[asterisk-users] Sip from ip address
Is there a way to specify which IP address to originate calls from in a peer on sip.conf? I need to send calls from 10.1.3.10 which is a routed network through openvpn, but it's using 10.39.0.10 which is a vpn IP address - the asterisk box is the same box as the vpn bridge for the 10.1.3.0/24 network. I can't set the host as 10.39.x.x as it is dynamic. i can't change bindaddr since i need to be able to receive connections from the external ip address as well as the internal address - unless there's a way to specify 2 ip's to use? For now i will use friend with a dynamic host instead of peer, but would prefer to use peer without having to use username and passwords. Cheers Geraint -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A108 PCIe V2.0
i suppose that depends on the number of eggs and baskets you have... but i'm guessing not many of either since you're considering using a desktop board for this... but, email sangoma support, they will tell you. On 17 September 2010 12:47, John Novack jnov...@stromberg-carlson.orgwrote: Anita Hall wrote: Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? Ask Sangoma They are very helpful The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience with multiple A108 with PCIe on the same motherboard that supports PCIe 2.0 ? Any comments will be helpful. Lot of eggs in one basket! John Novack Thanks, Anita Hall Simmortel Voice. -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR on Transfer...
to get accurate cdr's i just use a border server to send every call through that logs cdr... doesn't matter how many times it gets transferred internally the border server still gets accurate records of the whole call. On 27 August 2010 21:07, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Carlos Chavez cur...@telecomabmex.com writes: I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensión the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a mention that it should be fixed in 1.6 which it is not (at least on 1.6.2.11). You can set a TRANSFERCONTEXT. In that context you can try to use ForkCDR and its companions to get the records right. If you come up with a setup which acts perfectly no matter the scenario I would be happy to hear about it. Note that TRANSFERCONTEXT is not invoked when the phone does a SIP redirect before the call is answered, AFAIK. Notice that it's been a long time since I battled with this part of Asterisk, and I didn't check that I remembered correctly. This will all be a lot more sane with Channel Event Logging in 1.8.x, but at that point you need to run mediation before you get CDR's you can use for billing. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Connect problem...
I would like to figure out why but can't really switch back now it works since to replicate the problem... whatever it may be... i'd need to leave it running live and wait for the live system to die... which obviously isn't what i really want to happen :) On 19 August 2010 08:11, Sherwood McGowan sherwood.mcgo...@gmail.comwrote: On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee gera...@gmail.com wrote: This is what I ended up doing, working fine now. Cheers On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote: Avoid to use MySQL dialplan application, instead write an AGI script for this purpose LOL, I hate to say this but writing an AGI script just adds yet another application layer to your total solution. OP, if you'd like to figure out WHY that was happening instead of abandoning the ship, I'd be glad to work with you to discover the cause. I've been using the MySQL Addon since the early days of ViaTalk back when 1.4 was still trunk code and the ARA was considered VERY experimental. I've never come across a problem with it that I couldn't figure out within a day so long as I stepped back and worked the logical path model of problem solving... Drop me a line, I think that I can figure it out within 20 questions and maybe a peek at a log ;-) Slainte, Sherwood McGowan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec choice
i do this by having 2 peers setup, one has a call limit of 10 and uses g729, the rest of the calls get sent to the second peer which uses ulaw. all calls attempt peer 1 if there's channels available it uses it if not it just moves through the dialplan to use the second one. On 19 August 2010 09:14, Deepika Nijhawan deepika.nijha...@oxygen8.comwrote: Hi, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Connect problem...
This is what I ended up doing, working fine now.
Cheers
On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote:
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee gera...@gmail.com wrote:
Right, I'm baffled.
I have:
exten = s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten = s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12))
exten = s,n,MYSQL(Query RESULT1 ${DB1} SELECT\ LAST_INSERT_ID())
exten = s,n,MYSQL(Fetch FOUND1 ${RESULT1} VALUE1)
exten = s,n,MYSQL(Clear ${RESULT1})
exten = s,n,MYSQL(Disconnect ${DB1})
exten = s,n,MixMonitor(${VALUE1}.wav)
exten = s,n,Set(CALLERID(all)=xxx)
exten = s,n,Dial(SIP/prov1/${ARG1})
in a macro to dial numbers...
Every few hours or so every call hangs on the s,1 MYSQL(Connect) and won't
work until i restart asterisk.
The mysql server has a maximum connections of 2048 (of which around 90 are
in use) so it's not a mysql connection limit problem from what i can tell
since while asterisk is stuck i can still log in to mysql just fine, as
can the web server.
Does anyone have any suggestions what could be causing asterisk to get
stuck here? i don't see anything in cli and core show channels just shows
everyone stuck in state ring on the connect string with no errors.
Cheers
Geraint
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[asterisk-users] MySQL Connect problem...
Right, I'm baffled.
I have:
exten = s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten = s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12))
exten = s,n,MYSQL(Query RESULT1 ${DB1} SELECT\ LAST_INSERT_ID())
exten = s,n,MYSQL(Fetch FOUND1 ${RESULT1} VALUE1)
exten = s,n,MYSQL(Clear ${RESULT1})
exten = s,n,MYSQL(Disconnect ${DB1})
exten = s,n,MixMonitor(${VALUE1}.wav)
exten = s,n,Set(CALLERID(all)=xxx)
exten = s,n,Dial(SIP/prov1/${ARG1})
in a macro to dial numbers...
Every few hours or so every call hangs on the s,1 MYSQL(Connect) and won't
work until i restart asterisk.
The mysql server has a maximum connections of 2048 (of which around 90 are
in use) so it's not a mysql connection limit problem from what i can tell
since while asterisk is stuck i can still log in to mysql just fine, as
can the web server.
Does anyone have any suggestions what could be causing asterisk to get stuck
here? i don't see anything in cli and core show channels just shows everyone
stuck in state ring on the connect string with no errors.
Cheers
Geraint
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Re: [asterisk-users] installing with yum
it would be far easier to just use the source... but... yum search asterisk might get you on your way, although i can't see anything that looks like samples in there. On 13 August 2010 19:08, Albert Bonomo apeto2...@gmail.com wrote: Hi, I'm trying to install Asterisk with yum. I have followed the instructions on http://www.asterisk.org/downloads/yum I discovered that the repositories that describe there, don't exist. So thar running yum install asterisk16 won't install al all. Some guy from Fedora mailing list suggested me to run yun install asterisk It worked great! ( I had to remove the repositories added before, as instructed in http://www.asterisk.org/downloads/yum ) The problem is that no examples nor sounds where installed. I'm afraid something else is missing and Astrisk won't work properly. Can anybody advice me on how to install the rest of the Asterisk ? add-ons and other stuff ? Thanks Alberto. -- @apetob at Tweeter There are only 10 types of people in the world: Those who understand binary and those who don't -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?
try looking in extensions.ael On 25 June 2010 12:25, Eyal Goltzman egoltz...@gmail.com wrote: Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what asterisk is dialing. How can I trace\debug my dialplan? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to randomly use provider?
look at Random() 2009/12/12 Landy Landy landysacco...@yahoo.com Hello List. I would like to know how I can use two or more service providers with asterisk to be used randomly for ei, if an user tries to make a call I would like to randomly use a provider. It doesn't matter where the call is destined to. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DeadAgi
1) does the file exist
2) is it chmod'd to 755 (not sure if this matters though)
3) do you have something like #!/usr/bin/php at the start of the php file?
Cheers
Geraint
2009/9/17 Anahi Ludueña a_ludu...@hotmail.com
Hi people, I have the following dialplan:
[context]
exten = s,1,Noop(Start)
...
exten = h,1,Noop(Ending)
exten = h,n,DEADAGI(finconf.php,${ARG1},${ARG2})
When it is running, the asterisk gives the following error:
-- Launched AGI Script /var/lib/asterisk/agi-bin/finconf.php
== finconf.php|800|: Failed to execute
'/var/lib/asterisk/agi-bin/finconf.php': No such file or directory
But the file is there. The command ls -l returns:
*-rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php*
Why does it return the error?
Thanks,
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Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
MeetMe agreed, but depending on how many people you expect to be listening, i think you can do this on a virtual server with minimal bandwidth, you can probably do this very very cheaply, or even find someone that will host it for free since it's non profit, unless of course you're talking about hundreds of people listening... but for ~20 i don't think it will cost too much at all, and i'm talking not much in the non profit sense, so i don't think you'd need hundreds... one of the servers i manage costs £40/month (1and1) which is currently handling over 100 calls with no complaints at all, so you should certainly be able to get something much cheaper than that, and i'm sure i've seen ISPs doing free services for non profit organisations in the past. Cheers Geraint 2009/9/2 li...@mgreg.com li...@mgreg.com Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short-term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They are looking to allow callers to listen in to public speaking sessions. They currently have a single phone line with call waiting and are using an archaic one-person switch to then allow folks to call-chain via 3-way calling. What they want is basically a switchboard that allows multiple people (5 to 10) to call in at a time of their choosing and begin listening to the in-progress session. My first question would be: Is Asterisk the proper tool for this job (or is there something else you'd recommend)? A follow-up question would be: What kind of cost is involved in a small setup of this nature? Your input is much appreciated. Best, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
Asterisk is perfectly capable of it, your limiting factor will be bandwidth if you want to do it in-house... you'll obviously need enough bandwidth for all of your callers to be able to hear... unless of course you'll be using real phone lines, in which case you'll need to buy the appropriate hardware for your phone lines. Cheers Geraint 2009/9/2 li...@mgreg.com li...@mgreg.com On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote: Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j Thanks for the posts thus far! In all honesty I'm looking for a complete in house solution. I don't mind spending up to $500-600 on equipment if necessary. I just want to know that when I'm done there are no residual costs, etc. Is Asterisk capable of this kind of setup/ management? As for labor, I'm willing to donate as much as is necessary. Thanks again, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
On another note... have you considered using a simple shoutcast setup instead? There will be a way (many ways probably) to hook this in with asterisk if necessary. You may have better results if it's simply listening the callers need to do, and depending on the audience that will be listening may work out easier and cheaper too. 2009/9/2 li...@mgreg.com li...@mgreg.com On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote: Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j Thanks for the posts thus far! In all honesty I'm looking for a complete in house solution. I don't mind spending up to $500-600 on equipment if necessary. I just want to know that when I'm done there are no residual costs, etc. Is Asterisk capable of this kind of setup/ management? As for labor, I'm willing to donate as much as is necessary. Thanks again, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Autodialer
For someone who is developing an 'autodialer' you are asking for an awful lot! I would recommend getting to grips with asterisk before even considering developing a dialer... question 1 - aren't you developing your own so why would you need documentation for another? or... why not use the other? question 2... you shouldn't be writing a dialer if you can't come up with a way to test it on your own question 3... check the mailing list question 4 I'm not sure if you've heard of it... but there's this search engine called google, i hear you can search for things just like that on there! Of course the question you're trying to ask may have been that you want to implement a dialer and want people to recommend one, i think that's the question you should be asking if not. Cheers Geraint 2009/8/25 Sanjoy Rath sanjoy_r...@hotmail.com Hello, I am developing an asterisk autodialer. I am looking for the following information: 1. Detailed Configuration Documentation for Asterisk Autodialer 2. Volume Testing Strategy 3. Lessons Learnt from past Asterisk Autodialer configuration 4. What are the different asterisk autodialer functionality that have been implemented Your response will be appreciated. Thanks, Sanjoy. -- Send and receive email from all of your webmail accounts - right from your Hotmail inbox! http://go.microsoft.com/?linkid=9671351 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk???
Good luck with the N95... my experiences of the N95 and SIP haven't been great... the phone likes to restart... regularly. Nokia may well have fixed these glitches by now though. Getting it configured was a bit of a mission too... and as expected the battery life shoots down when it's enabled... so... again... good luck, and hope that nokia have fixed all those issues :p saying that though, to be fair, it did work most of the time, just had to put up with annoying restarts, sometimes at very inconvenient times... like when the phone rang. As for skype, I can see how it could be useful, I've worked with a few developers who's choice of communication method has been skype (I hate it myself though!), so being able to let them call a skype number and have it direct to a real phone would be quite useful. 2009/8/18 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Tue, 18 Aug 2009, Geoff Lane wrote: On Tuesday, August 18, 2009, Remco Barendse wrote: But then again, who needs Skype for business purposes anyways, i don't think there is a huge market for it. Me ... at least in theory! Our cellphones have built-in Skype, so a Skype gateway should give me call forwarding and diversion to our cellphones free of charge. So far Skype as implemented on our mobiles has proved too unreliable period for business use. It seems only available when we can get a 3G/HSDPA signal and even then the system regularly logs us out of Skype and sometimes doesn't log us back in. However, if and when my cellular provider get Skype sorted out on their system ... I was under the impression that Three (who I guess you're using) placed a regular call over their network then Skyped it at their HQ - rather than have the Skype client actually reside in the handset.. (And I'm suspecting their 3G limitation is that they want to use their own 3G network rather than pay Orange for the call over their 2G network) But then again, wifey's just gotten a new Three mobile (N95 - end of line, but a cheap deal) It has built in SIP via Wi-Fi and skype, so I might have a play with it when I can prise it out of her hands... It does worry me that I see many so-called business people advertising Skype numbers on their business cards, etc. To me it rings of cheapness. I can almost always tell when someone calls me using their Skype out service - the quality is dreadful, and I end up calling them on their regular landline. Cheapskates who can't/won't pay for a decent Internet service. I'm going to ask my customers if they want to be able to call Skype numbers, but I'd probably have to charge for it to justify the cost of the license(s) required. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues load balancing
Take a look at: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random You should be able to do what you want with this, it obviously won't take in to account the actual amount of people still in the queue (for example if someone hangs up while on hold). I'm sure there'd be a way of integrating this in to it using some different functions, but for a quick fix random will do just fine. Cheers 2009/7/20 Joao Gomes Pereira gomespere...@startel.pt Hello I have 2 queues (queue_1 and queue_2 ) in my Asterisk, and I want to send 2/3 of the calls to queue_1 and 1/3 of the calls to queue_2 How can I do that load balancing in extensions.conf? I have something like this: exten = 123,1,Ringing exten = 123,2,Wait(1) exten = 123,3,Answer ; 2 in 3 calls go to queue_1 exten = 123,x,Queue(queue_1) ; 1 in 3 calls go to queue_2 exten = 123,x,Queue(queue_2) But how can I configure this call distribution? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is Asterisk reliable for a call center application??
yes, when done correctly. 2009/7/13 gergis.rasmy gergis.ra...@gmail.com i am asked to implement a call center of 50 seats for my company , and i was wondering if Asterisk can fit this as a relaibale and low price system is it mature enough for this task?? best regards Gers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
search the mailing list, this question has been asked and answered several times. But it's all dependent on hardware, codecs, bandwidth. If you mix the right technologies there is no limit to how many calls you could handle, you just have to do it in the right way with multiple servers obviously. Cheers 2009/7/3 abdelkader abdelkader2...@gmail.com Hello, What is the maximum number of simultaneous calls supported by asterisk. thks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK Vodafone femtocells now available
or maybe i misread :) 2009/7/1 Mike Dent mcd...@gmail.com 2009/7/1 Geraint Lee gera...@gmail.com: agreed. extended o2 coverage would be very useful, especially for Wales! I like the idea, i don't like the idea of paying, if they want mobile traffic it should be possible to buy your own hardware controlled in the same method as wireless AP's allowing you to connect for free to the service and not be tied to a contract; or pay a very much reduced rate with an optional addon to your service for £2 or £3/month. I thought I read on the Vodafone site it was to be included with any 3g contracts over £25 per month? Maybe I misread? Mike Looking forward to seeing what the other networks will have to offer! 2009/7/1 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Wed, 1 Jul 2009, Dean Collins wrote: For those of you who have been waiting for ATT to announce the public availability of their femtocell appliance in order to fix the shitty ATT network coverage this will interest you. It's getting a lot of press and a bit of a mixed reaction over here. Some are complaining that they shouldn't have to pay to extend the networks coverage, others wanting to jailbreak their iPhones to take advantage of poor O2 coverage where they are... (but moving to voda is a backward step IMO ;-) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK Vodafone femtocells now available
agreed. extended o2 coverage would be very useful, especially for Wales! I like the idea, i don't like the idea of paying, if they want mobile traffic it should be possible to buy your own hardware controlled in the same method as wireless AP's allowing you to connect for free to the service and not be tied to a contract; or pay a very much reduced rate with an optional addon to your service for £2 or £3/month. Looking forward to seeing what the other networks will have to offer! 2009/7/1 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Wed, 1 Jul 2009, Dean Collins wrote: For those of you who have been waiting for ATT to announce the public availability of their femtocell appliance in order to fix the shitty ATT network coverage this will interest you. It's getting a lot of press and a bit of a mixed reaction over here. Some are complaining that they shouldn't have to pay to extend the networks coverage, others wanting to jailbreak their iPhones to take advantage of poor O2 coverage where they are... (but moving to voda is a backward step IMO ;-) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling non-extension numbers issue
I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dail in modem
If i understand correctly you need users to be able to dial in using a modem to your servers then you are going to share your internet connection with those who dial your server. So, no, it has nothing to do with asterisk... you want to be looking at wvdial for the clients (assuming they are linux) and whatever the equivalent server would be (don't know as i've never done it). Good luck 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Geraint lee I also dont know .what kind of requirements are these :P i am just looking if it can happen On Fri, Jun 19, 2009 at 9:33 PM, Geraint Leegera...@gmail.com wrote: is it just me or am i right in thinking this has nothing to do with asterisk? 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Hello Actually i am required to make two application 1) that user use 2) that is deployed on server Application for user will be just like the windows standard connection using dail up modem but user will dail my PSTN number instead of the number we inter provided by ISP. on deployed server side we will get he usename and pass and other parameters of application and then use them in java code is it possible ? (nothing is impossible but for a Asterisk and java developer with limited time frame) Thanks On Fri, Jun 19, 2009 at 7:24 PM, Bob Piercepier...@westmancom.com wrote: On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dail in modem
is it just me or am i right in thinking this has nothing to do with asterisk? 2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com Hello Actually i am required to make two application 1) that user use 2) that is deployed on server Application for user will be just like the windows standard connection using dail up modem but user will dail my PSTN number instead of the number we inter provided by ISP. on deployed server side we will get he usename and pass and other parameters of application and then use them in java code is it possible ? (nothing is impossible but for a Asterisk and java developer with limited time frame) Thanks On Fri, Jun 19, 2009 at 7:24 PM, Bob Piercepier...@westmancom.com wrote: On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. Why do you need a modem? What will be dialing into the Asterisk system, a human or a machine? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Soft Phone
twinkle. 2009/6/15 Manoj Panicker - FOES manoj.panic...@emirates.com Hi Guys, Any suggestions on any open source soft phones that has IAX and SIP support. I would also like to some programming over it and interface it with address book or LDAP in order to make the call making easier for the users. Thanks Manoj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP phone recommendation
i quite like the aastra 55i phones, i find the sound quality is better than the polycom sound stations on loud speaker, and handset quality is perfect. 2009/6/3 Christian Stredicke christian.stredi...@snom.de Check out the snom 300 or the snom 820... CS -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] Im Auftrag von Rilawich Ango Gesendet: Mittwoch, 3. Juni 2009 09:45 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] IP phone recommendation Hi all, Any good recommendation of IP phone in term of sound quality and price (reasonable) using with asterisk? ango ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can asterisk work here
Yes, that should work fine, just remember you need a crossover cable to go from the a102 to the legacy system 2009/6/3 Jim Dickenson dicken...@cfmc.com I have a potential client that currently has a T1 circuit that feeds into an Adtran 750. Their phone sets are connected to the 24 ports on the 750. I was wondering if I could take an Asterisk system with a Sangoma A102de in it and plug the T1 into one port of the A102 and the 750 into the second port? Would I then have 24 voice channels that I could manage for the 24 phone sets? The only thing I know about the T1 is that it uses wink start signaling. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 Channel Information
I don't quite understand what you're trying to achieve, but if it's a
firewall wouldn't using something like iptables make more sense and be far
more secure?
Cheers
2009/6/3 Lee Spenadel spena...@gmail.com
I’m trying to isolate the IP address of inbound calls to my switch over
IAX2. Is the proper way to get that information as follows:
${IAXPEER(IP)}
If the caller was inbound via SIP, this works:
${SIPCHANINFO(PEERIP)}
So I’m looking to return the IP address of the caller via IAX2.
Thanks
Lee
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Re: [asterisk-users] IP phone recommendation
I personally find the snom phones to be generally ugly and un-finger-friendly, in terms of reliability and quality, never had any trouble, good phones all in all, i just can't get past the tacky look and feel so don't buy them. 2009/6/3 Darrick Hartman dhart...@djhsolutions.com On 06/03/2009 11:47 AM, Jeff LaCoursiere wrote: On Thu, 4 Jun 2009, Rob Hillis wrote: Jeff LaCoursiere wrote: We are still talking about a $175 phone. How about the Polycom IP 320? $85 at 888voipstore. Can't go wrong with Polycom for voice quality. True, Polycom's are brilliant for voice quality, but unlike the Snom, a Polycom /will/ reboot on the drop of a hat /and/ take a damned long time to do it (~45-60 seconds) In addition, the web interface should be taken away and shot - the only real way to configure them is through (T)FTP. They are however, extraordinarily configurable through the XML config and they are very stable. Once they're configured they work very nicely. The lack of a decent number of BLF keys (even with a very expensive sidecar you only get two more keys than a standalone Snom320) puts me off a little. However, for a conference phone, the Polycom's can't be easily beaten. Their handsfree call quality is in a league of it's own. Mainly I suggest it because the OP asked for an inexpensive quality phone. I agree on the provisioning - the web interface is useless, and unless you know how to setup the XML files properly you are doomed to a very frustrating experience. The Polycom 320/330's are nice little phones for the price. There are several resources for configuring the phones from the XML config files. If the config files are sane, the phones don't take that long to reboot. This is probably one of the better examples: http://www.kfife.com/voip/ Karl did a good job commenting in the config files where he made changes. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP phone recommendation
you could say it has, and you're right, it probably has :) but i personally find these threads help make the day pass a little faster 2009/6/3 John Novack jnov...@stromberg-carlson.org Hasn't this religious argument/discussion gone on long enough?? zoach...@securax.org wrote: I personally find the snom phones to look quite good compared to the american and chinese brands, might be a european thing though :) Zoa Geraint Lee wrote: I personally find the snom phones to be generally ugly and un-finger-friendly, in terms of reliability and quality, never had any trouble, good phones all in all, i just can't get past the tacky look and feel so don't buy them. 2009/6/3 Darrick Hartman dhart...@djhsolutions.com mailto:dhart...@djhsolutions.com On 06/03/2009 11:47 AM, Jeff LaCoursiere wrote: On Thu, 4 Jun 2009, Rob Hillis wrote: Jeff LaCoursiere wrote: We are still talking about a $175 phone. How about the Polycom IP 320? $85 at 888voipstore. Can't go wrong with Polycom for voice quality. True, Polycom's are brilliant for voice quality, but unlike the Snom, a Polycom /will/ reboot on the drop of a hat /and/ take a damned long time to do it (~45-60 seconds) In addition, the web interface should be taken away and shot - the only real way to configure them is through (T)FTP. They are however, extraordinarily configurable through the XML config and they are very stable. Once they're configured they work very nicely. The lack of a decent number of BLF keys (even with a very expensive sidecar you only get two more keys than a standalone Snom320) puts me off a little. However, for a conference phone, the Polycom's can't be easily beaten. Their handsfree call quality is in a league of it's own. Mainly I suggest it because the OP asked for an inexpensive quality phone. I agree on the provisioning - the web interface is useless, and unless you know how to setup the XML files properly you are doomed to a very frustrating experience. The Polycom 320/330's are nice little phones for the price. There are several resources for configuring the phones from the XML config files. If the config files are sane, the phones don't take that long to reboot. This is probably one of the better examples: http://www.kfife.com/voip/ Karl did a good job commenting in the config files where he made changes. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information from ESET NOD32 Antivirus, version of virus signature database 4124 (20090602) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __ Information from ESET NOD32 Antivirus, version of virus signature database 4124 (20090602) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Domains
It might be worth clarifying what the question is, i'm pretty lost. Cheers Geraint 2009/5/27 Adrian Marsh adrian.ma...@ubiquisys.com Noone can give me a clue on this ? How Domains are used within Asterisk ? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Adrian Marsh *Sent:* 26 May 2009 12:14 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Domains Hi, I’m trying to understand an issue I’m seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then the call works, and is placed through to server As logic successfully. But if I call in as client 2000, then the call fails, server A shows no log at all of the call (even a sip set debug ip ip showed nothing – though tcpdump did show the inbound invite). However if I remove the definition of client 2000 from server A, then the call succeeds. So I think that for a defined account server A is wanting to challenge for a password, even though the inbound call is not a local account – hence my trying now to understand if and how Asterisk uses Domains. If I define a serverA.company.com domain on server A, will it ignore the challenge for an INVITE coming from server B ?? Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS
There is indeed... well i was about to say there was, but it turns out the one i've got is an fxo adapter, have a look and see if sangoma have any fxs adapters in the series, it seems to be called the usbfxo u100 2009/5/26 Diogo Saad diogos...@gmail.com What is the easiest way to connect my black phone to a PC running asterisk? I don't need multiple extensions, I've got just 1 phone. Is there any USB FXS adapter? Thanks -- Diogo Saad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...
have you checked /var/log/maillog to see what the error might be? 2009/5/22 David da...@linuxcrazy.com -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Here is mine if it helps; [general] format=wav49|gsm|wav serveremail=asterisk attach=yes skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 emaildateformat=%A, %B %d, %Y at %r sendvoicemail=yes [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/Chicago|'vm-received' Q 'digits/at' IMp central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours' military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] = ,David Abbott,x...@.net Thats all I have in there, asterisk will use my SMTP client without me doing anything. I am using asterisk 1.4 - -david - -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com pgp.mit.edu -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkoWvecACgkQcZ+z4vAcSszhlQCeKpnBggDU75DVsI0dj1/m8UVx 6+wAn2Z+gRGatPscWNJvOWR7qxJVRXOy =w4YA -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...
ignore me! i've just realised half this thread was deleted :) 2009/5/22 Geraint Lee gera...@gmail.com have you checked /var/log/maillog to see what the error might be? 2009/5/22 David da...@linuxcrazy.com -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Here is mine if it helps; [general] format=wav49|gsm|wav serveremail=asterisk attach=yes skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 emaildateformat=%A, %B %d, %Y at %r sendvoicemail=yes [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/Chicago|'vm-received' Q 'digits/at' IMp central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours' military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] = ,David Abbott,x...@.net Thats all I have in there, asterisk will use my SMTP client without me doing anything. I am using asterisk 1.4 - -david - -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com pgp.mit.edu -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkoWvecACgkQcZ+z4vAcSszhlQCeKpnBggDU75DVsI0dj1/m8UVx 6+wAn2Z+gRGatPscWNJvOWR7qxJVRXOy =w4YA -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
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2008/12/11 Dave Fullerton dfullertaster...@shorelinecontainer.com
Brent Davidson wrote:
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd like to somehow make the CallerID more useful. Currently if an
extension at Office1 dials an extension at Office2 the CID on the phone
at Office2 says Office1. The same thing happens if a person at
Office1 transfers an incoming call to Office2. The caller ID at Office2
always just says Office1.
What I would like to happen would be when Bob at Extension 12 at Office1
calls Office2 the caller ID at office 2 would say Bob in the name
files and 12 in the number field. If Bob does a blind transfer to an
extension at Office2 I would like the caller ID on the Office2 phone to
display the original caller's name and number.
I've read most of the documentation on the CallerID variables, but am
still having a bit of trouble wrapping my head around the necessary
logic to accomplish what I need to do, (mainly because I'm in the middle
of a totally unrelated project and am having trouble multi-tasking).
Could anyone give me a starting point?
Thanks,
Brent
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
additionally if you want to have the callerid to include office1 when
calling office2, you could change the callerid using
Set(CALLERID(name)=${CALLERID(name)} Office 1)
just before sending through to office 2
Something along those lines anyway, not entirely sure on the syntax or if
there's a better way to do it.. but i'm sure someone will correct me if i'm
wrong :)
Geraint
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div class=gmail_quote2008/12/11 Dave Fullerton span dir=ltrlt;a
href=mailto:dfullertaster...@shorelinecontainer.com;dfullertaster...@shorelinecontainer.com/agt;/spanbrblockquote
class=gmail_quote style=border-left: 1px solid rgb(204, 204, 204); margin:
0pt 0pt 0pt 0.8ex; padding-left: 1ex;
div class=Ih2E3dBrent Davidson wrote:br
gt; I have several branch offices all running Asterisk PBX#39;s that
registerbr
gt; to each other via SIP so that calls can be transferred from office tobr
gt; office. nbsp;Everything is working great on the office to office
transfers,br
gt; but I#39;d like to somehow make the CallerID more useful. nbsp;Currently
if anbr
gt; extension at Office1 dials an extension at Office2 the CID on the phonebr
gt; at Office2 says quot;Office1quot;. nbsp;The same thing happens if a
person atbr
gt; Office1 transfers an incoming call to Office2. nbsp;The caller ID at
Office2br
gt; always just says quot;Office1quot;.br
gt;br
gt; What I would like to happen would be when Bob at Extension 12 at
Office1br
gt; calls Office2 the caller ID at office 2 would say quot;Bobquot; in the
namebr
gt; files and quot;12quot; in the number field. nbsp;If Bob does a blind
transfer to anbr
gt; extension at Office2 I would like the caller ID on the Office2 phone tobr
gt; display the original caller#39;s name and number.br
gt;br
gt; I#39;ve read most of the documentation on the CallerID variables, but
ambr
gt; still having a bit of trouble wrapping my head around the necessarybr
gt; logic to accomplish what I need to do, (mainly because I#39;m in the
middlebr
gt; of a totally unrelated project and am having trouble multi-tasking).br
gt; Could anyone give me a starting point?br
gt;br
gt; Thanks,br
gt; Brentbr
br
/divCheck the entries for office1 and office2 servers in sip.conf. If theybr
have a callerid= entry comment it out and do a SIP reload. When it isbr
set asterisk overrides the caller ID sent to
it./blockquotedivbradditionally if you want to have the callerid to
include office1 when calling office2, you could change the callerid
usingbrbrSet(CALLERID(name)=${CALLERID(name)} Office 1)br
brjust before sending through to office 2brbrSomething along those lines
anyway, not entirely sure on the syntax or if there#39;s a better way to do
it.. but i#39;m sure someone will correct me if i#39;m wrong :)br
brGeraintbr/div/div
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Re: [asterisk-users] Where I get free VoiP-in numbers?
that's a bit lazy isn't it? google and the list archive should reveal all. 2009/4/27 almidos...@gmail.com almidos...@gmail.com Hi list, Anyone knows how to get free VoiP-in numbers from USA or Canada, I have found some links for example sipnumber.com but it does not run. Also I want to know how to configure it in my asterisk server. Thanks in advance. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function originate
You could use 2 originate commands and connect both of them to a meetme room? But surely what you're trying to do is going to confuse the person anyway if they don't hear anyone when they answer? Wouldn't it just be better to play a message after party a answers and then start ringing party b so that party a knows what's going on? 2009/4/24 Rilawich Ango maillist...@gmail.com Hi, Feature originate can be used make call thro' the web. There is a parameter ,Async, in it. I set it to true but there is no effect. Actually, I want to do the following. What I know the function originate is: originate call --- party A party A rings party A answers call party B rings, party A still hear ring party B answers and A B connected. party A will feel weird when she will still hear ring after answering a call until party B answers it. Below is what I want to do: originate call --- party A party A rings party B rings party A answers call A B connected. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record in mp3
you probably don't want to record directly to mp3 as there will be an overhead in converting the audio on the fly and this will probably break your call recordings... you should either record in the codec you are using for phone calls (i think?) or in .wav and then convert afterwards (correct me if i'm wrong someone!). 2009/4/24 Jose Enes Mateus jemat...@yahoo.com.br But have you tried to record directly in mp3, without to covert the file? --- Em *qui, 23/4/09, Danny Nicholas da...@debsinc.com* escreveu: De: Danny Nicholas da...@debsinc.com Assunto: Re: [asterisk-users] Record in mp3 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Data: Quinta-feira, 23 de Abril de 2009, 17:33 The way I read to do this is to use sox to create a wav file, then use lame to convert the wav to mp3. I did this for some MOH files. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jose Enes Mateus *Sent:* Thursday, April 23, 2009 3:28 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Record in mp3 *Somebody knows* if I can save files in mp3 with the Record command on Asterisk? I try to recompile sox to suport mp3 but Asterisk return the folowing message when I use the Record command: - Executing [...@liberado15:15] Record(SIP/1201-083453c8, /var/spool/asterisk/alarme/alarme-1201-200905121212:mp3) in new stack -- SIP/1201-083453c8 Playing 'beep' (language 'pt_BR') [Apr 23 17:22:36] ERROR[4494]: format_mp3.c:283 mp3_rewrite: I Can't write MP3 only read them. [Apr 23 17:22:36] WARNING[4494]: file.c:378 fn_wrapper: Unable to rewrite format mp3 [Apr 23 17:22:36] WARNING[4494]: file.c:1092 ast_writefile: Unable to rewrite /var/spool/asterisk/alarme/alarme-1201-200905121212.mp3 [Apr 23 17:22:36] WARNING[4494]: app_record.c:272 record_exec: Could not create file /var/spool/asterisk/alarme/alarme-1201-200905121212 I'am doing something wrong? Thanks -- Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10http://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/- Celebridadeshttp://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/celebridades/- Músicahttp://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/m%C3%BAsica/- Esporteshttp://br.rd.yahoo.com/mail/taglines/mail/*http:/br.maisbuscados.yahoo.com/esportes/ -Anexo incorporado- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Veja quais são os assuntos do momento no Yahoo! + Buscados: Top 10http://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/- Celebridadeshttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/celebridades/- Músicahttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/m%C3%BAsica/- Esporteshttp://br.rd.yahoo.com/mail/taglines/mail/*http://br.maisbuscados.yahoo.com/esportes/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Capacity
Hi Guys, I have a strong feeling the loads on my servers will be shooting up soon... anyone got any idea how many calls i can expect to put through a DL360: Dual Quad Core 2.33ghz 4gb RAM with 1gb allocated for a ramdisk (call recordings) This server is recording calls (mixmonitor), codec is gsm (no conversion). I know there's a lot of other things to consider like AGI scripts and such things but i'd like to know what the capacity should be simply for sip registrations (which are in conf files) and calls (usually between 20 and 60 concurrent calls at present (around 12,000 calls a day - so relatively low volume). No voicemail or meetme. I expect to be pushing 300-400 concurrent calls within the next 2 months. Next question... do i need to be looking at openSIPS or something similar to handle registrations? Any hints, tips and things to watch out for with a larger volume would be great. Cheers Geraint ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Capacity
Thanks for that, it's pretty much confirming what i first anticipated... my intentions are as follows: agents register with opensips, opensips clusters a set of call recording servers which then connect to our border servers which will save cdr and choose the sip/iax provider to send the call to. and for my predictive dialer, each server will spool as many calls as they can before i see performance issues when they have an answer they too will connect to the opensips server to get a call recording server which in turn will pass it on to the agent again via opensips. simples :) looks like i need to install and learn opensips since this whole scenario seems to be heavily relying on it :) Cheers 2009/4/23 z gringo z_gri...@hotmail.com You don't say how many SIP registrations you are doing, but I have several servers with betwen 1000 and 1200 simultaneous registered users 24/7. When we had the registrations in realtime (cached) with the mysql connector, everything started failing around 600 users. With the ODBC connector we have not had that problem. Ditto for putting the users in .conf files. My servers all have around 300 to 400 simultaneous calls during peak periods, and I have a 1GB ramdisk for recordings.We are only recording a tiny percentage of those calls. MySQL is running on a separate server dedicated to Databases. The asterisks connect to the realtime DB via a private network on a second nic. My thoughts are these: 1. Asterisk is not going to be able to handle much more registration traffic than around 1200 registered users. (this depends on a whole lot of things though). Eventually, it will need to be offloaded to something like OpenSIPs 2. Somewhere around 800 simultaneous calls is about the most asterisk is going to be able to push. 3. Your problem is going to be the call recording. If you are trying to record all the calls on your server or even a large percentage of them, that is going to be your first problem area. Another important thing to consider is how many calls you are setting up and tearing down each second. If you have a bunch of users dialing manually and making long calls, that will be a lot easier to handle than if you have 3 predictive dialers running against your server trying to bring up 30 calls per second. If you are doing something like that, you will probably need to distribute accross multiple servers. -- Date: Thu, 23 Apr 2009 12:12:35 +0100 From: gera...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Capacity Hi Guys, I have a strong feeling the loads on my servers will be shooting up soon... anyone got any idea how many calls i can expect to put through a DL360: Dual Quad Core 2.33ghz 4gb RAM with 1gb allocated for a ramdisk (call recordings) This server is recording calls (mixmonitor), codec is gsm (no conversion). I know there's a lot of other things to consider like AGI scripts and such things but i'd like to know what the capacity should be simply for sip registrations (which are in conf files) and calls (usually between 20 and 60 concurrent calls at present (around 12,000 calls a day - so relatively low volume). No voicemail or meetme. I expect to be pushing 300-400 concurrent calls within the next 2 months. Next question... do i need to be looking at openSIPS or something similar to handle registrations? Any hints, tips and things to watch out for with a larger volume would be great. Cheers Geraint -- Rediscover Hotmail®: Now available on your iPhone or BlackBerry Check it out.http://windowslive.com/RediscoverHotmail?ocid=TXT_TAGLM_WL_HM_Rediscover_Mobile2_042009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI PHP script
Check you can run the script from th ecommand line and successfully send
email... have you considered using phpagi for your scripts?
2009/4/23 James A. Shigley j...@answeringserv.com
I have the below script that doesn’t seem to be working. I don’t know if
I have something in the script wrong that I am just missing. Or if I don’t
have the php.ini set correctly for emailing
This is the CLI output
-- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1, newhire,s,1)
in
new stack
-- Goto (newhire,s,1)
-- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack
-- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack
-- Executing [...@newhire:3] Monitor(DAHDI/50-1,
wav,/var/lib/asterisk/soun
ds/NewHire/Newhire-1240503071.15148-4099819213-s,o) in new stack
-- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php
-- DAHDI/50-1AGI Script newhire.php completed, returning 0
-- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/50-1'
Here is the script
#!/usr/bin/php5
?php
// Get AGI vars from *
$agivars = array();
while (!feof(STDIN)) {
$agivar = trim(fgets(STDIN));
if ($agivar === '') {
break;
}
$agivar = explode(':', $agivar);
$agivars[$agivar[0]] = trim($agivar[1]);
}
extract($agivars);
// Variable Declarations
$agi_uniqueid;
$agi_callerid;
$agi_calleridname;
$agi_extension;
$agi_uniqueid;
$UNIQUEID = $agi_uniqueid;
$CALLERID = $agi_callerid;
$EXTEN = $agi_extension;
$attachment =
/var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav;
$from = �...@xxx.com;
$to =j...@answeringserv.com ;
$subject=New Applicant;
$headers = From: $from;
$message =$UNIQUEID , $CALLERID , $EXTEN , $attachment;
mail($to,$subject,$message,$headers);
?
So is it anything obviously wrong with the script I’m missing?
Besides something not being configured in php.ini correctly any other
ideas?
James Shigley
*Monroe Telephone Answering Service*
409-981-9213**
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
Ecreator:2.21, eResponse 1.1.7
Webportal,WebApps,
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Re: [asterisk-users] Asterisk Database
i'd use mysql... and i do use mysql for this... 2009/4/21 Sriram d_r_sri...@hotmail.com My setup : Trixbox 2.6.1 TE410P running well .: 1. I need to store the CallerId of the PSTN caller with his language preference so that next time he is played the prompt in his language that he chose the first time.What would be better - storing his number in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial plan and quering it everytime to see the callers record ? how many records can AstDB handle safely ? In my case the total records wont exceed 20,000 since there are many repeat callers ? rgds Sriram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alcatel OmniPCX Enterprise + Asterisk with E1
I haven't worked with the omnipbx's but i have with an alcatel 4400 and used a sangoma A108 and A104.. the sangoma cards work perfectly and if you have nay issues sangoma support are always more than happy to help - and they actually know what there talking about as apposed to having someone reading a script. as for different devices, you could use one of those vegastream devices, i've got a vega 400 but have never used it and so can't comment on how good / bad they are... but the vega 400 allows up to 4 (i think) e1/t1 connections converted to SIP. 2009/4/17 Sebastian Milioto smili...@gmail.com Hi all, I'm new in the forum, and although I have some experience in Asterisk, I've never work with Asterisk FXO, FXS, E1... cards. I have several costumers with ATAs working with my SER. However one of them bought an Alcatel PBX OmniPCX Enterprise and now he wants I give him a E1 interface for interconection with its new PBX. I understand I need a E1-IP gateway which could be Asterisk I think. So the network would look like this: Extensions-OmniPCX---E1---AsteriskIP--SER About that, I have a few question: 1. Anybody has done this interconection? How does Asterisk and PBX OmniPCX work together through an E1 interface? Any problems or bugs? 2. What E1 card should I buy for Asterisk? Is the physical interface (conectors) E1 identical as T1? 3. If cost wasn't a problem, do you suggest another interconection way technically better? May be replacing Asterisk with another device with an in-box E1? Thanks very much in advance, Sebastian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Calls at a time
i haven't understood any of this thread... but i'm going to throw
busy-level in sip.conf in to the mix... i have no idea if this is a useful
contribution... but i felt i should contribute something :)
2009/4/16 David @ULC ucoms2...@gmail.com
My SIP config is below :
[sip64]
type=peer
username=fiduci
fromuser=fiduci
authuser=fiduci
secret=pass
host=64.33.22.11
nat=no
canreinvite=yes
insecure=very
disallow=all
allow=g729
allow=ulaw
context=default
dtmfmode=rfc2833
Now, I need to add another element as call-limit=1 and this should solve
my problem ?
If yes. Great. Kindly advice.
But will that allow 3 party conference ?
On Thu, Apr 16, 2009 at 10:22 PM, David @ULC ucoms2...@gmail.com wrote:
call-limit in sip.conf
Can you elaborate please and how to set that.
Lets presume I have 10 agents and dial ratio is 4.
On Thu, Apr 16, 2009 at 10:06 PM, David @ULC ucoms2...@gmail.com wrote:
Even I thought so thats why I tried with 4 VOIP provider and things
didn't change. :-(
On Thu, Apr 16, 2009 at 8:36 PM, David @ULC ucoms2...@gmail.com wrote:
Many time we face an issue where even if an agent is on Call, another
call comes in.
Sometimes, even if agent hang up the call, call stays back and another
come sin and then both customers can hear each other { which i think is
VERY
dangerous [image: Wink] }
Also, this thing happens even when we have just 5 agents on a single
server. [image: Sad]
Our version is Asterisk 1.2.27
Any Solutions ?
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Re: [asterisk-users] Sangoma and BT single lines
sangoma support are amazing, they've solved nearly all the problems i've experienced with PRI, except for one which turned out to be a bug in SWIX (some rubbish windows based voip pbx, full of bugs and generally crap!). there also quite happy to log in to your systems and have a look themselves if you want them to, or if it's a particularly mind boggling problem. 2009/4/8 John Novack jnov...@stromberg-carlson.org Steve Davies wrote: 2009/4/6 Ed W li...@wildgooses.com: We have found that using Residential settings as a starting point,and then asking for Disconnect clear time to be set to 800ms is all that is needed. That one setting allows the hangup to be detected reliably. We do also use the dialtone detection of Asterisk to be sure we're dialling on a line that is ready to take a call. When was this added to Asterisk?? For years now, outbound dialing begun WITHOUT detecting dial tone, requiring multiple w to be inserted in the dial string. Dial tone detection was/is long overdue. Anyone know when this was added? Wading through cryptic change logs makes no mention of addition of this feature. This should have been put in bold red letters! Also Sangoma provides really great support for their cards. If all else fails, consider contacting Sangoma John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Originate Command
Use the Local/ channel type(?) Local/0123456...@outbound-route 2009/3/24 Nhadie nha...@gmail.com Hi All, I'm trying to use the orginate cmd. I have it working if originate is from a user e.g. SIP/ originate SIP/ extension 987654...@outbound-route What i'd like to be able to is instead of a local extensions i would call an outside number then connect it another outside number. e.g. originate SIP/85431...@outbound-route extension 987654...@outboudn-route is this possible? thanks. regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended USB Headsets ?
We've had no end of trouble with usb headsets on linux (especially cmedia chipset), as soon as you touch the volume control the sound settings all mess up... i'm sure there'll be an alsa seting somewhere which would solve this but i'm not that clued up on alsa so opted for using standard 2 connection headsets which work great (with a good soundcard). Geraint 2009/3/23 Edward Gray eg...@tucows.com Hi, we are looking to roll out PBX IN A Flash at our office. The first group will be using Soft Phones (X-Lite appears to be the best and works in Windows, Apple Linux). There are many types of USB Headsets to choose from and a fairly broad price range. Is there any USB headsets people would recommend? I'm specifically interested in acceptable audio (speaker and microphone) quality for business calls but am sensitive to price as well. In reading online, the Logitech Premium headset does get some good reviews but the reviews appear to be more from consumer based. I'd much prefer real experience from the good people who are operating their own Asterisk implementations. Any advice? Thanks! -- Ed ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended USB Headsets ?
I disagree with your opinion on softphones, i think they're great, saved thousands in cabling, switch and phone costs. I've had 50 agents running diskless/pxe linux (fedora 8), firefox, thunderbird and twinkle and never had any problems, in the next few months i expect to have at least 250 agents using this solution. I must admit though that i did share your opinion that hardphones are the only way to go, but having actually taken the time to get everything configured properly i see no advantage to having hard phones, only a huge unneccesary cost - that's the same way i look at using windows for agent pc's, completely unneccesary unless they *have* to use some sort of software that will only run on windows. Cheers 2009/3/23 Steve Totaro stot...@first-notification.com My advice? If mandated with a USB device and softphone, I would certainly go with Plantronics. My question is why not pick up some real Polycom 430s or something and realize that you really just saved yourself a great deal of time and money in all reality. Softphones, like inkjet printers, should be used at home or ONLY when REQUIRED. Thanks, Steve Totaro On Mon, Mar 23, 2009 at 10:09 AM, Edward Gray eg...@tucows.com wrote: Thank, is there advantages to Zoiper? The interface didn't seem that great, I haven't checked to see if it's compatible on Linux or Apple yet. Edward Gray Director, Vendor Management Tucows.com Co. eg...@tucows.com Direct : (416) 538-5483 Work : (416) 535-0123 Ext. 1277 Fax : (416) 531-5584 zoach...@securax.org wrote: Edward Gray wrote: Hi, we are looking to roll out PBX IN A Flash at our office. The first group will be using Soft Phones (X-Lite appears to be the best and works in Windows, Apple Linux). tsk tsk tsk :P (I'm working for the zoiper.com :p ) There are many types of USB Headsets to choose from and a fairly broad price range. Is there any USB headsets people would recommend? I'm specifically interested in acceptable audio (speaker and microphone) quality for business calls but am sensitive to price as well. In reading online, the Logitech Premium headset does get some good reviews but the reviews appear to be more from consumer based. I'd much prefer real experience from the good people who are operating their own Asterisk implementations. Forget about logitech, they are toys, go for plantronics or gn netcom if this is for business use, the logitechs will probably fall apart in a month. (Been there done that:) Any advice? Thanks! -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended USB Headsets ?
I think we can conclude that hardphones should be used if you cannot under any circumstances loose the call (power goes down in building, phones still powered by PoE switches on UPS) or if you prefer/don't mind spending the extra on hardphones... and softphones if it doesn't make a difference. All this from a usb headset recommendations thread :) but on that subject... plantronics all the way, they seem to realise that agents will complain if the headsets hurt (too tight, pulls hair etc etc) and that agents don't really care about the equipment they are using and so need to be strong. we use non usb plantronics (no idea what model) headsets, never had one break except for chairs running over cables, but a few cable ties stopped that from happening ever again though! 2009/3/23 Steve Totaro stot...@totarotechnologies.com On Mon, Mar 23, 2009 at 12:39 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Mar 23, 2009 at 12:03:42PM -0400, Steve Totaro wrote: On Mon, Mar 23, 2009 at 11:44 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Mar 23, 2009 at 03:09:54PM +, Gordon Henderson wrote: On Mon, 23 Mar 2009, Tzafrir Cohen wrote: A lot of the issues I've seen have been more to do with comfort than quality... If you're going to wear something all day then it had better be comfortable to use and easy to clean... A hardware phone is way less screen space to use for the user interface. And that is bad how? A small app, screen pop, or whatever would work very well and not potentially kill a phone call, sale, or lose you money. Seems like you have a lousy window manager. If a phone is so important it should be on top (or above that). You first asked, Can you just send it to the background so it does not occupy and [screen] space while not in use? Trivial for a software phone. Then you said Seems like you have a lousy window manager. If a phone is so important it should be on top (or above that). Circular and tiring doublespeak trying to impersonate some kind of logic. You ask the users with those fancy keyboard with the extra 20 buttons not to use one of them for Answer call? You ask them to actually get their hands off the keyboard (and mouse?) to answer a call? How very productive. I don't ask user's to use softphones, obviously you have missed the point. I was offering app alternatives that could control the Real phone and the other way around, period. Please don't try to spin my words and especially your own. I have caught you doing this many, many times over the years and it doesn't work with me. Selectively snipping or misquoting is deceitful and you are guilty of it regularly. Check the archives. Can you change a theme of a hardware phone? How many people actually would do this? In any fortune 500 or higher company I have worked in (first, softtphones would never even be considered) and support of skins themes would not be entertained. What would it take you to put the right icons on a Cisco phone so the dumb secretary could understand what they mean? Could you group the function buttons in logical groups? I am sure I could if you gave some example other than right icons and group function buttons Lack of clarity leads to no answers. What would it take you to get a nice Polycom phone but with the big buttons the old Grandstream Bug-tone has, so that granny can use it? An ATA with big ole buttons, they exist, I see commercials for them all the time. Only people I see changing Themes of phones are teenie boppers putting bling on their cellies. Or granny[1]. Or maybe a PHB who happens to also have some pretty thick glasses? Aunt Tilly was laid-off, anyone else can simply use an ATA and a specialized phone. Can you just send it to the background so it does not occupy and [screen] space while not in use? Trivial for a software phone. Yes, just push the hard phone to the side. 1. Messing your desk in the process. 2. Moving parts. Increases the chance that the wire in the back will disconnect. Causing you eventually to lose a call. Which we cannot afford in a F500C. 1. Moving a phone on a clean desk does not make it messy. 2. Park the call, but I have moved thousands of phones and unless someone has the wire pulled so tight, there was absolutely no chance of pulling out the power. If you mean a patch cable, then buy decent patch cables. All moot points, broken down one by one. Other posters don't seem to much care either.. [1] which happens to be aunt Tilly? Those silly examples make me think of the Aunt Tilly threads. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- Thanks,
Re: [asterisk-users] Recommended USB Headsets ?
Just noticed you said DECT headsets... so what i wrote had nothing to do with them, but i've used them too i think, excellent quality, tested them with an aastra phone and worked great. 2009/3/23 Geraint Lee gera...@gmail.com hehe, nice. i've used those headsets hooked up to an old 4400 (well, via an alcatel phone obviously)... not bad at all and i know the support department could fix most of them - usual problems were recrimping the rj11 connections and resoldering the bits inside the volume box thingy (assuming we're thinking of the same ones). On that subject... those hardphones suffered a lot of use and needed fixing regularly (not from abuse) but from general use (pick up the phone, dialing phone numbers etc etc) and after a while the connections start to fall off the board they are connected to, so maybe another reason why hardphones aren't so good? 2009/3/23 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 23 Mar 2009, Geraint Lee wrote: but on that subject... plantronics all the way, they seem to realise that agents will complain if the headsets hurt (too tight, pulls hair etc etc) and that agents don't really care about the equipment they are using and so need to be strong. Interesting little anecdote here... Had a client want a cordless headset for their desk phone (A Snom, but that's not important). I asked if he had a preference and his answer was The ones the girls use on the late night Sky channels ... He reckoned if they could wiggle about on-screen with the headsets taking calls from punters for several hours then they must be OK... So some market research was required here ;-) ... and he got Plantronics DECT headsets to go with the Snom desk phones. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended USB Headsets ?
hehe, nice. i've used those headsets hooked up to an old 4400 (well, via an alcatel phone obviously)... not bad at all and i know the support department could fix most of them - usual problems were recrimping the rj11 connections and resoldering the bits inside the volume box thingy (assuming we're thinking of the same ones). On that subject... those hardphones suffered a lot of use and needed fixing regularly (not from abuse) but from general use (pick up the phone, dialing phone numbers etc etc) and after a while the connections start to fall off the board they are connected to, so maybe another reason why hardphones aren't so good? 2009/3/23 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 23 Mar 2009, Geraint Lee wrote: but on that subject... plantronics all the way, they seem to realise that agents will complain if the headsets hurt (too tight, pulls hair etc etc) and that agents don't really care about the equipment they are using and so need to be strong. Interesting little anecdote here... Had a client want a cordless headset for their desk phone (A Snom, but that's not important). I asked if he had a preference and his answer was The ones the girls use on the late night Sky channels ... He reckoned if they could wiggle about on-screen with the headsets taking calls from punters for several hours then they must be OK... So some market research was required here ;-) ... and he got Plantronics DECT headsets to go with the Snom desk phones. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-tenant with receptionist features for managed service
We can put about 9/10 calls using SIP/gsm through our BT Business Network ADSL package connection (832kbit upstream, £65/month) before you notice the quality starting to drop, but you could always get two connections and bond them together into one using openvpn or some other method if you wanted to. 2009/3/17 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 16 Mar 2009, Gavin Henry wrote: Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. Gavin, You won't get 12 concurent G711 calls over a standard ADSL line in the UK. If you're on an ADSL2+ service you may get up to 1.1Mb/sec upload speed, but even then, 12 * 80 = 960Kb/sec which is really pushing it, so use G729, or get that 2Mb SDSL line in. Make sure it's a decent ISP too. Using IAX will give you a few extra channels though as the IP overhead is less. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. Have a look at: http://www.astassistant.com/ rather than FOP. Even has a Linux client which is nice... If anyone has any ideas on the best way to put this together, I'm all ears ;-) The consultant in me says Pay someone to do it for you :) However it's not that hard to do and setup if youve done something similar in the past - and your budget is tight. If you know you're going to get more of these, then go for it - spend your time on the software and front-end for the the first one, then the rest are clones... I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. When budgets tight - I've deployed a lot of Grandstream phones - might give you a bit more breathing space if you use (eg) GXP280's for the client phones and a GXP2000 + button box for the receptionist. You can save money by building your own hardware too. Atom mobo, 1GB of RAM and an OpenVox card running oslec is still overkill for this. I mostly use 1GHz VIA boards for these sort of projects with up to 60 extensions. Billings a PITA and other than what I've written myself, have never found anything that works the way I'm happy with... Good luck! I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Personally I'd stick the box on-site and have a central peering server or 2 in the DC - well that's how I do it ;-) You'll struggle to get properly redundant links in that budget range too - one JCB can ruin everyones day! Cheers, Gordon -- www.drogon.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Direct Dial-Out and CDR destination numbers
what about relogging the information using:
Set(CDR(customfield)=${CDR(originalfield)})
i think?
who knows, i might be wrong with all of this but i guess it will work...
2009/3/17 Matthias Urlichs matth...@urlichs.de
Hi,
as German phone numbers are variable_length, I need to use direct dial-out.
The problem is that only the part which appears in extensions.ael (and thus
in the argument to Dial()) is logged to the call data record.
What I want, obviously, is for the Dial() app to append the additional
digits to the CDR's destination number.
Is that possible?
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Re: [asterisk-users] Help Inbound number
are you sure calls from this provider are going to context 'default' ? sip.conf [procall] type=peer username=XX secret=XX context=default 2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and WebIntegration
I reverse the inbound calls so they appear as outbound calls for agents, all of our calls are managed by the dialer i've written and integrate directly to our CRM, essentially asterisk is only providing the SIP/IAX functionality to me everything else is done via php... so... inbound call comes in and gets parked in a php script stores in database as an outbound call, agents screen then pops and checks the database for the CLI so we can try to guess who's calling us and opens up all of their details. php script that is parking the inbound call then dials the allocated agents extension and connects the call. also on the dial command i have used Dial(SIP/1234,,A(beep)) so that the agent hears a beep when they get a call. Hope this enlightens you a bit on handling inbounds in this situation :) Cheers 2009/3/12 Kurian Thayil kurianmtha...@gmail.com Hi Geriant, My apologies for the delay in reply. We won't be using php but Perl and there is an AGI module for perl Asterisk::AGI. I may be using Manager API for sending Hangup signal. Im planning to write a bash script which perl invokes when hangup button is pressed in the web interface. Bash script telnets and sends Hangup signal to the manager API. I am not yet able to acheive sending commands via bash script using telnet. But I am trying. One thing that's confusing me is if in future, incoming facility needs to be activated and since Auto Answer feature in EyeBeam/Twinkle is ON, don't you think that would be a problem? I think for that, the possible work around will be using 2 softphones, say EyeBeam and Xlite together in the same PC. Configuring one extension in EyeBeam to make outbound calls (with Auto Answer enabled) and configuring Xlite with an extension which receives inbound calls. Do you have any suggestion on that? Regards, Kurian Mathew Thayil. On Tue, Mar 10, 2009 at 7:32 PM, Geraint Lee gera...@gmail.com wrote: If you're using a php i'd take a look at phpagi - there are others around for various different languages too. our agents use twinkle with auto-answer, the only reason they need to look at twinkle is if they need to perform a transfer (that too will soon be done from the web browser), you can do pretty much anything with the asterisk manager (originate the call and hangup the call and a load of other useful stuff) Cheers 2009/3/10 Kurian Thayil kurianmtha...@gmail.com Hi Steve, That worked beautifully. Thank you so much. But one question though. Imagine if I keep a Hangup Button in the interface and it should terminate the call. Will that be possible? This scenario happens when the user gets connected to an invalid phone number where the user have to manually disconnect. I don't plan to confuse the user by asking them to use eyebeam to disconnect the call. If it could be integrated to the web interface they just have to stick on to that alone. Is there any way? Regards, Kurian Mathew Thayil. On Tue, Mar 10, 2009 at 4:51 PM, Steve Totaro stot...@first-notification.com wrote: On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil kurianmtha...@gmail.com wrote: Hi All, Is there a way that I can include call dialing functionality in a webinterface. I have EyeBeam configured with a SIP user say 8440. Will I be able to design an inteface which agent can choose a number and the Dial without punching in the number in Eyebeam. I tried using the .call file. ie The agent can choose which number to dial from a web interface. Then, a .call file is created with the following informations. Channel: Zap/g2/9444204943 Context: inbound_support Extension: 8440 Priority: 0 Now, in the extensions.conf file, I mentioned the following under inbound_support context. [inbound_support] exten =8440,1,Dial(SIP/8440,55,tTo) exten =8440,2,Answer exten =8440,3,Hangup But, here the call gets connected only when the receiver end receives the call. When the receiver end picks up the phone, SIP/8440 rings. Is there any other way to implement this. I am not ready to use Vicidial (AstGUIClient) because the interface to be designed is too custom and the agent should have the list of numbers in front of them while they dial which cannot be done using Vicicial. Regards, Kurian Mathew Thayil. The following will ring the internal support personnel (8440) first, after answered, it will then dial the customer (14109850123) (Are you in Maryland?) Turn on auto-answer and it should be seamless. Stolen from Wiki: To create a call to 14109850123 on a SIP phones called bt101, here's the file you'd create in /var/spool/asterisk/outgoing (whatever name is good, of course must be accessible and deletable by asterisk GNU/Linux user): Channel: SIP/8440 MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [outgoing] # Context: outgoing # Extension: 14109850123 # Priority: 1 -- Thanks, Steve Totaro
Re: [asterisk-users] Outbound routing
If it's anything like the UK, it won't make a difference... for example: o2 mobile number ported to orange mobile... On most providers you still pay the o2 rate. three mobile ported to o2... you still pay the three rate (which isn't so good since it's far more expensive than o2). Cheers 2009/3/13 Asterisk aster...@abraxas.si Dear All, I have a small call center in which I have to define least cost routing for outbound calls. For now I have always done this by routing numbers to different providers according to the number prefix. However, a new law became effective now which allows people to switch between providers without changing their telephone numbers. This makes least cost routing based on number prefixes much less effective. Are there any known solutions for this? Thanks in advance, Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and WebIntegration
If you're using a php i'd take a look at phpagi - there are others around for various different languages too. our agents use twinkle with auto-answer, the only reason they need to look at twinkle is if they need to perform a transfer (that too will soon be done from the web browser), you can do pretty much anything with the asterisk manager (originate the call and hangup the call and a load of other useful stuff) Cheers 2009/3/10 Kurian Thayil kurianmtha...@gmail.com Hi Steve, That worked beautifully. Thank you so much. But one question though. Imagine if I keep a Hangup Button in the interface and it should terminate the call. Will that be possible? This scenario happens when the user gets connected to an invalid phone number where the user have to manually disconnect. I don't plan to confuse the user by asking them to use eyebeam to disconnect the call. If it could be integrated to the web interface they just have to stick on to that alone. Is there any way? Regards, Kurian Mathew Thayil. On Tue, Mar 10, 2009 at 4:51 PM, Steve Totaro stot...@first-notification.com wrote: On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil kurianmtha...@gmail.com wrote: Hi All, Is there a way that I can include call dialing functionality in a webinterface. I have EyeBeam configured with a SIP user say 8440. Will I be able to design an inteface which agent can choose a number and the Dial without punching in the number in Eyebeam. I tried using the .call file. ie The agent can choose which number to dial from a web interface. Then, a .call file is created with the following informations. Channel: Zap/g2/9444204943 Context: inbound_support Extension: 8440 Priority: 0 Now, in the extensions.conf file, I mentioned the following under inbound_support context. [inbound_support] exten =8440,1,Dial(SIP/8440,55,tTo) exten =8440,2,Answer exten =8440,3,Hangup But, here the call gets connected only when the receiver end receives the call. When the receiver end picks up the phone, SIP/8440 rings. Is there any other way to implement this. I am not ready to use Vicidial (AstGUIClient) because the interface to be designed is too custom and the agent should have the list of numbers in front of them while they dial which cannot be done using Vicicial. Regards, Kurian Mathew Thayil. The following will ring the internal support personnel (8440) first, after answered, it will then dial the customer (14109850123) (Are you in Maryland?) Turn on auto-answer and it should be seamless. Stolen from Wiki: To create a call to 14109850123 on a SIP phones called bt101, here's the file you'd create in /var/spool/asterisk/outgoing (whatever name is good, of course must be accessible and deletable by asterisk GNU/Linux user): Channel: SIP/8440 MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [outgoing] # Context: outgoing # Extension: 14109850123 # Priority: 1 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple asterisks in a server
yes, you need to make sure bindaddr is set correctly in iax.conf, sip.conf, dundi.conf, manager.conf and any other files that might include bindaddr for BOTH instances of asterisk, you can't allow one to bind to all ip's and the other just to bind to one - it won't work. 2009/2/25 Rilawich Ango maillist...@gmail.com It seems better to install once with multiple instances. Do we need to take care the port or IP of each instance? - Show quoted text - On Wed, Feb 25, 2009 at 5:36 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Klaus Darilion wrote: Rilawich Ango wrote: Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Just to have several Asterisk instances on a single server you do not need to install it multiple times. Install it once and start it multiple times. Of course you have to have a dedicated configuration for each server, eg: /etc/asterisk/instance1/* /etc/asterisk/instance2/* /etc/asterisk/instance3/* Then you start the Asterisk process and specify the location of the asterisk.conf file. asterisk -C /etc/asterisk/instance1/asterisk.conf asterisk -C /etc/asterisk/instance2/asterisk.conf asterisk -C /etc/asterisk/instance3/asterisk.conf Further, in asterisk.conf specify for each asterisk instance a different location of: spool directory, PID file, btw: I use a common /var/lib/asterisk/ as I want to have the same sounds for all instances. This gives a problem when you use 1.4, as 1.4 can not configure the location of astdb. For these you have to apply this patch: http://bugs.digium.com/view.php?id=14257 regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple asterisks in a server
Yes it's possible.. When you install use... ./configure --prefix=/usr/local/asterisk2 or something like it. I had to change astrundir (in asterisk.conf) as well. One thing to watch out for is that if you run make samples it will overwrite the ones stored in /etc/asterisk and not where you'd expect them to be in /usr/local/asterisk2/etc/asterisk (or at least it di dwhen i did it!). and for a helping hand i symlinked /usr/local/asterisk2/sbin/asterisk to /usr/local/sbin/asterisk2 and /usr/local/asterisk2/sbin/safe_asterisk to /usr/local/sbin/safe_asterisk2 Cheers Geraint You will also need to look at asterisk.conf in the new installation directory and as a quickfix to get it running, use a different location for astrundir 2009/2/24 Rilawich Ango maillist...@gmail.com Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple asterisks in a server
Almost forgot, you need to make sure you bind each instance to either it's own IP address or different ports on the same ip, i used 2 IP's for it and never hda a problem. 2009/2/24 Geraint Lee gera...@gmail.com Yes it's possible.. When you install use... ./configure --prefix=/usr/local/asterisk2 or something like it. I had to change astrundir (in asterisk.conf) as well. One thing to watch out for is that if you run make samples it will overwrite the ones stored in /etc/asterisk and not where you'd expect them to be in /usr/local/asterisk2/etc/asterisk (or at least it di dwhen i did it!). and for a helping hand i symlinked /usr/local/asterisk2/sbin/asterisk to /usr/local/sbin/asterisk2 and /usr/local/asterisk2/sbin/safe_asterisk to /usr/local/sbin/safe_asterisk2 Cheers Geraint You will also need to look at asterisk.conf in the new installation directory and as a quickfix to get it running, use a different location for astrundir 2009/2/24 Rilawich Ango maillist...@gmail.com- Show quoted text - Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple asterisks in a server
I do it for CDR, when using the originate command via the manager and initiate a call to a phone which then connects to an agi script upon answer, the cdr stops at the point of answer and there is no other created, which of course is useless for billing customers - there may very well be a way to make the cdr continue after it seems to stop logging, or is it a bug? either way, the quickest solution for me was to install a second copy and send all calls out on a second installation with accurate cdr logging. 2009/2/24 David Backeberg dbackeb...@gmail.com On Tue, Feb 24, 2009 at 2:59 AM, Rilawich Ango maillist...@gmail.com wrote: Hi all, Is it possible to install more than 1 asterisk in a single server? Can somebody help me understand why you would want to do this? I suppose development versus production, but wouldn't you also want better separation, like virtualization? - Show quoted text - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple asterisks in a server
should have thought of that one lol Cheers for the tip... will be changing my setup to this lol 2009/2/24 Klaus Darilion klaus.mailingli...@pernau.at Rilawich Ango wrote: Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Just to have several Asterisk instances on a single server you do not need to install it multiple times. Install it once and start it multiple times. Of course you have to have a dedicated configuration for each server, eg: /etc/asterisk/instance1/* /etc/asterisk/instance2/* /etc/asterisk/instance3/* Then you start the Asterisk process and specify the location of the asterisk.conf file. asterisk -C /etc/asterisk/instance1/asterisk.conf asterisk -C /etc/asterisk/instance2/asterisk.conf asterisk -C /etc/asterisk/instance3/asterisk.conf Further, in asterisk.conf specify for each asterisk instance a different location of: spool directory, PID file, regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you use? .conf or AEL?
.conf all the way, purely because i only noticed that extensions.ael even existed a couple of months back, i should pay more attention really :p but until it's broke, i can't be bothered to fix it. 2009/2/10 Alan Lord (News) alansli...@gmail.com Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed new and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security issue
what about something along the lines of... iptables -A INPUT -p udp --dport 5060 -j DROP iptables -A INPUT -p udp -s 192.168.0.0/24 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -s 10.0.0.0/8 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -s 66.66.66.66 --dport 5060 -j ACCEPT Cheers 2009/2/9 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Fri, 6 Feb 2009, oumar ndiaye wrote: Is there a way to restrict connection to my asterisk server to users based on their IP addresses, and not just password. I have some hackers who connect to my server to make illegitimate solicitation calls to people. I had to shutdown the server for now until I find a solution. ANY HELP? I'm curious about hackers getting in when you have username and passwords set. How are they cracking the passwords in the first place? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security issue
well, you got the general idea :) 2009/2/9 Tzafrir Cohen tzafrir.co...@xorcom.com On Mon, Feb 09, 2009 at 11:09:34AM +, Geraint Lee wrote: what about something along the lines of... iptables -A INPUT -p udp --dport 5060 -j DROP iptables -A INPUT -p udp -s 192.168.0.0/24 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -s 10.0.0.0/8 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -s 66.66.66.66 --dport 5060 -j ACCEPT Err... I guess you meant: iptables -A INPUT -p udp -s 192.168.0.0/24 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -s 10.0.0.0/8 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -s 66.66.66.66 --dport 5060 -j ACCEPT # only if previous three did not match: iptables -A INPUT -p udp --dport 5060 -j DROP -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Michael Graves post
Still doesn't work but i'm guessing it's to do with not being friends with Michael? 2009/2/9 Dean Collins d...@cognation.net http://tinyurl.com/c4qbcj is that better for you? Cheers, Dean -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Monday, 9 February 2009 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Michael Graves post On 9 Feb 2009, at 14:20, Dean Collins wrote: Michael Grave just posted a question about surround conferences. http://www.facebook.com/notes.php?id=564633430#/note.php? note_id=50097263908id=564633430index=0 I didn't see it posted on the ast-list, what do you think? Does something like this have potential? I'd love to listen in on one of these calls to see how it actually sounds if someone builds a trial version of 'N' deviations. Can we have a sensible link please? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
Could you not use some iptables to do this? I don't know the exact command you'd need but it could work something like... If the destination port is 5060 and destination ip is xxx then route via the default ip (so do nothing) If the destination port is 5061 and destination ip is xxx change the destination port back to 5060 and set secondary ip as the source? Just a thought... i'm guessing this would be able to do the job.. not sure what issues you might run in to by changing 5060 to 5061... but if it came to it you could try it by using an alternate ip and changing it back. Who knows... not sure if i've even read enough to understand the problem :) Cheers Geraint 2009/2/1 Mike l...@virtutel.ca At the risk of seeming impolite (I really am not), why not? Isn't Asterisk able to send packets using another interface using bindaddr? The problem, for the two of us, is that bindaddr is Asterisk-wide, and not per-peer. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Sunday, February 01, 2009 14:56 To: bilal ghayyad Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine Ah, that makes more sense. Asterisk binding to another IP is not the issue, actually, and even running another instance will not do what you need. Your problem is that the OS itself will stamp outbound packets with the main source IP of the main interface. Asterisk could be modified to send packets with specific IP source, but I don't think that would be a simple change. j On Sun, 1 Feb 2009, bilal ghayyad wrote: OK, if I send for my provider (the destination), it will authenticate based on the IP ONLY, this is the provider system. And once authenticated me based on that IP, it will give me all the schema related to this account. Sometimes I need to use another schema for some calls, I am not able until send for the provider from another IP. Did u get what I need? Regards Bilal --- On Sun, 2/1/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, February 1, 2009, 12:44 PM I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] CentOS and BAT File
learn to use google. http://tldp.org/LDP/Bash-Beginners-Guide/html/sect_02_02.html 2009/1/25 David @ULC ucoms2...@gmail.com *1) What name I have to save it.Like what extension ?* 3) How I save it ? *2) How to run it to execute it ?* Should i do vi autobatch and then type and then save it ? *** have to save. And Most importantly how to run it ?* On Sun, Jan 25, 2009 at 10:04 PM, David @ULC ucoms2...@gmail.com wrote: In windows, we use BAT file to execute few series of command , which help us in not writing each command manually everytime we want to execute those commands. In CentOS, I want to do the same thing. Any Advice ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
nload will show you current bandwidth usage, but i guess that isn't what you're looking for? http://sourceforge.net/projects/nload/ Cheers Geraint 2008/12/11 Shaun Wingrin [EMAIL PROTECTED] Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dies when external access is lost
just an idea, could it have something to do with DNS being unavailable, but that wouldn't really explain why it would die when ADSL is down... h. Cheers Geraint 2008/12/11 Phil Knighton [EMAIL PROTECTED] Hello Looking for some help with a rather odd problem. We have Asterisk 1.4.10 running on a Linux box, within our Windows domain. Our Domain Controller is a Windows 2003 server, providing the normal Windows domain functions, such as DHCP and DNS. When we lose either our Domain Controller (for a reboot/maintenance) or external ADSL access, Asterisk drops all SIP registrations - even internal SIP calls within the building no longer function. All of our SIP clients are assigned static IP addresses, and our incoming lines are via a Zaptel card using (currently) analog lines from our national telco. When the SIP registrations drop, Asterisk will still answer incoming calls via the Zap channels, but can't forward them anywhere. What is most confusing is a recent issue when our ADSL connections were all offline, and despite everything internal to the network working perfectly, all of the SIP phones stopped working and left us without phones for 4 hours. I'm suspecting that these two issues (losing connectivity when DC is unavailable and losing connectivity when ADSL drops) are related, but I can't figure out how? I'm sure I'm missing something simple in the config, but I've been tinkering with this issue since we were using Asterisk 1.2 and I've still not resolved it. Any help or comments would be appreciated... Thanks in advance Phil Phil Knighton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP CallerID Question
2008/12/11 Dave Fullerton [EMAIL PROTECTED]
Brent Davidson wrote:
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd like to somehow make the CallerID more useful. Currently if an
extension at Office1 dials an extension at Office2 the CID on the phone
at Office2 says Office1. The same thing happens if a person at
Office1 transfers an incoming call to Office2. The caller ID at Office2
always just says Office1.
What I would like to happen would be when Bob at Extension 12 at Office1
calls Office2 the caller ID at office 2 would say Bob in the name
files and 12 in the number field. If Bob does a blind transfer to an
extension at Office2 I would like the caller ID on the Office2 phone to
display the original caller's name and number.
I've read most of the documentation on the CallerID variables, but am
still having a bit of trouble wrapping my head around the necessary
logic to accomplish what I need to do, (mainly because I'm in the middle
of a totally unrelated project and am having trouble multi-tasking).
Could anyone give me a starting point?
Thanks,
Brent
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
additionally if you want to have the callerid to include office1 when
calling office2, you could change the callerid using
Set(CALLERID(name)=${CALLERID(name)} Office 1)
just before sending through to office 2
Something along those lines anyway, not entirely sure on the syntax or if
there's a better way to do it.. but i'm sure someone will correct me if i'm
wrong :)
Geraint
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Re: [asterisk-users] Execute AGI after answered Dial() has ended
use deadagi on the h extension maybe? Cheers Geraint 2008/12/10 Martin Tirsel [EMAIL PROTECTED] Hello, I am googling for a while but google seems to be broken today, no solution yet :D I have a PHP script which needs to be started after Dial() has ended. If there is no answer, busy, etc., it is not a problem, because dialplan continues after the Dial() application, but when the call is established (i call macro in Dial() with AGI execution, working ok) and after the call ends, dialplan execution stops on the Dial(). But I need dialplan to continue after call end and execute the AGI script. Is there any way how to do it? Thanks for help, mt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy strangeness...
Right after a bit of investigation i've found that it's because we're running a mysql database on the same server, it was fine all morning with a relatively low load on the server, now the rest of the agents have logged in the problem has returned! Time to buy a new database server... mystery solved! Cheers Geraint 2008/12/2 Thomas Kenyon [EMAIL PROTECTED] Geraint Lee wrote: Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on at the time. This is happening on mixmonitor recordings even if you're not listening in on chanspy too. Any suggestions? I don't have any suggestions, but this is similar to something I am experiencing with Chanspy in 1.4.21.1. If I spy on a call, then progressively throughout the call a delay is introduced. By the end of the call I can be listening to sound that is 10 seconds out of sync. (Then I don't get to hear the end of the call when the call is finished). This also leaves stale channels open. (the entry in show channels doesn't go away until the asterisk process is restarted). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy strangeness...
Doesn't look like anyone has any suggestions though, guess it's time to play until it's fixed then :) 2008/12/2 Thomas Kenyon [EMAIL PROTECTED] Geraint Lee wrote: Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on at the time. This is happening on mixmonitor recordings even if you're not listening in on chanspy too. Any suggestions? I don't have any suggestions, but this is similar to something I am experiencing with Chanspy in 1.4.21.1. If I spy on a call, then progressively throughout the call a delay is introduced. By the end of the call I can be listening to sound that is 10 seconds out of sync. (Then I don't get to hear the end of the call when the call is finished). This also leaves stale channels open. (the entry in show channels doesn't go away until the asterisk process is restarted). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor and ChanSpy strangeness...
Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on at the time. This is happening on mixmonitor recordings even if you're not listening in on chanspy too. Any suggestions? Cheers Geraint ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conneting Asterisk to Swyx pri
For the purposes of google since i'm sure someone is going encounter the same problem at some point in time... I have now fixed the problem, it seems asterisk and swyx refuse to talk properly with euroisdn.. it now works by changing it to use qsig... on the swyx side: windows device manager select the isdn card (sx2 dualpri) advanced isdn parameters tick use additional d-channel features tick enable q.sig restart the server asterisk side: modify /etc/asterisk/zapata.conf and set switchtype=qsig for your swyx trunks and one more thing... for wanpipe you need to set tdmv_dchan = 0 instead of 16 job done! Geraint 2008/10/7 Geraint Lee [EMAIL PROTECTED] I don't mean to be a pain, but i could really do with a heads up on this... does anyone have ANY ideas? I've trawled through google and come up with nothing except for questions with no answers... Cheers Geraint 2008/10/6 Geraint Lee [EMAIL PROTECTED] Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT - Swyx The above setup works fine... what i'm trying to achieve is BT SIP Trunks - Asterisk - Swyx I have connected to our BT (2 x ISDN30 UK) with asterisk and have no errors and can make and receive calls and it never dies... the problem comes when i try and connect asterisk to swyx... I can make calls from asterisk to the swyx system with no problems or errors, but... when i try and place a call from Swyx to asterisk i receive the following error: [Oct 6 10:56:20] ERROR[9794]: chan_zap.c:8250 zt_pri_error: !! Unexpected Channel selection 3 The call does complete as normal but after about 2 or 3 hours of calls passing through this setup i start receiving errors like the following: [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:9245 pri_dchannel: Hangup on bad channel 0/30 on span 3 [Oct 6 10:55:55] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use And eventually no more calls can be placed from swyx to asterisk... time for some configs... and before anyone says something about wanpipe3 and 4 having dchan=0, i tried with dchan=16 and no calls can be placed... I hope someone can point me in the right direction as we're trying to get rid of swyx since we're tied down by limiting software and excessive licensing costs. Thanks! Geraint pri show spans shows all spans as up and active. zap show status shows all as ok wanrouter status shows all as connected wanpipe1 and 2: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 16 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF= NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES wanpipe3 and 4: [devices] wanpipe3 = WAN_AFT_TE1, Comment [interfaces] w3g1 = wanpipe3, , TDM_VOICE, Comment [wanpipe3] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 16 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 3 TE_CLOCK= MASTER TE_REF_CLOCK= 1 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 3 TDMV_DCHAN = 0 TDMV_HW_DTMF= NO [w3g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES zaptel.conf: loadzone=uk defaultzone=uk #Sangoma A104 port 1 [slot:1 bus:16 span:1] wanpipe1 span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 hardhdlc=16 #Sangoma A104 port 2 [slot:1 bus:16 span:2] wanpipe2 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 hardhdlc=47 #Sangoma A104 port 3 [slot:1 bus:16 span:3] wanpipe3 span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 #Sangoma A104 port 4 [slot:1 bus:16 span:4] wanpipe4 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 I have also tried with hardhdlc=109 and have the same problem. zapata.conf: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes
Re: [asterisk-users] Conneting Asterisk to Swyx pri
I don't mean to be a pain, but i could really do with a heads up on this... does anyone have ANY ideas? I've trawled through google and come up with nothing except for questions with no answers... Cheers Geraint 2008/10/6 Geraint Lee [EMAIL PROTECTED] Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT - Swyx The above setup works fine... what i'm trying to achieve is BT SIP Trunks - Asterisk - Swyx I have connected to our BT (2 x ISDN30 UK) with asterisk and have no errors and can make and receive calls and it never dies... the problem comes when i try and connect asterisk to swyx... I can make calls from asterisk to the swyx system with no problems or errors, but... when i try and place a call from Swyx to asterisk i receive the following error: [Oct 6 10:56:20] ERROR[9794]: chan_zap.c:8250 zt_pri_error: !! Unexpected Channel selection 3 The call does complete as normal but after about 2 or 3 hours of calls passing through this setup i start receiving errors like the following: [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:9245 pri_dchannel: Hangup on bad channel 0/30 on span 3 [Oct 6 10:55:55] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use And eventually no more calls can be placed from swyx to asterisk... time for some configs... and before anyone says something about wanpipe3 and 4 having dchan=0, i tried with dchan=16 and no calls can be placed... I hope someone can point me in the right direction as we're trying to get rid of swyx since we're tied down by limiting software and excessive licensing costs. Thanks! Geraint pri show spans shows all spans as up and active. zap show status shows all as ok wanrouter status shows all as connected wanpipe1 and 2: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 16 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF= NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES wanpipe3 and 4: [devices] wanpipe3 = WAN_AFT_TE1, Comment [interfaces] w3g1 = wanpipe3, , TDM_VOICE, Comment [wanpipe3] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 16 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 3 TE_CLOCK= MASTER TE_REF_CLOCK= 1 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 3 TDMV_DCHAN = 0 TDMV_HW_DTMF= NO [w3g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES zaptel.conf: loadzone=uk defaultzone=uk #Sangoma A104 port 1 [slot:1 bus:16 span:1] wanpipe1 span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 hardhdlc=16 #Sangoma A104 port 2 [slot:1 bus:16 span:2] wanpipe2 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 hardhdlc=47 #Sangoma A104 port 3 [slot:1 bus:16 span:3] wanpipe3 span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 #Sangoma A104 port 4 [slot:1 bus:16 span:4] wanpipe4 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 I have also tried with hardhdlc=109 and have the same problem. zapata.conf: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes musiconhold=default rxgain=0.0 txgain=0.0 immediate=no ; BT switchtype=euroisdn group=1 context=from-bt signalling=pri_cpe ; Port 1 - BT channel = 1-15,17-31 ; Port 2 - BT channel = 32-46,48-62 ; Swyx overlapdial=yes group=2 context=from-swyx signalling=pri_net ; Port 3 - Swyx channel = 63-77,79-93 ; Port 4 - Swyx channel = 94-108,110-124 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
[asterisk-users] Conneting Asterisk to Swyx pri
Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT - Swyx The above setup works fine... what i'm trying to achieve is BT SIP Trunks - Asterisk - Swyx I have connected to our BT (2 x ISDN30 UK) with asterisk and have no errors and can make and receive calls and it never dies... the problem comes when i try and connect asterisk to swyx... I can make calls from asterisk to the swyx system with no problems or errors, but... when i try and place a call from Swyx to asterisk i receive the following error: [Oct 6 10:56:20] ERROR[9794]: chan_zap.c:8250 zt_pri_error: !! Unexpected Channel selection 3 The call does complete as normal but after about 2 or 3 hours of calls passing through this setup i start receiving errors like the following: [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:9245 pri_dchannel: Hangup on bad channel 0/30 on span 3 [Oct 6 10:55:55] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use And eventually no more calls can be placed from swyx to asterisk... time for some configs... and before anyone says something about wanpipe3 and 4 having dchan=0, i tried with dchan=16 and no calls can be placed... I hope someone can point me in the right direction as we're trying to get rid of swyx since we're tied down by limiting software and excessive licensing costs. Thanks! Geraint pri show spans shows all spans as up and active. zap show status shows all as ok wanrouter status shows all as connected wanpipe1 and 2: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 16 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF= NO [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES wanpipe3 and 4: [devices] wanpipe3 = WAN_AFT_TE1, Comment [interfaces] w3g1 = wanpipe3, , TDM_VOICE, Comment [wanpipe3] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 16 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 3 TE_CLOCK= MASTER TE_REF_CLOCK= 1 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 3 TDMV_DCHAN = 0 TDMV_HW_DTMF= NO [w3g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES zaptel.conf: loadzone=uk defaultzone=uk #Sangoma A104 port 1 [slot:1 bus:16 span:1] wanpipe1 span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 hardhdlc=16 #Sangoma A104 port 2 [slot:1 bus:16 span:2] wanpipe2 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 hardhdlc=47 #Sangoma A104 port 3 [slot:1 bus:16 span:3] wanpipe3 span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 #Sangoma A104 port 4 [slot:1 bus:16 span:4] wanpipe4 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 I have also tried with hardhdlc=109 and have the same problem. zapata.conf: [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=yes restrictcid=no usecallingpres=no threewaycalling=yes callreturn=yes transfer=yes cancallforward=yes musiconhold=default rxgain=0.0 txgain=0.0 immediate=no ; BT switchtype=euroisdn group=1 context=from-bt signalling=pri_cpe ; Port 1 - BT channel = 1-15,17-31 ; Port 2 - BT channel = 32-46,48-62 ; Swyx overlapdial=yes group=2 context=from-swyx signalling=pri_net ; Port 3 - Swyx channel = 63-77,79-93 ; Port 4 - Swyx channel = 94-108,110-124 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conneting Asterisk to Swyx pri
brilliant idea - except it would be a sunday morning and another problem the handsets that come with swyx aren't sip compatible :S Cheers Geraint 2008/10/6 Gordon Henderson [EMAIL PROTECTED][EMAIL PROTECTED] On Mon, 6 Oct 2008, Geraint Lee wrote: Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT - Swyx The above setup works fine... what i'm trying to achieve is BT SIP Trunks - Asterisk - Swyx I have connected to our BT (2 x ISDN30 UK) with asterisk and have no errors and can make and receive calls and it never dies... the problem comes when i try and connect asterisk to swyx... I can make calls from asterisk to the swyx system with no problems or errors, but... when i try and place a call from Swyx to asterisk i receive the following error: [Oct 6 10:56:20] ERROR[9794]: chan_zap.c:8250 zt_pri_error: !! Unexpected Channel selection 3 The call does complete as normal but after about 2 or 3 hours of calls passing through this setup i start receiving errors like the following: [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use [Oct 6 10:55:25] WARNING[9794]: chan_zap.c:9245 pri_dchannel: Hangup on bad channel 0/30 on span 3 [Oct 6 10:55:55] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't fix up channel from 63 to 92 because 92 is already in use And eventually no more calls can be placed from swyx to asterisk... time for some configs... and before anyone says something about wanpipe3 and 4 having dchan=0, i tried with dchan=16 and no calls can be placed... I hope someone can point me in the right direction as we're trying to get rid of swyx since we're tied down by limiting software and excessive licensing costs. So go in one Saturday morning, wire it up as you want (BT - Asterisk) and the re-configure all the SIP phones to talk directly to the asterisk box and not the swyx box, then arrange the the swyx box to misteriously die, then tell everyone what a good job it was that you were in on the weekend to re-configure the phones to use the asterisk box ;-) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
Linksys SRW248P or something like that... something from linksys anyway are quite capable of all you mentioned... maximum 24 port powered though iirc. Geraint 2008/10/6 Ken D'Ambrosio [EMAIL PROTECTED] Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
yes, thats the one i mean, 224p, the one i mentioned isn't capable of vlans properly (which was strange, since it said it did)... i never had any problems with them powering phones and cisco access points. 2008/10/6 Chris Bagnall [EMAIL PROTECTED] We've used Linksys SRW224P units at quite a few places without issue. For a little lower cost, we've also used Netgear FS726 series switches. Personally, I prefer the Linksys ones - they have a serial port for administration rather than relying on you doing it over the LAN (though they have a pretty web interface, too). The pretty web interface is less fussy than the Netgear one (which seems unreliable in non-Internet Exploder browsers). On the other hand, the Netgear is substantially less deep (an issue in some wallmount cabinets) and definitely a lot quieter. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
You can get incoming numbers from voipon.co.uk and a load of other companies in the UK... 0800 is free for them to ring but you have to pay for the call, you can also get regional numbers which are charged as a local call for them - stay away from 070 numbers though. 2008/9/29 Babcock, Michael Alex [EMAIL PROTECTED] hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
bt give an annoying message before it connects your call, well, annoying if you actually are using 070 as a personal number and callers aren't charged stupid amounts of money to call it. virgin(old ntl) and h3g don't give any warning message at all though. 2008/9/29 asterisk [EMAIL PROTECTED] Ofcom banned end user revenue share on 070 numbers several years ago although the provider makes money. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: 29 September 2008 18:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] uk tole-free dids? On Mon, 29 Sep 2008, Babcock, Michael Alex wrote: what are 70 numbers? Prefix 070 (then 8 more digits) These are so-called personal numbers. They're a blot and an anomaly. They are expensive to call and the recipient usually gets revenue from the calls. ie. they are premium rate, revenue generating numbers in disguise. In disguise becasue a lot of people (in the UK) don't realise this because they look like mobile numbers - which start 07[1-9] then 8 more digits, so they think they're calling a mobile, when in-fact it's costing them much more. Gordon On Sep 28, 2008, at 10:26 PM, Geraint Lee wrote: You can get incoming numbers from voipon.co.uk and a load of other companies in the UK... 0800 is free for them to ring but you have to pay for the call, you can also get regional numbers which are charged as a local call for them - stay away from 070 numbers though. 2008/9/29 Babcock, Michael Alex [EMAIL PROTECTED] hi; i do not know how it works in the uk, but is there an equalivent to our 866-877-888-800 numbers for london for say? I have some friends in london and want them to be able to call me in the states. Please help with where i can get the numbers, what they start with, how much they are, and what not. Thanks mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor + Originate
Hi everyone,
I'm trying to get calls to record with the following setup:
Using phpagi originate is called from a web application:
$asm-originate(Local/ . $row['extension'] . @sip-standard,
$row['phone_number'], sip-standard, 1, , , 7000);
The agent being called is extension Local/[EMAIL PROTECTED] and the number
originated for the agent is [EMAIL PROTECTED]
MixMonitor records fine up until 100 answers then the recording stops, but
the CLI output suggests that the call is still being recorded...
extensions.conf and CLI output below...
Anyone have any ideas?
extensions.conf:
exten = 100,1,MixMonitor(test.wav)
exten = 100,2,Dial(SIP/${EXTEN})
exten = _1XX,1,Dial(SIP/${EXTEN})
Output from CLI:
== Manager 'amis' logged on from 192.168.0.180
-- Executing [EMAIL PROTECTED]:1] Dial(Local/[EMAIL PROTECTED],2,
SIP/101) in new stack
-- Called 101
-- SIP/101-096f7ff8 is ringing
-- SIP/101-096f7ff8 answered Local/[EMAIL PROTECTED],2
-- Executing [EMAIL PROTECTED]:1]
MixMonitor(Local/[EMAIL PROTECTED],1, test.wav) in new stack
== Begin MixMonitor Recording Local/[EMAIL PROTECTED],1
-- Executing [EMAIL PROTECTED]:2] Dial(Local/[EMAIL PROTECTED],1,
SIP/100) in new stack
-- Called 100
-- Local/[EMAIL PROTECTED],1 requested special control 20, passing
it to SIP/100-09706218
== Manager 'amis' logged off from 192.168.0.180
== Spawn extension (sip-standard, 101, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
-- SIP/100-09706218 is ringing
-- SIP/100-09706218 answered SIP/101-096f7ff8
== Spawn extension (sip-standard, 100, 3) exited non-zero on
'SIP/101-096f7ff8'
== End MixMonitor Recording Local/[EMAIL PROTECTED],1
Cheers
Geraint
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Re: [asterisk-users] Reliable wireless SIP phones
I've used several hitachi dmp330's they work great, roam between wireless access points with no loss of audio or connection for that matter. it will be a great shame if hitachi stop producing them, they are the most reliable wireless sip phones i've come accross... stay well away from pirelli phones, they are very buggy. Cheers Geraint 2008/8/28 Cory Andrews [EMAIL PROTECTED] Just a heads up, Hitachi is effectively ceasing production of their IP5000 and IP3000 WiFi SIP Phonesproduct availability is next to nil on these. They also have no plans apparently to continue producing WiFi phones. Cory J. Andrews Director New Market Initiatives VoIP Supply, LLC. 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE [EMAIL PROTECTED] Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, August 28, 2008 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Reliable wireless SIP phones On Thursday 28 August 2008 08:06:37 Jaap Winius wrote: Are there any reliable wireless SIP phones available on the market yet? I've gotten a Hitachi WIP3000, which works great. Supports b g, all the wireless encryption standards, scans networks, everything a laptop softphone would do, but in a wireless handset. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] phpagi
You don't install it as such, you just include the files from your php scripts. On 19/03/2008, Carlos Carvalhar [EMAIL PROTECTED] wrote: Hello, How do I install phpagi? http://phpagi.sourceforge.net/ I couldn't find any info about setup in that site, and I couldn't email the developers…so I'm lost. I know it isn't a real question for this list, but I suppose many people here already have installed it. So, how can I install it? Thanks Carlos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
