Re: [Asterisk-Users] Are messages censored on this board?
On Monday 05 January 2004 16:23, John Coll wrote: I've submitted a message twice this evening and it has not appeared. Are messages censored on this board? Have patience. The mailing list is just a little slow today. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Are messages censored on this board?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Coll Sent: Tuesday, 6 January 2004 9:23 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Are messages censored on this board? I've submitted a message twice this evening and it has not appeared. Are messages censored on this board? I've got all 3 messages, the first two being two (2) copies of a message of despair. Cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to monitor calls initiated by .call file using manager interface?
Hi Here is my problem: I initiate a conference call by placing several .call files into /var/spool/asterisk/outgoing/ directory Asterisk starts calls and I can see events in the manager interface. At the same times there are other calls going on and there are many more events in the manager interface. How can I identify events that are related to the calls started via spool? I tried to pass additional variables in the call using SetVar: statement, but they do not get propagated into events. Is there a way to do what I need? Thank you Serge _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=dept/featurespgmarket=en-caRU=http%3a%2f%2fjoin.msn.com%2f%3fpage%3dmisc%2fspecialoffers%26pgmarket%3den-ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This is a test
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Tuesday, 6 January 2004 11:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] This is a test It appears that my replies aren't getting to the list. Just testing to see what is going on. Sean Works for me. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identifying the Originating Cisco SIP Gateway
On Monday 05 January 2004 19:45, Ray Burkholder wrote:
I have several Cisco SIP gateways sending calls to Asterisk. Because the
gateways don't have user-agents, they don't authenticate with Asterisk.
And because they don't authenticate, they use the default context in the
sip.conf file.
Is there a way to either:
A) identify the inbound gateway with a variable, in channel info, or the
manager interface? If there was a ${SIPDOMAIN} for the originator rather
than the destination, that would be cool, or
B) make the inbound gateway use the sip.conf file section belonging to it
via the host= line in the sip.conf file without user authentication, or
Last time I checked this, it worked as you want only if your SIP Gateways are
on port 5060. I was able to differentiate via the host=line, but as soon as
I used a Gateway on a port other than 5060, Asterisk was not able to match
the sip.conf entry and it used the default. I did not open a bug report as
this was not something we needed at the time. Give it a try and let me know
if you see the same thing.
Regards,
Andres
http://www.telesip.net
C) some other way I have yet to fathom
I'm trying to differentiate between legitimate gateways that initiate calls
vs other gateways that should get a very limited inbound capability.
Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunified.net
704 644 6999 x2002
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RE: [Asterisk-Users] I stumbled on this list...
Hi All, I am new to the list and have ordered myself a Asterisk Developer's Kit (TDM). I am just waiting on the order from a reseller of digium. I have yet to play around with the system when it arrives. And I haven't looked into the manual yet. I do hope I have all the hardwares to test first before I go and buy a sipura spa-2000 unit. I have my sister who is living in Germany. We both use broadband. I know you can use msn and yahoo to do voice, but I want to explore this thing as I am interested in this. Novice question Suppose I were to set up a debian with the Asterisk software, and I get a additional line(line2) locally. Can I make the one line(line2) be connected to a Asterisk and another connected to the broadband internet connection. Make it so that when I dial from any phone in locally to line2, it rings to the broadband phone hooked via the sipura device in Germany? Or Can I just get a ip phone for her? Dinesh. -Original Message- From: Nick Bachmann [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 8:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] I stumbled on this list... Hi there, I stumbled on this list mostly by accident. I came across Asterisk * as a means to help me get a better handle on my soaring telephone costs. Each month I look at my phone bills and my stomach just turns because I can not find any competition to Verizon which is the local anointed phone company around here. Trust me, it could be worse. You could have Ameritech, for example. Or Quest. In fact, I'd say Verizon is probably the best of the major RBOCs/ILECs (I'm in former GTE country, not NYNEX/Bell Atlantic, which might be different). But even then, they do leave something to be desired. Since I am a neophyte at all this I was wondering if some kind soul would confirm/disconfirm my assumptions about this software called Asterisk *. 1) Am I correct to assume that there is a way to dump Verizon and strictly go VOIP in a SOHO situation? Yes. 2) Can 1-800 numbers terminate to a VOIP assigned number? This a service of the provider that many do offer. 3) With VOIP am I under the assumption that one must also purchase licenses for such service to work. No, you don't need to purchase licenses. Asterisk is published under the GNU General Public License. 4) Who are the companies I can purchase VOIP service from? I need numbers in my local area code, plus I need some kind of unlimited VOIP service Asia - mainly to Taiwan. Google can help with this. There are a couple of Big providers such as NuPhone, Voicepulse, etc. 5) Am I being unrealistic in my savings by implementing an Asterisk * PBX in our SOHO situation. There is not doubt, that VoIP can save you money. If you're talking about a number or two, you'd be best to stick with just getting an ATA (a little box that turns your phone line into VoIP) from your provider, and not messing with a full fledged PBX. But if you have a PBX or Key System already, you might consider using Asterisk. As a warning, however, if you don't know much about (pick two) telephony, Linux, or software development, and you're still interested, you might to well to find somebody (like, say, me :-) who can give you a turnkey system. If you do want to go it alone, there is a book at http://www.asteriskdocs.org/, a wiki at http://www.voip-info.org, and a search engine at http://search.voip-forum.com/. Please utilize these, as they will answer most of your questions about *. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: This email is confidential and may be privileged. If you are not the intended recipient, please delete it and notify us immediately. Please do not copy or use it for any purpose, or disclose its contents to any other person as it may be an offence under the Official Secrets Act. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Fwd: reject connect from iaxtel.com]
I have just resolved the problem and included in this email, hoping it will help other people of get incoming Iaxtel working. Asterisk, at the moment, does not work with FWD. But Iaxtel and FWD can interconnect therefore having an Iaxtel account is the same as FWD account. I have read through a lot of google search and the materials are scare. I think it would be useful to document it. For Iaxtel to direct calls into your *, you will need to set up your iax.conf as follows: Under general section, the default will be fine. You need to register iax with iaxtel so that FWD user can contact you. [general] register: dkwok:[EMAIL PROTECTED] But you have to set up a client to allow Iaxtel to put calls through. [iaxtel] type=friend host=iaxtel.com context=from-iaxtel For type, type=friend * will allow both incoming and outgoing type=user * will allow incoming type=peers * will allow outgoing context will direct the incoming call to the context section in extensions.conf. In this case, [from-iaxtel]. You will need to setup exten under [from-iaxtel]. When Iaxtel sending through call it is in the format [EMAIL PROTECTED]/[EMAIL PROTECTED], therefore, you have to set up your context in extensions.conf as follows: [from-iaxtel] exten =s,1,Wait(1) exten =s,2,Answer exten =_.,3,Dial(sip/1001,20,tr) exten =_.,4,Hangup Have fun with *, thanks to Mark and his team. David Kwok ---BeginMessage--- Hi All I have problem trying to receive incoming calls from iaxtel.com. The error message is rejected connect from ip address - iaxtel.com. I have set up the iax.conf file as follow: port=5036 allow=gsm register=dkwok:[EMAIL PROTECTED] [dkwok] type=friend context=from_iaxtel My extensions.conf is as follows: [from_iaxtel] exten = 17001813482,1,Dial(sip/1001,20,tr) iax2 show registry 69.73.19.178:4569 dkwok 203.219.xxx.xxx:1200 60 Registered But when connection is attempted, the console says : File chan_iax2.c, Line 4301 (socket_read): Rejected connect attempt from 69.73.19.178. Any pointer will be appreciated. Also when iaxtel put call through local iax server does it send username password and is it matched up with the client section in iax.conf? David Kwok smime.p7s Description: S/MIME Cryptographic Signature ---End Message--- smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Interfacing Asterisk with PSTN network (Nortel SL100 PBX)
Hello, I am at a stage when I can start interfacing my standalone asterisk system with our organization PSTN network. We are using Nortel SL 100 PBX. I have already install T100P T1 card. Could anyone please give me pointers on how to go about interfacing it. What are the things I should keep in mind while doing it. Best regards, Tony
[Asterisk-Users] Everyone is busy at this time message ?
Hello I can call SNOM 200 from Analog Phone connected to TFM400P card. But whenever I try calling same SNOM 200 from CISCO IP 7905G phone I get Everyone is busy at this time message and call goes to voicemail. Could anyone tell where might be the problem, why I am getting BUSY message though the SNOM 200 phone is not in use. I have installed the latest firmware version on SNOM 200 (SNOM200-SIP 2.03h) Regards. Tony
RE: [Asterisk-Users] FW: This newbie gives up for now - sadly (2)
[EMAIL PROTECTED] wrote: This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let the two speak or get back the original caller - an utterly fundamental office procedure on a PBX. [SNIP] I found comments like This is where it might come down to redesigning the way calls are dealt with in an organization. Sometimes new phone systems do this, and hopefully the company sees new efficiencies with dealing with the customer in general. unhelpful and out of touch with user's and managers needs: Actually, this feature is extremely simple to use, and I don't understand why you might have asked the question and got anything other than the simple instructions on how to make it work. In fact, the default sample conf files already includes the needed config details. Of course, that implies that you are using hardware that supports that function. AFAIK, only Zapata connected hardware and some IP phones support that feature. So, if you are trying to use CAPI, or I4L, or IP phones, then maybe you are having that problem, and maybe it isn't possible. Other people on this list are better qualified to respond, but I am pretty sure the cisco and snom phones are capable of this. So, it might be that you have chosen non-optimal hardware to 'test' asterisk with. It would be 'better' to choose the easiest hardware to learn the software, and then after you know about the software, try with more complex/less supported hardware. It would seem to me that a lot of people (and hey, I did this too), try to use non-digium hardware to 'test' asterisk with before going out and buying the digium hardware for full production use. However, this makes it much more difficult to 'test' asterisk because it is harder to configure, and often causes problems you wouldn't normally have (ie, echo, etc). (Of course, there are STILL valid reasons for not using digium hardware!! Ie, in Australia, it is still illegal to use any of their hardware (for PSTN connectivity) because they do not have the relevant approval. Yes, the E400P is supposedly approved, but where is the paperwork/stickers/etc? Is that approval going to carry across to the TE405P ?? In fact, where is the TE405P?) [SNIP] help other newbies to get going but I think its time to give up and re-visit Asterisk in some months time. I am really disappointed not to be able to use asterisk now. This can often work surprisingly well. Just going away and coming back in a few months allows two things: A) You have time to mature/learn new things about Linux/IP Telephony. B) The project has time to mature, new/better documentation + more features + more bug fixes. This doesn't just apply to you, but hopefully the above makes you sit up and consider that you are blaming your problems/difficulty on asterisk when in fact you should blame to in-compatible hardware or even the protocols you are forcing asterisk to use (ie, SIP/H323) Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR Question
Hello In my IVR menu whenever user select the option number 1 then it should jump to echo context, I think call did jump to echo context but I always get the following warning and I hear couple of beeps and then call hung ups. -- Goto (echo,s,1) WARNING[1227879616]: File pbx.c, Line 1160 (pbx_extension_helper): No application ' Background ' for extension (echo, s, 1) extension.conf has following lines under echo context [echo] exten = s, 1, Background (demo-echotest) exten = s, 2, Echo exten = s, 3, Background (demo-echodone) exten = s, 4, Goto(mainmenu,s,6) Could you please tell me where I could be wrong ? Regards, Tony
Re: [Asterisk-Users] I stumbled on this list...
before buying spa-2000 i would recommend ur sister to use one of the softphones (X-tern etc). First make sure she can connect to ur asterisk then try to establish a softphone-to-softphone communication. X-tern (Ur sister) --- * - X-Tern (You). Get this setup working first. -B - Original Message - From: Birlasekaran Dinesh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 05, 2004 9:24 PM Subject: RE: [Asterisk-Users] I stumbled on this list... Hi All, I am new to the list and have ordered myself a Asterisk Developer's Kit (TDM). I am just waiting on the order from a reseller of digium. I have yet to play around with the system when it arrives. And I haven't looked into the manual yet. I do hope I have all the hardwares to test first before I go and buy a sipura spa-2000 unit. I have my sister who is living in Germany. We both use broadband. I know you can use msn and yahoo to do voice, but I want to explore this thing as I am interested in this. Novice question Suppose I were to set up a debian with the Asterisk software, and I get a additional line(line2) locally. Can I make the one line(line2) be connected to a Asterisk and another connected to the broadband internet connection. Make it so that when I dial from any phone in locally to line2, it rings to the broadband phone hooked via the sipura device in Germany? Or Can I just get a ip phone for her? Dinesh. -Original Message- From: Nick Bachmann [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 8:37 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] I stumbled on this list... Hi there, I stumbled on this list mostly by accident. I came across Asterisk * as a means to help me get a better handle on my soaring telephone costs. Each month I look at my phone bills and my stomach just turns because I can not find any competition to Verizon which is the local anointed phone company around here. Trust me, it could be worse. You could have Ameritech, for example. Or Quest. In fact, I'd say Verizon is probably the best of the major RBOCs/ILECs (I'm in former GTE country, not NYNEX/Bell Atlantic, which might be different). But even then, they do leave something to be desired. Since I am a neophyte at all this I was wondering if some kind soul would confirm/disconfirm my assumptions about this software called Asterisk *. 1) Am I correct to assume that there is a way to dump Verizon and strictly go VOIP in a SOHO situation? Yes. 2) Can 1-800 numbers terminate to a VOIP assigned number? This a service of the provider that many do offer. 3) With VOIP am I under the assumption that one must also purchase licenses for such service to work. No, you don't need to purchase licenses. Asterisk is published under the GNU General Public License. 4) Who are the companies I can purchase VOIP service from? I need numbers in my local area code, plus I need some kind of unlimited VOIP service Asia - mainly to Taiwan. Google can help with this. There are a couple of Big providers such as NuPhone, Voicepulse, etc. 5) Am I being unrealistic in my savings by implementing an Asterisk * PBX in our SOHO situation. There is not doubt, that VoIP can save you money. If you're talking about a number or two, you'd be best to stick with just getting an ATA (a little box that turns your phone line into VoIP) from your provider, and not messing with a full fledged PBX. But if you have a PBX or Key System already, you might consider using Asterisk. As a warning, however, if you don't know much about (pick two) telephony, Linux, or software development, and you're still interested, you might to well to find somebody (like, say, me :-) who can give you a turnkey system. If you do want to go it alone, there is a book at http://www.asteriskdocs.org/, a wiki at http://www.voip-info.org, and a search engine at http://search.voip-forum.com/. Please utilize these, as they will answer most of your questions about *. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: This email is confidential and may be privileged. If you are not the intended recipient, please delete it and notify us immediately. Please do not copy or use it for any purpose, or disclose its contents to any other person as it may be an offence under the Official Secrets Act. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This newbie gives up for now - sadly
TC wrote: Hmm what channel type (Zap, SIP, H323 ??) on a Zap channel I just hook flash (this puts call 1 on hold), then i hear dial tone, I dial another end pt talk to that extension then, hook flash again now we are on a 3-way call, at that point can stay on the call or simply hang up New wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20zap%20transfer Linked from: http://www.voip-info.org/tiki-index.php?page=Asterisk%20PBX%20functions I need help updating the PBX functions page. I have not worked that much with PBXs, so I don't really understand all functions. I think it would be beneficial for all of us if we explained Asterisk from a PBX function list, so a potential customer/user could compare Asterisk with a commercial PBX. There's a lot of built in functions, some functions solved by extensions.conf magic and some functions supported in some channels, but not in others. Thank you for your assistance with completing/correcting this page! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Everyone is busy at this time message ?
tony banks wrote: Hello I can call SNOM 200 from Analog Phone connected to TFM400P card. But whenever I try calling same SNOM 200 from CISCO IP 7905G phone I get Everyone is busy at this time message and call goes to voicemail. Could anyone tell where might be the problem, why I am getting BUSY message though the SNOM 200 phone is not in use. I have installed the latest firmware version on SNOM 200 (SNOM200-SIP 2.03h) Please do a CLI sip debug trace and mail us. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR Question
tony banks wrote: Hello In my IVR menu whenever user select the option number 1 then it should jump to echo context, I think call did jump to echo context but I always get the following warning and I hear couple of beeps and then call hung ups. -- Goto (echo,s,1) WARNING[1227879616]: File pbx.c, Line 1160 (pbx_extension_helper): No application ' Background ' for extension (echo, s, 1) extension.conf has following lines under echo context [echo] exten = s, 1, Background (demo-echotest) exten = s, 2, Echo exten = s, 3, Background (demo-echodone) exten = s, 4, Goto(mainmenu,s,6) Could you please tell me where I could be wrong ? The error message assist you, curiously enough. Beware of spaces! exten = s,1,Background(demo-echotest) should work. Is there a reason why Asterisk doesn't just remove spaces in these places? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to monitor calls initiated by .call file using manager interface?
You can use the asterisk management interface to query for extension status etc - see http://www.voip-info.org/wiki-Asterisk+manager+API. You may need to supply a channel number for the device you want to monitor. This is usually derived from the name you supplied for the extension in the relevant .conf (eg sip.conf). Iain --On Monday, January 5, 2004 11:05 pm -0500 Serge Mankovski [EMAIL PROTECTED] wrote: Hi Here is my problem: I initiate a conference call by placing several .call files into /var/spool/asterisk/outgoing/ directory Asterisk starts calls and I can see events in the manager interface. At the same times there are other calls going on and there are many more events in the manager interface. How can I identify events that are related to the calls started via spool? I tried to pass additional variables in the call using SetVar: statement, but they do not get propagated into events. Is there a way to do what I need? Thank you Serge _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=dept/featurespgmarket=en-caRU=http%3a%2f%2fjo in.msn.com%2f%3fpage%3dmisc%2fspecialoffers%26pgmarket%3den-ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not working with session border controller
Hi All My company has bought a license for a session border controller called Middle. It look like this software acts like a proxy and registers devices to the asterisk. the problem (As per Middle guys) is that Middle sends a registration request to the Asterisk asking it to reply to the port 7062, but the Asterisk sends back the packet to the originating port 7060. This confuses the Middle and the registration process stalls @ Proxy Authentication Required. When i talked to person responsible, he told me that Asterisk is not according to the RFC and he asked me to find a solution to fix this problem. The only info i got about middle is here (http://sip-martini.homeip.net/venkat/middle.html) Please help me in fixing this problem. I really dont know what to do as i'm not fully aware of SIP protocol or Middle. Thanks Venkat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I stumbled on this list...
Hi! I have my sister who is living in Germany. We both use broadband. If you have a dynamic IP or are not permanently on-line: Let your sister register with FWD (Free World Dialup, fwd.pulver.com) and use that to call her. You can do the same with your Asterisk box, i.e. let Asterisk register with FWD. If you have a static IP and are on-line 24/7: She can register her SIP software phone or hardware phone with your Asterisk directly. Anyone else who wants to call her has to call your Asterisk server. Hope this helps, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queue and Agent Statistics
Dear Group, I need to write a couple of reporting tools for my Call Center Asterisks implementation. I have multiple call queues with multiple agents that can sign in and based on gain access to multiple queues based on their assignments. I would like to write a script to collect call statistics for the agents the queues and the calls, and to put these into MySQL for reporting purposes. I'm thinking that each one of my customers would have their own table with the relevant information. Some of the statistics I'm looking for is; Which agents took the call. Average Call time Average Hold Time (How long was the call in the queue). In addition I'm looking at developing a couple of web pages; To show the agents that are logged into the system To show users that is registered. i.e. which interfaces are logged in and what is their status. I was wondering if anyone on this list was doing anything similar and would be able to share their ideas/code with me. I have written a number of large scale web based administration tools based on Perl and MySQL and would like to release this code to the Asterisk community once it is completed, to act as a call center tool. Warm Regards and Thanks --- Shad Mortazavi US Technical Manager Nexus Management
[Asterisk-Users] FW: Matrix Orbital (usbl LCD or VFD) (oops, wrong list I think)
Sorry bout that -Original Message- From: Kris Edwards [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 3:38 AM To: '[EMAIL PROTECTED]' Subject: Matrix Orbital (usbl LCD or VFD) This probably isnt practical for anyone other than home users, but I would like to use a USB LCD display in my case to display things such as: Answering Caller ID Info Current Context Etc I am very new to asterisk (in fact, I wont even be getting my digium hardware until the 15th), so Im sorry if this question isnt up to par with the other discussions going on. Does anyone know of any info on this? If not, is there a particluar file that I can grep out what I need and send to the display? Kris Edwards icq*5661686 smime.p7s Description: S/MIME cryptographic signature
Re: [Asterisk-Users] Asterisk not working with session border controller
And you say this is a commercial product that you purchased and they asked you to fix the problem? Where did you purchase it from or are you developing it and need help? More docs might help. - Original Message - From: Venkat Venkataraju [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 3:17 AM Subject: [Asterisk-Users] Asterisk not working with session border controller Hi All My company has bought a license for a session border controller called Middle. It look like this software acts like a proxy and registers devices to the asterisk. the problem (As per Middle guys) is that Middle sends a registration request to the Asterisk asking it to reply to the port 7062, but the Asterisk sends back the packet to the originating port 7060. This confuses the Middle and the registration process stalls @ Proxy Authentication Required. When i talked to person responsible, he told me that Asterisk is not according to the RFC and he asked me to find a solution to fix this problem. The only info i got about middle is here (http://sip-martini.homeip.net/venkat/middle.html) Please help me in fixing this problem. I really dont know what to do as i'm not fully aware of SIP protocol or Middle. Thanks Venkat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Trunking
On Mon, 05 Jan 2004 15:42:25 -0600 Steven Critchfield [EMAIL PROTECTED] wrote: On my lab tests, SIP with gsm uses 26kB/s, since the link is a frame-relay and cisco routers, I've used cisco rtp header compression, and got 16kB/s per channel. Something sounds fishy here. Asterisk sends out 50 packets a second of audio(20ms). If your numbers above are per channel, you achieved a 10k reduction in 50 packets, or 204.8 bytes average per packet. Since a GSM audio packet contains 33 bytes of audio, this large header compression sounds fishy. If you are talking bits, not bytes, then it isn't that impressive. You still will probably find more efficiency in IAX. Try it and tell us your results before shooting it down. Sorry, the results are in bits per second, not bytes. my mistake. I'm doing measure tests with SIP and IAX2 trunking. I'll finish today and post the results. Thanks for the tips -- Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems compiling cdr_pgsql
Hi, Having installed postgresql-devel-7.4-0.3 and postgresql-libs-7.4-0.3 I'm having probs. compiling cdr_pgsql, can anyone offer any pointers as to what I might be missing? I'm hoping I've just missed out something like postgresql-wibblewobble-7.4-0.3 or something ... Below is the result of a make in the cdr source dir which may help those of you in the know thanks... Andy [EMAIL PROTECTED] cdr]# make cc -o cdr_pgsql.so cdr_pgsql.o -lpq -lz -L/usr/lib /usr/lib/gcc-lib/i386-redhat-linux/3.2/../../../crt1.o: In function `_start': /usr/lib/gcc-lib/i386-redhat-linux/3.2/../../../crt1.o(.text+0x18): undefined reference to `main' cdr_pgsql.o: In function `pgsql_log': cdr_pgsql.o(.text+0x168): undefined reference to `ast_log' cdr_pgsql.o(.text+0x192): undefined reference to `ast_log' cdr_pgsql.o(.text+0x4c1): undefined reference to `ast_log' cdr_pgsql.o(.text+0x506): undefined reference to `ast_log' cdr_pgsql.o(.text+0x535): undefined reference to `ast_cdr_disp2str' cdr_pgsql.o(.text+0x5c4): undefined reference to `ast_log' cdr_pgsql.o(.text+0x60e): undefined reference to `ast_log' cdr_pgsql.o(.text+0x65c): undefined reference to `ast_log' cdr_pgsql.o(.text+0x6ae): undefined reference to `ast_log' cdr_pgsql.o(.text+0x6d8): undefined reference to `ast_log' cdr_pgsql.o(.text+0x771): more undefined references to `ast_log' follow cdr_pgsql.o: In function `my_unload_module': cdr_pgsql.o(.text+0x988): undefined reference to `ast_cdr_unregister' cdr_pgsql.o: In function `my_load_module': cdr_pgsql.o(.text+0x9b7): undefined reference to `ast_load' cdr_pgsql.o(.text+0x9ed): undefined reference to `ast_log' cdr_pgsql.o(.text+0xa0e): undefined reference to `ast_variable_browse' cdr_pgsql.o(.text+0xa3f): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xab6): undefined reference to `ast_log' cdr_pgsql.o(.text+0xae9): undefined reference to `ast_log' cdr_pgsql.o(.text+0xb11): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xb88): undefined reference to `ast_log' cdr_pgsql.o(.text+0xbbb): undefined reference to `ast_log' cdr_pgsql.o(.text+0xbe3): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xc5a): undefined reference to `ast_log' cdr_pgsql.o(.text+0xc8d): undefined reference to `ast_log' cdr_pgsql.o(.text+0xcb5): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xd2c): undefined reference to `ast_log' cdr_pgsql.o(.text+0xd5f): undefined reference to `ast_log' cdr_pgsql.o(.text+0xd87): undefined reference to `ast_variable_retrieve' cdr_pgsql.o(.text+0xdfe): undefined reference to `ast_log' cdr_pgsql.o(.text+0xe31): undefined reference to `ast_log' cdr_pgsql.o(.text+0xe4b): undefined reference to `ast_destroy' cdr_pgsql.o(.text+0xe78): undefined reference to `ast_log' cdr_pgsql.o(.text+0xea5): undefined reference to `ast_log' cdr_pgsql.o(.text+0xedb): undefined reference to `ast_log' cdr_pgsql.o(.text+0xf08): undefined reference to `ast_log' cdr_pgsql.o(.text+0xf35): undefined reference to `ast_log' cdr_pgsql.o(.text+0xf62): more undefined references to `ast_log' follow cdr_pgsql.o: In function `my_load_module': cdr_pgsql.o(.text+0x1079): undefined reference to `ast_cdr_register' cdr_pgsql.o(.text+0x10a9): undefined reference to `ast_log' collect2: ld returned 1 exit status make: *** [cdr_pgsql.so] Error 1 [EMAIL PROTECTED] cdr]# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI Scripting
Hi!. Is there any way to know which extension answered a call , when dialing from an AGI Script?? Thanks! Luciano
Re: [Asterisk-Users] I stumbled on this list...
1) Am I correct to assume that there is a way to dump Verizon and strictly go VOIP in a SOHO situation? Yes. 5) Am I being unrealistic in my savings by implementing an Asterisk * PBX in our SOHO situation. There is not doubt, that VoIP can save you money. Since others have already answered your questions, I'll simply add that regardless of which voip provider you pick, the service levels will be somewhat lower then your local provider. Part of that is due to Internet connectivity (from your location all the way to your voip service provider as an example) is not likely to be as consistent as local dial tone, and the quality of the connectivity can vary from day-to-day, etc. That is meant only to level-set expectations and not to stop you from considering voip. It certainly doesn't cost much to run both local and voip services in parallel for a month or two, and compare the two under your facilities and conditions. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Matrix Orbital (usbl LCD or VFD) (oops, wrong list I think)
Sorry 'bout that. -Original Message- From: Kris Edwards [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 3:38 AM To: '[EMAIL PROTECTED]' Subject: Matrix Orbital (usbl LCD or VFD) This probably isn't practical for anyone other than home users, but I would like to use a USB LCD display in my case to display things such as: Answering Caller ID Info Current Context Etc. I am very new to asterisk (in fact, I won't even be getting my digium hardware until the 15th), so I'm sorry if this question isn't up to par with the other discussions going on. Does anyone know of any info on this? If not, is there a particluar file that I can grep out what I need and send to the display? Kris Edwards icq*5661686 Kris, Thats an interesting thought... Since the source code is available you could always modify it to either send the data to the serial port or into a file that you could monitor and then extract what you are looking for. Also the Manager Interface (http://www.voip-info.org/wiki-Asterisk+GUI) might be another source of the data. When I start * I redirect the console log to a file.. That file could be displayed on the LCD as an indication of current activity. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Scripting
Luciano Ramos wrote: Hi!. Is there any way to know which extension answered a call , when dialing from an AGI Script?? Thanks! Luciano I don't think so, the AGI script will hand off the call at the point it dials it and will no longer participate in the session so from that point on it will not have any information about the call.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This is a test
- Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Monday, January 05, 2004 11:06 PM Subject: RE: [Asterisk-Users] This is a test From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland Sent: Tuesday, 6 January 2004 11:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] This is a test It appears that my replies aren't getting to the list. Just testing to see what is going on. Sean Yesterday at least, emails were delayed for some time... - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This newbie gives up for now - sadly
Robert Hajime Lanning: He is using SIP phones. Supervised Transfers do not really work with SIP. He wants, on a SIP phone (I think he had Grandstream phones), to: o hit transfer o dial new extension o talk to new extension * this part does not work * o hit transfer to complete the transfer or some cancel button to abort Yes that is exactly what I want - thanks for clarifying. -- Derek Irwin: I guess what I'm saying is from the start, * continues to surprise and impress. If you put in the time to learn it, you will be rewarded with a feature-rich system that can go head-to-head with the commercials system out there. Well perhaps Derek but my experience so far, and I'm not talking rocket-science requests, is that Asterisk just does not do the most basic of things out of the box and that the documentation is so dispersed and incomplete that it needs a massive effort to get even the most basic stuff running. And in some cases even the most basic stuff turns out not to work - yet. I will come back to asterisk when it is leading edge and not bleading edge. This is not a criticism of asterisk - just that its clearly not at the stage where an average linux sysadmin can use it for normal PBX applications with a reasonable time investment - if at all. I am sure it will get there and I am very keen to come back on board when it does. I hope I have not offended any developers by these comments - I know I am just sitting here while you guys do all the work. Please keep up the good work - and thanks for the comments. john quote who=Tilghman Lesher On Monday 05 January 2004 13:44, John Coll wrote: This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let the two speak or get back the original caller - an utterly fundamental office procedure on a PBX. I don't know why you'd need to implement that, as it's as simple as turning on two options in zapata.conf. Actually, I think both of those options are on by default in the sample configuration files. And I've spent the requisite few hours on Google and all the docs I have printed out. Eventually I found the thread transfer with three-way calling (circa Mon, 15 Dec 2003 20:45:08 -0600) and it seems that I can't do that basic operation in Asterisk. Why not? Are you not able to send a flash hook? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Weirdness with CALLERID / CALLERIDNAME from incoming PRI
Thanks for the reply. Yup, sure enough it appears the calling party name is in the facility message. I get the following, where the 'ATLANTA' and 'GA' sections are the calling party name. Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 2 (reference 527/0x20F) (Originator) Message type: FACILITY (98) Facility (len=25) [ Facility (len=25) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x13, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0b, 'ATLANTA', 0x2c, 0x20, 'GA' Facility (len=25) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x13, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0b, 'ATLANTA', 0x2c, 0x20, 'GA' ] -- Processing IE 28 (Facility) I don't have a good understanding for the Q.931 signaling process, but is it possible the call is being presented and handled by the extension logic prior the facility message coming through? Or is this just the way the pri debug span X output is given? I guess the next question is, are there any commands that could map the facility message to the calling party name before sending the call to the extension? Try doing a Wait(1) before you do the dial command. The facility is sent *after* the initial setup in a totally separate message, so unless you wait first, we won't get the name. We *do* have some code to try to extract it from FACILITY, but since we don't know the spec, it's just a hack based on our observations on one particular switch. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Policies - deny some nubers
Hi, I had asterisk installed, ISDN-adapter, some x-lite software-phones and I can call betweens the softphone- and 'normal' phones during the ISDN-card. 2 questions now 1) Is it posible to create policies, so that some SIP-users can dial ALL numbers, and some SIP-users not are allowed to dial eg. 900xxx-numbers, 30xx (mobilphones), 40(long distance) and if possible on time basic- so that from 18.00-0800 it is possible to dial all numbers ? 2) when dialing in to asterisk via ISDN get a new dialtone so it's posibble to dial all sipphones, or get voiceresponce saying press 1 to dial manager, press 2 to dial dad, and 3 to leave a message ? /Hans-Henrik _ Tired of slow downloads? Compare online deals from your local high-speed providers now. https://broadband.msn.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Nat Issue
Here's the problem my sipura 2000 is setup on Nat Network in my office and my Asterisk Server is setup also on Nat Network at home the sipura can register and get calls but no audio comes in and out of the sipura and when i dial local extensions on the sipura i get this error message. any suggestions on what i can try as work around. *CLI NOTICE[1158921008]: File chan_sip.c, Line 5394 (handle_request): Unknown SIP command 'NOTIFY' from '205.158.181.200'WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)NOTICE[1158921008]: File chan_sip.c, Line 5394 (handle_request): Unknown SIP command 'NOTIFY' from '205.158.181.200'WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)
Re: [Asterisk-Users] Policies - deny some nubers
Hans-Henrik Andresen wrote: Hi, I had asterisk installed, ISDN-adapter, some x-lite software-phones and I can call betweens the softphone- and 'normal' phones during the ISDN-card. 2 questions now 1) Is it posible to create policies, so that some SIP-users can dial ALL numbers, and some SIP-users not are allowed to dial eg. 900xxx-numbers, 30xx (mobilphones), 40(long distance) and if possible on time basic- so that from 18.00-0800 it is possible to dial all numbers ? Yes this is controlled by your dial plan.. If you haven't already I suggest taking a read through the hand book which will help a lot to get you started.. 2) when dialing in to asterisk via ISDN get a new dialtone so it's posibble to dial all sipphones, or get voiceresponce saying press 1 to dial manager, press 2 to dial dad, and 3 to leave a message ? Yes but you won't do it with a dial tone, you will use voice prompts to tell the person calling in what options are availible to them.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cant load drivers for TE410P cards
hello, I have been using the T1 card with my asterisk for a while now, but an attemp to upgrade the system to use a TE410P card ( using the T1 option) i have a 3.3V motherboard. but when i try to load the module it gives the following errors: #modprobe zaptelO.k # modprobe wct1xxp gives /lib/modules/2.4.18-4/misc/wct1xxp.0:init module:no such device Hint: insmod errors cab caused by incorrect module parameters, including invalid IO and IRQ parameters. /lib/modules/2.4.18-4/misc/wct1xxp:: insmod wct1xxp failed. is there a way of installing the TE410P module? regarsd -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Scripting
Use the callmanager their u can use the link event Michael Van: Luciano Ramos [mailto:[EMAIL PROTECTED] Verzonden: di 6/01/2004 13:56 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] AGI Scripting Hi!. Is there any way to know which extension answered a call , when dialing from an AGI Script?? Thanks! Luciano winmail.dat
[Asterisk-Users] Asterisk feature list: spreadsheet
http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106.xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the common features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I figured I'd take as many of the unique features as others had offered, and put them together in a big list, and then also include some of the features that are unique to Asterisk. I think this list will be of some use to persons evaluating Asterisk against their existing PBX platform, or other vendors of new VoIP systems. Note that I took some liberties with the standard feature syntax: often, things that seemed _very_ easy for me to do with Asterisk's scripting features I listed as Standard, even though it would require a few minutes of work. Other features which seemed to be a little more complex, or which would require some web programming, I listed with estimates of how much time it would take to build them. I assumed use of Cisco 7960 phones, so some of the features which are really deskset options I listed as Standard if I were to use 7960 devices. Caveat emptor for other desksets. This is not truly a comparison, since there are no other columns in the spreadsheet. However, it's good fodder for you to whack your VoIP or PBX vendor on the nose with, since they typically will not be able to match the feature list. Olle: feel free to add to the Wiki, since you asked for this type of list. Anyone else: feel free to send me updates in cut/pasteable Excel form if you have things you'd like to add to the list. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to flash hook when there is no hook ?
Hi all, I'm using regular analog phones with * and an ATA186, and I plan to move to cordless phones. But cordless phones usually have no way to flash hook cause there is no hook :-) I seem to recall there is a way to simulate a flash hook using dtmf tones in *, but I can't find which combination to use. More generally, where can I find all those magic cominations to put people on hold, use pickup grouping, redirect calls,... ? Thanks, -- Rémi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cant load drivers for TE410P cards
The TE410P does not use the wct1xxp driver -- it uses wct4xxp... Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:16 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cant load drivers for TE410P cards hello, I have been using the T1 card with my asterisk for a while now, but an attemp to upgrade the system to use a TE410P card ( using the T1 option) i have a 3.3V motherboard. but when i try to load the module it gives the following errors: #modprobe zaptelO.k # modprobe wct1xxp gives /lib/modules/2.4.18-4/misc/wct1xxp.0:init module:no such device Hint: insmod errors cab caused by incorrect module parameters, including invalid IO and IRQ parameters. /lib/modules/2.4.18-4/misc/wct1xxp:: insmod wct1xxp failed. is there a way of installing the TE410P module? regarsd -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cant load drivers for TE410P cards
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 06 January 2004 15:15, [EMAIL PROTECTED] wrote: I have been using the T1 card with my asterisk for a while now, but an attemp to upgrade the system to use a TE410P card ( using the T1 option) i have a 3.3V motherboard. but when i try to load the module it gives the following errors: #modprobe zaptelO.k # modprobe wct1xxp gives You need to use wct4xxp for the TE410. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/+spC2TEAILET3McRAlAUAJ9tkYKn2zOo0ttgb0k5amKVTJiWBwCcCy17 ExUKWL1WtXX/RuXTvZd9qws= =TK2K -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message waiting indicator
What is required to get the mwi to work? Is it more of a phone subject or *? I have the mailbox= line in sip.conf, but only one extension is named, and in some of the examples, I have seen that there are two... What is that all about and how does it affect the extensions.conf and voicemail.conf? I think the examples that you might have looked are suggesting that when a voicemail is left for a single extension, you can place definitions in your sip.conf file that turn on the MWI (message waiting indicator) LED on more then one phone. (I'll leave that up to you to figure out whether that is a feature of use to you.) Asterisk will occasionally look in the /var/spool/asterisk/voicemail/default/3008/INBOX directory (where 3008 represents the extension number), and if a certain file exists, send a sip message to the extn(s) that you defined in sip.conf as mailbox=3008. The sip message sent to the phone (in hex using a packet sniffer) looks like: 0020: c1 5b 13 c4 13 c4 01 e2 58 97 4e 4f 54 49 46 59 | Á[.Ä.Ä.âXNOTIFY 0030: 20 73 69 70 3a 33 30 30 38 40 32 30 35 2e 32 31 | sip:[EMAIL PROTECTED] 0040: 32 2e 31 39 33 2e 37 31 20 53 49 50 2f 32 2e 30 | 2.173.91 SIP/2.0 0050: 0d 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 | ..Via: SIP/2.0/U 0060: 44 50 20 32 30 35 2e 32 31 32 2e 31 39 33 2e 31 | DP 205.212.193.1 0070: 30 31 3a 35 30 36 30 3b 62 72 61 6e 63 68 3d 7a | 01:5060;branch=z 0080: 39 68 47 34 62 4b 33 63 31 63 61 35 65 31 0d 0a | 9hG4bK3c1ca5e1.. 0090: 46 72 6f 6d 3a 20 22 61 73 74 65 72 69 73 6b 22 | From: asterisk 00a0: 20 3c 73 69 70 3a 61 73 74 65 72 69 73 6b 40 32 | sip:[EMAIL PROTECTED] 00b0: 30 35 2e 32 31 32 2e 31 39 33 2e 31 30 31 3e 3b | 05.212.193.101; 00c0: 74 61 67 3d 61 73 35 37 63 63 64 33 32 65 0d 0a | tag=as57ccd32e.. 00d0: 54 6f 3a 20 3c 73 69 70 3a 33 30 30 38 40 32 30 | To: sip:[EMAIL PROTECTED] 00e0: 35 2e 32 31 32 2e 31 39 33 2e 39 31 3e 0d 0a 43 | 5.212.193.91..C 00f0: 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 61 73 74 | ontact: sip:ast 0100: 65 72 69 73 6b 40 32 30 35 2e 32 31 32 2e 31 39 | [EMAIL PROTECTED] 0110: 33 2e 31 30 31 3e 0d 0a 43 61 6c 6c 2d 49 44 3a | 3.101..Call-ID: 0120: 20 34 34 66 39 31 38 36 64 34 30 62 30 31 35 33 | 44f9186d40b0153 0130: 30 36 37 39 32 30 31 39 64 33 36 39 35 66 36 31 | 06792019d3695f61 0140: 36 40 32 30 35 2e 32 31 32 2e 31 39 33 2e 31 30 | [EMAIL PROTECTED] 0150: 31 0d 0a 43 53 65 71 3a 20 31 30 32 20 4e 4f 54 | 1..CSeq: 102 NOT 0160: 49 46 59 0d 0a 55 73 65 72 2d 41 67 65 6e 74 3a | IFY..User-Agent: 0170: 20 41 73 74 65 72 69 73 6b 20 50 42 58 0d 0a 45 | Asterisk PBX..E 0180: 76 65 6e 74 3a 20 6d 65 73 73 61 67 65 2d 73 75 | vent: message-su 0190: 6d 6d 61 72 79 0d 0a 43 6f 6e 74 65 6e 74 2d 54 | mmary..Content-T 01a0: 79 70 65 3a 20 61 70 70 6c 69 63 61 74 69 6f 6e | ype: application 01b0: 2f 73 69 6d 70 6c 65 2d 6d 65 73 73 61 67 65 2d | /simple-message- 01c0: 73 75 6d 6d 61 72 79 0d 0a 43 6f 6e 74 65 6e 74 | summary..Content 01d0: 2d 4c 65 6e 67 74 68 3a 20 33 37 0d 0a 0d 0a 4d | -Length: 37M 01e0: 65 73 73 61 67 65 73 2d 57 61 69 74 69 6e 67 3a | essages-Waiting: 01f0: 20 79 65 73 0a 56 6f 69 63 65 6d 61 69 6c 3a 20 | yes.Voicemail: 0200: 31 2f 30 0a | 1/0. where if you look closely at the text on the right side, you can see data like Missages-Waiting: yes in the packet. (You might want to read the RFC that defines what the sip protocol is all about, it will help you understand.) The hardware/software sip phone is supposed to translate that sip msg and turn on the LED on the phone's panel. Exactly how each phone implements that function is up to the phone manufacturer, and like many things in the voip space, some get right while others seem to struggle with the simple things in life. If you don't have a packet sniffer, then from the asterisk command line, enter sip debug and dig through the display for the equivalent message. The interesting piece will look something like: To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 37 Messages-Waiting: yes Voicemail: 1/0 (no NAT) to 205.212.193.91:5060 Sip read: I SIP/2.0 200 Ok = If your phone's display or MWI LED doesn't function, then use the above to diagnose which component is not working right. The SIP RFC and sip debug command are your friends; get to know them. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to flash hook when there is no hook ?
I just looked at my cordless phone and it has a flash button. It's a Panasonic... Remi Letot wrote: Hi all, I'm using regular analog phones with * and an ATA186, and I plan to move to cordless phones. But cordless phones usually have no way to flash hook cause there is no hook :-) I seem to recall there is a way to simulate a flash hook using dtmf tones in *, but I can't find which combination to use. More generally, where can I find all those magic cominations to put people on hold, use pickup grouping, redirect calls,... ? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Policies - deny some nubers
Hi!
1) Is it posible to create policies, so that some SIP-users can dial
ALL numbers, and some SIP-users not are allowed to dial eg.
900xxx-numbers, 30xx (mobilphones), 40(long distance)
and if possible on time basic- so that from 18.00-0800 it is possible
to dial all numbers ?
Yes this is controlled by your dial plan.. If you haven't already I
suggest taking a read through the hand book which will help a lot to get
you started..
Look at contexts and the include statement. Read the draft handbook
linked from www.asterisk.org, support section. Or look here:
http://www.voip-info.org/wiki-Asterisk+howto+dial+plan
2) when dialing in to asterisk via ISDN get a new dialtone so it's
posibble to dial all sipphones, or get voiceresponce saying press 1 to
dial manager, press 2 to dial dad, and 3 to leave a message ?
Yes but you won't do it with a dial tone, you will use voice prompts to
tell the person calling in what options are availible to them..
For the dialtone: You can very well use separate context for that with a
Background() announcement and something like
[dial_what_you_want_context]
exten = s,1,Background(enter-an-extension)
exten = _.,1,Dial(Local/${EXTEN})
Apart from that if you wish to dial out again look at/ search for DISA.
http://www.voip-info.org/wiki-Asterisk+cmd+DISA
Greetings, Philipp
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[Asterisk-Users] URGENT - micronet asterisk on h323
hello, my situation is h323gw - gatekeeper - asterisk - SIP client my problem is, that I can't make call from h323gw, when this GW is Micronet (sp5004). A --- CUT --- -- Executing Wait(H323/ip$62.152.225.18:52434/20702, 1) in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' --- CUT --- On the other side, when the h232gw is Cisco ATA186, Cisco 7905 or Cisco AS5300 all is working good. I'm using standart h323 modul, which is included in the *. OH323 modul allways crashes. I can make call from SIP client to H323 network. please help me, this is urgent best regards Peter Hudec -- Company: PosTel a.s.Position: IP network manager Borska 6 Bratislava 841 04 mail: [EMAIL PROTECTED] www: [http://www.postel.sk] phone: [+421 02 50203169]mobil: [+421 905 997203] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to flash hook when there is no hook ?
Switch Hook is also known as FLASH. Pretty much all phones support this. Most cordless phones support this as well. On Tue, 2004-01-06 at 08:35, Remi Letot wrote: Hi all, I'm using regular analog phones with * and an ATA186, and I plan to move to cordless phones. But cordless phones usually have no way to flash hook cause there is no hook :-) I seem to recall there is a way to simulate a flash hook using dtmf tones in *, but I can't find which combination to use. More generally, where can I find all those magic cominations to put people on hold, use pickup grouping, redirect calls,... ? Thanks, -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] small question from a new user
Dear all, Is Asterisk able to function straightly as a Voip transit softswitch (class 4) ? Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi-line help
Thank you all for your responses. Since I was a phone installer (previous life) and installed Lucent Partner and Merlin systems, I was on the key system mode of thinking. On the Polycom phones each line button is a registration, so I wonder how I could program a SIP registration to speed dial a number? Would that be done through exten.conf like: [button2] Exten = 1,dial(zap/g1/5551212) ??? So then, carrying over to key system terms, I would basically be setting up line pool buttons... Basically with my small office (2 phones, and one * box with 2 x100p cards) I would just use the first button (or whatever) for my registration with my * and call it good... I am thinking of proposing this system to my partner corp which would entail around 13 extensions and 6 lines... How would I give someone upstairs the ability to view if each user was on the phone or not? -- should probably be a new thread Currently they have 18 button phones that are programmed with the incomming lines, then the users (LED's glow when user is on). This is so much fun! (no really!) Sean -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Monday, January 05, 2004 6:55 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Multi-line help Sean, Basically I guess I am thinking of the traditional key systems approach which is to have the CO lines appear on the phone. The problem it appears with SIP (not really *) and the particular phones, is to have the reporting. I guess what I was looking for was to have the buttons not only represent the incoming lines, but to also show their status (busy, hold, etc...). As you've already mentioned, what you've described is a key system, and not a pbx. (There might be an open source key system out there somewhere.) On that note, what (typically with SIP/*) are the multiple line phone buttons used for? I know you have to have at least one for access to the asterisk system, but what is the point of the multiple registrations? Several reasons depending upon the actual requirements... 1. Small office, Customer Service appears on line 1, President on line 2. Answering line 2 with an appropriate messages (when he's not around) is different then answering line 1 as a Customer Service person. 2. Shared tenant service: five different businesses in the same small complex. The receptionist has all five lines on her phone, and answers with an appropriate message for each business when their lines are unanswered. 3. Home-boy (no asterisk) subscribes to two different VoIP providers with two different rate plans. Line 1 registers with provider 1, and line 2 with provider 2. You choose which service you want based on your knowledge of what your trying to accomplish (not necessarily programmable if an * system was included). 4. On the Cisco 7960, I have one of the line buttons programmed as a speed dial to a certain extn as I'm calling it often. 5. Remote intercom: place a speaker phone by the front door and configure it for auto answer. When the doorbell rings, push one of your preprogrammed buttons to speak to who's at the door. 6. You could probably program * to open the garage door with one of the buttons. ;) As Steve pointed out earlier today, there are many ways to accompish the same function within asterisk, therefore some of the items listed above might be done a different way. That's fine. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Matrix Orbital (usbl LCD or VFD) (oops, wrong list I think)
From: Kris Edwards [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 06, 2004 3:38 AM
To: '[EMAIL PROTECTED]'
Subject: Matrix Orbital (usbl LCD or VFD)
This probably isn't practical for anyone other than home users, but I
would like to use a USB LCD display in my case to display things such
as:
if you grab the perl modules from citats site http://asterisk.gnuinter.net/
These scripts are used with MO VFD, to display call counts in the q usng the
manager interface
you can set another callback events grab the caller id from a stateup
event on the device of interest
then write that to the LCD ...
then save this include
---cut along dotted line
astq.pl--
#!/usr/bin/perl
#
# Example script to show how to use Asterisk::Manager
#
# Written by: James Golovich [EMAIL PROTECTED]
#
#
use lib './lib', '../lib';
use Asterisk::Manager;
use mo;
$|++;
my $astman = new Asterisk::Manager;
$astman-user('lcd');
$astman-secret('test');
$astman-host('localhost');
$astman-connect || die $astman-error . \n;
$astman-setcallback('Leave', \queue_callback);
$astman-setcallback('Join', \queue_callback);
display_init ( );
display_clear ( );
display_write ( User Count: );
$astman-eventloop;
$astman-disconnect;
sub hangup_callback {
print STDERR hangup callback\n;
}
sub default_callback {
my (%stuff) = @_;
foreach (keys %stuff) {
print STDERR $_: . $stuff{$_} . \n;
}
print STDERR \n;
}
sub queue_callback {
my (%stuff) = @_;
$cnt = 0;
foreach (keys %stuff) {
print STDERR $_: . $stuff{$_} . \n;
if ($_ eq Count) {
$cnt = $stuff{$_};
}
elsif ($_ eq xCount) { #
$i=1;
}
}
print STDERR I got $cnt \n;
display_clear ( );
display_write ( User Count: $cnt );
display_init_large_digit ( );
display_large_digit ( 16,3 );
}
---cut along dotted line
mo.pm--
#
# mo.pm
# you should set the $serial_port to the correct tty below.
# all cmds start with 254 decimal 0xFE
package mo;
use Exporter;
use POSIX;
use IO;
# editable section
#
$serial_port = /dev/ttyS0; # usually /dev/ttyS0 or /dev/ttyS1
#
# end editable section
@ISA = qw ( Exporter );
@EXPORT = qw ( moInit moWrite moClear moClose
moPosition moBlinkOn moBlinkOff
moInitLargeDigit moLargeDigit
);
#
# moInit ( );
# opens the serial port in bi-directional mode and sends the commands to
# turn the ( display on for a vfd | backlight on for an lcd ) and clears
# the screen.
#
sub moInit
{
open ( DISPLAY, + $serial_port ) || die Couldn't open $_: $!;
$fd_disp = fileno ( DISPLAY );
$termios = POSIX::Termios-new ( );
$termios-getattr ( $fd_disp );
$termios-setispeed ( B19200 );
$termios-setospeed ( B19200 );
$cflag = $termios-getcflag ( );
$lflag = $termios-getlflag ( );
$oflag = $termios-getoflag ( );
$iflag = $termios-getiflag ( );
$iflag = ~(IGNBRK|BRKINT|PARMRK|ISTRIP|INLCR|IGNCR|ICRNL|IXON);
$oflag = ~OPOST;
$lflag = ~(ECHO|ECHONL|ICANON|ISIG);
$cflag = ~(CSIZE|PARENB|HUPCL);
$cflag |= CS8|CLOCAL;
$termios-setcflag ( $cflag );
$termios-setlflag ( $lflag );
$termios-setoflag ( $oflag );
$termios-setiflag ( $iflag );
$termios-setattr ( $fd_disp, TCSANOW );
$clear = sprintf ( %cX, 0xfe );
$noblink = sprintf ( %cT, 0xfe );
$dim = sprintf ( %cY%c, 0xfe, 0x03 );
$on = sprintf ( %cB%c, 0xfe, 0x00 );
$repeat = sprintf ( %c~%c, 0xfe, 0x00 );
print DISPLAY $on$repeat$noblink$dim$clear;
}
#
# moWrite ( hi there! );
# prints the selected text to the display at the current cursor position.
#
sub moWrite
{
my ( $text ) = @_;
print DISPLAY $text\n;
}
#
# moClear ( );
# clears the display
#
sub moClear
{
$clear = sprintf ( %cX, 0xfe );
print DISPLAY $clear\n;
}
#
# moClose ( );
# clears the display and turns ( it off for a vfd | the backlight off for
# an lcd )
#
sub moClose
{
$clear = sprintf ( %cX, 0xfe );
$off = sprintf ( %cF, 0xfe );
print DISPLAY $clear\n$off\n;
close ( DISPLAY );
}
#
# moInitLargeDigit ( );
# init the display for large digits
#
sub moInitLargeDigit
{
$largedigits = sprintf ( %cn, 0xfe );
print DISPLAY $largedigits\n;
}
#
# moLargeDigit ( $x, $digit );
# writes large digit at column $x given.
#
sub moLargeDigit
{
my ( $x, $digit ) = @_;
$digitpos = sprintf %c#%c%c, 0xfe, $x, $digit;
print DISPLAY $digitpos;
}
#
# moPosition ( $x, $y );
# homes the cursor to the x and y coordinates given.
#
sub moPosition
{
my ( $x, $y ) = @_;
$pos = sprintf %cG%c%c, 0xfe, $y, $x;
print DISPLAY $pos;
}
# moBlinkOn ( );
# turns the cursor blinking on.
#
sub
[Asterisk-Users] ring tone
Hi ! I have a small problem. When switching a call (pstn - sip user), I get the sip phone ringing - ie. everything is OK, but I do not get a ringtone in the handset on the pstn side. Can anyone help me out in how to make * play tones ? My setup: E1 IP pstn -- Asterisk -- sip phone Regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This newbie gives up for now - sadly
On Tue, 2004-01-06 at 06:20, John Coll wrote: Robert Hajime Lanning: He is using SIP phones. Supervised Transfers do not really work with SIP. He wants, on a SIP phone (I think he had Grandstream phones), to: o hit transfer o dial new extension o talk to new extension * this part does not work * o hit transfer to complete the transfer or some cancel button to abort Yes that is exactly what I want - thanks for clarifying. It sounds to me like this is a problem with the Grandstream phones in particular, and not Asterisk. Supervised transfers work *GREAT* with the Cisco 7960 phones... I use them almost every day. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Weirdness with CALLERID / CALLERIDNAME from incoming PRI
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Spencer Try doing a Wait(1) before you do the dial command. The facility is sent *after* the initial setup in a totally separate message, so unless you wait first, we won't get the name. We *do* have some code to try to extract it from FACILITY, but since we don't know the spec, it's just a hack based on our observations on one particular switch. Works great here too Mark, thanks. FYI, the switch we speak with is a 5ESS (BellSouth). If you need any pri debugs, feel free to ask. Regards, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Multi-line help AOL Messenger Style PBX Navigation
Sean, I am thinking of proposing this system to my partner corp which would entail around 13 extensions and 6 lines... How would I give someone upstairs the ability to view if each user was on the phone or not? -- should probably be a new thread Currently they have 18 button phones that are programmed with the incomming lines, then the users (LED's glow when user is on). =20 This is something I've been trying to work out. I bought a Cisco 7914 for the Cisco 7960's we have, which according to the box, allows users to see the status of other lines. Hmmm... In closer view, it seems the Cisco Call Manager itself pushes updates to the phone... Something that a standlone application polling Asterisk could do... Better yet, chan_sccp is beginning to support more and more features of the Cisco Call Manager's SCCP implementation - so it's possible that some kind of line status could be pushed out via SCCP to the phones with right kit (namely the 7914). SIP, as far as I can tell, is really an HTTP style standard that is better suited to carrying media over IP, rather than just specifically voice. Of course, it's been developed over the years, and is now getting more and more advanced, but it will always be designed from a different point of view, using technology that doesn't have the same limitations! My solution, is to use something like Gastman as an advanced console for the Operator and power users. It may even be conceivable to have a small version of Gastman that sits in the bottom of your screen, and keeps you informed on the status of people and their lines. Could actually be a handy piece of software - not just for Voice. I'd love to know where the hell people are, without having to phone them! Almost AOL style! Imagine that! An AOL Messenger style application. You can chat with people via conventional text chat, you can set your status - like Gone to lunch, or At My Desk, even Busy, but accepting Priority calls. Then, double click on their name entry, and ta-da, your phone extension starts ringing. Pick it up, and it connects you though to them. Or for the softphone users; it has an integrated soft phone. Cisco technicall refer's to the 7914 as being a speed-dial and line extension unit, which just happens to report back on lines status. It does state, that it is by no means, an Operator/Attendant's console. Instead, they have a dedicated piece of software for that. Web Attendant is an ugly Java-made browser-based application (AstGuiClient is Kate Moss compared to the Cisco application) which provides live status on all lines, both incoming/outgoing and internal, with an XML micro-directory. The directory is basic in comparison to the XML based directories that 7960's can use, and seems like an after-thought. Because it's browser based, the user still has to have a softphone or hardphone (with a headset of course) to actually act as an attendant. Think of it as a visual display/console - not as an attendant station. It's more that feasible to take an IP Phone with a large display, that's capable of running Java, and to write an attendant solution to sit on it. Better yet, take this console maxim further and use AstGuiClient or Gastman as console's and transfer managers, and just use hardware for the speaking-bit. This is so much fun! (no really!) W...ww...Work, what is this thing? :-) Best, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi-line help
On Tue, 2004-01-06 at 09:42, Sean Garland wrote: Thank you all for your responses. Since I was a phone installer (previous life) and installed Lucent Partner and Merlin systems, I was on the key system mode of thinking. On the Polycom phones each line button is a registration, so I wonder how I could program a SIP registration to speed dial a number? Would that be done through exten.conf like: [button2] Exten = 1,dial(zap/g1/5551212) Button 2 would have some form of instant connect to asterisk, then in the context it comes in you would have. [button2] exten s,1,dial(blah/blah) Conversely, your phone may support the extra information such as the extension to dial in it's config. So then, carrying over to key system terms, I would basically be setting up line pool buttons... Basically with my small office (2 phones, and one * box with 2 x100p cards) I would just use the first button (or whatever) for my registration with my * and call it good... I am thinking of proposing this system to my partner corp which would entail around 13 extensions and 6 lines... How would I give someone upstairs the ability to view if each user was on the phone or not? -- should probably be a new thread Currently they have 18 button phones that are programmed with the incomming lines, then the users (LED's glow when user is on). If you use gastman, you not only know what lines are in use, but who is on each one, and whether the call was incoming or outgoing along with CallerID or called number for the calls. You also will have the option of grabbing a call and dumping it on any other user you choose even if it is currently in progress. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino
I'm running a few machines on RH9. Only one has exhibited crashing during normal operation. It is the only machine that uses voicemail - and music on hold. I'm suspecting mpg123... Of course I just stuck in the export. If it is still up in a couple of weeks, I'll be pretty sure that is what it was. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James H. Cloos Jr. Sent: Friday, January 02, 2004 1:03 AM To: JR Richardson Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino JR == JR Richardson [EMAIL PROTECTED] writes: JR I ran that export command you JR suggested, then launched *, everything worked fine. I'm still JR looking for info on what that command actually does. Can you shed JR some light please? Exporting LD_ASSUME_KERNEL=2.4.1 tells libc to use the old-style 'int 80' method of doing syscalls to the kernel, as well as the old-style type of thread support. RH9 and 2.6 kernels support newer, faster methods of syscalls and threads on amd64 and recent ia32 cpus. The need to assume an earlier kernel version indicates that some part of * or a lib it (or one of its modules) is linked to breaks when using the newer routines. Eventually such bugs should be eradicated and LD_ASSUME_KERNEL will not be required. (Eg, my gentoo laptop only supports nptl threads and I have no problems running * there.) -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk feature list: spreadsheet
Great list, Thank you, John! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA call
Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz 8 headers, 0 lines Retransmitting #1 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=4bd7a841 Content-Length: 0 290Ñ to 200.167.103.219:1025 Sip read: LI INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Proxy-Authorization: Digest username=porto,realm=asterisk,nonce=4bd7a841,uri=sip:[EMAIL PROTECTED] .77,response=1ecb99d4d5e23be179a9eb55eb33c62a Expires: 300 Content-Length: 250 Content-Type: application/sdp v=0 o=porto 3642 3642 IN IP4 192.168.0.150 s=ATA186 Call c=IN IP4 192.168.0.150 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.150 : 5060 (NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 256, them - 268/0, combined - 256 Non-codec capabilities: us - 1, them - 1, combined - 1 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:200.167.103.219:1025 SIP/2.0 Via: SIP/2.0/UDP 200.170.156.77:5060;branch=z9hG4bK1937468f From: asterisk sip:[EMAIL PROTECTED];tag=as5566fcc8 To: sip:200.167.103.219:1025 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 200.167.103.219:1025 Sip read: LI ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Content-Length: 0 8 headers, 0 lines Sip read: LI INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Contact: sip:[EMAIL PROTECTED]:5060;transport=udp User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Proxy-Authorization: Digest username=porto,realm=asterisk,nonce=514a024a,uri=sip:[EMAIL PROTECTED] .77,response=adb7da64c3f557d1db20b699c04f6d84 Expires: 300 Content-Length: 250 Content-Type: application/sdp v=0 o=porto 3692 3692 IN IP4 192.168.0.150 s=ATA186 Call c=IN IP4 192.168.0.150 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 192.168.0.150 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 256, them - 268/0, combined - 256 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as046b1041 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=6512ffab Content-Length: 0 to 200.167.103.219:1025 Sip read: LI ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.150:5060;received=200.167.103.219 From: sip:[EMAIL PROTECTED];tag=3346186142 To: sip:[EMAIL PROTECTED];user=phone;tag=as36ac1b92 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA 186 v2.16.1 ata18x (030709a) Content-Length: 0 8 headers, 0 lines Retransmitting #1 (no NAT): OPTIONS
Re: [Asterisk-Users] URGENT - micronet asterisk on h323
Peter Hudec wrote: --- CUT --- -- Executing Wait(H323/ip$62.152.225.18:52434/20702, 1) in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' --- CUT --- First off you are going to have to provide more debug than just that and secondly a Wait,1 doesn't do anything but wait one second. I suggest checking your extensions.conf file, you have a missing priority number or whole exten line. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 911 and lawsuits
Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling cdr_pgsql
On Tuesday 06 January 2004 06:16, Andy Powell wrote: Having installed postgresql-devel-7.4-0.3 and postgresql-libs-7.4-0.3 I'm having probs. compiling cdr_pgsql, can anyone offer any pointers as to what I might be missing? I'm hoping I've just missed out something like postgresql-wibblewobble-7.4-0.3 or something ... Below is the result of a make in the cdr source dir which may help those of you in the know [EMAIL PROTECTED] cdr]# make snip Don't run make directly from the cdr subdirectory. Run make from the directory one higher and it will build just fine. The main Makefile defines several flags which are passed to the cdr subdirectory. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
On Tue, 2004-01-06 at 10:56, Jim Flagg wrote: Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. Good question otherwise. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Got SIP response 482 Loop Detected
Hello Today I observed this strange problem, as soon as I called from my SNOM IP phone (910) to CISCO IP Phone 7905G (810), I got following warning messages and call didn't connect. But after couple of minutes this problem was gone, without me doing anything..Has anyone observed this thing before... Called 810 -- SIP/810-b6dc is ringing -- SIP/810-b6dc answered SIP/910-6c4e -- Attempting native bridge of SIP/910-6c4e and SIP/810-b6dc WARNING[1227879616]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 = 524302 is not codec1 = 524302, can't do reinvite -- Got SIP response 482 Loop Detected back from 129.82.44.226 WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Response) WARNING[1142106560]: File chan_sip.c, Line 2329 (__transmit_response): Unable to determine sequence number from '' Thanks Tony
RE: [Asterisk-Users] 911 and lawsuits
I've always been advised that personal injury liability waivers are of limited value in either avoiding a lawsuit or limiting damages, in the US. Can't hurt to have such an agreement, but probably would not help under our tort system. Outside the US, might be a different story! Regards Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, January 06, 2004 5:13 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 911 and lawsuits On Tue, 2004-01-06 at 10:56, Jim Flagg wrote: Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. Good question otherwise. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Hi Doug, I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my ATA on show sip peers. What I can see is: - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits and redundancy
This is esp true of any VoIP PBX system. In fact I think many of them run Windows. I do have a related question about how * users are creating redundancy in thier setups? I am going live in a few days with a single office setup where I have patched the * PBX in front of our existing legacy phone system, giving us auto attendent and voice mail, plus the potential to do a large scale test of IP phones. If successful the next step is a 150-400 station multi-office setup. Most calls are inter-building such that we currently only need 6 outbound lines to the PSTN. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Steven Critchfield [EMAIL PROTECTED]: On Tue, 2004-01-06 at 10:56, Jim Flagg wrote: Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. Good question otherwise. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Hi Doug, I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my ATA on show sip peers. What I can see is: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN Just one thing which I did not mention on the last email is that my ATA is behing NAT. Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Got SIP response 482 Loop Detected
Code you gave the audio settings of your 7905 and which type and firmware for the snom Van: tony banks [mailto:[EMAIL PROTECTED] Verzonden: di 6/01/2004 18:23 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] Got SIP response 482 Loop Detected Hello Today I observed this strange problem, as soon as I called from my SNOM IP phone (910) to CISCO IP Phone 7905G (810), I got following warning messages and call didn't connect. But after couple of minutes this problem was gone, without me doing anything..Has anyone observed this thing before... Called 810 -- SIP/810-b6dc is ringing -- SIP/810-b6dc answered SIP/910-6c4e -- Attempting native bridge of SIP/910-6c4e and SIP/810-b6dc WARNING[1227879616]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 = 524302 is not codec1 = 524302, can't do reinvite -- Got SIP response 482 Loop Detected back from 129.82.44.226 WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Response) WARNING[1142106560]: File chan_sip.c, Line 2329 (__transmit_response): Unable to determine sequence number from '' Thanks Tony http://clients.rediff.com/signature/track_sig.asp winmail.dat
Re: [Asterisk-Users] Asterisk not working with session border controller
Steve Totaro wrote: And you say this is a commercial product that you purchased and they asked you to fix the problem? guys @ middle claim that the Asterisk is breaking the RFC and they are asking me to fix it on Asterisk as our company wants to use Asterisk. The middle does seems to work with many other SIP proxies. Where did you purchase it from or are you developing it and need help? More docs might help. I'm not developing it, but i'm working with a company thats trying to bring consumer VoIP by using Asteisk. I'm working on call processing system that uses astman. But i was pulled into this to get asterisk work with Middle. I dont have any other docs. i've asked for it. i'll post it if i get one. I've got logs from Middle, but i'm not sure how it may help. for the request REGISTER sip:sip-x.homeip.net SIP/2.0 Via: SIP/2.0/UDP 68.#.#.84:7062;branch=z9hG4bK-middle-4178 Via: SIP/2.0/UDP 68.#.#.125:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=1722079273 To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED]:7062 Call-ID: [EMAIL PROTECTED] CSeq: 6 REGISTER Content-Length: 0 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Asteisk is supposed to respond back to the IP on port 7062 (i was told. i'vnt read the RFC), but it send the request back on the port 7060, the originating port of the request. Thanks Venkat - Original Message - From: Venkat Venkataraju [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 3:17 AM Subject: [Asterisk-Users] Asterisk not working with session border controller Hi All My company has bought a license for a session border controller called Middle. It look like this software acts like a proxy and registers devices to the asterisk. the problem (As per Middle guys) is that Middle sends a registration request to the Asterisk asking it to reply to the port 7062, but the Asterisk sends back the packet to the originating port 7060. This confuses the Middle and the registration process stalls @ Proxy Authentication Required. When i talked to person responsible, he told me that Asterisk is not according to the RFC and he asked me to find a solution to fix this problem. The only info i got about middle is here (http://sip-martini.homeip.net/venkat/middle.html) Please help me in fixing this problem. I really dont know what to do as i'm not fully aware of SIP protocol or Middle. Thanks Venkat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk not working with session border controller
Venkat Venkataraju wrote: Steve Totaro wrote: And you say this is a commercial product that you purchased and they asked you to fix the problem? guys @ middle claim that the Asterisk is breaking the RFC and they are asking me to fix it on Asterisk as our company wants to use Asterisk. The middle does seems to work with many other SIP proxies. Where did you purchase it from or are you developing it and need help? More docs might help. I'm not developing it, but i'm working with a company thats trying to bring consumer VoIP by using Asteisk. I'm working on call processing system that uses astman. But i was pulled into this to get asterisk work with Middle. I dont have any other docs. i've asked for it. i'll post it if i get one. I've got logs from Middle, but i'm not sure how it may help. for the request REGISTER sip:sip-x.homeip.net SIP/2.0 Via: SIP/2.0/UDP 68.#.#.84:7062;branch=z9hG4bK-middle-4178 Via: SIP/2.0/UDP 68.#.#.125:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=1722079273 To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED]:7062 Call-ID: [EMAIL PROTECTED] CSeq: 6 REGISTER Content-Length: 0 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Asteisk is supposed to respond back to the IP on port 7062 (i was told. i'vnt read the RFC), but it send the request back on the port 7060, the originating port of the request. Venkat, Please add a SIP DEBUG trace of a registration, so I see how Asterisk responds. That's a weird Contact: header... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
ever notice the spec sheets from semiconductor manufacturers specifically exclude the device from being used for medical applications ? do something similar with asterisk - put a sticker on the box saying not 911 rated or something, use at your own risk. I wouldn't be caught dead (well maybe I would be :) ) without a plain old phone set plugged directly into one of my analog lines to use in an emergency. Lots of telco equipment comes with an emergency jack as well where if the device loses power, or self destructs or whatever, this is mechanically shunted over to the primary analog line with a relay that drops out when it loses power. The phone does not have to necessarily be at the pbx either, it could be brought out to the reception desk etc. On Tue, 2004-01-06 at 10:56, Jim Flagg wrote: Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. Good question otherwise. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
It's just as well that here in Hong Kong employers don't have to worry about being sued by their staff tripping over their own laces ; or microwave oven manufacturers getting sued by old ladies drying off their poodle ; or supermarket owners getting sued by stupid customers who trip over their own kids. In most countries cases such as these would be thrown out the minute they are filed. Of course, these are slight exaggerations insofar as asterisk is concerned - because being able to dial 911 (or 999 as it is in this part of the world) is a much more 'genuine' problem. But nonetheless, it should be the responsibility of the implementor of such a system to ensure that there are adequate measures taken against system failure - such as UPS, or even a primitive analogue phone line somewhere in the home/office. Though I cannot possibly comment regarding 'fear of being prosecuted', simply because I have no reason to fear (i'm not under jurisdiction of a ridiculous judicial system) - I would say that it is a huge shame that a group of people all with the common goal of contributing towards free software projects such as this should even have to worry about things such as lawsuits. If there are people out there who have problems with asterisk, I suggest they just don't use it. To go as far as suing - that is just taking the piss! (sorry, can't think of equivalent non-British term). Terence Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 12:12 PM Subject: Re: [Asterisk-Users] 911 and lawsuits I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Agreed, Guess I should have said traditional computer. Most PBXs would only use a hard drive for voice mail. A hard drive failure would not cause the PBX to stop working. Also, with something like Asterisk that is changing so often, there is always the possibility of a typo that is not discovered until you need to use one of those rarely used features like calling 911. Most business would have lots of cell phones around but in many metal building they do not work. They also don't provide the address information that a land line phone provides. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pls confirm
Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711?
Re: [Asterisk-Users] ATA call
Your ATA is not registering. I have a sample ATA-186 at http://www.fnords.org/~eric/asterisk/ On Tue, 2004-01-06 at 17:41, Osvaldo Mundim Junior wrote: Hi Doug, I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my ATA on show sip peers. What I can see is: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN Just one thing which I did not mention on the last email is that my ATA is behing NAT. Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco to Cisco - poor quality
I have never used Cisco phones, but I have had problems in the past relating to * RTP talking to a widget with VAD turned on. * RTP stack can not run on its own. It relies on receiving RTP packets for doing its timing. A simple test is to sniff the line to make sure the phones always send packets. If you see pauses, you may need to disable some type of VAD setting on the phone. Or just never quit talking when using the Cisco phone. Terence Parker wrote: I have set canreinvite=no in the sip.conf for each user (well, there are only two) using a cisco phone. What does this imply? As for whether the problem is due to the phones or asterisk however, indications would suggest both, because: - Voicemail works fine (and is clear) - I can initiate a call between MSN and Cisco, and that would sound fine. This might suggest a problem with my phones. However : - When using Vocal previously, Cisco to Cisco conversation was fine. This has led me to be completely stumped! I notice some mention elsewhere about asterisk lacking certain codecs because of license restrictions? Is this anything to do with me? Or should the phones still - in theory - be able to talk to each other without any problems? I have tried the cisco phone on both g729a and g711ulaw. I'm currently *trying* to get ahold of an updated firmware for my phone. I will see if this fixes the problems. Thanks again, Terence -- How are the phones talking to each other? Directly, or through asterisk? (canreinvite=what? in the sip.conf for each of them?). What I'm trying to get at here is, it is a problem between the phones, or are you having a problem possibly with the asterisk box? Some other things to know: are you running voicemail yet? If so and you can dial into it from either of the phones, how does it sound? If not, how about anything from the * boxlike the demo annoucment stuff? Daryl - Thanks for the replies. My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work fine under vocal though - which was strange. Is this definitely nothing to do with asterisk? I do note however that my firmware is fairly old... except cisco aren't exactly generous with firmware upgrades. I have tried both g729a (default on my phone) and g711ulaw with no success. But i'll have another fiddle and try to get it to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits and redundancy
- Original Message - From: Jonathan Moore [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 12:34 PM Subject: Re: [Asterisk-Users] 911 and lawsuits and redundancy This is esp true of any VoIP PBX system. In fact I think many of them run Windows. Or VOIP in general. This is what Vonage makes you agree to in their Terms of Service. 2.4 Requires Activation: You acknowledge and understand that 911 dialing does not function unless you have successfully activated the 911dialing feature by following the instructions from the Dial 911 link on your dashboard, and until such later date that such activation has been confirmed to you through a confirming email. You acknowledge and understand that you cannot dial 911 from this line unless and until you have received a confirming email. 2.5 Failure to Designate the Correct Physical Address When Activating 911 Dialing: Failure to provide the current and correct physical address and location of your Vonage equipment will result in any 911 communication you may make being routed to the incorrect local emergency service provider. 2.6 Requires Re-Activation if You Change Your Number: You acknowledge and understand that 911 dialing does not function if you change your phone number unless and until you have successfully activated the 911 dialing feature following the instructions from the Dial 911 link on your dashboard, and until such later date that such activation has been confirmed to you through a confirming email. 911 dialing must be re-activated. Although you may have activated 911 dialing with your former Vonage phone number, you must separately activate 911 dialing for any new number. 2.7 Change of Physical Location of Vonage Equipment: You acknowledge and understand that 911dialing does not function properly or may not function at all if you take your equipment with you away from the address or physical location that you have designated. 2.8 Requires Re-Activation if You Move: You acknowledge and understand that 911 dialing does not function properly or at all if you move or change the physical location of your Vonage equipment to a different street address, unless and until you have successfully activated the 911 dialing feature following the instructions from the Dial 911 link on your dashboard, and until such later date that such activation has been confirmed to you through a confirming email. 911dialing must be re-activated although you may have activated 911 dialing using your former address, and you must separately activate 911 dialing for any new physical address. Failure to provide the current and correct physical address and location of your Vonage equipment will result in any 911 dialing you may make being routed to the incorrect local emergency service provider 2.9 Possibility of Network Congestion and/or Reduced Speed for Routing 911: Due to the manner in which it is technically possible to provide the 911 dialing feature for Vonage DigitalVoice at this time, you acknowledge and understand that there is a greater possibility of network congestion and/or reduced speed in the routing of a 911 communication made utilizing your Vonage equipment as compared to traditional 911 dialing over traditional public telephone networks. You acknowledge and understand that 911 dialing from your Vonage equipment will be routed to the general telephone number for the local emergency service provider, and will not be routed to the 911 dispatcher(s) who are specifically designated to receive incoming 911 calls at such local provider's facilities when such calls are routed using traditional 911 dialing. You acknowledge and understand that there may be a greater possibility that the general telephone number for the local emergency service provider will produce a busy signal or will take longer to answer, as compared to those 911 calls routed to the 911 dispatcher(s) who are specifically designated to receive incoming 911 calls using traditional 911 dialing. 2.10 Automated Number Identification: At this time in the technical development of Vonage 911 dialing, it may or may not be possible for the Public Safety Answering Point (PSAP) and the local emergency personnel to identify your phone number when you dial 911. Vonage's system is configured in most instances to send the automated number identification information; however, the phone system routes the traffic to the PSAP and the PSAP itself must be able to receive the information and pass it along properly, and they are not yet always technically capable of doing so. You acknowledge and understand that PSAP and emergency personnel may or may not be able to identify your phone number in order to call you back if the call is unable to be completed, is dropped or disconnected, or if you are unable to speak to tell them your phone number and/or if the Service is not operational for any reason, including without limitation those listed elsewhere
Re: [Asterisk-Users] ATA call
Some times the sip show peers shows me: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN and some times shows me: Name/usernameHost Mask Port Status porto/porto 200.167.103.219 (D) 255.255.255.255 1025 LAGGED (815 ms) Does the port supposed to be 5060? Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
I would ask the same question about zero SLA Broadband Internet providers. How could an Asterisk installers determine if the Broadband latency reached a level were the IP network was not available to a VoIP subscriber at time of a 911 call. this is a log clip of a SIP UA connecting across a Cable modem. Jan 5 17:48:57 NOTICE[-1127097424]: File chan_sip.c, Line 4682 (handle_response): Peer '6400' is now TOO LAGGED! Jan 5 17:49:07 NOTICE[-1127097424]: File chan_sip.c, Line 4677 (handle_response): Peer '6400' is now REACHABLE! Jan 5 17:51:09 NOTICE[-1127097424]: File chan_sip.c, Line 4682 (handle_response): Peer '6400' is now TOO LAGGED! Jan 5 17:51:19 NOTICE[-1127097424]: File chan_sip.c, Line 4677 (handle_response): Peer '6400' is now REACHABLE! Jan 5 17:59:20 NOTICE[-1127097424]: File chan_sip.c, Line 4682 (handle_response): Peer '6400' is now TOO LAGGED! Jan 5 17:59:30 NOTICE[-1127097424]: File chan_sip.c, Line 4677 (handle_response): Peer '6400' is now REACHABLE! This subscriber would have a Best Effort 911 service. Doug Jim Flagg wrote: Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk feature list: spreadsheet
John I looked at your spreadsheet with interest. You grade each item with an indication of the amount of effort needed to use the feature. Level 1 effort is 1-6 hours of development, plus testing. Level 2 effort is 6-20 hours of development, and/or external purchase of tools. S=Standard, O=Optional, NA=Not Available This implies to me that items marked as S (standard) will take less than 1 hour of development effort. For a traditional PBX a Standard feature can generally be pretty easily configured in half an hour or so. As we know that is not the case for Asterisk and this therefore seems somewhat disingenuous. I see the need and value of a list like this - it is really helpful. However in fairness to anyone evaluating Asterisk from the perspective of someone who is familiar with a traditional PBX, should one not add a note explaining that ... Standard features are likely to take several hours to configure and may well only work with certain hardware / software combinations. Concise documentation to enable you to rapidly provision many of the features with particular hardware is not available. I would prefer that potential users are given a balanced and realistic expectation of what asterisk offers today on a like-for-like basis and to do that there needs to be a clear disclaimer/explanation such as that shown above. john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: 06 January 2004 14:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk feature list: spreadsheet http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106 .xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the common features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I figured I'd take as many of the unique features as others had offered, and put them together in a big list, and then also include some of the features that are unique to Asterisk. I think this list will be of some use to persons evaluating Asterisk against their existing PBX platform, or other vendors of new VoIP systems. Note that I took some liberties with the standard feature syntax: often, things that seemed _very_ easy for me to do with Asterisk's scripting features I listed as Standard, even though it would require a few minutes of work. Other features which seemed to be a little more complex, or which would require some web programming, I listed with estimates of how much time it would take to build them. I assumed use of Cisco 7960 phones, so some of the features which are really deskset options I listed as Standard if I were to use 7960 devices. Caveat emptor for other desksets. This is not truly a comparison, since there are no other columns in the spreadsheet. However, it's good fodder for you to whack your VoIP or PBX vendor on the nose with, since they typically will not be able to match the feature list. Olle: feel free to add to the Wiki, since you asked for this type of list. Anyone else: feel free to send me updates in cut/pasteable Excel form if you have things you'd like to add to the list. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
On Tue, 6 Jan 2004, Jon Pounder wrote: The phone does not have to necessarily be at the pbx either, it could be brought out to the reception desk etc. On Definity systems, we used a device called something like Emergency Cut-over. When power from the switch was lost, the device threw a bunch of relays cutting CO lines over to fax machines that were specifically chosen to allow dialing without power (many fax machines won't dial unless there is power) or to fax machines with a Y adapter connected to a $9 Wal-Mart phone. Normally, these fax machines would go through the switch, but if the switch had problems, it would cut over. We put big signs above all the fax machines indicating that they were EMERGENCY PHONES. I've also seen pay-phones installed in some areas to serve this function as well (typically shop environments where personal phone calls using company equipment were frowned upon). -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This newbie gives up for now - sadly
Hi John, I'd like to point out that there are several SIP phones that handle supervised transfers well. Try the Polycom phones (500 or 600). They work great. You must remember that SIP is a fat client protocol, which is to say, the phones have a considerable amount of intelligence in them. When the phones are natively bridged (the voice streams are not traversing Asterisk), then it makes sense that the phones handle their own transfer and conferencing functionality - Asterisk is totally out of the equation. It lessens the load on the server. Unfortunately, it also means that SIP phones don't tend to interoperate super well between brands. So, buy the right phones, and you'll be set. Christian On Tuesday 06 January 2004 05:20, John Coll wrote: Robert Hajime Lanning: He is using SIP phones. Supervised Transfers do not really work with SIP. He wants, on a SIP phone (I think he had Grandstream phones), to: o hit transfer o dial new extension o talk to new extension * this part does not work * o hit transfer to complete the transfer or some cancel button to abort Yes that is exactly what I want - thanks for clarifying. -- Derek Irwin: I guess what I'm saying is from the start, * continues to surprise and impress. If you put in the time to learn it, you will be rewarded with a feature-rich system that can go head-to-head with the commercials system out there. Well perhaps Derek but my experience so far, and I'm not talking rocket-science requests, is that Asterisk just does not do the most basic of things out of the box and that the documentation is so dispersed and incomplete that it needs a massive effort to get even the most basic stuff running. And in some cases even the most basic stuff turns out not to work - yet. I will come back to asterisk when it is leading edge and not bleading edge. This is not a criticism of asterisk - just that its clearly not at the stage where an average linux sysadmin can use it for normal PBX applications with a reasonable time investment - if at all. I am sure it will get there and I am very keen to come back on board when it does. I hope I have not offended any developers by these comments - I know I am just sitting here while you guys do all the work. Please keep up the good work - and thanks for the comments. john quote who=Tilghman Lesher On Monday 05 January 2004 13:44, John Coll wrote: This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let the two speak or get back the original caller - an utterly fundamental office procedure on a PBX. I don't know why you'd need to implement that, as it's as simple as turning on two options in zapata.conf. Actually, I think both of those options are on by default in the sample configuration files. And I've spent the requisite few hours on Google and all the docs I have printed out. Eventually I found the thread transfer with three-way calling (circa Mon, 15 Dec 2003 20:45:08 -0600) and it seems that I can't do that basic operation in Asterisk. Why not? Are you not able to send a flash hook? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
Jon Pounder wrote: ever notice the spec sheets from semiconductor manufacturers specifically exclude the device from being used for medical applications ? As does Microsoft's standard software license. Don't use this for any life-or-death application. (I believe medical and nuclear plant applications are specifically mentioned, but I haven't seen an MS license lately.) -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Geek In Charge * 888.480.4NET (4638) * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo with polycom phones
Hey Sean, Are you using Zap modem cards (X100P)? There can be bad echo with those things. The echo canceling effect you are hearing comes from the Polycom phones - they dynamically learn from the echo at the beginning of the call and adjust the echo cancellation accordingly. We are using a Mediatrix SIP gateway now - no echo at all. Christian On Monday 05 January 2004 13:28, Sean Garland wrote: I have soundpoing ip 500 phones and the first few seconds of every call has echo, which then goes away. Is there a way to have the echo cancel on at the beginning? It seems like it is testing at the beginning but it would be nice if I could have it start closer Thanks Sean Garland Siskiyou Technology Consultants 205 N. Mt. Shasta Blvd. Suite 100 Mt. Shasta, CA 96067 Phone: (530)926-1489 FAX: (530)926-6296 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This newbie gives up for now - sadly
John, Jared is right. I have a co-worker who has coughed up the money for the Cisco 7960 SIP phones. These have a soft button for Supervised Transfer. And, it works. I only have the Grandstream BT101 phones, and their Transfer button only implements Blind Transfer. So, to get it to work, you will need to upgrade to non-budget phones. Not ideal, but Asterisk does support the feature, just Grandstream does not. quote who=Jared Smith On Tue, 2004-01-06 at 06:20, John Coll wrote: Robert Hajime Lanning: He is using SIP phones. Supervised Transfers do not really work with SIP. He wants, on a SIP phone (I think he had Grandstream phones), to: o hit transfer o dial new extension o talk to new extension * this part does not work * o hit transfer to complete the transfer or some cancel button to abort Yes that is exactly what I want - thanks for clarifying. It sounds to me like this is a problem with the Grandstream phones in particular, and not Asterisk. Supervised transfers work *GREAT* with the Cisco 7960 phones... I use them almost every day. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pls confirm
Jess Magnaye wrote: Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? It can do G.723 between endpoints (passthrough).. It can do G.729a with the purchase of an additional licence of $10 per channel.. Yes you can use G.711 with a provider, some providers offer GSM, iLBC and Speex as alternative codecs.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Message waiting indicator
Thanks, the phones that I have Polycom Soundpoint IP 500's. In the specific config file for the phone itself, there are some lines that have to do with MWI and there are three settings to set. Here is the section of the manual for the phone msg.mwi.x.subscribe ASCII encoded string containing If non-Null, the telephone digits (the user part of a SIP will send a URL) or a string that constitutes SUBSCRIBE request a valid SIP URL (6416 orto this contact after [EMAIL PROTECTED]) boot-up. msg.mwi.x.callBackMode contact or registration If set to contact, a call will be placed to the contact specified in the callback attribute when the user invokes message retrieval. If set to registration, a call will be placed using this registration to the contact registered (the telephone will call itself). msg.mwi.x.callBack ASCII encoded string containing Contact to call when digits (the user part of a SIP retrieving messages URL) or a string that constitutes for this registration. a valid SIP URL (6416 or [EMAIL PROTECTED]) Does this mean that if the sip entry comes out to [EMAIL PROTECTED], is that what I put in for the subscribe and callback? I don't understand the connection between the SUBSCRIBE feature and the NOTIFY Anyone with Polycom experience with MWI? I will have to check to see if the NOTIFY is even happening... Sean -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 6:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Message waiting indicator What is required to get the mwi to work? Is it more of a phone subject or *? I have the mailbox= line in sip.conf, but only one extension is named, and in some of the examples, I have seen that there are two... What is that all about and how does it affect the extensions.conf and voicemail.conf? I think the examples that you might have looked are suggesting that when a voicemail is left for a single extension, you can place definitions in your sip.conf file that turn on the MWI (message waiting indicator) LED on more then one phone. (I'll leave that up to you to figure out whether that is a feature of use to you.) Asterisk will occasionally look in the /var/spool/asterisk/voicemail/default/3008/INBOX directory (where 3008 represents the extension number), and if a certain file exists, send a sip message to the extn(s) that you defined in sip.conf as mailbox=3008. The sip message sent to the phone (in hex using a packet sniffer) looks like: 0020: c1 5b 13 c4 13 c4 01 e2 58 97 4e 4f 54 49 46 59 | Á[.Ä.Ä.âX-NOTIFY 0030: 20 73 69 70 3a 33 30 30 38 40 32 30 35 2e 32 31 | sip:[EMAIL PROTECTED] 0040: 32 2e 31 39 33 2e 37 31 20 53 49 50 2f 32 2e 30 | 2.173.91 SIP/2.0 0050: 0d 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 | ..Via: SIP/2.0/U 0060: 44 50 20 32 30 35 2e 32 31 32 2e 31 39 33 2e 31 | DP 205.212.193.1 0070: 30 31 3a 35 30 36 30 3b 62 72 61 6e 63 68 3d 7a | 01:5060;branch=z 0080: 39 68 47 34 62 4b 33 63 31 63 61 35 65 31 0d 0a | 9hG4bK3c1ca5e1.. 0090: 46 72 6f 6d 3a 20 22 61 73 74 65 72 69 73 6b 22 | From: asterisk 00a0: 20 3c 73 69 70 3a 61 73 74 65 72 69 73 6b 40 32 | sip:[EMAIL PROTECTED] 00b0: 30 35 2e 32 31 32 2e 31 39 33 2e 31 30 31 3e 3b | 05.212.193.101; 00c0: 74 61 67 3d 61 73 35 37 63 63 64 33 32 65 0d 0a | tag=as57ccd32e.. 00d0: 54 6f 3a 20 3c 73 69 70 3a 33 30 30 38 40 32 30 | To: sip:[EMAIL PROTECTED] 00e0: 35 2e 32 31 32 2e 31 39 33 2e 39 31 3e 0d 0a 43 | 5.212.193.91..C 00f0: 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 61 73 74 | ontact: sip:ast 0100: 65 72 69 73 6b 40 32 30 35 2e 32 31
Re: [Asterisk-Users] 911 and lawsuits
quote who=Jim Flagg Most business would have lots of cell phones around but in many metal building they do not work. They also don't provide the address information that a land line phone provides. My company gets over the issue of the incorrect address information for the true location of the caller, by requiring that people inside the building dial a special extension (posted on every phone). This rings an emergency phone(s) at the central security office. We currently use a couple of Nortel PBXs. As for the PBX not working at all during the time of emergency, I don't know what we actualy do. (I am not telecom at my company, I manage firewalls.) -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI Scripting
Which Call Manager??? Luciano -Mensaje original-De: Michael Devenijn [mailto:[EMAIL PROTECTED]En nombre de Michael DevenijnEnviado el: Martes 6 de Enero del 2004 10:10Para: [EMAIL PROTECTED]Asunto: RE: [Asterisk-Users] AGI Scripting Use the callmanager their u can use the link event Michael Van: Luciano Ramos [mailto:[EMAIL PROTECTED] Verzonden: di 6/01/2004 13:56 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] AGI Scripting Hi!. Is there any way to know which extension answered a call , when dialing from an AGI Script?? Thanks! Luciano attachment: winmail.dat
Re: [Asterisk-Users] Pls confirm
quote who=Jess Magnaye Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? If you purchase the non-royalty free g729 codec, you can use g729. g723 is even more expensive, I believe. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Oz, Is the * server on a real Internet IP? NAT traversal may require you to redirect (PAT) SIP port 5060 to the inside IP or your ATA 186. try something like this, extensions.conf exten = 3000,1,Dial(SIP/3000,20,tr) exten = 3000,2,Voicemail,u3000 exten = 3000,102,Voicemail,b3000 sip.conf [3000] type=friend username=3000 secret= host=dynamic mailbox=3000 qualify=200 nat=yes Doug Osvaldo Mundim Junior wrote: Hi Doug, I do use the SIP 2.16x on my ATA 186. But I can not see the IP address of my ATA on show sip peers. What I can see is: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN Just one thing which I did not mention on the last email is that my ATA is behing NAT. Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
The question wasn't that someone had problems with asterisk, but was asking a question all consultants eventually have to think about. If you touch someone else's hardware, you are now playing a risk game. The risks are that you haven't clued your customer in fully on what to expect and therefore they think they are getting more than they they are. The risk is that you may not have written a tight proposal that limits what your customer could expect for the cash laid out. You also are risking that someone has an industrial accident as the power failed due to construction or storm and it wipes out the phone lines. All of this comes under the heading of risk management. One way of reducing exposure to risk is to have clients sign waivers that prove they are aware of the risks that they are assuming. This even allows the customer to make a informed descision as to whether the cost difference is worth it to them. After that, there is insurance. I understand Insurance is a very British thing, basically the business of gambling and spreading of risk. On Tue, 2004-01-06 at 11:42, Terence Parker wrote: If there are people out there who have problems with asterisk, I suggest they just don't use it. To go as far as suing - that is just taking the piss! (sorry, can't think of equivalent non-British term). Terence Just curious if any of the Asterisk installers are doing anything special to protect themselves from a possible lawsuit caused by 911 failure during a Asterisk/computer crash? I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Anybody requiring customers to acknowledge and sign any kind of waiver? Just the legal fees of defending yourself in a lawsuit could sink most Asterisk installers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hpw to enable Voicemail Indicator on IP/Analog Phone ?
Hello Whenever I receive voicemail on CISCO or SNOM or Analog Phones (Scitec), I would like to have some kind of indication in terms of beep sound or blinking voicemail indicatorCould you please tell me the way to enable this feature through Asterisk. Regards Tony
[Asterisk-Users] Scaleable Solution for small office
Hi, Have posted to this list a couple of times and have always received great responses and help. I have a basic * system setup Using 3 X100P cards with 6 Snom200 IP phones. It was a bit of a struggle getting everything up and running but have been pretty happy with the flexibility and ease of *. My major problem is one that has been discussed on this list many times before. The echo from the X100P cards is completely irritating to my users. I have tweaked the system as much as I can, IE: switching tip and ring, using aggressive suppressor, tweaking the TX/RX settings. There is still echo in the first 0-8 seconds of the call while the echo cancellation is catching up. I understand the problem is that you do not hear this echo in analog systems even though it is present and it is only heard in the digital system because it is not as fast as analog. So here is my problem. Our CEO wants me to get rid of the system unless I can provide a solution that will expand and work just as good as an analog replacement PBX. He came to me with this decision after trying to use the Snom200 speaker phone for a conference call, not a good idea as we all know the speaker on this phone is not a conference phone replacement. He also was frustrated when he tried to patch a call that came to another extension, which transferred fine, but was disconnected when the receiving party made a switch on their PBX to another extension and it was dropped. Currently we have 4 POTS lines and I have 1 setup for DSL and the other 3 hooked into the X100P cards, 2 outgoing and 2 incoming, using the DSL as incoming rollover. We are planning on expanding our business pretty rapidly and he wants a system that will be easy to setup and scale. I know asterisk and VOIP phones are great for this but the little glitches in the phones, hardware wise, are not supporting my Asterisk decision. I would like to upgrade to a channel bank, and was wondering if anyone has had any echo issues with other digium hardware. I know that the X100P issue comes from not having ECAN DSP in the card. I was also wondering if anyone had any luck adding a conference phone such as the Polycom Soundstation Conference Phone. Any suggestions would be appreciated Ryan R. Fligg Secured Digital Storage, Inc. 104 SW 4th St. Des Moines, IA 50309 Phone: (515)-244-6290 Cell: (720)-841-5802 Website: www.dstorage.com E-Mail: [EMAIL PROTECTED]
Re: [Asterisk-Users] 911 and lawsuits
On Tue, 2004-01-06 at 11:46, Jim Flagg wrote: - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 12:12 PM Subject: Re: [Asterisk-Users] 911 and lawsuits I realize that any traditional PBX or even a phone line can fail but, anything running on a computer is probably going to be less reliable than most PBXs. What do you think most PBXs are? Maybe not a x86, but it is a computer. Agreed, Guess I should have said traditional computer. Most PBXs would only use a hard drive for voice mail. A hard drive failure would not cause the PBX to stop working. Also, with something like Asterisk that is changing so often, there is always the possibility of a typo that is not discovered until you need to use one of those rarely used features like calling 911. Most business would have lots of cell phones around but in many metal building they do not work. They also don't provide the address information that a land line phone provides. In the US they do now. Most Cell phones now either have a GPS unit built in, or will identify via some form of cell tower information. I think the requirement right now is to know where the phone is to within 100 feet or so. As for the metal building, you'd be surprised how well they work. The only troubles I had seen before where related to wireless devices that used similar radio space. When I worked for Ingram Book Company, the warehouse used wireless terminals to deal with inventory tracking and movements. These terminals used 900-930Mhz spread spectrum. This trampled all over the beeper frequencies that were available. Where ever the transmitters where strongest, you absolutely had no chance of being beeped. Move out farther into the warehouse where fewer transmitters where and you could get some through. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pls confirm
- Original Message - From: Jess Magnaye [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 1:00 PM Subject: [Asterisk-Users] Pls confirm Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? Do you mean any specific provider? Asterisk can be a g729 endpoint if you have the licenses. It can connect two g729 devices without a license as long as it doesn't have to get in the middle for *any* reason. g723: It might, I just don't know. See the wiki: http://www.voip-info.org/ - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems compiling cdr_pgsql
Hi, Having installed postgresql-devel-7.4-0.3 and postgresql-libs-7.4-0.3 I'm having probs. compiling cdr_pgsql, can anyone offer any pointers as to what I might be missing? I'm hoping I've just missed out something like postgresql-wibblewobble-7.4-0.3 or something ... Below is the result of a make in the cdr source dir which may help those of you in the know thanks... Andy [EMAIL PROTECTED] cdr]# make Run make install from /usr/src/asterisk directory Girish _ Send DD, pay no commission. http://server1.msn.co.in/msnleads/suvidha/dec03.asp?type=hottag Click here. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer Function in *
Is there way to program the keys to transfer calls on analog fones instead of using the pound sign i've notice while i was calling to check voice mail at work that when i hit the pound key i get transfer message i want to use something elese besides pound to transfer calls. any insight would be great
[Asterisk-Users] 911
FYI there is a way to do 911 its called E-911 enhanced 911 the user has to set it up with the local emergency services to it and you setup your pbx to xmit the data. Here is the fcc rule about it http://www.fcc.gov/911/enhanced/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Heads up v2.03h on snom 200
Christian, Just a quick heads up warning that something seems to be amiss on the Snom 200 running v2.03h code. The Message Waiting Indicator is blinking (due to a voicemail left in Asterisk) as its suppose to. Some time later the LED stops blinking (as though no voicemail); then later it starts blinking again. I left a voicemail to x3008 (button 2) this morning at 8:48am, and purposefully have not picked it up. I'm waiting on recurrance right now. I'm also running a sniffer packet trace to validate whether the phone is being told to turn off the LED, or whether something in the phone is doing it by itself. Pure guess: since there are two lines now defined on this phone, I'd bet Asterisk turns the LED on for MWI on x3008, and since no voicemail exists on x3002 (button 1), it turns it off. (Or something like that.) I don't seem to have this issue with the Cisco 7960's on the same asterisk box, so I'm jumping to the conclusion that something is amiss on the Snom 200 code. But we both know what assume means. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA call
Sometimes Cisco nat changes the port, and * can't contact to ATA. I see that behavior some days ago, and I can't resolve that. :( Regards, Gus - Original Message - From: Osvaldo Mundim Junior [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:15 PM Subject: Re: [Asterisk-Users] ATA call Some times the sip show peers shows me: Name/usernameHost Mask Port Status porto/porto (Unspecified) (D) 255.255.255.255 0UNKNOWN and some times shows me: Name/usernameHost Mask Port Status porto/porto 200.167.103.219 (D) 255.255.255.255 1025 LAGGED (815 ms) Does the port supposed to be 5060? Oz - Original Message - From: Doug Shubert [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 9:09 AM Subject: Re: [Asterisk-Users] ATA call Is your ATA running SIP if so, what version (2.16?) With SIP, then * extensions.conf and sip.conf files are configured you should see the following asterisk3*CLI sip show peers Name/usernameHost Mask Port Status 3000/300010.0.0.30 (D) 255.255.255.255 5060 OK (15 ms) 9000/900010.0.0.90 (D) 255.255.255.255 5060 OK (47 ms) ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960 to test an extension from the CLI CLIdial ext. # you should hear your ATA ring Doug Osvaldo Mundim Junior wrote: Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz Name: ast_log.txt ast_log.txtType: Plain Text (text/plain) Encoding: quoted-printable -- FREE Unlimited Worldwide Voip calling set-up an account and start saving today! http://www.voippages.com ext. 7000 http://www.pulver.com/fwd/ ext. 83740 free IP phone software @ http://www.xten.com/ http://iaxclient.sourceforge.net/iaxcomm/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk feature list: spreadsheet
For some of us, the estimations are dead on. I don't necessarily think the intended audience was for the telecom newbie. The time involved is definitely experience related. You have proven you aren't ready to administer your own asterisk machine, please whip out your check book as you will either be paying a asterisk consultant to fill in your gaps, or you will be paying for a person to install a PBX that has some other name on it. Understand that only people who have invested very significant time into learning the telecom world and asterisk will be able to turn a profit with asterisk. Maybe you might want to stop making comments here before too many of them become archived and your companies name tarnished. If you continue, we will have to assume you are a troll since you don't care about your company. On Tue, 2004-01-06 at 12:19, John Coll wrote: John I looked at your spreadsheet with interest. You grade each item with an indication of the amount of effort needed to use the feature. Level 1 effort is 1-6 hours of development, plus testing. Level 2 effort is 6-20 hours of development, and/or external purchase of tools. S=Standard, O=Optional, NA=Not Available This implies to me that items marked as S (standard) will take less than 1 hour of development effort. For a traditional PBX a Standard feature can generally be pretty easily configured in half an hour or so. As we know that is not the case for Asterisk and this therefore seems somewhat disingenuous. I see the need and value of a list like this - it is really helpful. However in fairness to anyone evaluating Asterisk from the perspective of someone who is familiar with a traditional PBX, should one not add a note explaining that ... Standard features are likely to take several hours to configure and may well only work with certain hardware / software combinations. Concise documentation to enable you to rapidly provision many of the features with particular hardware is not available. I would prefer that potential users are given a balanced and realistic expectation of what asterisk offers today on a like-for-like basis and to do that there needs to be a clear disclaimer/explanation such as that shown above. john -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: 06 January 2004 14:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk feature list: spreadsheet http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106 .xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the common features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I figured I'd take as many of the unique features as others had offered, and put them together in a big list, and then also include some of the features that are unique to Asterisk. I think this list will be of some use to persons evaluating Asterisk against their existing PBX platform, or other vendors of new VoIP systems. Note that I took some liberties with the standard feature syntax: often, things that seemed _very_ easy for me to do with Asterisk's scripting features I listed as Standard, even though it would require a few minutes of work. Other features which seemed to be a little more complex, or which would require some web programming, I listed with estimates of how much time it would take to build them. I assumed use of Cisco 7960 phones, so some of the features which are really deskset options I listed as Standard if I were to use 7960 devices. Caveat emptor for other desksets. This is not truly a comparison, since there are no other columns in the spreadsheet. However, it's good fodder for you to whack your VoIP or PBX vendor on the nose with, since they typically will not be able to match the feature list. Olle: feel free to add to the Wiki, since you asked for this type of list. Anyone else: feel free to send me updates in cut/pasteable Excel form if you have things you'd like to add to the list. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
