[Asterisk-Users] Directory App Weirdness
I noticed tonight, when doing a demo of the Directory app, that something mighty odd is going on. I have one Zap FXS channel and a SIP channel (Grandstream B101). When I invoke that app on the Zap phone things work normally. When I invoke it from the GS phone, the CLI shows that it is playing the intro, but instead I get a consistent ring signal, and then it cuts off. This is completely reproducible. The GS phone works just fine for incoming and outcoming calls of all sorts. Is I doing something nutty or have I tickled a bug? Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SwissVoice MGCP IP10S
At 23:49 31-10-2003 +0100, you wrote: Hi! MGCP works on IP basis, it has no userid's or passwords. Ouch - that means MGCP and NAT w/ dynamic IP (of the router) is a No-No? Correct. Use IAX :) Florian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO modules for TDM400P?
Any details yet? -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] msn messenger
At 01:43 1-11-2003 +0300, you wrote: Is msn messenger capable of using asterisk as it's gateway? Yes, provided you are using MSN 4.7, and not 5.0 or higher. Configure the Communications Service under the Options/Accounts pane. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi, At 05:03 30-10-2003 +0300, you wrote: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK your device is not registered on * , [] name in mgcp.conf must be exactly as gw name in your case you have configured gw-name as 'ip10' in mgcp.conf but on your device it is '[192.168.0.5]' change it on device to ip10 or in * to [[192.168.0.5]] Actually, if we are talking about swissvoice phones then I must say I have not needed this. By the way, the exact gateway name is 192.168.0.5, without brackets (see log above). So this still does not explain why its not talking. I get the idea Asterisk is simply not writing anything back on the port to respond to the request. Are you up to date with CVS code ? Could you try and TCPDUMP to see what is communicated between Asterisk and the phone ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
Ray Burkholder wrote: Does any one else have problems with huge, random silence breaks between an X-Lite and Cisco 7960 SIP phone? Both are running g.711. Softphone to/from softphone works, softphone to/from iax2 works, iax2 to/.from cisco phone works. However, voice as heard on X-Lite is just fine from Cisco, but voice as heard on Cisco from X-Lite has random silent breaks of one or two or three second duration on a very regular basis. Any ideas on how to get rid of the random silent breaks? Ray Burkholder [EMAIL PROTECTED] _http://www.oneunified.net_ 704 576 5101 By default X-Lite now has silence supression turned on.. Go to Advanced System Settings Audio Settings Silence Settings and change Transmit Silence to Yes.. See if that helps.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1 out of 0
Does any one know what below means? - DEBUG[6151]: File chan_sip.c, Line 4904 (handle_request): Check for res for 2298 DEBUG[6151]: File chan_sip.c, Line 973 (find_user): Call from user '2298' is 1 out of 0 --- I presume it is something to do with codec. It started happening since I have changed allowed codec for sip . Also, now extensions can not place calls. Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk FAQ
I'll see what I can do to upgrade the speed of www.voip-info.org Traffic has been going up as it gets more popular. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Michael Wood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 6:44 AM Subject: Re: [Asterisk-Users] asterisk FAQ It has been extremely slow for me too. Regards, Mike On Fri, 31 Oct 2003 15:46:49 +0100, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote : I just went there. Do they share a single isdn B channel with 50 other servers? it was sloow. I'll put it there, eventually On Fri, 2003-10-31 at 15:21, Rich Adamson wrote: Roy, I've started to write an FAQ for asterisk, available here: http://asterisk.pronto.tv/faq.php Please help me fill it up with the good stuff :) Why don't you put it here: http://www.voip-info.org/tiki-index.php and folks can updated/edit online? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pcphoneline
Yes, we have! Nice device, works fine on public IP but behind NAT it has problems. PCphoneline are sorting out NAT problems as far as I know. Ta Senad
RE: [Asterisk-Users] 1 out of 0
Found the answer. It was not codec, but instead missing [ in local context. Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NAT router and off-premise SIP audio problem
Our network is connected to a cablemodem using a dynamic DNS service to resolve our address.The Asterisk server has been alternately set up behind a NATrouter andwithout a NAT router -- that is, with two NICs, one of which isproviding NAT to the rest of the network; the office SIPsare behind that with static private IP addresses. Off-premise SIPs are all behind simple NAT routers. Off-premise SIPs have been able to receive calls from and make calls through the PSTN. No problem. Calls between on-premise SIPs, not a problem. Calls between off-premise SIPs and any other SIPs connected to the server are a problem... they ring up but no audio is passed in either direction. SIP.CONF has NAT=YES. We presume that a dedicated IP address for the Asterisk server would resolve this but we would like to avoid the extra expense. What are we missing? TIA. Jim Greenfield
Re: [Asterisk-Users] NAT router and off-premise SIP audio problem
Jim Greenfield, Computer Troubleshooters Metro NY/NJ wrote: Our network is connected to a cablemodem using a dynamic DNS service to resolve our address. The Asterisk server has been alternately set up behind a NAT router and without a NAT router -- that is, with two NICs, one of which is providing NAT to the rest of the network; the office SIPs are behind that with static private IP addresses. Off-premise SIPs are all behind simple NAT routers. Off-premise SIPs have been able to receive calls from and make calls through the PSTN. No problem. Calls between on-premise SIPs, not a problem. Calls between off-premise SIPs and any other SIPs connected to the server are a problem... they ring up but no audio is passed in either direction. SIP.CONF has NAT=YES. We presume that a dedicated IP address for the Asterisk server would resolve this but we would like to avoid the extra expense. What are we missing? TIA. Jim Greenfield Try adding canreinvite=no in the config of the remote phones.. This will force the audio path through Asterisk.. Also I would suggest that you NOT put the Asterisk server behind NAT.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT router and off-premise SIP audio problem
Look at RTP (/etc/asterisk/rtp.conf) packets, and its firewall configuration.
Re: [Asterisk-Users] asterisk FAQ
James, You can make mirror of your site at our facilities. To support Asterisk community we can host mirror of your site, or make it primary hosting whatever is more convinient for you. We can do it duting this weekend. Let me know Alexander Unofficial Asterisk Forums URL : http://asterisk.xvoip.com Registration is : http://asterisk.xvoip.com/profile.php?mode=register New XVOIP network , get your +1 777 number today. [EMAIL PROTECTED] - Original Message - From: James H. Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 01, 2003 6:25 AM Subject: Re: [Asterisk-Users] asterisk FAQ I'll see what I can do to upgrade the speed of www.voip-info.org Traffic has been going up as it gets more popular. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Michael Wood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 6:44 AM Subject: Re: [Asterisk-Users] asterisk FAQ It has been extremely slow for me too. Regards, Mike On Fri, 31 Oct 2003 15:46:49 +0100, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote : I just went there. Do they share a single isdn B channel with 50 other servers? it was sloow. I'll put it there, eventually On Fri, 2003-10-31 at 15:21, Rich Adamson wrote: Roy, I've started to write an FAQ for asterisk, available here: http://asterisk.pronto.tv/faq.php Please help me fill it up with the good stuff :) Why don't you put it here: http://www.voip-info.org/tiki-index.php and folks can updated/edit online? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is iaxtel.com down for 700 #'s?
I've not been able to register with iaxtel.com for the last couple of days. Is anyone else seeing this, or did I miss something? Same here I have not been able to get any calls nor do any calling through them! Mark indicated yesterday that digium changed the IP address of the iaxtel box that we're all registering with. It was changed from 12.37.165.130 to 69.73.19.178, which is now working. I'm somewhat confused with the caching though. My iax.conf had: register = npi:[EMAIL PROTECTED] in it (which worked fine prior to their IP change). On the same * machine, if I ping iaxtel.com now, the dns resolves to 69.73.19.178 however * continues to try 12.37.165.130 regardless. I've not found any config file or database (as yet) that is maintaining that old IP address. I changed the register statement to: register = npi:[EMAIL PROTECTED] and the iax connection occurs as expected. If I change that IP addr back to iaxtel.com (and restart *), it continues to work indicating the destination is cached somewhere that I'm missing. Anyone know where that is? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound PSTN Calls
Hi All, Is it possible to show which line a call has come in on in *. My scenario is 8 incoming lines, 6 lines are trunked to one number and the other 2 are individual lines. I would like to pass the trunked lines to one set of extensions, and the other lines to two other set of extensions. Also with the outgoing calls I would like to send the call out on the correct line for the extension group. I need to get this clear in my head before I go to a friend of mine who is looking for a new switch and tell him that * can do the job. The total system for a start will consist of, 8 PSTN (analogue lines and 25 extensions, with the possibility of expansion for remote SIP phones globally). Thanks in anticipation for any advice/recommendations. Dave PS. This may appear again as it was held for moderation as I had a picture attached. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax vs iax2 connections
Been meaning to ask this for some time... no big deal, but curious. I have a single register statement in my iax.conf for iaxtel like: [general] port=5036 register = npi:[EMAIL PROTECTED] snip However, when I restart *, I see: Registered to '69.73.19.178', who sees us as 205.221.193.101:5036 Registered to '69.73.19.178', who sees us as 205.221.193.101:4569 which suggests both iax and the iax2 protocols in use. The 'iax show registry' and 'iax2 show registry' confirms it as well. I only see this with iaxtel.com links, not with other iax links. If I remove the register statement, neither iax or iax2 is active. Is this something specific to the iaxtel.com box, something that I'm missing in my config, or what? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is iaxtel.com down for 700 #'s?
On Saturday 01 November 2003 09:21, Rich Adamson wrote: I'm somewhat confused with the caching though. My iax.conf had: register = npi:[EMAIL PROTECTED] in it (which worked fine prior to their IP change). On the same * machine, if I ping iaxtel.com now, the dns resolves to 69.73.19.178 however * continues to try 12.37.165.130 regardless. I've not found any config file or database (as yet) that is maintaining that old IP address. I changed the register statement to: register = npi:[EMAIL PROTECTED] and the iax connection occurs as expected. If I change that IP addr back to iaxtel.com (and restart *), it continues to work indicating the destination is cached somewhere that I'm missing. Anyone know where that is? Yep, if you look at struct iax2_peer, you'll find a member named addr. That address is resolved at reload time. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
rnc Info Lists wrote: Hi, -Original Message- The portion of extensions.conf is: exten = 3001,1,Dial(MGCP/aaln1,20) exten = 3001,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20) Or aaln/1@ip should do just fine. However this doesn't explain why there is no dialtone on the phone.. Oh, one thought: Did you set your toneconfiguration to Europe or US ? If you choose custom you need to configure it another way... Florian Update: I changed the tone config to USA to match Asterisk. No change. I did notice that when I booted up everythign tonight that the MGCP SHOW ENDPOINTS now shows: Gateway 'ip10' at 0.0.0.0 (Dynamic) -- 'aaln/[EMAIL PROTECTED] in 'from-sip' is idle In the messages at start up there is: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) -- MGCP Auditing endpoint aaln/[EMAIL PROTECTED] for hookstate [chan_iax2.so]NOTICE[163851]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.0.5' (and thus its endpoint 'aaln/1') does not exist -- Setting hookstate of aaln/[EMAIL PROTECTED] to ONHOOK MGCP DEBUG shows the below lines repeating every couple of seconds: from 192.168.0.5:2427MGCP read: RSIP 1375 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 RM: restart from 192.168.0.5:2427Verb: 'RSIP', Identifier: '1375', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Still no dialtone and not able to send or receive calls. Evidently there is a problem finding the phone. I can ping it from the Asterisk server so isn't a raw IP issue. On the phone there is the message Waiting for call manager Additional ideas are appreciated. Will keep plugging away at it. in sending you my mgcp.conf file, my ip10s mostly working fine... regards Marian ---mgcp.conf- [general] port = 2427 bindaddr = 192.168.1.253 [192.168.1.92] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=192.168.1.92 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = John 92 line = aaln/1 [192.168.1.91] threewaycalling=yes transfer=yes callwaiting=no callwaitingcallerid=no host=192.168.1.91 context=local nat=no ;dtmf=inband disallow=all allow=g711 allow=alaw callerid = Mary 91 line = aaln/1 Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
Could be comfort noice ? Check for PT 13 or 19 Michael Ray Burkholder wrote: Huge silence breaks between Cisco 7960 phone X-Lite Does any one else have problems with huge, random silence breaks between an X-Lite and Cisco 7960 SIP phone? Both are running g.711. Softphone to/from softphone works, softphone to/from iax2 works, iax2 to/.from cisco phone works. However, voice as heard on X-Lite is just fine from Cisco, but voice as heard on Cisco from X-Lite has random silent breaks of one or two or three second duration on a very regular basis. Any ideas on how to get rid of the random silent breaks? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses dangerous content at One Unified and is believed to be clean.
[Asterisk-Users] Outbound SIP Provider Nikotel Ringback
Title: Huge silence breaks between Cisco 7960 phone X-Lite I hear no ring back tone when I place a call using Nikotel as my outbound provider to a PSTN telephone number. When I call to a Vonage telephone number I get a ring back tone. Any suggestions as to why I do not receive ring back tone when calling PSTN numbers?
RE: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
Title: Message By default X-Lite now has silence supression turned on.. Go to Advanced System Settings Audio Settings Silence Settings and change Transmit Silence to "Yes".. I played with this. Still problems. Where do I check for PT 13 or 19? Could be comfort noice ? Check for PT 13 or 19 Does any one else have problems with huge, random silence breaks between an X-Lite and Cisco 7960 SIP phone? Both are running g.711. Softphone to/from softphone works, softphone to/from iax2 works, iax2 to/.from cisco phone works. -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
Re: [Asterisk-Users] problem DG-104S not call
Javier Rios wrote: hello you can help me with a problem I have dlink DG-104S already and this registered in asterisk but not to call... between in ports you can help with an example the configuration me of mgcp.conf extensions.conf ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.100.2 disallow=all allow=g729 allow=alaw ; tos=lowdelay [noc] host = 192.168.100.10 context = noc musiconhold=default pickupgroup=0 callgroup=0 cancallforward=1 transfer=1 pickupgroup=0 callgroup=0 mailbox=121 callerid=Vitaly Bulakhov 121 canreinvite=0 line = aaln/1 mailbox=122 callerid=Smena 122 canreinvite=0 line = aaln/2 mailbox=123 callerid=123 123 line = aaln/3 mailbox=124 callerid=124124 line = aaln/4 extention.conf ; [noc] exten = 120,1,Dial,MGCP/aaln/[EMAIL PROTECTED]MGCP/aaln/[EMAIL PROTECTED]MGCP/aaln/[EMAIL PROTECTED]MGCP/aaln/[EMAIL PROTECTED] exten = 121,1,Dial,MGCP/aaln/[EMAIL PROTECTED] exten = 122,1,Dial,MGCP/aaln/[EMAIL PROTECTED] exten = 123,1,Dial,MGCP/aaln/[EMAIL PROTECTED] exten = 124,1,Dial,MGCP/aaln/[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Soundpoint IP600
Default User Password is 123 Default Admin Password is 456 -sb -Original Message- From: Roman Pelikh [mailto:[EMAIL PROTECTED] Sent: Friday, October 31, 2003 11:54 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Polycom Soundpoint IP600 Does anyone have the Admin password for the phone in order to change configuration Roman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
Hi! However, voice as heard on X-Lite is just fine from Cisco, but voice as heard on Cisco from X-Lite has random silent breaks of one or two or three second duration on a very regular basis. Any ideas on how to get rid of the random silent breaks? X-Lite (build 1082 and possibly later) and choppy sound: In X-Lite go to -- Advanced Setup -- Audio Settings -- Silence Settings -- set Transmit Silence to yes to solve this issue. P.S.: Looks like I have to post this once a day now. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SwissVoice MGCP IP10S
Hi! in sending you my mgcp.conf file, my ip10s mostly working fine... Could you explain mostly in your sentence, and maybe - if you can - give quick overview of Grandstream vs. SwissVoice (except for the pending SIP implementation, of course)? Thanks, Philipp! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk: Reloaded
At 10:23 PM 10/31/2003, Bryan Nolen wrote: System execute asterisk -rx reload ? Yes, correct. --Ernest -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Saturday, 1 November 2003 5:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk: Reloaded Hello, Pretend I had a Perl script that did something to an Asterisk conf file... How can I [from Perl] ask Asterisk to reload? ;) Ben __ Benjamin Wakefield [EMAIL PROTECTED] http://www.dcsi.net.au/ DCSI - We do Internet. 64 Queen Street Warragul, VIC 3820 AU Ph: (+61) 1300 665 575 Fx: (+61) 1300 556 595 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO modules for TDM400P?
i m interested too? [EMAIL PROTECTED] wrote: Any details yet? -Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound PSTN Calls
At 07:42 AM 11/1/2003, you wrote: Hi All, Is it possible to show which line a call has come in on in *. Yes, absolutely. In asterisk each line is a channel. The channel information is VITAL to the call and is available (and used) everywhere in asterisk. Channels look like this: ZAP/1-1, which means Zaptel card, line 1, call 1. My scenario is 8 incoming lines, 6 lines are trunked to one number and the other 2 are individual lines. I assume you mean that they are six analog lines set up with a rollover. If you use eight FXO cards, then each line is a separate asterisk channel. Configure each channel with a different default context in the zaptel.conf file. I believe the same is true if you use a channel bank, in which case each T1 will be 23 channels (1-23, 24-47, etc) context=default signalling=fxs_ks channel=1 channel=2 context=notdefault channel=3 The total system for a start will consist of, 8 PSTN (analogue lines and 25 extensions, with the possibility of expansion for remote SIP phones globally). If you use a VoIP gateway, then you need to configure the gateway with a different user for each group of lines. I can't help you with this, as it depends on the gateway, but I'm told it's possible (and I'll be doing it myself soon). --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
At 08:54 AM 11/1/2003, you wrote: P.S.: Looks like I have to post this once a day now. You should post this (or I'll do it for you, with permission, as I already have an account) on the Asterisk wiki at www.voip-info.org. You might still have to post, but at least it will be out there... Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on remote end when using NuFone
At 10:42 AM 10/31/2003, you wrote: I have the same problem and it was solved setting: # Uncomment for aggressive residual echo supression under # MARK2 echo canceller # KFLAGS+=-DAGGRESSIVE_SUPPRESSOR This creates a very nasty click when I talk into the SNOM (but no long echo!). It's like having a conversation with a compulsive interjector who never finishes his sentences :) Do you have this problem? If so, do you recall how you solved it? Thanks, --Ernest in the makefile of zaptel and recompiling. miklos - Original Message - From: Ernest W. Lessenger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 4:21 PM Subject: [Asterisk-Users] Echo on remote end when using NuFone I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a separate issue, I am hearing a bad echo when using my Digium X100P to connect to the PSTN. I've tried tweaking the tx/rx gain to no real effect. I've also tried changing the volume on the SNOM phone, changing the codec to g711u, and decreasing the packet size. Any other things to try? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Huge silence breaks between Cisco 7960 phone X-Lite
However, voice as heard on X-Lite is just fine from Cisco, but voice as heard on Cisco from X-Lite has random silent breaks of one or two or three second duration on a very regular basis. Any ideas on how to get rid of the random silent breaks? X-Lite (build 1082 and possibly later) and choppy sound: In X-Lite go to -- Advanced Setup -- Audio Settings -- Silence Settings -- set Transmit Silence to yes to solve this issue. Sorry, this didn't fix the problem. I put my microphone up to a continuous music source, and the drop outs still occur. I checked, the auto gain controls are off. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
[EMAIL PROTECTED]
[Asterisk-Users] FWD connection
Title: Leterhead Hi All, I have a FWD number and wish to connect it to Asterisk to receive my FWD calls. How I do? Is it a register in sip.conf or iax.conf? Regards Dave Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED]
Re: [Asterisk-Users] FWD connection
As far as I know they do only SIP. If your Asterisk box is behind a NAT firewall then you probably will have problems. Hi All, I have a FWD number and wish to connect it to Asterisk to receive my FWD calls. How I do? Is it a register in sip.conf or iax.conf? Regards Dave html xmlns:v=urn:schemas-microsoft-com:vml xmlns:o=urn:schemas-microsoft-com:office:office xmlns:w=urn:schemas-microsoft-com:office:word xmlns=http://www.w3.org/TR/REC-html40; head meta http-equiv=Content-Type content=text/html; charset=us-ascii meta name=ProgId content=Word.Document meta name=Generator content=Microsoft Word 9 meta name=Originator content=Microsoft Word 9 link id=Main-File rel=Main-File href=cid:[EMAIL PROTECTED] !--[if gte mso 9]xml o:shapedefaults v:ext=edit spidmax=2051/ /xml![endif]--!--[if gte mso 9]xml o:shapelayout v:ext=edit o:idmap v:ext=edit data=1/ /o:shapelayout/xml![endif]-- /head body lang=EN-GB link=blue vlink=purple div style='mso-element:header' id=h1 div cite=mid:Unknown20031101T182611378; p class=MsoHeaderfont size=3 color=black face=Times New Romanspan style='font-size:12.0pt;color:black;mso-color-alt:windowtext'!--[if gte vml 1]v:shapetype id=_x_t75 coordsize=21600,21600 o:spt=75 o:preferrelative=t path=[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@5xe filled=f stroked=f v:stroke joinstyle=miter/ v:formulas v:f eqn=if lineDrawn pixelLineWidth 0/ v:f eqn=sum @0 1 0/ v:f eqn=sum 0 0 @1/ v:f eqn=prod @2 1 2/ v:f eqn=prod @3 21600 pixelWidth/ v:f eqn=prod @3 21600 pixelHeight/ v:f eqn=sum @0 0 1/ v:f eqn=prod @6 1 2/ v:f eqn=prod @7 21600 pixelWidth/ v:f eqn=sum @8 21600 0/ v:f eqn=prod @7 21600 pixelHeight/ v:f eqn=sum @10 21600 0/ /v:formulas v:path o:extrusionok=f gradientshapeok=t o:connecttype=rect/ o:lock v:ext=edit aspectratio=t/ /v:shapetypev:shape id=_x_s1026 type=#_x_t75 style='position:absolute; margin-left:68.4pt;margin-top:.55pt;width:300pt;height:93pt;z-index:1' o:allowincell=f v:imagedata src=cpclear/ w:wrap type=topAndBottom/ /v:shape![endif]--/span/font/p /div /div div style='mso-element:footer' id=f1 div cite=mid:Unknown20031101T182611378; p class=MsoFooterb style='mso-bidi-font-weight:normal'font size=3 color=navy face=Times New Romanspan style='font-size:12.0pt;color:navy; font-weight:bold'![if !supportEmptyParas]nbsp;![endif]o:p/o:p/span/font/b/p p class=MsoFooter align=center style='text-align:center'b style='mso-bidi-font-weight: normal'font size=3 color=navy face=Times New Romanspan style='font-size: 12.0pt;color:navy;font-weight:bold'Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ.o:p/o:p/span/font/b/p p class=MsoFooter align=center style='text-align:center'b style='mso-bidi-font-weight: normal'font size=3 color=navy face=Times New Romanspan style='font-size: 12.0pt;color:navy;font-weight:bold'Company Registration Number: - 03807643.span style=mso-spacerun: yesnbsp; /spanVAT Registration Number: - 734-3363-42o:p/o:p/span/font/b/p p class=MsoFooter align=center style='text-align:center'b style='mso-bidi-font-weight: normal'font size=3 color=navy face=Times New Romanspan style='font-size: 12.0pt;color:navy;font-weight:bold'Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957o:p/o:p/span/font/b/p p class=MsoFooter align=center style='text-align:center'b style='mso-bidi-font-weight: normal'font size=3 color=navy face=Times New Romanspan style='font-size: 12.0pt;color:navy;font-weight:bold'a href=http://www.codepipe.ltd.uk/;http://www.codepipe.ltd.uk/a / a href=http://www.codepipe.com/;http://www.codepipe.com/a / E-Mail: - [EMAIL PROTECTED]/span/font/bb style='mso-bidi-font-weight:normal'font color=navyspan cite= style='color:navy;font-weight:bold'o:p/o:p/span/font/b/p /div /div /body /html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD connection
Title: Leterhead You have to input your info in your sip.conf -- its in your examples Checkout this site for examples www.fnords.org/~eric/asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David J Carter Sent: Saturday, November 01, 2003 1:28 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FWD connection Hi All, I have a FWD number and wish to connect it to Asterisk to receive my FWD calls. How I do? Is it a register in sip.conf or iax.conf? Regards Dave Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED]
[Asterisk-Users] sizing - conference room
dear all gurus, i am looking into setting a fair size conference room system and would be most grateful for any advise, experience, recommendation on the following: 1) what is a reasonable real world max. channels conference room that 1 * server can handle? with what kinda h/w in server? 2) is it possible to have 1 conference room straddle over 2 * servers linked by IAX? if not directly, any workarounds? what kinda performance penalty if any? 3) is it possible to have 2 conference room on 2 * servers linked by IAX with inbound participants from either * going into either conference room? what i mean is which parts drain more processing resource; inbound channels? the conference room? or pretty much 50/50? 4) is there a significant difference in terms of max. conference channels on whether the inbound participants join from a T1 channel as suppose to a SIP channel? 5) if some inbound SIP participants use G.729 codecs whilst others use G.711 either DS0 or SIP, how many G.729 license i should get for the * server? how will this affect the max. channels? 6) if network speed is the only limitation (SIP participants), how many SIP inbound conference channels can a 10 Mbps symmetrical network handle? 7) is an * server suitable for this kinda application? thank you in advance for your input. gracefully yours, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Making a Skinny phone talk to Asterisk
Title: Making a Skinny phone talk to Asterisk I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. How do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure about this as the OS79XX.TXT makes newly arriving phones load the SIP image). Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses & dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] FWD connection
Title: Leterhead Sip.conf [general] register=FWDNUMBER:[EMAIL PROTECTED]/EXTENSION [fwd] type=friend username=FWDNUMBER secret=FWDPASSWORD host=fwd.pulver.com context=YOURINBOUNDCONTEXT extensions.conf [inboundcontext] exten = EXTENSION,1,Dial(SIP/SOMEOTHEREXT) Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED] Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED] Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED]
Re: [Asterisk-Users] Making a Skinny phone talk to Asterisk
Last I checked skinny firmware would try to connect to a host that would resolve to CiscoCM1 bkw On Sat, 1 Nov 2003, Ray Burkholder wrote: I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. How do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure about this as the OS79XX.TXT makes newly arriving phones load the SIP image). Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Making a Skinny phone talk to Asterisk
If it doesn't find a host named ciscoccm1 then it will try to connect to whatever host it got it's DHCP lease from. (assuming it's using DHCP, of course) On Sat, 2003-11-01 at 15:07, Brian West wrote: Last I checked skinny firmware would try to connect to a host that would resolve to CiscoCM1 bkw On Sat, 1 Nov 2003, Ray Burkholder wrote: I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. How do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure about this as the OS79XX.TXT makes newly arriving phones load the SIP image). Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
I can also confirm chan_h323 and g.729 work well to 5300s, but we had to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP which works fine--even to i4l interfaces. On Friday, October 31, 2003, at 01:57 AM, Jeremy McNamara wrote: John Todd wrote: I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be the case that it is the same question as I have seen already posted (with no 'definitive' answer): Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? Yes, g729r8 If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. Others seem to have massive issues with chan_h323 and G.729, but i've dealt a dozen or so 5300s of which I haven't had any trouble whatsoever, with nothing other than the code that is currently in the cvs. However, I have only terminated calls from Asterisk to the 5300, never from the 5300 to Asterisk. If Asterisk is going to be encoding G.729, yes you will need licenses from Digium. Jeremy McNamara P.S. I'm biased and cannot comment about that other driver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD connection
David J Carter wrote: I have a FWD number and wish to connect it to Asterisk to receive my FWD calls. See the Asterisk FAQ at http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ You'll find pointers to several Asterisk - FWD configurations there. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS What to do?
hi fred, i don't know if this question has been already answered... i haven't tested it whit asterisk YET, (i have to) check the following links: http://luxik.cdi.cz/~devik/qos http://www.ibiblio.org/pub/Linux/docs/HOWTO/other-formats/html_single/ADSL-Bandwidth-Management-HOWTO.html and tell me if you have found a solution -- santiago josé ruano rincón administración servidores y servicios de internet red de datos universidad del cauca http://www.unicauca.edu.co/~santiago/llaves/santiago_pub.asc hay 10 tipos de personas, las que entienden binario y las que no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] broadcast voicemail msg ??
How does one send a broadcast message to all voice mail boxes? I want to send a single message to every mailbox on the system informing them of changes, etc. any thoughts ?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quick Question
Apologies if there is a cleanly written and searchable FAQ that I could be directed to. I have no problem to RTFM if I can find the FM... Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... BTW, where would I find a useful FM? David -- David J. Sussman, MBA email: [EMAIL PROTECTED] web:http://www.processdevelopmentgroup.com phone: 248-212-7293 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Question
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... yes, asterisk under rh 9.0 works good. I have 3 systems running with that distro. BTW, where would I find a useful FM? http://www.digium.com/handbook-draft.pdf and several unofficial websites with a lot of infos... search the ML for them Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] broadcast voicemail msg ??
On Sat, 1 Nov 2003, Brian West wrote: OH great idea... feature request on bugs.digium.com? On Sat, 1 Nov 2003, John Brown (CV) wrote: How does one send a broadcast message to all voice mail boxes? I want to send a single message to every mailbox on the system informing them of changes, etc. any thoughts ?? It would be nice to have an administrator mail box. that would be able to: 1. send broadcast messages 2. change a vm users passowrd Also it would be nice to have a key to press to skip the time and date when listen to messages ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Question
At 05:15 PM 11/1/2003, you wrote: Apologies if there is a cleanly written and searchable FAQ that I could be directed to. I have no problem to RTFM if I can find the FM... Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... Asterisk works VERY well under RH9. Be sure to install kernel-sources and keep them up-to-date along with the rest of the system. BTW, where would I find a useful FM? Um, yeah. (1) Search the mailing list archives. (2) Check out http://www.voip-info.org/tiki-index.php?page=Asterisk. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quick Question
Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... I recall in the archives somewhere, and through someone's post earlier today, that there is some sort of problem with RH9 with Zaptel (hardware) drivers and that RH8 is preferred. Asterisk works VERY well under RH9. Be sure to install kernel-sources and keep them up-to-date along with the rest of the system. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Question
There are links to several other Asterisk related sites at the bottom of the page at http://www.fnords.org/~eric/asterisk/ On Sat, 2003-11-01 at 18:26, Brancaleoni Matteo wrote: Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... yes, asterisk under rh 9.0 works good. I have 3 systems running with that distro. BTW, where would I find a useful FM? http://www.digium.com/handbook-draft.pdf and several unofficial websites with a lot of infos... search the ML for them Matteo -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i disconnected a running DID line from our PBX and did a bunch of tests on it and found the following: * line from telco has NO voltage * the port from pbx is supplying the power(voltage) but no dial tone *the moment i disconnected the DID line from the PBX port, an alarm is triggered at the telco CO * i can attach an ordinary analog phone to the PBX PORT, pick up the handset and send (dial) 4 dtmf digits (being the last 4 digits of our DID number), the PBX will bridge me to the appropriate extension phone. * if or when the extension phone picks up, the PBX reverses the polarity on the line what type of signaling should i be using for such a line? many thanks in advance, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quick Question
At 05:03 PM 11/1/2003, you wrote: Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... I recall in the archives somewhere, and through someone's post earlier today, that there is some sort of problem with RH9 with Zaptel (hardware) drivers and that RH8 is preferred. Do you recall what kind of problem? The only problem I have is an annoying echo that I haven't yet gotten rid of. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NetJet Cards
Hello,
I am trying to use 2 netjet cards under asterisk and isdn4linux. I am
having a hard time trying to get them to work in terms of dial out. Does
anyone have a working config I could look at for even one card (tried
that, not much luck either).
When i dial out:
-- Accepting AUTHENTICATED call from 172.16.11.2, requested format =
2, actual format = 2
-- Executing Dial([EMAIL PROTECTED]:5036]/2,
modem/g1/v{EXTEN:1}) in new stack
-- Called g1/v{EXTEN:1}
-- Modem[i4l]/ttyI0 is busy
-- Hungup 'Modem[i4l]/ttyI0'
== Everyone is busy at this time
-- Executing Congestion([EMAIL PROTECTED]:5036]/2, ) in new
stack
== Spawn extension (international, 90412463080, 2) exited non-zero on
'[EMAIL PROTECTED]:5036]/2'
-- Hungup '[EMAIL PROTECTED]:5036]/2'
-- Registered 'menger' (AUTHENTICATED) at 172.16.11.2:5036
I am using devices /dev/ttyI0 for the first card and ttyI1 for the
second card. I have loaded the module up using:
modprobe hisax type=20,20 protocol=2,2 id=HiSax
I am running RedHat Linux 9.0 wth latest kernel from redhat.
Any help apart from buy a capi card would be most appreciated.
Thanks,
Matthew Enger
[EMAIL PROTECTED]
--
Matthew Enger [EMAIL PROTECTED]
Xintegration
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Re: [Asterisk-Users] NetJet Cards
You are missing a $ in front of {EXTEN:1}. It should be ${EXTEN:1}
On Sat, 2003-11-01 at 22:55, Matthew Enger wrote:
Hello,
I am trying to use 2 netjet cards under asterisk and isdn4linux. I am
having a hard time trying to get them to work in terms of dial out. Does
anyone have a working config I could look at for even one card (tried
that, not much luck either).
When i dial out:
-- Accepting AUTHENTICATED call from 172.16.11.2, requested format =
2, actual format = 2
-- Executing Dial([EMAIL PROTECTED]:5036]/2,
modem/g1/v{EXTEN:1}) in new stack
-- Called g1/v{EXTEN:1}
-- Modem[i4l]/ttyI0 is busy
-- Hungup 'Modem[i4l]/ttyI0'
== Everyone is busy at this time
-- Executing Congestion([EMAIL PROTECTED]:5036]/2, ) in new
stack
== Spawn extension (international, 90412463080, 2) exited non-zero on
'[EMAIL PROTECTED]:5036]/2'
-- Hungup '[EMAIL PROTECTED]:5036]/2'
-- Registered 'menger' (AUTHENTICATED) at 172.16.11.2:5036
I am using devices /dev/ttyI0 for the first card and ttyI1 for the
second card. I have loaded the module up using:
modprobe hisax type=20,20 protocol=2,2 id=HiSax
I am running RedHat Linux 9.0 wth latest kernel from redhat.
Any help apart from buy a capi card would be most appreciated.
Thanks,
Matthew Enger
[EMAIL PROTECTED]
--
Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/
BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643
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Re: [Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
Hi Patrick, You are in the UK, right (at least DDI strongly suggests that)? This is the commonest signalling for a DDI line on an analogue pair. The line is behaving just like the main exchange is a telephone. It picks up the line, by applying a 600ohm loop, and dials (with pulses per second or DTMF) into your PBX. Your PBX port is behaving like it is a public exchange, with a phone attached. Electrically, Digium's FXS card should do the job you need, but others will have to tell you whether * has the software features needed to make this work (it should certainly be pretty close). Regards, Steve hkirrc.patrick wrote: as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i disconnected a running DID line from our PBX and did a bunch of tests on it and found the following: * line from telco has NO voltage * the port from pbx is supplying the power(voltage) but no dial tone *the moment i disconnected the DID line from the PBX port, an alarm is triggered at the telco CO * i can attach an ordinary analog phone to the PBX PORT, pick up the handset and send (dial) 4 dtmf digits (being the last 4 digits of our DID number), the PBX will bridge me to the appropriate extension phone. * if or when the extension phone picks up, the PBX reverses the polarity on the line what type of signaling should i be using for such a line? many thanks in advance, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
