[asterisk-users] Asterisk and Solaris
Has anybody tried running Asterisk on Solaris on a SUN SparcStation ? Or maybe the alternative of running Asterisk on a Linux Distro on a SUN SparcStation? I am asked to do this but I think it's almost impossible work to make it happen. Regards, Jorge A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP Bandwidth questions
Capacity is planned using Erlang Formulae which is a medium complexity statistical model mainly used for voice communications trunk occupation and switching capacity. Some idea of bandwith usage might be obtained using the simple calculators at www.voipcalculator.com Regards, Jorge A. Erick Perez wrote: This one will surely heat up. Usually the telcos have to calculate the subscribers vs telco capacity. I use simple figures, so extrapolate this to millions of customers, millions of lines, peak amount of calls at any given time of the day and of course houndreds,thousands of millions of dollars in equipment. For example: Telco A has 100 subscribers to his phone service in a city (home and business), so he needs to ask himself a- Will the telco buy a switch that can handle 100 calls simultaneously? So he can provide service to his subscribers 100% of the time at any time of the day even during riots,panic,flood,etc? b- Or will the telco go for a balance and guess that at the peak time of the day he will have 75 simultaneous call, so he goes out and buy a switch that handles 75-80 calls at the same time? c- how many trunks will the Telco have to talk to other telcos? So telco in City A can communicate with Telco in city B (or even in the same city)? International voice providers suffer from this kind of problem. Some sell plastic cards with a local phone number and a pin so you call them to call to other cities/countries but that cheap voice provider has, let's say, ten thousand long distance lines and ten thousand local phone numbers, but they sell 100k plastic cards a month with a peak usage 3 times every ten days of 12thousand lines? obviously 2 thousand callers wont get connected (only 3 times every ten days in a specific time range) but the other 7 days the peak usage is 10thousand calls? Every ten days the provider try to connect 106k calls but fail to connect 6k calls, that's 6% failure rate every ten days (100% in a 7 days period and 98% in those 3 days). Can you live with that failure ratio? that's up to you. I don't work for a Telco, but a Telco may apply the dialup-internet rule (and they live happy with it) for 30subscribers-to-1line home users and 10(or 5)subscribers-to-1line for business. (correct me if I'm wrong please it will be nice to know real figures). So apply the same rule to you VoIP hosting. -What codec will you use? let say g711 and let's say it uses 100kilobits per leg. -How many subscribers will you have in a 6 month period? 500 -So to provide all of them with service you will need 48Megabits of bandwith all the time just to connect to your Telco equipments. - But you decide that you analyzed the usage patterns of your service and you will have only 125 subscribers calling other 125 subscribers (this is called On-Net) at peak time every day at 6pm (rush hour). So, go out and buy 24mbits of bandwidth only. - But you suddenly have the option to hire burst IP service where your IP carrier can provide you with more bandwidth if your usage starts to rise in any given time of the day. So you calculate again that your minimum constant usage at any time of the day is 40 users On-Net, so go out and buy 5mbits (for a total of 50 calls) of bandwidth with burst IP enabled from 6pm to 8pm of 48mbits (or 24mbits). This scenario is only subscriberyour_companysubscriber. you also need to calculate subscriber--your_companyother_telcos And the last but most important question is: how much money do you have to burn on this? 100% Uptime full-service, Top Carrier Class performance (and even they get busy sometimes)? or almost perfect service with the once-in-awhile glitch of we're sorry all circuits are busy, please try again. Hope this helps, How many times (at least in my country) haven't you suffered from Im sorry all circuits are busy, please try again during christmas midnight, new years eve, election days or similar behaviors that cause massive amounts of calls being initiated and received? So the answer to your question On 11/2/06, mail-lists [EMAIL PROTECTED] wrote: Hello everyone, This probably isn't the correct place to ask this but I thought I'd check here first. We're getting ready to roll out a hosted pbx solution on a very limited trial basis (some company employees are going to get voip service at home). Our main issue is of course bandwidth. We have enough bandwidth (spread across two locations) to accommodate the few employees (around 10) for the near future but we're worried about how this is going to scale. Obviously at some point we'll need to consider 'real' bandwidth. My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is
[asterisk-users] Articulation Palm client and Asterisk
Hello, Has anybody configured Asterisk and the Articulation palm client to work ? I can make calls but I cannot make it register to receive calls. It does not register to the box. There are so few parameters that I think Asterisk sip.conf must be changed somewhat. I do not pass any parameters here because my box works perfectly with polycom, grandstream and linksys/sipura, and I know what to touch. The articulation software has only SERVER,DOMAIN, DISPLAY NAME, USER, PASSWORD, codecs are configured correctly (it only supports G711u and GSM), and I configured SERVER=DOMAIN (ip address) since it does not try to register until DOMAIN has something in i, Regards, Jorge Alayon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Fax Question
Another reason that people hang to it is that on some countries fax is a legal document while e-mail is not. Same reason why Telex is still used were e-mail is available but fax is not (some fishing vessels to my knowledge), communication media that has a legal status. Legislation changes slower than technology, so fax will be with us for a while. Regards, Jorge A. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Lee Howard Enviado el: Viernes, 03 de Marzo de 2006 05:18 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Asterisk Fax Question Michael Sampson wrote: Your best bet is to just not use fax machines. They are outdated technology. It is older technology, true... but certainly it's not useless technology. Certainly there is nothing yet to replace it properly. And I could argue this on a technological standpoint, and I have before on this list, but since you've such a closed-minded attitude towards older technology I don't think that there would be a point to it. E-mail is good and fun and has its uses. In some ways it has been able to replace fax communication, just like e-mail has been able to replace communications of other kinds as well. However, fax still has a purpose and a place, and many businesses still use it like crazy. With email there is little reason to use fax machines anymore. But for some reason people just feel the need to hang on to them. There still are many reasons to use fax, and yes, one of these is because so many people still have them and an analog phone line to boot. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on Hold Quality
Music on hold audio should be resampled to 8 bits 16 Khz mono and preencoded, so audio distortion is minimized in Asgerisk encoding. This theory has worked form me on other commercial platforms, but not yet on Asterisk, because MP3s cannot be resampled that way. If anyone figures it out, please advice. Regards, Jorge Alayón -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Matt Enviado el: Miércoles, 28 de Septiembre de 2005 06:42 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Music on Hold Quality I have heard this issue when on hold with Cisco and Vonage... I don't think it's an asterisk problem I htink it's a G711 problem... or gsm problem.Basically they are made for voice, and I think the music goes outside their encoding ranges... sound logical? On 9/28/05, canuck15 [EMAIL PROTECTED] wrote: I was on hold at Digium and noticed they had that EXACT problem. It was REAL bad on the one occasion I was on hold with them a couple weeks ago. If Digium has it then there must be some inherent issues with Asterisk that need to be worked out. I personally think this should be given a HIGH priority. Just my opinion as I am not a coder and cannot contribute in that way. From: Justin Selleck [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 28, 2005 11:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music on Hold Quality Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information? Thanks! -Justin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 300 with latest 1.5.3 firmware not registering
Hello, I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the reseller. This is my first experience with Polycom and I cannot make them register in my Asterisk Box. I followed every advice I found (including separating [user] and [peer] in sip.conf. Using ethereal, I found that it tries to SUBSCRIBE to the asterisk box and it receives a 403 FORBIDDEN message. I compared to a Grandstream registration, and it tries to REGISTER to the asterisk receiving a 200 Message response and effectively registering. Finding in the packet capture no other great difference, I believe that SUBSCRIBE requires a different authentification approach, maybe related to the voIpProt.SIP.requestValidation.digest.realm parameter in sip.cfg. I Tried the Polycom default, empty, default (voicemail context), from-internal (Extension context), the IP of the asterisk box, the name of the asterisk box, asterisk, etc, with no result. I tried different approaches documented in the wiki and related pages with no result. I can makke calls but I cannot receive them. I've seen mails stating that some installations have more than 100 phones working perfectly, can someone point me in the right direction to solve this ? Regards, Jorge A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk for Voicemail Server
I did it rerouting Forward Busy and Forward Noanswer from Meridian to a number in Asterisk that was a prefix+extension, and taking that as DIDs in asterisk directly to the voicemail of the extension. Of course there was no flasshing light on Meridian phones, but voicemail arrives via e-mail or web (using AMPortal). Regards, Jorge A. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Jonathan k. Creasy Enviado el: Miércoles, 31 de Agosto de 2005 05:04 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Asterisk for Voicemail Server How does one go about connecting Asterisk to a Meridian PBX to handle voicemail? I have a customer who is out of capacity on their voicemail system (which connects to their meridian via several FXS cards) and I would like to see if I could use Asterisk to handle their voicemail. -Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another problem on queues
n' logged off from
127.0.0.1 dialparties.agi: Extension 8521 has call waiting
disabled dialparties.agi: Max calls of 1 exceeded - deleting from
dial dialparties.agi: Dial string is empty - nothing to do
dialparties.agi: Was direct call, jumping to priority 26
-- AGI Script Executing Application: (NoOp) Options: () --
AGI Script dialparties.agi completed, returning 0 --
Executing Wait("Local/[EMAIL PROTECTED],2",
"1") in new stack -- Executing Playback("Local/[EMAIL PROTECTED],2",
"vm-nobodyavail") in new stack -- Local/[EMAIL PROTECTED],1
answered SIP/XXX.XXX.XXX.XXX-43921110 -- Stopped music on
hold on SIP/XXX.XXX.XXX.XXX-43921110 -- Playing
'vm-nobodyavail' (language 'en') -- Executing Playback("Local/[EMAIL PROTECTED],2",
"allison7/pls-try-call-later") in new stack -- Playing
'allison7/pls-try-call-later' (language 'en') -- Executing
Hangup("Local/[EMAIL PROTECTED],2",
"") in new stack == Spawn extension (macro-exten-vm, novm, 5) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
in macro 'exten-vm' == Spawn extension (from-internal, 8521, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
-- Executing Macro("Local/[EMAIL PROTECTED],2",
"hangupcall") in new stack -- Executing ResetCDR("Local/[EMAIL PROTECTED],2",
"w") in new stack -- Executing NoCDR("Local/[EMAIL PROTECTED],2",
"") in new stack -- Executing Wait("Local/[EMAIL PROTECTED],2",
"5") in new stack -- Executing Hangup("Local/[EMAIL PROTECTED],2",
"") in new stack == Spawn extension (macro-hangupcall, s, 4) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2'
Any
help will be appreciated.
Regards,
Jorge
Alayon
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A problem with queues
Hello, I am implementing a small call center with 1 to 4 agents. For some reason I don't understand, if an agent is busy, and it is his/her turn in the queue round, asterisk gives an all destinations are busy message and hangs up the call. Agents are SIP lines registered with an audiocodes MP108FXS which registers each line independently. Ringing strategy is RoundRobin (most of this configured using AMPortal, but checked that is consistent with documentation on queues on the wiki). Supposedly, the roundrobin strategy is used on available agents, not busy ones, am I correct ? Where can the failure be ? The best way I was able to replicate the problem is using only two agents, one received a call that stayed on for several minutes, the second received another call that waqs a short one, and the third call did not reach the second agent as it wanted to reach the first agent (per roundrobin) and failed. I was expectin it to ring again on the second agent. Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial Option A(file.gsm)
Hello,
I am trying to let someone know that is being called from a specified location.
For that, the command:
exten = _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm))
should let the called person hear Anounce.gsm as soon as he/she answers.
(Only calls with prefix 107 are given this notice).
The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this
file in every sound directory, but no luck.
Has anyone used this feature ? Are there any additional parameters or
restrictions ?
Regards,
Jorge A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Option A(file.gsm)
Thank you, it worked !!!, all the manuals and instructions I lokked at showed
the extension on this particular switch.
Now my problem is another, it gets played before the other party answers
because it believes the SIPGW (which is connected to the PSTN using ISDN)
answers the call as soon as it is accepted by it although not yet answered.
That is a SIP issue, but I don't understand why it happens. Other calls that
are normally dialed show ringing. This one with anouncement does not.
Regards,
Jorge A.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Eric
Wieling aka ManxPower
Enviado el: Jueves, 30 de Junio de 2005 05:22 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Dial Option A(file.gsm)
Jorge Alayon wrote:
Hello,
I am trying to let someone know that is being called from a specified
location.
For that, the command:
exten = _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm))
should let the called person hear Anounce.gsm as soon as he/she answers.
(Only calls with prefix 107 are given this notice).
The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this
file in every sound directory, but no luck.
Has anyone used this feature ? Are there any additional parameters or
restrictions ?
You never provide a file extension for those sort of stuff. Asterisk
will figure that out.
Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce))
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple E1s on one box
Answering both questions: 1) I am connecting to a Meridian usin a SIP E1 Gateway in R2. I just bought the card and one of my test will be direct R2 connection. Have not tried yet. 2) I was told you can do 12 E1 as long as it is G.711, but nobody is telling me how many E1s per box doing G.729. I have read twice that 80-90 ports is possible, but others tell me that no more than 30 is possible. Of course, the biggest one box CPU in consideration is a Dual XEON 3.0 with 1 GB RAM. Regards, Jorge A. -Mensaje original- De: Moises Silva [mailto:[EMAIL PROTECTED] Enviado el: Miércoles, 08 de Junio de 2005 06:02 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Multiple E1s on one box Franco: Why is not possible to handle 8 E1?? then is not possible to use 3 PCI cards with 4XE1 ports, hence having 12 E1? i have never installed an E1, but i tought it was possible when i saw this: http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduc t=TE405P. thanks On 6/8/05, Franco Bellagamba [EMAIL PROTECTED] wrote: Jorge, As far as I've read, you won't be able to handle 8 E1 in one box. By the way, have you had success with interconnecting E1 R2 argentina? I´m having trouble with a Meridian... I can only make calls from asterisk, but the other way arround... Tks Franco - Original Message - From: Jorge Alayon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 07, 2005 11:13 AM Subject: [Asterisk-Users] Multiple E1s on one box Hello all, Has anyone tried 8xE1 in one box using Asterisk and Digium boards ? What is the CPU needed for sustained performance in this capacity ? Is this affected if G.729 codec is used ? Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple E1s on one box
Hello all, Has anyone tried 8xE1 in one box using Asterisk and Digium boards ? What is the CPU needed for sustained performance in this capacity ? Is this affected if G.729 codec is used ? Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem starting RX_FAX and TX_FAX Module
Hello all, After compiling successfully Asterisk and AMPortal, I cannot make the fax module work. Asterisk does not start (unless I remove the modules or mark them as Noload in modules.conf) with the following error: Jun 3 20:55:25 VERBOSE[3328]: [app_rxfax.so]Jun 3 20:55:25 WARNING[3328]: /usr/local/lib/libspandsp.so.0: undefined symbol: dds_modf Jun 3 20:55:25 WARNING[3328]: Loading module app_rxfax.so failed! Could not find any references to dds_modf. I'm Using version 1.0.7 of Asterisk, and spandsp 0.0.2pre18. Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Audiocodes 108 FXS
Hello all, Has anybody cofigured in SIP the Audiocodes MP108 FXS in a way that each port is an extension of the Asterisk Box ? So each port can have it's own mailbox, etc ? Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation Fautl / Core Dump with G.729
Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal 1.0.0.8 SIP.CONF --- ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding nat=1 to each peer definition to ; solve translation problems. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=alaw allow=g729 allow=g723 context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown language=es #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf --- SIP_ADDITIONAL.CONF --- [as5300] type=peer qualify=yes host=xxx.xxx.xxx.xxx (AS5300 box) --- AS5300 relevant Config --- ... ! voice class codec 1010 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g723ar63 codec preference 4 g723r63 ! ... ! dial-peer voice 1010 voip destination-pattern 85.. progress_ind setup enable 3 progress_ind progress enable 8 voice-class codec 1010 session protocol sipv2 session target ipv4:xxx.xxx.xxx.xxx (ASterisk Box) dtmf-relay cisco-rtp rtp-nte h245-signal h245-alphanumeric ! --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASterisk OH323.CONF Gateway Gatekeeper
Hi, Does anybody knows how to konfigure oh323.conf to allow calls comming from a peering gateway (i.e.: Cisco 5300) which is not connected to a gatekeeper, and also from the gatekeeper to which Asterisk is registered ? Something like: GK(Carrier1)Registered to:-AS5300(carrier 1)-peer GW2GW--Asterisk---registered to:---GK(Carrier2) I Would like to receive calls from both carriers. Registering AS5300 to GK from carrier 2 is not an option. Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and clarent
AS seen on the web, Clarent is owned by Verso Technologies. www.verso.com Regards, Jorge A. -Mensaje original- De: Danny N [mailto:[EMAIL PROTECTED] Enviado el: Jueves, 07 de Abril de 2005 12:40 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Asterisk and clarent Are you saying Clarent could also upgrade to using SIP? I have a partner who is also using old Clarent on H.323. It'd be great if Clarent can do SIP. Who owns Clarent now, and how to get support? Thanks, Danny - Original Message - From: Jay Ray [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 11:39 AM Subject: Re: [Asterisk-Users] Asterisk and clarent Use SIP instead of H323 on Clarent and it will work fine --- Jorge Alayon [EMAIL PROTECTED] wrote: Hello, I have a carrier that is offering me service in H.323, but their platform is Clarent and I am not being able to connect my Asterisk box to it using the parameters they give me (H.323 ID, GK IP and GK ID) with oh323 registering to their gatekeeper. I have successfuly done it with Cisco GK and other vendors. Has anyone had previous experience with this Clarent platform and Asterisk ? Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Personals - Better first dates. More second dates. http://personals.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how can i connect a cost display on asterisk
In my country payphone solutions for Call Shops are implemented using FXS SIP or H.323 gateways that implement the Polarity reversal feature that reverse polarity as soon as the other party answers. I have done this in several VoIP platforms but Asterisk. Regular Payphones and Call Shop metering systems rely on polarity reversal for proper call billing implemented in a local table. Call Shop uses mainly a central billing unit (a PC or stand Alone) and serveral metering boxes with display that connects to the central unit by RS485 bus. These boxes connects to the lines (FXS on the gateways or special payphone lines provided by local carrier) and to normal phones. a Call Shop can have form 2 to 16 call boxes. It it good bussiness above 8. Billing systems are so simple electronically that they are manufactured here. Regards, Jorge A. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Enviado el: Miércoles, 06 de Abril de 2005 11:01 a.m. Para: asterisk-users@lists.digium.com Asunto: Re: [Asterisk-Users] how can i connect a cost display on asterisk [EMAIL PROTECTED] wrote: Johannes, I would be curious to know if there is a solution for this. Another solution is that you buy a call meter. Which is a small box that can be placed in front of phone phone and that can display costs. FXS-- call meter -- analog phone This call meter needs to be programmed with a table inside and a rate for each destination. It depends on the type of cost meter. One of BT's products is the Meter Pulsing Facility which sends a short 50Hz longitudinal tone on supervision, and just before a unit has been consumed. BT scrapped unit charging in the mid-90s but this particular bit of legacy remains. It's intended for payphones where you charge, say, 10p for a unit and want to know when the 10p has been consumed. That's why you can sometimes hear a buzz on a BT payphone a few seconds before the credit drops, because the longitudunal pulse sometimes breaks through into the audio path, even though shouldn't. I suspect this is because they payphone isn't properly earthed. A cost meter (or paypgone) that determines cost without exchange assistance will suffer from inaccurate pricing information and an inability to determine the start of supervision. Still, given that BT charge a hefty wedge for the MPF, some people just stick a COCOT on a standard exchange line and hope it's good enough that they don't get ripped off. -- Common sense is the collection of prejudices acquired by age 18. - Albert Einstein ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and clarent
Unfortunately this carrier does not support SIP on Clarent, maybe they are using an old version ? It seems that H.323 is the only way. Regards, Jorge A. -Mensaje original- De: Jay Ray [mailto:[EMAIL PROTECTED] Enviado el: Lunes, 04 de Abril de 2005 03:40 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Asterisk and clarent Use SIP instead of H323 on Clarent and it will work fine --- Jorge Alayon [EMAIL PROTECTED] wrote: Hello, I have a carrier that is offering me service in H.323, but their platform is Clarent and I am not being able to connect my Asterisk box to it using the parameters they give me (H.323 ID, GK IP and GK ID) with oh323 registering to their gatekeeper. I have successfuly done it with Cisco GK and other vendors. Has anyone had previous experience with this Clarent platform and Asterisk ? Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Personals - Better first dates. More second dates. http://personals.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and clarent
Hello, I have a carrier that is offering me service in H.323, but their platform is Clarent and I am not being able to connect my Asterisk box to it using the parameters they give me (H.323 ID, GK IP and GK ID) with oh323 registering to their gatekeeper. I have successfuly done it with Cisco GK and other vendors. Has anyone had previous experience with this Clarent platform and Asterisk ? Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VAD (Silence suppresion problem)
Hello, I'm trying to use Asterisk as a SIP PBX with H.323 trunk connectivity. Everything works except that calls that comes from the H.323 side do not get audio both ways. Since the other way round works fine (calls to H.323 side), I suspect the problem to be in the way VAD or Silence suppresion is negotiated. Is there a way to disable VAD in the Asterisk for H.323 gatekeeper connectivity ? I have tried with both H.323 and OH323 modules with no success. Regards, Jorge A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Diagnosing codecs
Hello, I am trying a setup that is the following: SIP Phone (Zultys) -- Asterisk --- H.323 GK (Cisco) PSTN Any calls from H.323 GW through GK goes to PSTN, no problem. SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem. SIP Phone to PSTN, rings normally, on the PSTN, then connects when the PSTN phone picks up, no audio on both directions. PSTN GW support both G.723.1 and G.729. Zultys suposedly supports G.729, G711u and a. I Have successfully compiled in Asterisk G.723.1 and G.729 following a mail from the list, and codecs appears in 'SHOW TRANSLATION'. Also, both codecs are configured in H323.conf and sip.conf. Is there a way to know what is happening on the audio or RTP stream by means of the asterisk CLI ? All I know (by protocol analyzer) is that SIP Phone sends stream to Asterisk, but none goes to PSTN GW. GK is not doing proxy. Regards, Jorge A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper
I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets. Is that correct ? Regards, Jorge A. -Mensaje original- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper
Thank you, I will need a SIP client with G723 and/or G.729 then. Do you know any sip clients that do both ? Regards, Jorge A. -Mensaje original- De: kido noagbodji [mailto:[EMAIL PROTECTED] Enviado el: Monday, November 22, 2004 8:42 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hi Jorge, The oh323 channel and h323 channel by NuFone are different. As far as your problem, this looks like a codec problem i had. Try to look that way. K. - Original Message - From: Jorge Alayon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 11:06 AM Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets. Is that correct ? Regards, Jorge A. -Mensaje original- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper
Thank you, I will see into it. Regards, Jorge A. -Mensaje original- De: Paul Davidson [mailto:[EMAIL PROTECTED] Enviado el: Monday, November 22, 2004 12:12 PM Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper Message: 4 Date: Sun, 21 Nov 2004 17:56:10 -0800 From: Paul Mahler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have been working with this precise same issue, under bug number 0002659. I've seen this problem all the way up to CVS-HEAD-11/21. In my case, I'm using the gnuGK gatekeeper, and connecting to cisco callmanager 3.3.3. While callmanager can call in to Asterisk via the gateway, calls do not proceed in the other direction- the only difference between this setup and my own (aside from a different gatekeeper) is that mine is 100% H.323 with IAX softphones used to attempt the call. I've been bouncing stuff back and forth with JerJer on this isse- one thing that might help you (it didn't help me) is to use CVS-HEAD, which will require an update to OpenH323 and PWLIB (that was a long evening). Not much help- but at least know you're not alone. -pbd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and H.323 Gatekeeper
Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
