Re: [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?
It _seems_ to only pick up the line when the VoIP end answers. At least for me it doesn't stop ringing until I see the log entry in Asterisk say it picked up the call. Now I just have to figure out how to get Asterisk to _not_ override the incoming caller id with the SIP information from sip.conf. The SPA-3000 says it sends the CID from the PSTN through, but Asterisk is just show the SIP extension number. On Wed, 2004-08-04 at 12:12 +1200, Andrew Gordon wrote: Andres wrote: The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk mainmenu context (or ext I guess). Configure an auto-dial number in the SPA to that it corresponds to something in the mainmenu context. Like: PSTN_Caller_Default_DP[2] 2 ; Dial_Plan_2[2](S0:551155) ; When a call comes in the FXO port, the SPA automatically dials 551155 via your Proxy[2] settings.. Does it answer the line first then ring the 551155 or does it ring the 551155 and only pick up the line when the VoIP end answers? ___ 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 -- Mike Benoit [EMAIL PROTECTED] ___ 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] Rodopi Billing
On Tue, 3 Aug 2004, Tom wrote: At 07:08 PM 8/3/2004, you wrote: That sigh will turn to cursing after a couple of months. We currently use Rodopi, have for 10 years but the inflexability is too much to deal with anymore so we are moving away from it. To what? I am also a cursed Rodopi owner. :-( Tom We use Platypus from Boardtown, which was just acquired by Tucows. Although it has it's quirks, having seen Rodopi, Emerald, Prism and ISPEasy in action, I'll take Platypus ANY day! I have a method for hacking VoIP per minute billing into Platypus, but I haven't executed it yet. Basically, we dump all of our CDR records to a database. On a daily basis, we can tally up the the per-minute LD totals for each customer and then insert a Radius Start/Stop record w/ the total billing seconds for that day. Platypus's built in rate tables take care of the rest. We provide 1,000 minutes of long distance with each account, so at the end of the month when Platypus tallies up it's overage charges, if the usage exceeds the limit, it bills the customer accordingly. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Cisco SIP Phone 7960 DTMF Problem
Hi, When we use BudgeTone where the DTMF is set to via RTP (RFC2833) all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to remote pick-up the call through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Best Regards, Miroslav Nachev ___ 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] Call pickup (group)
Hi, Asterisk has a feature called pickupgroup, meaning you can pickup the call that is ringing on your collegues phone. Can this type of behaviour be emulated in extension logic or AGI (maybe together with manager login) ? We need the group settings to be tied into a database which makes it a little more dynamic :- Any suggestions are welcome. Florian ___ 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] about sip.conf
HI all, Is there any possible to add sip entry 7004 from CLI without open sip.conf like [7004] type=friend username=7004 secret=123 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=11 nat=yes Thanks in advance Regards Murali
[Asterisk-Users] avm c4
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi there, now c4 does work :) i plugged isdn cable in the fourth controller instead of the first one; now, the problem is: why the 4th does work and the 1th does not? i will try the 2th and 3th in the morning 10x - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBEJU94Q/49nIJTlwRAjDWAJ9yHX72cUhA0txJg6G6DtgwbM8o4QCdGSIu GazjEpC45MpcUcoh3JT4kug= =VGAm -END PGP SIGNATURE- ___ 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] capturing a call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ddoes it feasible with * to capture a call? when arrives a call, floor bells ring and everyone can hear them in the company, then everyone can answer 'capturing' the call m. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBEJYq4Q/49nIJTlwRAsZtAJ9jDbfeLg9ia2n3yYy6RR3NBidY/wCcDgML O3ViqrM+ypEzAra3UOfZTVM= =0csq -END PGP SIGNATURE- ___ 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] rxfax killed asterisk
HI All. I'm using tiff-v.3.5.7 and spandsp-0.0.1k with latest * cvs on Slackware-10.0. Here is debug messages from * console. Please advise. Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1699.96 (66) Training error 2.228910 Training succeeded (constellation mismatch 4.905620) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page Killed P.S. Have the same installation on Fedora Core 2 and everything works ok. But I need it on Slackware :) -- Best regards Vlad ___ 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] capturing a call
How about simply making that floor bell extension part of a call queue? If you use ringall it will ring along with all handsets in same queue. This is what I'm thinking about doing - once I get the PRI up here :) Just an idea - I haven't tried it yet... but I hope you can use it somehow. Regards - avizion Quoting Maurizio Marini [EMAIL PROTECTED]: Ddoes it feasible with * to capture a call? when arrives a call, floor bells ring and everyone can hear them in the company, then everyone can answer 'capturing' the call -- avizion on irc.freenode.org #asterisk ___ 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] PRI/H323 gateway
Hi, I ve got a problem whenI do this : usr/src/asterisk/channels/h323# make There are a lot of errors with ast_h323.cpp and .h. And at the end, I've got this: make ***[ast_h323.o] Error 1 In fact, I want a sample PRI/H323 gateway. Asterisk ___ |___| ISDN(PRI) VOIP (H323) -- |E1||RJ45.|--- |___| || |___| SoI've configured zapetel.conf and zapata.conf for an E1 card. I 've installed pwlib and openh323 like in their README. I 've changed the h323.conf to allow g723.1 and to disable the gatekeeper. And for the expension.conf, at first,I decided to put all in comments. BecauseI just want to take the channels which come from the ISDN to use H323 in order to have VOIP and reciprocally. Can someone help me? Thanks for your answers
[Asterisk-Users] German sounds
Hi *, are there already some free German sounds for Asterisk? Regards Bastian ___ 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] about sip.conf
Not that I know of, although one possibility is have * read from a mysql database instead of sip.conf then all you would have to do is 'sip reload' Duane Cox - Original Message - From: Murali To: [EMAIL PROTECTED] Sent: Wednesday, August 04, 2004 2:24 AM Subject: [Asterisk-Users] about sip.conf HI all, Is there any possible to add sip entry 7004 from CLI without open sip.conf like [7004] type=friend username=7004 secret=123 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=11 nat=yes Thanks in advance Regards Murali ___ 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] VoIP experiences with Cable and DSL
On Tue, Aug 03, 2004 at 07:48:13PM -0700, Chris said: - Original Message - From: Steve Szmidt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 7:04 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL Assuming you're talking about Random Early Detection, are you saying that all cable providers use it? No, not saying that, just that since it's so CPU friendly and can handle large bandwidth, it's an attractive choice... however because VoIP packets are so tiny and very latency sensitive, RED is their worst nightmare :( -Chris This is an interesting thread, but it's VERY difficult to follow when quoting is done incorrectly. If I had not read previous messages, I would not know who said what. I urge all outlook and outlook express users to install quotefix which fixes Outlook and OE's horribly broken behavior. http://home.in.tum.de/~jain/software/oe-quotefix/ and http://home.in.tum.de/~jain/software/outlook-quotefix/ ___ 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] German sounds
On Wed, 4 Aug 2004, Bastian Schern wrote: Hi *, are there already some free German sounds for Asterisk? Try here: http://www.voip-info.org/wiki-Asterisk+sound+files+international Christoph ___ 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] German sounds
Bastian Schern wrote: Hi *, are there already some free German sounds for Asterisk? Regards Bastian Take a look at this site http://www.stadt-pforzheim.de/asterisk/index.html . Chris ___ 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] Who is calling me ?
Hi, I made a follow me from my phone (111) to my softphone (222)... when someone call me, softphone show my phone number (111).. I´d like to asterisk to forward the calling number, so I could know how is calling me. Thanks a lot. Andrei. _ MSN Messenger: converse com os seus amigos online. http://messenger.msn.com.br ___ 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] Congested link
I try to dial a number 3 rings and then busy tone in h.323 trace writes Congested link to 10.1.105.3 what this means and how to make this connection work ?
Re: [Asterisk-Users] VoIP experiences with Cable and DSL
On Tue, 3 Aug 2004 12:07:14 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: For a few years now I've operated with cable as the obvious choice, at least in my area where RoadRunner really built up a good network. It could be that for nation wide implementation VoIP really should be on DSL. (Unless of course you need a big pipe where a split T is the only higher option.) I currently use Cogeco cable in Oakville, ON, Canada. It has been fantastic! I don't think I've used a provider with as much available throughput (exactly as advertised). Only occasionally does the service go up and down, but that is infrequent. I have an external modem, and am using a pure VoIP setup with IAX trunking to my VoIP/PSTN gateway. Only occasionally do I get a dropped packet or something, but nothing to worry about. I spoke with my parents for an hour over the connection, and there was no problems (actually... I was getting some echo, but Asterisk nicely took care of it, and my parents were not aware of any echo cancelling going on until I told them what Asterisk was doing on my end, as I could hear it working). I will be using Cogeco again for me internet (cable) so that I don't have to pay Bell any money. Unfortunately my buzzer isn't going to work in the apartment, so I'll have to let guests in, but hey, I'll do a bit of leg work just to save any of my money going to the greater of two evils :) Leif Madsen. http://www.asteriskdocs.org ___ 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] 2 sip servers
Good day all I have figured out most/basics of asterisk.I went with sip and made my own sip.conf and extensions.conf No I have 2 servers running sip in different towns.Both is connected with static ip so thats fine,but now. Lets say someone want to call someone else in the other town.How do I get asterisk to know,for instance sip extension 101 is on another sip server on a different ip. And I even want to say if someone in town A calls a town B code it should go out threw town B's pstn card,so it will only be charge for a local call?Can this be done Its hard to explain for me and my english is not that good.Please Help Thanks Altus ___ 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] Who is calling me ?
On Wed, 04 Aug 2004 12:48:03 +, Andrei Goncalves
[EMAIL PROTECTED] wrote:
Hi,
I made a follow me from my phone (111) to my softphone (222)... when
someone call me, softphone show my phone number (111)..
I´d like to asterisk to forward the calling number, so I could know how is
calling me.
Set the CallerID to the ${CALLERID} of the line, then you should be
able to make the call appear to be coming from the incoming call as
opposed to the extension number that is calling you from Asterisk.
What I ended up doing was specifying a callerID argument in my macro,
then I just passed the ${CALLERID} variable as the argument. I then
did a SetCallerID(${ARG2}) where ${ARG2} ended up being ${CALLERID}.
At least this is how I did it in my follow me script. I'm sure there
are at least a couple of ways of doing this.
HTH,
Leif Madsen.
http://www.asteriskdocs.org
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[Asterisk-Users] SIP pickupgroup
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Any reason why pickupgroup has been limited to 31? 31 groups are quickly used up when you have multiple companies on the same server. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFBEOMg32si/nlrQ5gRAu3+AJ9FkeGMgb1JaAy2WjY8wBNEsN4WnwCeMFP0 tn8MacTrFSK8ySN/yNbG/8I= =Erm8 -END PGP SIGNATURE- ___ 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] 2 sip servers
On Wed, 04 Aug 2004 15:03:59 +0200, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I have figured out most/basics of asterisk.I went with sip and made my own sip.conf and extensions.conf No I have 2 servers running sip in different towns.Both is connected with static ip so thats fine,but now. Lets say someone want to call someone else in the other town.How do I get asterisk to know,for instance sip extension 101 is on another sip server on a different ip. And I even want to say if someone in town A calls a town B code it should go out threw town B's pstn card,so it will only be charge for a local call?Can this be done This sounds like you want to use switch = In your dialplans. http://www.voip-info.org/wiki-Asterisk+config+extensions.conf http://www.voip-info.org/wiki-Asterisk+-+dual+servers These are links I found on the wiki with a little bit of googling (the wiki search function isn't very good. Doing a site:voip-info.org and a couple of terms will get you what you want) HTH, Leif Madsen. http://www.asteriskdocs.org ___ 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] No incoming audio on incoming SIP calls
Hi Dave, Long time no see... I have been looking at the various packets going in/out of my network with regards to my SIP phones. I usually use the ethereal network sniffer on my network and it has wonderful support for SIP/RTP/RTSP/SDP analysing. In the setup phase of a call the RTP packet shows exactly what IPs and ports that your call is supposed to use. When I have these problems, I sniff the network and it shows a private IP instead of the public (external) IP in the voice data stream. Then it's easy to see what needs to be fixed. What kind of router/firewall do you use ? I can send you some examples later on tonight if you need them. Best regards, Johan Landerholm, Stockholm, Sweden (ex. SCO) Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to Stanaphone, FWD, Gossiptel and PSTN via an X100P. For incoming calls, an 0870 number from CallUK routes to my FWD account, and an 0870 number from Gossiptel routing to my Gossiptel account. Outbound calls all work fine ... I get audio in both directions, no problem. Incoming calls on either 0870 number connect fine, and audio goes from the softphone to the caller, but not the other way ... I hear no audio on the softphone from the caller's phone. I'm getting no alerts from my firewall that it's dropping anything. I know my way around packet sniffers, but I don't know what to look for here. What should the inbound audio packets look like? Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] VoIP experiences with Cable and DSL
Leif Madsen wrote: On Tue, 3 Aug 2004 12:07:14 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: For a few years now I've operated with cable as the obvious choice, at least in my area where RoadRunner really built up a good network. It could be that for nation wide implementation VoIP really should be on DSL. (Unless of course you need a big pipe where a split T is the only higher option.) I believe this is a 'religious' discussion. I deployed a widespread (phoenix/california/hawaii) telecommuting setup for 50 employees using H.323 (not Asterisk - Altigen at the time). This was across probably 15 different providers networks and spread pretty equally between Cable modem/router and DSL. In all cases 'business class' services were ordered at the highest available speeds. The bottom line - after 2+ years we have had about equal amounts of trouble over both media types. When it's good it's just about perfect - when it's bad it's the same as bad cell phone connections. The bad times are infrequent on either media types. My .02$ ___ 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] Astetrisk connectet to PBX
I have got the change to set op an Asterisk PBX for one department in a large organisation. This means that in instead of connecting the Asterisk direct to the PSTN thought E1 interface (ISDN) I have to connect thought the big PBX central (IS3000).(http://pbcextra.net/commonnet/marcom/pdf/001_06221.pdf) Is there any body who has experience with this? The right size of connection between the Asterisk and the IS3000 is a E1 interface (30 channels). Which cards can I use? Are the cards on (http://asterisk.org/index.php?menu=hardware) the only ones or are they the only ones you have tested? The next part is of course the connections whit the IS 3000. In another department they have some matra boxes (I'm not sure - but I think they are using the LAN for transporting phone conversations matra to matra) this matra main box is connected to the IS3000 with an E1 interface in both boxes. Yo keep the overall control of the telephone network these two boxes are using Q.SIG. - this way the system shoud be transeprant. Maybe I'm mixing things - pleas help me. Overall the main think is that the Asterisk must be transperant so that the services on IP-telephones that is hook op on the Astrisk is the same as the services on traditional telephones that is hooked op on the IS 3000. (Login in and out, forwarding, show callerID) So my question is - Is Q.SIG the way for connection the Asterisk to the IS3000? and do this give any problems. (I have seen that this have been an issue on the list about a year ago). Regards/Claus ___ 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] Rodopi Billing
I second this. Our Primary (not the one we bought) system is radiator, with mysql for the backend. I've never found something it couldn't do. -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Saliel Figueira Filho Sent: Tuesday, August 03, 2004 7:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Rodopi Billing I know it's OT here ... but anyone looking for a Radius server should consider getting Radiator - http://www.open.com.au/radiator/. It's not free, but at a very decent price you get the full source code (it's written in Perl), but you seldom will need to tweak the code, as it is flexible by design. No, I don't work or represent them, just an old-time happy customer. Back to my lurking now. Saliel ___ 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] 2 sip servers
I setup extension 105 on my Asterisk server to Dial(SIP/[EMAIL PROTECTED]) and then defined [sipserver_b] in the sip.conf So then I setup extension on sipserver_b's extensions.conf file to answer with the auto attendant, and it simply plays a message asking what number I want to dial. It then puts me in a context where all outbound calls are pushed through the pstn card in it. So now from my system, I dial 105 and I immediately get asked what number to call, and once I enter it, the call goes through. The same thing works from asterisk_b to asterisk_a, now our cities are linked. Not familiar with switch but I saw references to it before, will investigate further. - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 04, 2004 9:23 AM Subject: Re: [Asterisk-Users] 2 sip servers On Wed, 04 Aug 2004 15:03:59 +0200, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I have figured out most/basics of asterisk.I went with sip and made my own sip.conf and extensions.conf No I have 2 servers running sip in different towns.Both is connected with static ip so thats fine,but now. Lets say someone want to call someone else in the other town.How do I get asterisk to know,for instance sip extension 101 is on another sip server on a different ip. And I even want to say if someone in town A calls a town B code it should go out threw town B's pstn card,so it will only be charge for a local call?Can this be done This sounds like you want to use switch = In your dialplans. http://www.voip-info.org/wiki-Asterisk+config+extensions.conf http://www.voip-info.org/wiki-Asterisk+-+dual+servers These are links I found on the wiki with a little bit of googling (the wiki search function isn't very good. Doing a site:voip-info.org and a couple of terms will get you what you want) HTH, Leif Madsen. http://www.asteriskdocs.org ___ 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] Logging into Multiple Call Queues on two * Servers and Voice Mail option.
-Original Message- From: Shad Mortazavi [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 9:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Logging into Multiple Call Queues on two * Servers and Voice Mail option. Dear All, I have two objectives that I need to meet; 1. I need to be able to log into two separate call queues on two different Asterisk servers, servicing two data centers. I seem to have problems configuring my SNOM phone to actively register with both servers. Has anyone got a working configuration for this? I have not implemented this exact scenario, but I have setup agents in multiple queues. In queues.conf you can either specify the agents that belong to the group or use agent groups. Since each agent can be in multiple queues all you have to do is specify that agent in the second queues section. As far as getting it working with two servers all you should have to do is use the switch = statement to forward the calls back and forth. Basically the agent would have to log in to both servers, but once finished he/she could receive calls from both. 2. I need to have an option for the user to press a button when in the call queue to go to voicemail. Has anyone got a working configuration for this? You need to specify the context option in your queues.conf for that queue. When you have a context defined whatever buttons the caller hits is sent directly to that context. This can work very similarly to an auto attendant where you can have them hit one to leave a message or two to receive a callback, etc... I appreciate all the help. No Problem, I hope this qualifies. Warm Regards Shad Mortazavi Robert Jackson ___ 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] Problems with E100P
Hi, I'm having trouble configuring and E1 link , I know the E1 is a PRI and the switchtype is 5ess. The problem is that everytime I try to dial I got and error saying that it was unable to open the zap channel . Devices are created in the /dev/zap directory and I can open them with a cat /dev/zap/1. I can also see the channels in /proc/zaptel/1. Thanks in advance [zaptel.conf] span=1,1,0,cas,ami bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us [zapata.conf] trunkgroup = 1,16 spanmap = 1,1 [channels] switchtype=5ess context = default signalling = pri_cpe group = 1 channel = 1-15,17-31 cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 AMI/ RED 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 Clear 26 WCT1/0/26 Clear 27 WCT1/0/27 Clear 28 WCT1/0/28 Clear 29 WCT1/0/29 Clear 30 WCT1/0/30 Clear 31 WCT1/0/31 Clear ___ 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] Re: Analog FXO Card
-= On 15 Sep 2003 11:09:38 -0600, tom [EMAIL PROTECTED] said: And interestingly, the Digium card looks a lot like a product sold by Tigerjet, called the Personal Phone Gateway. I'm purely speculating on this, but Digium could have used Tigerjet's reference design for their own board. Steve Haehnichen replied: That's kindof how the industry goes. No point in rehashing designs or trying to beat volume manufacturers at their own game. The FCC Reg# on the board is for AMIGO Technology Co of Taiwan. I'm guessing the FXO board is a lot like an AMI-IA92: http://www.amigo.com.tw/products/modem/AMIIA92_IE92.htm You can zoom in here: http://www.amigo.com.tw/catalogue/Modem.pdf The same right down to the AMI-IA92/IE92 on the FXO silkscreen. :) For the record: I bought an XP 100 so that I could too get the support that I expect that I will need. However, I like to know what I buy when I purchase hardware. I do not understand what is this notion of AMI-IA92 - this is being labeled as Intel's software base solution for a V.92 modem under windows. However, this is still showing up in my /proc/pci as a TigerJet 300 Communications Controller. Does this mean that Intel software works for the TigerJet 300? If I boot into windows, could I use this board as a modem? What about T.38 and Fax support for this board, is this envisionable? What I am interested in knowing is whether the sound i/o on this board is down through PCI DMA or its being done through a serial port on a PCI bus. -=Francois=- ___ 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
OT: Re: [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????
On Tue, Aug 03, 2004 at 02:25:03PM -0500, Brian Capouch wrote: Despite using spam control, I still have to hit delete fifty times or so a day to get rid of those disgusting sex ads. Why is it any harder to do the same with messages that, upon swift perusal, aren't of interest? That's it! Train a spam filter to block those messages. For everyone too troubled to hit the delete key, they can just train a spam filter to block Broadvoice complaints. Better yet, maybe if they just dropped in all of the clueless questions they don't like, it would automatically filter it out for them! Mm. I'm should patent that. Something like Patent for Using Spam Filters to Block Stuff That Isn't Spam or something equally obvious. I like it. ___ 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] 2 sip servers
-Original Message- From: Deon Rodden [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 04, 2004 9:46 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 sip servers Not familiar with switch but I saw references to it before, will investigate further. I believe that it is for use with IAX, but I could be wrong. Robert Jackson ___ 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] SIP pickupgroup
because a unsigned int has usually 32 bits? (I assume that the calling group is based on a bitmask) You don't need the callgroups anyway, you can call multiple phones with the Dial() application, e.g. Dial(TECH1/Phone1TECH2/Phone2) if you want to call two phones at once. ___ 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] Problems with E100P
Marcelo Rodriguez wrote: Hi, I'm having trouble configuring and E1 link , I know the E1 is a PRI and the switchtype is 5ess. The problem is that everytime I try to dial I got and error saying that it was unable to open the zap channel . Devices are created in the /dev/zap directory and I can open them with a cat /dev/zap/1. I can also see the channels in /proc/zaptel/1. cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 AMI/ RED You've got a red alarm - looks like there's a connection problem somewhere. -- Jon Stockill [EMAIL PROTECTED] ___ 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] 2 sip servers
to put it this way town A= sip 100+ ,local aria code = 022 . town B= sip 200+ ,local aria code = 145 . Now I want people from tow A to dial 202 and it should go threw the static ip to the town B server to the users 202,and same for town B And If people from town A want to call the local butcher of town B whose number is (145) 88 55 66,it should not go out threw town A's pstn card but threw the static ip to town B's server and then going out threw town B's pstn card since its a 145 number! Hope this makes better sence Thanks so far On Wed, 2004-08-04 at 15:45, Deon Rodden wrote: I setup extension 105 on my Asterisk server to Dial(SIP/[EMAIL PROTECTED]) and then defined [sipserver_b] in the sip.conf So then I setup extension on sipserver_b's extensions.conf file to answer with the auto attendant, and it simply plays a message asking what number I want to dial. It then puts me in a context where all outbound calls are pushed through the pstn card in it. So now from my system, I dial 105 and I immediately get asked what number to call, and once I enter it, the call goes through. The same thing works from asterisk_b to asterisk_a, now our cities are linked. Not familiar with switch but I saw references to it before, will investigate further. - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 04, 2004 9:23 AM Subject: Re: [Asterisk-Users] 2 sip servers On Wed, 04 Aug 2004 15:03:59 +0200, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I have figured out most/basics of asterisk.I went with sip and made my own sip.conf and extensions.conf No I have 2 servers running sip in different towns.Both is connected with static ip so thats fine,but now. Lets say someone want to call someone else in the other town.How do I get asterisk to know,for instance sip extension 101 is on another sip server on a different ip. And I even want to say if someone in town A calls a town B code it should go out threw town B's pstn card,so it will only be charge for a local call?Can this be done This sounds like you want to use switch = In your dialplans. http://www.voip-info.org/wiki-Asterisk+config+extensions.conf http://www.voip-info.org/wiki-Asterisk+-+dual+servers These are links I found on the wiki with a little bit of googling (the wiki search function isn't very good. Doing a site:voip-info.org and a couple of terms will get you what you want) HTH, Leif Madsen. http://www.asteriskdocs.org ___ 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] Making asterisk distributed
Trilogy India wrote: Hi, I want to know, if someone has tried to use clustering in asterisk to increase its scalability and make it distributed?? If yes, how easy it is to cluster? Can someone please ive me details about the same Asterisk will not benefit from clustering. Jeremy McNamara ___ 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] IAX2 'no authority found' problem
Hi everyone, I'm having some problem trying to set up an IAX connection between two * servers. The scenario is : serverA has an X100p card and will direct all calls from the X100p over IAX to a specific extension on serverB which is at the other end of an unfirewalled VPN connection. At the moment serverA tries to redirect the call to serverB but recieves this message (it appears on both servers) : -- Executing Dial(Zap/1-1, IAX2/test:[EMAIL PROTECTED]/cardiff) in new stack -- Called test:[EMAIL PROTECTED]/cardiff Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 6ms SCall: 1 DCall: 0 [192.168.1.250:4569] VERSION : 2 CALLED NUMBER : cardiff LANGUAGE: en USERNAME: test FORMAT : 2 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 151287361 Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569] CAUSE : No authority found Aug 4 14:50:02 WARNING[147465]: chan_iax2.c:5339 socket_read: Call rejected by 192.168.1.250: No authority found Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569] -- Hungup 'IAX2/192.168.1.250:4569/1' == No one is available to answer at this time Here are excerpts from the config files : ServerA: extensions.conf [incoming] exten = s,1,Dial(IAX2/test:[EMAIL PROTECTED]/cardiff) ServerB: iax.conf [cardiff] type=friend username=test secret=test context=sipfonescard extensions.conf [sipfonescard] exten = cardiff,1,Dial(SIP/1101) Has anyone got any suggestions on what might be the solution to the 'no authority found' problem, I'm convinced that it must be something pretty simple that I'm missing but I can't find any tips to point me in the right direction. Any suggestions would be appreciated, Thanks, Simon Ward ___ 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] Rodopi Billing
Greg Boehnlein wrote: We use Platypus from Boardtown, which was just acquired by Tucows. Although it has it's quirks, having seen Rodopi, Emerald, Prism and ISPEasy in action, I'll take Platypus ANY day! I have a method for hacking VoIP per minute billing into Platypus, but I haven't executed it yet. Basically, we dump all of our CDR records to a database. On a daily basis, we can tally up the the per-minute LD totals for each customer and then insert a Radius Start/Stop record w/ the total billing seconds for that day. Platypus's built in rate tables take care of the rest. We provide 1,000 minutes of long distance with each account, so at the end of the month when Platypus tallies up it's overage charges, if the usage exceeds the limit, it bills the customer accordingly. If that customer is still around to pay you. Billing for services such as VoIP should be in real-time, anything less is unacceptable in my book. Jeremy McNamara ___ 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] Barge in on to agents conversation
Hi, 1. When an agent is active on a call, i need the ablity for a third person to join the conversation. Basically barge in by a supervisor, participate in the conversation and then leave. 2. As an extension to the above, while on call, can the agent request a conference from another agent and later hang him up. 3. Is there any way for a call to be put in the queue destined for a specific agent only? I need this for a callback feature with auto dial out. I tries the wiki, google, a lil bit of the code but found no answers. Maybe i am bad at searching. Can somebody please put me on to the right track Thanx in advance. Regards Navnit ___ 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] PRI/H323 gateway
Asmine Ouloube wrote: Hi, I ve got a problem when I do this : usr/src/asterisk/channels/h323# make There are a lot of errors with ast_h323.cpp and .h. And at the end, I've got this: make ***[ast_h323.o] Error 1 So I've configured zapetel.conf and zapata.conf for an E1 card. I 've installed pwlib and openh323 like in their README. I 've changed the h323.conf to allow g723.1 and to disable the gatekeeper. And for the expension.conf, at first, I decided to put all in comments. Because I just want to take the channels which come from the ISDN to use H323 in order to have VOIP and reciprocally. Can someone help me? Read the README again and if you still can't figure it out send least one error message, but the README does cover how to setup your environment and how to properly compile the code. Jeremy McNamara ___ 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] rxfax killed asterisk
On Wednesday 04 August 2004 05:18, Vladyslav wrote: HI All. I'm using tiff-v.3.5.7 and spandsp-0.0.1k with latest * cvs on Slackware-10.0. Here is debug messages from * console. Please advise. I get the same problem with Slack91 - some faxes work, some don't, some segfault *. I'm no longer using rxfax, at least not without another machine handy so I don't have to worry about taking our entire phone system down :-) -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] Re: problems with'#' transfer after hold
On Tuesday 03 August 2004 23:57, Chris wrote: Check out bugs.digium.com, bug number 2010. Twisted (one of the bug marshalls) has written a patch that allows you to set the transfer key in features.conf to be anything you want... Was this patch updated so that if you have a double-character transfer sequence and you don't get the secondary character(s), it emits the primary character? i.e. if you have it set to ## and you hit #... after 0.5s or something it emits the # to the channel so you can pass it on (say a remote IVR). -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] VoIP experiences with Cable and DSL
On Tuesday 03 August 2004 21:16, Steve Szmidt wrote: I take it you paid $200 for the Sangoma? Yes I did, and it was the best $200 I ever spent on VOIP equipment. Relatively inexpensive and like I said it eliminated the guessing games and queueing garbage. Did you have to get through any hoops to get it up, or did it just autoconfigure, as advertized, and you were a happy camper? The autoconfigure did *not* work. Several little problems but all solveable. 1) 2.4.26 is not supported at the time of this email. It's coming, they say, but they have some other pressing issues with other equipment and customers. (2.4.26 is closer to 2.6.x in terms of some of the driver backend) 2) The autoconfigure says it'll compile in ADSL but it doesn't... I found I had to do a manual config and then specify all the protocols it said were default (ADSL among them) -- the screen looked as if I'd done an default install but I had to do it manually. That's it. After that it built, link went up without any hassle and it's been working great. Now, having said that, I've been having perfect audio for the past 3 weeks but this past week I am having choppy outgoing audio but my bandwidth consumption is well below the maximum so I'm trying to track down what changed. I don't believe it to be a problem with the sangoma card, though. -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] Re: problems with'#' transfer after hold
- Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 04, 2004 7:46 AM Subject: Re: [Asterisk-Users] Re: problems with'#' transfer after hold On Tuesday 03 August 2004 23:57, Chris wrote: Check out bugs.digium.com, bug number 2010. Twisted (one of the bug marshalls) has written a patch that allows you to set the transfer key in features.conf to be anything you want... Was this patch updated so that if you have a double-character transfer sequence and you don't get the secondary character(s), it emits the primary character? i.e. if you have it set to ## and you hit #... after 0.5s or something it emits the # to the channel so you can pass it on (say a remote IVR). Yep, sure was I'm using it right now, it's great! -Chris ___ 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] Problems with E100P
In zaptel.conf, I think the line should be: span=1,1,0,ccs,hdb3 With an optional ,crc4 at the end. This works for my E1 card. The code you're using sounds like for a T1. Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcelo Rodriguez Sent: Wednesday, August 04, 2004 6:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problems with E100P Hi, I'm having trouble configuring and E1 link , I know the E1 is a PRI and the switchtype is 5ess. The problem is that everytime I try to dial I got and error saying that it was unable to open the zap channel . Devices are created in the /dev/zap directory and I can open them with a cat /dev/zap/1. I can also see the channels in /proc/zaptel/1. Thanks in advance [zaptel.conf] span=1,1,0,cas,ami bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us [zapata.conf] trunkgroup = 1,16 spanmap = 1,1 [channels] switchtype=5ess context = default signalling = pri_cpe group = 1 channel = 1-15,17-31 cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 AMI/ RED 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 Clear 26 WCT1/0/26 Clear 27 WCT1/0/27 Clear 28 WCT1/0/28 Clear 29 WCT1/0/29 Clear 30 WCT1/0/30 Clear 31 WCT1/0/31 Clear ___ 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] Rodopi Billing
That sigh will turn to cursing after a couple of months. We currently use Rodopi, have for 10 years but the inflexability is too much to deal with anymore so we are moving away from it. To what? I am also a cursed Rodopi owner. :-( Tom We bought the source code to wirebill and are building our own platform. Gary ___ 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: OT: Re: [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????
On Wed, 2004-08-04 at 09:00, Jayson Vantuyl wrote: For everyone too troubled to hit the delete key, they can just train a spam filter to block Broadvoice complaints. Better yet, maybe if they just dropped in all of the clueless questions they don't like, it would automatically filter it out for them! Mm. I'm should patent that. Something like Patent for Using Spam Filters to Block Stuff That Isn't Spam or something equally obvious. I like it. I think I have documented Prior Art on that. Your message nearly tripped the spam filter and this one will probably do so. In fact this mailing list is the reason I noticed my spam filter classified annoying questions as spam. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Making asterisk distributed
On Wed, 2004-08-04 at 09:30, Jeremy McNamara wrote: Trilogy India wrote: Hi, I want to know, if someone has tried to use clustering in asterisk to increase its scalability and make it distributed?? If yes, how easy it is to cluster? Can someone please ive me details about the same Asterisk will not benefit from clustering. Not in the HPC sense, but it will in the high availability sense. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Channel Bank
Since it doesn't look like any of the FXS cards supported by asterisk support analog DID trunks, would it work if I used a T100P connected to an adtran channel bank (atlas 550?) with an FXS card installed? Anyone ever try this configuration? ___ 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] German sounds
Bastian Schern wrote: are there already some free German sounds for Asterisk? Yes, 2 sets: http://voip-info.org/wiki-Asterisk+sound+files+international F ___ 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] Gafachi?
Luke, I have used them and have been very happy with the service. They are the only ones I have found that seem to be able to process a call from Firefly that goes through 2 * servers. Nufone and Voicepulse are not able to process it - I think it is a firefly problem not a Nufone or Voicepulse problem, since everything works fine if I use IAXPhone. Anyway, Gafachi has worked well for me, and I do not think you need to register unless you are receiving incoming phone calls. Charles -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Tuesday, August 03, 2004 9:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Gafachi? Anybody use them... I signed up for $20 to see how there system works.. They're at $.02 per minute for US Termination and their other ITX rates aren't too shabby. Sadly my IAX registration is rejected... maybe a glitch, wondering if anyone's had a similar issue. Luke ___ 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] IAX2 'no authority found' problem
Simon, i was having the exact same problem, the only solution I found, was to remove the secret, then it worked great.. I thought I must have been missing something too, but apparently not. I'm not sure exactly what is causing this, as if i set the servers up to register with each other, they register fine, but the moment they try to pass a call to one another, they fail, unless there is no secret listed in iax.conf for the connections. -twisted Simon Ward wrote: Hi everyone, I'm having some problem trying to set up an IAX connection between two * servers. The scenario is : serverA has an X100p card and will direct all calls from the X100p over IAX to a specific extension on serverB which is at the other end of an unfirewalled VPN connection. At the moment serverA tries to redirect the call to serverB but recieves this message (it appears on both servers) : -- Executing Dial(Zap/1-1, IAX2/test:[EMAIL PROTECTED]/cardiff) in new stack -- Called test:[EMAIL PROTECTED]/cardiff Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 6ms SCall: 1 DCall: 0 [192.168.1.250:4569] VERSION : 2 CALLED NUMBER : cardiff LANGUAGE: en USERNAME: test FORMAT : 2 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 151287361 Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569] CAUSE : No authority found Aug 4 14:50:02 WARNING[147465]: chan_iax2.c:5339 socket_read: Call rejected by 192.168.1.250: No authority found Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569] -- Hungup 'IAX2/192.168.1.250:4569/1' == No one is available to answer at this time Here are excerpts from the config files : ___ 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] Barge in on to agents conversation
Hello, On Wed, 2004-08-04 at 11:35, Navnit Chachan wrote: Hi, 1. When an agent is active on a call, i need the ablity for a third person to join the conversation. Basically barge in by a supervisor, participate in the conversation and then leave. Asternic, the Flash Operator Panel can do this, but you need to open it on a web browser and use your mouse to drag the manager extension to any leg of an already bridged call, with some extensions logic and meetme in the mix. I'm not sure if it will fit your needs, but it might help... http://www.asternic.org Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ 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] Cisco SIP Phone 7960 DTMF Problem
Hello, On Wed, 2004-08-04 at 04:35, Miroslav Nachev wrote: Hi, When we use BudgeTone where the DTMF is set to via RTP (RFC2833) all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to remote pick-up the call through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Did you try by dialing just '*8' ? -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ 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] BT100 bad handset?
hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical SalesOpen Telephony Labs, LLC801.527.2284www.optellabs.com
Re: [Asterisk-Users] IAX2 'no authority found' problem
On Wed, 4 Aug 2004, Simon Ward wrote: -- Executing Dial(Zap/1-1, IAX2/test:[EMAIL PROTECTED]/cardiff) in new stack -- Called test:[EMAIL PROTECTED]/cardiff Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 6ms SCall: 1 DCall: 0 [192.168.1.250:4569] VERSION : 2 CALLED NUMBER : cardiff LANGUAGE: en USERNAME: test FORMAT : 2 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 151287361 Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569] CAUSE : No authority found Aug 4 14:50:02 WARNING[147465]: chan_iax2.c:5339 socket_read: Call rejected by 192.168.1.250: No authority found Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569] -- Hungup 'IAX2/192.168.1.250:4569/1' == No one is available to answer at this time Hi, Seeing that you are sending username test, you need an entry in iax.conf for [test] That's how I understand it anyway. Steve ___ 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] rxfax killed asterisk
Hi Vladyslav, Several people with these symtoms - crashing just as the reception of the actual page starts - found they had other versions of libtiff on their system, as well as 3.5.7. When the others were removed the problem when away. Regards, Steve Vladyslav wrote: HI All. I'm using tiff-v.3.5.7 and spandsp-0.0.1k with latest * cvs on Slackware-10.0. Here is debug messages from * console. Please advise. Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1699.96 (66) Training error 2.228910 Training succeeded (constellation mismatch 4.905620) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page Killed P.S. Have the same installation on Fedora Core 2 and everything works ok. But I need it on Slackware :) ___ 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[2]: [Asterisk-Users] Cisco SIP Phone 7960 DTMF Problem
Dear Nicolas, NG Did you try by dialing just '*8' ? I try, but the result is the same. The problems is in Cisco Phone, because the same account with BudgeTone is working well. -- Best regards, Miroslavmailto:[EMAIL PROTECTED] Wednesday, August 4, 2004, 7:23:31 PM, you wrote: NG Hello, NG On Wed, 2004-08-04 at 04:35, Miroslav Nachev wrote: Hi, When we use BudgeTone where the DTMF is set to via RTP (RFC2833) all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to remote pick-up the call through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? NG Did you try by dialing just '*8' ? ___ 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] Get MWI from Telco's voicemail
Howdy I have a question regarding support for picking up when the telco sends a MWI message. My client's setup is a small office with three incoming lines on a TDM400P with iaxy's and a Grandstream as extensions. I am using CVS Head from yesterday. (I was resolving a different issue.) Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco. What I am seeing is an event every half hour exactly, on each of the two voice lines. This causes the simple switch to kick in and ring the extensions. Of course there is no one there. I have put a workaround in the dialplan of using WaitForRing(0) as the first entry in the s extension. The downside is that it delays the ringing of the extensions by one ring which my clients don't really like. Since these events happen every half hour and only on the lines that have voicemail I am very confident that it is the telco sending a trigger to turn the MWI on the phones either on or off. I really don't want to have to try and find out from the telco as their support is much, much, much, less than knowledgeable or helpful. What I am wondering is if there is any way for Asterisk to pick up on these message from the telco. It would be great to set the MWI on the extensions and remove the delay in ringing. I have searched through the Wiki and lists.digium.com to try and track down information to no avail. I think I have looked at every message that has MWI, stutter or message waiting in it. Everything that I can find only relates to Asterisk's voicemail. Any ideas or pointers are appreciated. Cheers Scott Petersen Xavier Solutions Inc. 250-216-5407 ___ 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] Rodopi Billing
Hi Gary, WireBill looks interesting. You mentioned that you are using the source code to build your own platform, but how does it hold up on its own? Can I ask what it can't do that requires you to build your own? Thanks, - Darren On Wed, 2004-08-04 at 08:14, Gary Carr wrote: That sigh will turn to cursing after a couple of months. We currently use Rodopi, have for 10 years but the inflexability is too much to deal with anymore so we are moving away from it. To what? I am also a cursed Rodopi owner. :-( Tom We bought the source code to wirebill and are building our own platform. Gary ___ 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 Integration
Can some one give me a tip on how do I integrate two asterisks Thanks Felippe Kilian Martins NPD-UFSC This message was sent using IMP, the Internet Messaging Program. ___ 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 ISDN-card
Hi! If I install a CAPI-compatible ISDN-card in my server, will that: a) enable me to connect that server to the public phone net b) allow me to connect an ISDN phone to the server and use it as a SIP-phone c) all of the above? Regards, Evert ___ 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] problems with'#' transfer after hold...
Hi, (please be gentle - still learning :)) This may be similar to a problem I had today... Calling an automated phone system wanted me to press # to confirm a number - obviously * treats this a transfer - and relevant prompts were played... If you wait for this to 'time out' (or do an invalid extension) no more DTMF tones are sent down to the called number (I did a test calling to a mobile afterwards) - even a # doesn't provoke the correct response. A thought that came to mind on this was - how do you actually send down a # when requested by an automated service as * treats this as its own. But - looks like the question has been answered by Chris' reply with this ## patch :) - still - I dont know if that would fix the 'if you time out a transfer and then try to send DTMF tones they dont actually get sent' problem. So in answer to Chris question - I dont have this patch installed. Thanks Wayne. Stephen Hon wrote: Hi.. Has anybody been experiencing any problems with transfers using # to transfer after taking a call off of hold? Transfers using the # and music on hold work fine by themselves. However, when we place somebody on hold we can no longer use the # to transfer. This is a problem since we use the # button to park calls. So, say a call comes in, the operator is on a call already, places call on hold and answers the new call, places new call on hold, resumes old call and tries to transfer using the # button it wont work, itll just play the DTMF tone for the # button. At first, I thought somewhere along the line the Tt options must be messed up in a dial command somewhere.. but I double checked everywhere and ensured that I was enabling transfers. Does anybody have any suggestions? Thanks, Steve ___ 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] Making asterisk distributed
Jeremy McNamara wrote: (B (B Trilogy India wrote: (B I want to know, if someone has tried (B to use clustering in asterisk to increase (B its scalability and make it distributed?? (B If yes, how easy it is to cluster? (B (BAsterisk will not benefit from clustering. (B (BIt all depends on how one defines the term cluster. (B (BThere was a time not so long ago when clusters had little (Bor nothing to do with the kind of distributed parallel (Bprocessing that is so often (perhaps wrongly) called (Bclustering today. A more precise term would be grid (Bcomputing, not clustering. (B (BDEC made the term cluster fashionable in the 80s with (Btheir VAXcluster architecture. They pretty much coined and (Bowned the term back then. But those clusters where (Bdesigned for high availability and redundancy, not for (Bparallelising and distributing a compute job over multiple (BCPUs. (B (BOf course a VAXcluster would also increase scalability in (Bthe same way that mirrored web servers do, simply because (Bthey offer the same service on a single virtual network (Baddress. Connections to the service are then workload (Bbalanced between multiple nodes, but any given job is (Balways executing entirely on a single node, unless the (Bnode goes down while the job is processing, in which case (Bit is failed over to another node to continue there. (B (BIn the hayday of the VAXcluster, if you used the word (Bclustering for any bundling of computing resouces that did (Bnot meet the high standards set by VAXcluster technology, (Bmost IT folks would have lectured you like "That's not a (Bcluster, it doesn't do proper failover, it doesn't have (Bquorum, it doesn't have distributed locking" etc etc. (BConsequently, Unix vendors were extremely careful to avoid (Busing the word cluster. They would use terms like (Bworkstation farm, compute farm, hot standby, etc etc. (BHowever, during the 90s the term cluster has become a (Bcatch all for anything that somehow bundles computing (Bresources. (B (BIn this sense, running multiple Asterisk servers to offer (Bthe same service on a domain name representing multipe IPs (Bthrough round robin DNS or similar techniques is a form of (Bclustering, much more so than grid computing is, at least (Bin the original sense as it was defined by DEC. In the (Bsame sense, TDMoE is a form of clustering. (B (BOf course if your definition of clustering is grid (Bcomputing, then your statement is correct. Grid computing (Bdoes nothing for Asterisk. (B (Brgds (Bbenjk (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto-attendant with an IP trunk
Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have extensions from 2000 - 2010. What happens is that Asterisk jumps out at the first digit '2' and says that nothing found with '2'. It doesn't even read the rest of the digits '000'. I expect this to be a basic PBX function and I am sure I'm missing something. Any help would be greatly appreciated. Regards, Anil ___ 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] Cisco SIP Phone 7960 DTMF Problem
On Wed, 2004-08-04 at 04:35, Miroslav Nachev wrote: Hi, When we use BudgeTone where the DTMF is set to via RTP (RFC2833) all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to remote pick-up the call through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Is your cisco dial plan file set up to allow you to dial *8#? -- Joshua M. Thompson [EMAIL PROTECTED] ___ 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] Cisco SIP Phone 7960 DTMF Problem
Hiya, IIRC Cisco's take the '#' as being a 'send what ive dialed' key when there is no active call. for example you could dial 123456, wait for the phone to 'time out' then it sends/dials your number or you could dial 123456# to send 123456 as soon as you press the # key. So - I would guess that when you dial '*8#' - asterisk is only getting a '*8' and not knowing what to do with it. Dunno if you can change a cisco to not use # to 'send' - too new to all this at the mo - this is just what I've observed with playing at home :) Wayne. Nicolas Gudino wrote: Hello, On Wed, 2004-08-04 at 04:35, Miroslav Nachev wrote: Hi, When we use BudgeTone where the DTMF is set to via RTP (RFC2833) all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to remote pick-up the call through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Did you try by dialing just '*8' ? ___ 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] rxfax killed asterisk
On Wednesday 04 August 2004 13:31, Steve Underwood wrote: Several people with these symtoms - crashing just as the reception of the actual page starts - found they had other versions of libtiff on their system, as well as 3.5.7. When the others were removed the problem when away. Untrue. I am one such person and I posted something to that effect to this list when I was playing with rxfax. The crashes became MUCH less frequent but they still occured on occassion :-) -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] Gafachi?
I use Gafachi as well. They have killer international rates. On Wed, 4 Aug 2004 11:08:56 -0500, Charles Ellis [EMAIL PROTECTED] wrote: Luke, I have used them and have been very happy with the service. They are the only ones I have found that seem to be able to process a call from Firefly that goes through 2 * servers. Nufone and Voicepulse are not able to process it - I think it is a firefly problem not a Nufone or Voicepulse problem, since everything works fine if I use IAXPhone. Anyway, Gafachi has worked well for me, and I do not think you need to register unless you are receiving incoming phone calls. Charles -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Tuesday, August 03, 2004 9:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Gafachi? Anybody use them... I signed up for $20 to see how there system works.. They're at $.02 per minute for US Termination and their other ITX rates aren't too shabby. Sadly my IAX registration is rejected... maybe a glitch, wondering if anyone's had a similar issue. Luke ___ 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] Zultys ZIP2
Hello All, I'm having trouble getting a Zultys ZIP2 to work with Asterisk, along with some other troubles in general. I keep getting a Got SIP response 481 Call Leg/Transaction Does Not Exist back from x.x.x.x). Even when Asterisk reports that the ZIP2 registered correctly, I can't make any calls out from the phone, or calls into the phone. Occaisionally I get a busy tone when I try to dial also. This is the entry for the ZIP2 in sip.conf [phone10] type=friend username=phone10 host=dynamic dtmfmode=rfc2833 mailbox=110 context=sip Incidentally, when I place the ZIP2 in a local subnet 192.168.0.x, the web interface for the phone is quick. But when I place it on a public IP address, the web interface is all but unusable and times out 90% of the time. However when I ping the phone it comes back with a 1~5ms ping. Does anyone have any ideas? Has anyone got a ZIP2 to work wth Asterisk? Thanks in Advance, Jason ___ 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] Re: Integration with Altigen
- Original Message - Message: 15 From: Geoff Nordli [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Tue, 3 Aug 2004 11:36:05 -0700 Subject: [Asterisk-Users] Integration with Altigen Reply-To: [EMAIL PROTECTED] I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple of open ports. If I used a TDM40B (FXS ports) could I interface that with the Altigen system and connect the two of them together. your integration would be difficult via FX(x). the altigen is going to need to terminate the ringing line at a destination, ie vm AA that gives routing options for example. can't remember from my altigen days how well it does something like DISA but that maight work as well. going the other way would be fine, you would have to set up a dial code like 9 in the altigen that selected one of the ports connected to the *. once the port was opened you could dial anything in the dialplan in *. down and dirty but also cheap. good luck jason kawakami Open Telephony Labs, LLC www.optellabs.com Sorry Jason, could you expand on your answer a little bit. I understand the dialplan parts, but not sure of the connection options. I don't have that much background in the PBX arena. Thanks, Geoff ___ 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] PSTN Access Providers for Asterisk
Asterisk Users, I'm looking for U.S. providers that will provide access to the PSTN and allow me to easily use my Asterisk box with their services. I would prefer a provider that supports number portability, so that I can park my existing cell number on their network and later move it again, but I'm open to doing some funky stuff with call forwarding if I have to do that. Can anyone provide their recommendations or experience in using a VoIP provider, as opposed to a LEC, to provide Asterisk with PSTN access? Thanks, in advance, for your ideas. -- _ __ __ ___ _| | William R. Lorenz [EMAIL PROTECTED] \ V V / '_| | http://www.ohiolinux.org/ ; Free conference and event hosting \./\./|_| |_| Linux and OSS-related topics. October 2, 2004 - Columbus, OH. ___ 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] calling card on Zap outbound
iH i want to use a MCI calling card for long distance (outbound) calls from a Zap channel. how can i have the Dial command wait for a number of seconds before entering the pin? thanks - hcir ___ 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 Integration
[EMAIL PROTECTED] wrote: Can some one give me a tip on how do I integrate two asterisks Yes, sure... look at http://www.voip-info.org/tiki-index.php ___ 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 Integration
On Wednesday 04 August 2004 14:34, [EMAIL PROTECTED] wrote: Can some one give me a tip on how do I integrate two asterisks I would ask that you visit http://www.catb.org/~esr/faqs/smart-questions.html and after meditating up on the knowlege contained therein come back and ask a question someone can actually answer. It's not that we're disrespectful to newcomers, it's that newcomers typically have no idea how to ask questions that will get them useful responses. Regards, Andrew ___ 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] Auto-attendant with an IP trunk
Could you post the part of your extensions.conf in question? On Wed, 2004-08-04 at 11:47, [EMAIL PROTECTED] wrote: Hi: I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I have an IP trunk to voicepulse and my outgoing calls go over that. I can also receive calls on that voicepulse trunk and want it to an auto attendant. Everything works except on the following: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have extensions from 2000 - 2010. What happens is that Asterisk jumps out at the first digit '2' and says that nothing found with '2'. It doesn't even read the rest of the digits '000'. I expect this to be a basic PBX function and I am sure I'm missing something. Any help would be greatly appreciated. Regards, Anil ___ 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] H323 Call Dropping
Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) - SER - ASTERISK - GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call immediately after receiving it. Can anyone tell me what is the reason for this? Is it a codec issue or anything i have misconfigured? I would sincerly appreciate any help or guidence on this. I am using Nufone Network's chan_h323 driver. This is from the Asterisk console: -- Executing Dial(SIP/XXX.XX.XXX.XXX-080f5e78, h323/h323:[EMAIL PROTECTED]) in new stack -- Called h323:[EMAIL PROTECTED] == No one is available to answer at this time -- Executing Hangup(SIP/XXX.XX.XXX.XXX-080f5e78, ) in new stack == Spawn extension (default, 14083339452, 2) exited non-zero on 'SIP/XXX.XX.XXX.XXX-080f5e78' This is the gatekeeper log: ACF|XXX.XX.XXX.XXX:1723|3950_endp|5285|14083339452:dialedDigits|995041321:dialedDigits|false; DCF|XXX.XX.XXX.XXX|3950_endp|5285|normalDrop; Registration details on gatekeeper for Asterisk: ? AllRegistrations RCF|XXX.XX.XXX.XXX:1723|root:h323_ID|gateway|3950_endp This is from h323.conf: [general] port = 1723 disallow=all allow=g723.1 allow=ulaw allow=alaw allow=gsm This is from sip.conf: [general] context=default port=5070 disallow=all allow=g723.1 allow=ulaw allow=alaw allow=ilibc allow=gsm Extensions.conf has these entries in the default context: exten = _.,1,Dial(h323/h323:[EMAIL PROTECTED]) exten = _.,2,Hangup *CLI show version Asterisk 1.0-RC1 built by [EMAIL PROTECTED] on a i686 running Linux TIA... /G __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ 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 Integration
Sure -- read up on IAX (www.google.com, www.voip-info.org are good places for this) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 04, 2004 1:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Integration Can some one give me a tip on how do I integrate two asterisks Thanks Felippe Kilian Martins NPD-UFSC ___ 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] RE: No incoming audio on incoming SIP calls
Solved my own problem ... thought I'd record it here for any others who come across it. The problem arises since Asterisk is trying to get out of the way of the media stream, by doing a SIP re-INVITE to get the two ends of the conversation to talk directly. This won't work, as Asterisk is telling the calling party that the IP address to talk to is the private IP address of the softphone on the internal network. Adding canreinvite=no to the softphone's stanza in sip.conf solves the problem. It would be helpful if Asterisk noticed that it's about to tell the other end to use a private IP address ... the ranges are well known, and Asterisk could do an implicit canreinvite=no in this situation. The same problem didn't occur on outgoing calls as the Dial string includes a t for timeout - as per the wiki, this means that Asterisk must stay in the stream to be able to implement this. Of course, the other way to solve this would be to use a proper SIP proxy server which handles RTP stream port forwarding ... something I must get around to. -- David Gurr Congruity Ltd. Hemel Hempstead, UK -Original Message- From: David Gurr [mailto:[EMAIL PROTECTED] Sent: 04 August 2004 14:05 To: [EMAIL PROTECTED] Subject: No incoming audio on incoming SIP calls Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to Stanaphone, FWD, Gossiptel and PSTN via an X100P. For incoming calls, an 0870 number from CallUK routes to my FWD account, and an 0870 number from Gossiptel routing to my Gossiptel account. Outbound calls all work fine ... I get audio in both directions, no problem. Incoming calls on either 0870 number connect fine, and audio goes from the softphone to the caller, but not the other way ... I hear no audio on the softphone from the caller's phone. I'm getting no alerts from my firewall that it's dropping anything. I know my way around packet sniffers, but I don't know what to look for here. What should the inbound audio packets look like? Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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 ISDN-card
Citat Evert Meulie [EMAIL PROTECTED]: Hi! If I install a CAPI-compatible ISDN-card in my server, will that: a) enable me to connect that server to the public phone net b) allow me to connect an ISDN phone to the server and use it as a SIP-phone c) all of the above? Only a) would work. b) (or c) for that sake) can only be archived with HFC-S based passive ISDN cards at the moment and either zaphfc/qozap modules/channel drivers or mISDN/chan_mISDN, when talking BRI. Kind regards, Martin List-Petersen -- When he got in trouble in the ring, [Ali] imagined a door swung open and inside he could see neon, orange, and green lights blinking, and bats blowing trumpets and alligators blowing trombones, and he could hear snakes screaming. Weird masks and actors' clothes hung on the wall, and if he stepped across the sill and reached for them, he knew that he was committing himself to destruction. -- George Plimpton ___ 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] Problems with E100P
Well I turn on the pri intense debuging and the only message that I could find was this: [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended Also Scott was right I was using the wrong setting , changes the zaptel.conf file with this span=1,1,0.ccs,hdb3 the /proc/zaptel/1 now shows this : Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS ClockSource But I can't still get the Zap channel to dial , any other thoughts? Thanks in advance Marcelo Rodriguez On Wed, 2004-08-04 at 09:58, Horacio J. Pea wrote: Well as you sugested I changed to ccs, hdb3 and nothing happen. Well the message on /proc/zaptel/1 changes to Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW ClockSource But at the end when I try to dial I still have the same errors Any thoughts? I'm very new to asterisk, so take my words with a grain of salt. First, you should ask your provider what signalling type you should use (ami/hdb3, cas/ccs, crc4/nocrc) Then configure that on zaptel.conf On asterisk, pri intense debug span 1 will let you see anything that is happening over the E1. Saludos! Saludos? Si hablas castellano no nos gastemos en usar ingles :-) Aug 4 09:40:51 DEBUG[1217602880]: pbx.c:1255 pbx_extension_helper: Launching 'Dial' Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 1 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 2 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 3 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 4 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 5 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 6 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 7 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 8 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 9 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 10 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 11 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 12 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 13 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 14 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 15 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 17 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 18 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 19 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 20 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 21 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 22 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 23 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 24 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 25 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 26 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 27 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 28 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 29 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 30 Aug 4 09:40:51 DEBUG[1217602880]: chan_zap.c:6535 zt_request: Using channel 31 Aug 4 09:40:51 NOTICE[1217602880]: app_dial.c:714 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time Aug 4 09:40:51 DEBUG[1217602880]: app_dial.c:974 dial_exec: Exiting with DIALSTATUS=CHANUNAVAIL. On Wed, 2004-08-04 at 09:28, Horacio J. Pea wrote: [zaptel.conf] span=1,1,0,cas,ami Are you sure that cas,ami is correct? I'm using ccs,hdb3 Saludos, HoraPe --- Horacio J. Pea [EMAIL PROTECTED] [EMAIL PROTECTED] ___ 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] PSTN Access Providers for Asterisk
Try voicepulse connect connect.voicepulse.com This mailbox protected from junk email by MailFrontier Desktop from MailFrontier, Inc. http://info.mailfrontier.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William R. Lorenz Sent: Wednesday, August 04, 2004 3:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PSTN Access Providers for Asterisk Asterisk Users, I'm looking for U.S. providers that will provide access to the PSTN and allow me to easily use my Asterisk box with their services. I would prefer a provider that supports number portability, so that I can park my existing cell number on their network and later move it again, but I'm open to doing some funky stuff with call forwarding if I have to do that. Can anyone provide their recommendations or experience in using a VoIP provider, as opposed to a LEC, to provide Asterisk with PSTN access? Thanks, in advance, for your ideas. -- _ __ __ ___ _| | William R. Lorenz [EMAIL PROTECTED] \ V V / '_| | http://www.ohiolinux.org/ ; Free conference and event hosting \./\./|_| |_| Linux and OSS-related topics. October 2, 2004 - Columbus, OH. ___ 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] Re: IAX2 'no authority found' problem
In article [EMAIL PROTECTED], Josh Roberson [EMAIL PROTECTED] wrote: Simon, i was having the exact same problem, the only solution I found, was to remove the secret, then it worked great.. I thought I must have been missing something too, but apparently not. I'm not sure exactly what is causing this, as if i set the servers up to register with each other, they register fine, but the moment they try to pass a call to one another, they fail, unless there is no secret listed in iax.conf for the connections. It does work, but it took me a little digging to understand. The following is a simple one-way setup that should work: -- ServerA: extensions.conf [some-context] exten = some-extension,1,Dial(IAX2/userid:[EMAIL PROTECTED]/[EMAIL PROTECTED]) ServerB: iax.conf [userid] type=user secret=password context=acontext extensions.conf [acontext] exten = extension,1,Dial(wherever) -- You can make the dial string more concise by putting a peer section in the iax.conf for ServerA: -- ServerA: iax.conf [identifier] type=peer username=userid secret=password peercontext=acontext host=192.168.1.250 extensions.conf [some-context] exten = some-extension,1,Dial(IAX2/identifier/extension) ServerB: same as previous example. -- However, it is only in recent CVS versions (from 1.175 of chan_iax2.c on 2004/07/30) that it picks up the username as userid from the [identifier] section. In older versions it is still necessary to say Dial(IAX2/[EMAIL PROTECTED]/extension). In fact, when experimenting I also found it necessary to say [EMAIL PROTECTED], even when peercontext= was set. I haven't yet fully investigated why. If ServerB is on a dynamic IP address, then host=192.168.1.250 must be replaced with host=dynamic. It is then necessary for ServerB to register with ServerA by including the following in ServerB's iax.conf: [general] register = identifier:[EMAIL PROTECTED] where 123.123.123.123 is the IP address or hostname of ServerA. Note that it is identifier:password, not userid:password. I've deliberately kept the various names different to show which is dependent on which. That's also why I have keep peer and user separate instead of using type=friend. Bi-directional peering can be set up by swapping the above sections for ServerA and ServerB, changing names accordingly. With suitable choices of names, and the same secret in both directions, it is then possible to combine type=peer and type=user sections into a single type=friend, e.g. -- ServerA (static IP): iax.conf [serverB] type=friend host=dynamic username=serverA secret=password peercontext=from-serverA context=from-serverB qualify=yes notransfer=yes extensions.conf [outgoing-to-B] ;exten = outextenB,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = outextenB,1,Dial(IAX2/serverB/extenB) [from-serverB] exten = extenA,1,Dial(wherever-on-A) ServerB (dynamic IP): iax.conf [general] register = serverB:[EMAIL PROTECTED] [serverA] type=friend host=serverA.name.or.ip username=serverB secret=password peercontext=from-serverB context=from-serverA qualify=yes notransfer=yes extensions.conf [outgoing-to-A] ;exten = outextenA,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = outextenA,1,Dial(IAX2/serverA/extenA) [from-serverA] exten = extenB,1,Dial(wherever-on-B) -- As I said, it ought to be possible to omit the context from the IAX2 Dial commands if peercontext= is specified. I will investigate further. Hope this all helps! Cheers Tony (softins) -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] VoIP experiences with Cable and DSL
[EMAIL PROTECTED] wrote: On Tue, 3 Aug 2004 12:07:14 -0400, Steve Szmidt [EMAIL PROTECTED] wrote: For a few years now I've operated with cable as the obvious choice, at least in my area where RoadRunner really built up a good network. It could be that for nation wide implementation VoIP really should be on DSL. (Unless of course you need a big pipe where a split T is the only higher option.) I currently use Cogeco cable in Oakville, ON, Canada. It has been fantastic! I don't think I've used a provider with as much available throughput (exactly as advertised). Only occasionally does the service go up and down, but that is infrequent. I have an external modem, and am using a pure VoIP setup with IAX trunking to my VoIP/PSTN gateway. Only occasionally do I get a dropped packet or something, but nothing to worry about. I spoke with my parents for an hour over the connection, and there was no problems (actually... I was getting some echo, but Asterisk nicely took care of it, and my parents were not aware of any echo cancelling going on until I told them what Asterisk was doing on my end, as I could hear it working). I will be using Cogeco again for me internet (cable) so that I don't have to pay Bell any money. Unfortunately my buzzer isn't going to work in the apartment, so I'll have to let guests in, but hey, I'll do a bit of leg work just to save any of my money going to the greater of two evils :) Leif Madsen. http://www.asteriskdocs.org I am using Shaw for cable access and Primus as my VSP. I find on a fairly regular basis calls will just drop after 10 minutes or so. The call doesn't hangup but you can't hear each other talk. I am not using any QOS or bandwidth rate control. Any ideas why this might be happening? Thanks, Geoff ___ 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] Get MWI from Telco's voicemail
On Wed, 2004-08-04 at 14:21, Scott Petersen wrote: Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco. Maybe I am misunderstanding you but why does this force you to use telco voice mail instead of * voice mail? You can also free that third line up for voice if you use faxdetect. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Playback doesn't work whith h323
Hi Jeremy My entry in the extensions.conf is like this: exten = 011001,1,Playback(tt-monkey) I didn't asociate the cmd Dial whit this entry, so, I can't answer the line (sorry by my english) - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 7:32 AM Subject: Re: [Asterisk-Users] Playback doesn't work whith h323 M. Willigs wrote: Hi everybody. I install the new Asterisk 1-RC1 on my machine and I can't make the Playback function works through chan_h323 whith version 7.x's configs files Any ideas? Are you Answering the line before attempting to playback audio? Have you read AND FOLLOWED the README in asterisk/channels/h323? Jeremy McNamara ___ 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] Auto-attendant with an IP trunk
On Wed, 2004-08-04 at 14:47, [EMAIL PROTECTED] wrote: - one of the options is to allow the caller to press the extension that they would like to be connected to. I have extensions from 2000 - 2010. What happens is that Asterisk jumps out at the first digit '2' and says that nothing found with '2'. It doesn't even read the rest of the digits '000'. You might consider adding a DigitTimeout command at the beginning of your IVR. http://www.voip-info.org/wiki-Asterisk+cmd+DigitTimeout -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] RE: No incoming audio on incoming SIP calls
I must do the same with the proxy... one note... the t stands for transfer per the wiki: t : Allow the called user to transfer the call http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gurr Sent: Wednesday, August 04, 2004 3:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: No incoming audio on incoming SIP calls Solved my own problem ... thought I'd record it here for any others who come across it. The problem arises since Asterisk is trying to get out of the way of the media stream, by doing a SIP re-INVITE to get the two ends of the conversation to talk directly. This won't work, as Asterisk is telling the calling party that the IP address to talk to is the private IP address of the softphone on the internal network. Adding canreinvite=no to the softphone's stanza in sip.conf solves the problem. It would be helpful if Asterisk noticed that it's about to tell the other end to use a private IP address ... the ranges are well known, and Asterisk could do an implicit canreinvite=no in this situation. The same problem didn't occur on outgoing calls as the Dial string includes a t for timeout - as per the wiki, this means that Asterisk must stay in the stream to be able to implement this. Of course, the other way to solve this would be to use a proper SIP proxy server which handles RTP stream port forwarding ... something I must get around to. -- David Gurr Congruity Ltd. Hemel Hempstead, UK -Original Message- From: David Gurr [mailto:[EMAIL PROTECTED] Sent: 04 August 2004 14:05 To: [EMAIL PROTECTED] Subject: No incoming audio on incoming SIP calls Now this is really frustrating. Everything was working fine, and now it isn't ... I don't think I've changed anything that would affect this, but I guess you never can be too sure. My setup is as follows: SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall box. This is all on the internal network. Asterisk then dialing out through various means - SIP to Stanaphone, FWD, Gossiptel and PSTN via an X100P. For incoming calls, an 0870 number from CallUK routes to my FWD account, and an 0870 number from Gossiptel routing to my Gossiptel account. Outbound calls all work fine ... I get audio in both directions, no problem. Incoming calls on either 0870 number connect fine, and audio goes from the softphone to the caller, but not the other way ... I hear no audio on the softphone from the caller's phone. I'm getting no alerts from my firewall that it's dropping anything. I know my way around packet sniffers, but I don't know what to look for here. What should the inbound audio packets look like? Thanks -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ 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] problems with'#' transfer after hold...
The patch presents a dtmftimeout option in features.conf that deals with this issue. It defaults to half a second, but can be set to longer than that if necessary... If the length of time between # presses is longer than this value, then * assumes that it's NOT a transfer and sends the DTMF tone to the other end... if however the length is less then you hear Allison say Transfer and everything goes as you'd expect... -Chris - Original Message - From: Wayne [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 04, 2004 11:40 AM Subject: Re: [Asterisk-Users] problems with'#' transfer after hold... Hi, (please be gentle - still learning :)) This may be similar to a problem I had today... Calling an automated phone system wanted me to press # to confirm a number - obviously * treats this a transfer - and relevant prompts were played... If you wait for this to 'time out' (or do an invalid extension) no more DTMF tones are sent down to the called number (I did a test calling to a mobile afterwards) - even a # doesn't provoke the correct response. A thought that came to mind on this was - how do you actually send down a # when requested by an automated service as * treats this as its own. But - looks like the question has been answered by Chris' reply with this ## patch :) - still - I dont know if that would fix the 'if you time out a transfer and then try to send DTMF tones they dont actually get sent' problem. So in answer to Chris question - I dont have this patch installed. Thanks Wayne. Stephen Hon wrote: Hi.. Has anybody been experiencing any problems with transfers using # to transfer after taking a call off of hold? Transfers using the # and music on hold work fine by themselves. However, when we place somebody on hold we can no longer use the # to transfer. This is a problem since we use the # button to park calls. So, say a call comes in, the operator is on a call already, places call on hold and answers the new call, places new call on hold, resumes old call and tries to transfer using the # button it wont work, itll just play the DTMF tone for the # button. At first, I thought somewhere along the line the Tt options must be messed up in a dial command somewhere.. but I double checked everywhere and ensured that I was enabling transfers. Does anybody have any suggestions? Thanks, Steve ___ 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] Re: Auto-attendant with an IP trunk
Josh:
The configuration I have for the extensions.conf is the following:
[macro-oneline]
exten = s,1,Dial(${ARG1},20)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
[from-sip]
ignorepat = 9
exten = _91XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91XX,2,Playback(invalid)
exten = _91XX,3,Hangup
exten = s00123456,1,Wait,2
exten = s00123456,2,Answer
exten = s00123456,3,DigitTimeout,5
exten = s00123456,4,ResponseTimeout,30
exten = s00123456,5,Background(welcome-mainmenu)
; the welcome menu says press 1 for sales, 3 for support, 9 for
; directory and press the extension at any time to be directly
; connected. It plays this greeting when the caller calls the voice
; -pulse number and will also goto sales etc. on user input. the
; problem is when the user tries to enter the extensions i.e. 2000,
; 2001, 2002 at this point
exten = 1,1,Goto(sales,s,1)
exten = 3,1,Goto(support,s,1)
exten = 9,1,Directory(from-sip)
exten = 0,1,Goto(from-sip,2000,1)
exten = 2000,1,Macro(oneline,SIP/2000);
exten = 2001,1,Macro(oneline,SIP/2001);
exten = 2002,1,Macro(oneline,SIP/2002);
exten = 2003,1,Macro(oneline,SIP/2003);
exten = 2004,1,Macro(oneline,SIP/2004);
exten = 2005,1,Macro(oneline,SIP/2005);
exten = 2006,1,Macro(oneline,SIP/2006);
exten = 2007,1,Macro(oneline,SIP/2007);
exten = 2008,1,Macro(oneline,SIP/2008);
exten = 2009,1,Macro(oneline,SIP/2009);
exten = 2010,1,Macro(oneline,SIP/2010);
exten = 2011,1,Macro(oneline,SIP/2011);
exten = 2012,1,Macro(oneline,SIP/2012);
exten = 2013,1,Macro(oneline,SIP/2013);
exten = 2014,1,Macro(oneline,SIP/2014);
From: [EMAIL PROTECTED]
Date: 2004/08/04 Wed PM 01:47:27 CDT
To: [EMAIL PROTECTED]
Subject: Auto-attendant with an IP trunk
Hi:
I am trying to setup a simple auto attendant with Asterisk using SIP extensions. I
have an IP trunk to voicepulse and my outgoing calls go over that.
I can also receive calls on that voicepulse trunk and want it to an auto attendant.
Everything works except on the following:
- one of the options is to allow the caller to press the extension that they would
like to be connected to. I have extensions from 2000 - 2010. What happens is that
Asterisk jumps out at the first digit '2' and says that nothing found with '2'. It
doesn't even read the rest of the digits '000'.
I expect this to be a basic PBX function and I am sure I'm missing something. Any
help would be greatly appreciated.
Regards,
Anil
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Re: [Asterisk-Users] Gafachi?
I like gafachi... i have had no problems AT ALL with them. I havent had any with nufone either... broadvoice on the otherhand (sip)... On Wed, 4 Aug 2004 14:05:16 -0500, Chris Foster [EMAIL PROTECTED] wrote: I use Gafachi as well. They have killer international rates. On Wed, 4 Aug 2004 11:08:56 -0500, Charles Ellis [EMAIL PROTECTED] wrote: Luke, I have used them and have been very happy with the service. They are the only ones I have found that seem to be able to process a call from Firefly that goes through 2 * servers. Nufone and Voicepulse are not able to process it - I think it is a firefly problem not a Nufone or Voicepulse problem, since everything works fine if I use IAXPhone. Anyway, Gafachi has worked well for me, and I do not think you need to register unless you are receiving incoming phone calls. Charles -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luke Catranis Sent: Tuesday, August 03, 2004 9:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Gafachi? Anybody use them... I signed up for $20 to see how there system works.. They're at $.02 per minute for US Termination and their other ITX rates aren't too shabby. Sadly my IAX registration is rejected... maybe a glitch, wondering if anyone's had a similar issue. Luke ___ 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] Using Cisco SIP Phones with Asterisk
Are they still hurdles using Cisco phones with asterisk as mentioned at http://www.voip-info.org/wiki-Cisco+Phones ? We are looking for some cisco phones to test with. Gary ___ 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] Rodopi Billing
Mostly CLEC stuff like CDR imports for specific ILECS LD carriers as well as some ISP stuff like redirecting past due accounts to a payment page as well as any other stuff we may need. We plan to offer it to other service providers as a ASP model and for purchase. Regards, Gary WireBill looks interesting. You mentioned that you are using the source code to build your own platform, but how does it hold up on its own? Can I ask what it can't do that requires you to build your own? Thanks, - Darren On Wed, 2004-08-04 at 08:14, Gary Carr wrote: That sigh will turn to cursing after a couple of months. We currently use Rodopi, have for 10 years but the inflexability is too much to deal with anymore so we are moving away from it. To what? I am also a cursed Rodopi owner. :-( Tom We bought the source code to wirebill and are building our own platform. Gary ___ 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] Snom 200 Programmable Keys
I would like to use one of my Snom 200's 5 programmable keys to park calls. I am using image SIP 2.04g. I have tried a variety of combinations and have come to the conclusion that: 1) On the Key Mappings administration page, I must select the Transfer under the Break Keys option box to be able to successfully transfer calls using the Transfer button. 2) Since my parking extension is 700, I have tried placing 700 in the Number column and tried all the combinations of (Line, Destination, Intercom, Park Orbit, Voice Recorder). Once I have used the Save button to save my change, it changes the number to SIP:[EMAIL PROTECTED]. However, I have no luck in parking the call using the programmable key. According to the documentation, you can use DTMF in H.323 mode on the Snom's, but this does not seem to be an option under SIP mode. Anyone had any luck using the programmable keys for anything but transfering/calling sip url's? -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 [EMAIL PROTECTED] www.adstelecom.com ___ 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] CAC AB1 and Asterisk
On Wed, 2004-08-04 at 15:26, Ronan wrote: Any ideas on this? Have patience is the first idea. While there is a large subscriber base, there isn't a large base of knowledgable users with that hardware. You haven't let enough time pass to be sure people with the knowledge have seen it. I would suspect that you have a signalling problem. Configs would have to be shared for that to be diagnosed though. On Wed, 2004-08-04 at 01:06, Ronan wrote: Hi all, Don't know if anyone can help me. We just set up a CAC Access Bank 1 with Asterisk. Everything works great except, when we ring a Zap interface, the analog phone does not actually ring. The light blinks, and if you answer it you are connected to the person, but the actual phone does not ring. Using the Test option for the channel does make the phone ring, but not during an actual call. Has anyone had this prob before. I saw something about it on a list from back in 2002, but no answer. :( Any help would be appreciated. Thanks, -Ronan ___ 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 -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Get MWI from Telco's voicemail
id call telco and deactivate voicemail and use asterisk VM... more flexible... also if they need to check voicemail when out, give them a dial number (voip from a company like nufone, or land line) that calls into app VoicemailMain. On Wed, 04 Aug 2004 16:37:32 -0400, Seth Remington [EMAIL PROTECTED] wrote: On Wed, 2004-08-04 at 14:21, Scott Petersen wrote: Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco. Maybe I am misunderstanding you but why does this force you to use telco voice mail instead of * voice mail? You can also free that third line up for voice if you use faxdetect. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Integrating an old PBX with Asterisk
Hi all, I was thinking about integrating an old PBX with Asterisk and I was wondering some possible configurations. The question is: which is the best way to let the 2 systems interact ? I can imagine some possible scenarios: - scenario 1: I want to use other then old PBX terminations (ie I have to link the 2 systems with some internal number line) In this scenario I could think to give each user a dedicated old line number from old PBX to a 'dedicated' port of a TDM card. Pros: easy configuration (one - to - one mapping), no old PBX configuration changes, users with new SIP phone can still mantain their old extension. Dis: expensive (one TDM card each 4 ext), not scalable (2 limits:free extension on the old PBX and PCI slots in the * server to add TDM cards), when I receive a call from a old extension and I want to forward it to another old PBX extension I am actually using 2 lines between * and the old PBX. - scenario 2:I want to link the 2 PBX with a trunk of n linesnd use an arbitrary number of SIP phones being able to have # of SIP phones then # of lines. Pros: less expensive then scenario 1 because the number of lines I have to use between * and old PBX is based onblock probability I choose to have, more scalable for the same reason, virtually no limit to SIPextension number Dis: same call transfer problem of above, if the old PBX doesn't support some sort of DID between its extension I have to tell * to answer the line and then to ask the required extension, configuration changes to old PBX... I know that probably the best way should be to add a digital card to old PBX and havea trunk between two systems, but the PBX is really old and I'm not sure I can still find an expansion card. Any suggestion or tip ??? thanks marco ---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.732 / Virus Database: 486 - Release Date: 29/07/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 29/07/2004
Re: [Asterisk-Users] RE: No incoming audio on incoming SIP calls
On Wed, 2004-08-04 at 14:56, David Gurr wrote: Solved my own problem ... thought I'd record it here for any others who come across it. The problem arises since Asterisk is trying to get out of the way of the media stream, by doing a SIP re-INVITE to get the two ends of the conversation to talk directly. This won't work, as Asterisk is telling the calling party that the IP address to talk to is the private IP address of the softphone on the internal network. Adding canreinvite=no to the softphone's stanza in sip.conf solves the problem. It would be helpful if Asterisk noticed that it's about to tell the other end to use a private IP address ... the ranges are well known, and Asterisk could do an implicit canreinvite=no in this situation. What if both phones are on the private net? I'm sure something is being worked on. The same problem didn't occur on outgoing calls as the Dial string includes a t for timeout - as per the wiki, this means that Asterisk must stay in the stream to be able to implement this. t and T are for transfer, not timeout, case denotes which end can transfer. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Get MWI from Telco's voicemail
On Wed, 4 Aug 2004, Scott Petersen wrote: [snip] What I am seeing is an event every half hour exactly, on each of the two voice lines. This causes the simple switch to kick in and ring the extensions. Of course there is no one there. I have put a workaround in [snip] Since these events happen every half hour and only on the lines that have voicemail I am very confident that it is the telco sending a trigger to turn the MWI on the phones either on or off. I really don't want to have to try and find out from the telco as their support is much, much, much, less than knowledgeable or helpful. What I am I found that when I'm dealing with a technical issue on POTS, I get the best results by personally visiting my CO and hoping a nice person answers when I ring the bell. For example, I finally got my DSL to work through this approach: it was up and down often, apparently related to ambient outdoor temperature. I got to know the (then US west, now qwest) DSL tech who handled that CO and worked with him directly to resolve the issue. When he could see that he'd need to spend more than just a few minutes on a job, he'd take my phone number and tell me to call the 800-number service people. When they generated the work request and he received it, he'd call me directly to find out what the real problem is and get it addressed. So what I'm getting at is that you might find it advantageous to at least try talking with techs in your particular CO. They're much more likely to understand what you're asking about and to know the answer than a 800-number support rep. Just make sure you're nice and avoid taking more than just a few minutes of their time. Greg ___ 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] PSTN Access Providers for Asterisk
On Wed, 4 Aug 2004, William R. Lorenz wrote: I'm looking for U.S. providers that will provide access to the PSTN and allow me to easily use my Asterisk box with their services. I would prefer a provider that supports number portability, so that I can park my existing cell number on their network and later move it again, but I'm open to doing some funky stuff with call forwarding if I have to do that. http://www.voip-info.org/wiki-VOIP+Service+Providers Can anyone provide their recommendations or experience in using a VoIP provider, as opposed to a LEC, to provide Asterisk with PSTN access? After you go through the providers listed in the page above, reviewing each of their coverage areas, features, use policies, etc, you'll probably have narrowed it down to just a few who could meet your particular needs. At that point you can use google or another list archive search tool to find praises, rants, and probably config examples, for the ones you are interested in. Greg ___ 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
