agent does this, the caller get's transfered to the office phone, as expected. As far as I understand the documentation, the context that is assigned to a queue in queue.conf is only valid before an agent has accepted the call. I'm still running Asterisk 1.0.6, which is the current version for SuSE 9.3. Maybe Asterisk 1.2.x would help? Thanks for help hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth 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 for talking to other Asterisk machines
Just curious how most of you are defining SIP peers in sip.conf for Asterisk boxes talking to each other. Are most of you just making a type=friend connection and a single context or are you separating them out to in/out definitions and contexts? In other words Where voicegw1 is the Asterisk box with the TDM cards for talking to the PSTN, it will receive calls from the PSTN and forward to the appropriate Asterisk box as well as receive calls from the other Asterisk boxes to forward out to the PSTN. Do you on the Asterisk box that contains all the SIP phones define (ie the client to the PSTN Asterisk box and voicegw1 is the one with the PSTN connection) [voicegw1-in] type=user username=virtualpbx1-in secret=1234 host=192.168.1.99 context=voicegw1-in canreinvite=no nat=no qualify=yes allow=all [voicegw1-out] type=peer username=virtualpbx1-out secret=1234 host=192.168.1.99 context=voicegw1-out canreinvite=no nat=no qualify=yes allow=all or [voicegw1] Type=friend Blah Context=voicegw1 And use a single context for inbound/outbound routing? The same would apply to the PSTN Asterisk server. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FOP Installation help
Hi, You might need to recreate fop config files with amp tools. The script that does that is named retrieve_op_conf_from_mysql.pl. You might also have permissions problems on the directory or files inside them. They must be owned by user asterisk if I recall correctly. Good luck, On 9/18/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I had FOP installed previously on [EMAIL PROTECTED] which was working perfectly fine until today when I decided to upgrade it to its ver. 2.6 I ended up losing my old FOP and new one no success. First I backed up /var/www/html/panel to /var/www/html/panel_old. But after no sucess with new FOP, moved the files back to original location. But now the error I get is that it doesn't show all the extensions, and no trunks and queues at all. Please help me how to fix this problem. I didn't change anything in these files at all, then why all of a sudden they've stopped working. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth 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] ANI and Meetme...
Ok. First question is how to make it say my number back. Like if you call extension 1000 from extension 1001, I want it to say Number is 1,0,0,1 like an ANI number? Help. Also I want to setup a meetme conference so that it asks Enter conference number then execute meetme($entered_number) I feel dumb asking because these sound like they should be so easy, but I cant find any help with this. Thanks. Natambu Obleton Network Engineer FastTrack Communications [EMAIL PROTECTED] (970) 247-3366 office (970) 247-2426 fax ___ --Bandwidth 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] Cisco 7940 Problem (Mess)
You'll need to change your XP box to the default router of the phone (or, just change your XP to a static on your local net, then add a static ipunder advanced for the default route of the phone, that way you still have access to the internet from your XP box), and then add (under advanced) the IP address of the tftp server the phone is looking for. Since your XP box now knows about the local network of the phone, and the IP of the tftp server, it won't tell the phone to go elsewhere. That will at least get you to the point where the phone will try to access your tftp server. As far as the rest, you'll probably need to start out on Cisco's version 3 of SIP and keep upgrading it version by version (at least major version) until you get to the version you want to run. To get more info on this, you cansearch Cisco's website on upgrading from MGCP to SIP. I've done this a few times, but it seems like I had to try different things each time to get things working. There was nothing consistent enough for me to write down the steps I took. I had one phone take me most of the day, another I had pretty much given up on, and then suddenly it worked. Be persistent. Eventually you will get there. You will need a smartnet contract to do this, unless you already have access to all the firmware versions of the phone. On 9/18/06, Gary Guthary [EMAIL PROTECTED] wrote: I know this isn't directly Asterisk related. - But I do appreciate theresponses I get from you folks. - At least I don't get flamed like I've seen on the Java Perl-Mod lists (geesh!).Okay - Here's what I've got.A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'llwork on my Asterisk box (outside of the FXO FXS modules on the TDM card in the Asterisk server, I only run SIP on the hardphones).I don't know the phone's password (sound familiar?). - Have triedeverything, cisco, *##, etcNothing works.So I figured I would set up a .cnf or .tlv (or whatever file MGCP looks for) with a one-line entry something like 'phone_password:cisco' then let thephone TFTP the file and reset it's password. - Then I can get into the'innards' and set up my local stuff so I can upgrade/reload the phone with SIP and get it ready for my Asterisk box. - Sound do-able?This is going to be an exercise in 'Networking' for sure...The only catch is that per the phone's network settings:The phone uses a static IP of something like 192.168.0.220 with a Gateway of192.168.0.1. - Standard class 'C' netmask (255.255.255.0).The phone has DHCP DISABLED. The phone has it's TFTP server set to something like 62.120.xxx.xxx(something completely outside of the local network).My home LAN uses 10.0.0.xxx on the local side.But I can reset my XP-Box to 192.168.0.99 and ping the phone with no probs.But If I set my XP-Box to a static IP of the Phone's TFTP server, my'SolarWinds' TFTP server sees nothing. - Figured as much as the phone's IP and my TFTP server's IP are not in the same 'net.Bottom line. - I've got to figure out a way to build a 'mini-network' so thephone'll be happy but also set up a PC TFTP with the same address that's set in the phone. - Perhaps I can fake-out the phone into thinking it'shitting a TFTP box on the Internet ().I've got plenty of PC's even a spare Linksys router.Any ideas. - Or is there something simpler. - Or have I just bought an expensive paper-weight?Thanks in advance.Gary Guthary[EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy MooreAspendora, Inc. ___ --Bandwidth 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] Fedora
Hi, we're using Fedora Core 3-5 on all of our customer asterisk installations, as well as on other projects like mythtv, mailservers and general servers without any problems. We like Fedora's bleeding edge state. Guido Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. ___ --Bandwidth 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] Fedora
bilal ghayyad wrote: Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hrmwell, first, you should think about using CentOS, as Fedora is the development branch. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940 Problem (Mess)
A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll work on my Asterisk box (outside of the FXO FXS modules on the TDM card in the Asterisk server, I only run SIP on the hardphones). I don't know the phone's password (sound familiar?). - Have tried everything, cisco, *##, etc Nothing works. You could factory-reset the phone. try this http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml and this http://www.sokol-associates.com/?q=node/51 hth ___ --Bandwidth 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] Fedora
Centos is a free version of RHEL. Natambu Obleton Network Engineer FastTrack Communications [EMAIL PROTECTED] (970) 247-3366 office (970) 247-2426 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Monday, September 18, 2006 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fedora Yes. We use fedora as a test system before moving it onto RHEL. On Mon, 2006-09-18 at 11:34 -0700, bilal ghayyad wrote: Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make Polycom 501 go off hook when pressingany digits
Hi Mike, It's done using the digitmap feature of sip.cfg - email me offlist or come on #asterisk and I can help you with the specifics. CP On 18-Sep-06, at 11:08 AM, Mike wrote: I'm trying to make the Polycom 501 go off-hook (in speaker phone mode) when any digits is dialed and the handset hasnt been lifted. Is this possible? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dial and Timeout
Are you having this problem with an analog line or PRI ? David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tobias Wolf Envoyé : 18 septembre 2006 11:41 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Dial and Timeout Hi, we have experienced som troubles with the timeout option of the Dial-App. It seems the Dial startts counting down the timeout imediatly, but there are great differences when the called phone actually starts ringing. If i call a landline phone in my own country it is nearly the same, but if i want to call a cell phone it could take up to 10 secs until the phone begins to ring. So, if i want to dial with a timeout of 30 secs until i want to do something else, a landline user has 30 seconds of time to answer the call, but the cell user has only 20 seconds, more or less. In use cases where i don't know what type of phone is to be called it becomes quite difficult to set an appropriate timeout. Is there someway to get Dial() to start the countdown, when the channel state changes to ringing ? Thx for any advice, Tobias Wolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to make Polycom 501 go off hook whenpressingany digits
Mike, If you could send the answer here it would be greate. I would like to add this features to my Polycom phones too. David -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Anthony Rodgers Envoyé : 18 septembre 2006 19:09 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] How to make Polycom 501 go off hook whenpressingany digits Hi Mike, It's done using the digitmap feature of sip.cfg - email me offlist or come on #asterisk and I can help you with the specifics. CP On 18-Sep-06, at 11:08 AM, Mike wrote: I'm trying to make the Polycom 501 go off-hook (in speaker phone mode) when any digits is dialed and the handset hasnt been lifted. Is this possible? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Periodic announcements MySQL Realtime
Hi everybody, I'm trying to use the periodic-annouce and periodic-announce-frequency options. I use Asterisk 1.2.9.1-BRIstuffed-0.3.0 with MySQL realtime configuration. It seems that Asterisk Realtime queues doesn't support these options. When I try to add the 2 fields to the MySQL table, nothing happens ! Have you got any idea to solve this problem or static configuration is the only way for using these options ? Thank you by advance, Seb. ___ --Bandwidth 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] How to learn or teach VoIP QoE
Hi,How would you best learn VoIP Quality of Experience ?Before diving into packet loss and jitter, I would like to know what a toll-quality call is, what a rated 3.5 MOS call is like.I'm wondering how I should proceed. Shall I :- get pre-recorded sound files somewhere and simply stream them to a MOS enabled softphone (Counterpath sells eye-beam which includes a telchemy MOS rating module),- or shall I install some network impairment software, generate VoIP trafic and tweak myself jitter and other parameters so that I can associate network measures to call quality ? I've never heard of any sound files library aimed to learn what the impact of packet is like for end user experience.I've seen here and there network simulators (some of them free of charge) but it seems tricky to tune them to VoIP (is a 10% packet loss realistic or not ?). To make myself perfectly clear, my ultimate goal is to better undestand users testimonies when they warn me about poor quality phone calls.Regards ___ --Bandwidth 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 GUI?
So the press announcement said that the new Digium GUI will be available in v1.4 sometime in Oct. Is the GUI already there in Trunk or is there some other branch of development that the general public cannot access? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digium GUI?
You mean the menuselect ncurses screen? If yes, then yes... it's a gui. :) -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Monday, September 18, 2006 4:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Digium GUI? So the press announcement said that the new Digium GUI will be available in v1.4 sometime in Oct. Is the GUI already there in Trunk or is there some other branch of development that the general public cannot access? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Enabling Second Processor Trashes Audio Quality
Any thoughts on this one? IBM xSeries 330 processor running Debian 3.1 (2.6.8 kernel) with a TE406P board. Working fine (more or less) connected to a couple of PRIs. Rebuild kernel with support for second CPU and inbound (PRI - SIP) audio is badly garbled. Outbound (Asterisk - PRI) is fine. Rebooting a kernel with support for only a single CPU clears up the problem There is a small possibility that the TE406P card is acting up and that the audio problem is coincidental with the switch between dual-processor/single-processor kernels but thought I'd consult the list for advice. Will be swapping out the TE406P for a new TE407P in the next couple of days and will report findings then. g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) Hosted IP PBX Services for SOHO Small Businesses - www.ip-centrex.ca VoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca ___ --Bandwidth 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 Appliance, will Asterisk Business Edition be mandatory?
Just wondering about the Asterisk Appliance. I was waiting for hardware like this but the details are still kind of sketchy about how it will be sold. Will there be an option to buy it barebones without Asterisk Business Edition? Not even sure if it's feasable for a mere mortal such as myself to compile Linux/Asterisk on that Blackfin RISC processor it uses. ___ --Bandwidth 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] Starting Asterisk PBX: FATAL: Module ixj not found.
When I started Asterisk I get this error but it is working fine and should I be concerned. Error below: [EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk startStarting Asterisk PBX: FATAL: Module ixj not found. ___ --Bandwidth 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] Starting Asterisk PBX: FATAL: Module ixj not found.
Check /etc/asterisk/modules.conf and see if there is a line trying to load it. On 9/18/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk start Starting Asterisk PBX: FATAL: Module ixj not found. ___ --Bandwidth 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] create_addr: No such host:
I have created a context in extensions.conf and when I dial, it is suppose to ask me to enter pin number but instead this the error I get. Sep 18 18:11:54 WARNING[6514]: chan_sip.c:1968 create_addr: No such host: 4035Sep 18 18:11:54 NOTICE[6514]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Below is the channel information in extensions.conf [applicationxyz];CallingCard applicationexten = 4035,1,Answerexten = 4035,2,Wait,2exten = 4035,3,DeadAGI,applicationxyz.phpexten = 4035,4,Wait,2exten = 4035,5,Hangup ___ --Bandwidth 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] Starting Asterisk PBX: FATAL: Module ixj not found.
The only load I have is, load = chan_modem.soload = res_musiconhold.so [global]chan_modem.so=yes -- Original message -- From: "Justin Tunney" [EMAIL PROTECTED] Check /etc/asterisk/modules.conf and see if there is a line trying to load it. On 9/18/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]>wrote: [EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk start Starting Asterisk PBX: FATAL: Module ixj not found. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Rates
I've used this company now for over a year. It is part of Ipcb.net, so you got live support 24 hours a day every day. The quality is very good and the reliability is near perfect. You can have 1000 simultaneous calls. On the down side - The Signup is not so easy. I had to fax 7 papers to verify my account. And had to wait a couple of days for them to check it. You also have to be a company to register and send them your bank information. Fredrik I saw this termination company, www.BuyMin.com http://www.buymin.com/ the rates looks good. Has anyone any experience with this company? I use Gafachi, very reliable but expensive. ___ --Bandwidth 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] Termination Rates
Thanks I'll give them a trial. -- Original message -- From: "Insider KT" [EMAIL PROTECTED] I've used this company now for over a year. It is part of Ipcb.net, so you got live support 24 hours a day every day. The quality is very good and the reliability is near perfect. You can have 1000 simultaneous calls. On the down side - The Signup is not so easy. I had to fax 7 papers to verify my account. And had to wait a couple of days for them to check it. You also have to be a company to register and send them your bank information. FredrikI saw this termination company, www.BuyMin.com the rates looks good. Has anyone any experience with this company? I use Gafachi, very reliable but ex pensive. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
