[Asterisk-Users] patlooptest output
Does anyone know if patlooptest either doesn't work for fxo/fxs signaled channels or if you have to do it a different way? If I run ./patlooptest /dev/zap/25 60 with a config like: fxsks=25-32 fxoks=33-48 it gives me a bunch of output along the lines of: (Error 4071): Unexpected result, 254 != 255, 11 bytes since last error. (Error 4072): Unexpected result, 0 != 255, 1 bytes since last error. (Error 4073): Unexpected result, 10 != 11, 11 bytes since last error. (Error 4074): Unexpected result, 12 != 11, 1 bytes since last error. (Error 4075): Unexpected result, 22 != 23, 11 bytes since last error. (Error 4076): Unexpected result, 24 != 23, 1 bytes since last error. (Error 4077): Unexpected result, 34 != 35, 11 bytes since last error. (Error 4078): Unexpected result, 36 != 35, 1 bytes since last error. If I change the channels to be bchan/dchan however the test runs without errors. Thanks, --Dan-- ___ 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, Linux, ISDN
HI Mike, 2) I could add an isdn card to the Linux box. This seems to me to be the cleanest solution, I'd make my firewall also be the asterisk server, and hopefully gain some control of tcp flows that way to more highly prioritize voice traffic +apparent simplicity, maybe fax support -s it seems most of the ISDN cards in isdn4linux are not sold in the US, the technology is stagnant, and I'm less than enthused about statements like Any CAPI based ISDN card will work when I'd prefer something like ISDN card XXX tested on an opteron running kernel X.Y.Z, using multi-link ppp and and asterisk, no problems I am using a Traverse NetJet-S card with the ISDN4Linux drivers in Asterisk on my home firewall. I was using ISDN to connect to the 'Net, but I've just -- in the last couple of months -- managed to convice Telstra let me get ADSL provisioned. I decided to keep the ISDN line for voice, so I've got a personal number and a business number coming in the ISDN. The only problem I've had is that I have had absolutely no luck in getting fax support working with the i4l driver and my questions on this list in regards to that have gone unanswered on at least 3 occasions... You can find the Traverse site at http://www.traverse.com.au/. Last time I checked they had a card for the US market (you'd need to email them to ask if the NetSpider works with I4L). HTH, Darryl ___ 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] tdm400p static - out of ideas
On July 8, 2004 02:18 am, [EMAIL PROTECTED] wrote: On Wed, 7 Jul 2004, Ryan Courtnage wrote: Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. Attempts to send outbound calls on any Zap channel will result in hearing a loud 'static' noise on the line. On one occasion the problem actually occurred while someone was on an active call with the PSTN. 25 minutes into the call, this loud static noise occurred, and the call was dropped. Debug log files show nothing unusual. It's obvious that * is unaware that there is any problem with the card. I had a simmilar problem with an FXS card in a Compaq ML330. I would get a power reset message on my server console. Are you sure the noise is coming from the FXO ports? Are you using SIP phones, or ZAP? Using SIP phones. I think the 'static noise' or just a symptom of the card going into this non-working state - or visa-versa. I also believe that the tdm400p card (or driver) is to blame, and not the FXO modules on the card. Aside from your's, I have heard reports of people having similar problems with their cards loaded with FXS modules. I do not get these 'power reset' messages. What did you do to resolve the issue? ___ 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] Small Linux Distro
Does anyone have a current, stripped linux distro which has only asterisk and net drivers? If so do you have it available somewhere? I guess also, my question could be, does anyone know of a small distro, which will run asterisk. When I say small I mean 700Mb Also, anyone got any sites on hand which would point to ways to make linux start up faster? (BTW this is all in aid of making Asterisk boxes, with LCDs and buttons as opposed to keyboard and screen - i will also write an interface for Asterisk to LCDproc, so that it can be controlled from buttons mounted next to the screen, and make it GPL). Any help, pointers greatly appreciated. Matt ___ 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: What is the difference between queeu members and queue agents
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 07 July 2004 09:51 pm, Constantine Filin wrote: greetings I've read a lot on http://www.voip-info.org/wiki-Asterisk, but I cannot understand what the difference between a queue member and queue agent is. Agents would be people who's job it is to answer calls. An agent logs in=20 indicating that he's now available to take calls. Asterisk then sends calls= to each agent as they are free to take a call. Members are those calling in and piling up in the queue, waiting for an agent to answer. Oops! Members are also agents. The same thing... - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA7LlwljK16xgETzkRAnPXAJ4rE7Kr/W1L5vN6EugUnPre+053ugCdEQI0 9eTMQ8MGcWNFY28Lfw2arDQ= =0vSQ -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] Perl library to manipulate 'ini files'
iH from cpan, i use this module a lot !! http://search.cpan.org/~wadg/Config-IniFiles-2.38/IniFiles.pm - hcir On Jul 7, 2004, at 12:40 PM, kaiduan xie wrote: Hi, all, Can anyone tell where can I find the perl library for manipulating 'ini files'? Thanks, ___ 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 hackers gut Caller ID
what do you mean not quite right??? if the clid is supposed to be blocked then don't send it. if the far end is a law enforcement or emergency agency then the clid is NOT supposed to be blocked!! if the originating switch had the ability to send or not send, problem solved for voip providers from getting a blocked clid - hcir On Jul 7, 2004, at 1:47 PM, Steve Kennedy wrote: On Wed, Jul 07, 2004 at 07:57:36AM -0800, rich allen wrote: this is really simple, companies like Nortel, Lucent need to change their code for caller id, if the number should be blocked then dont transmit it to the far end switch Err, not quite right. There are a few circumstances when called ID can be blocked (it's rumoured certain spook agencies have this ability), however if a user withholds CID, then it's just flagged at the local switch and passed switch to switch with the withold CLI flag. The terminating switch should then NOT pass on CLI if the withold flag is set on to an end-user line. Of course some agencies will get CLI passed even if the withold flag is set (in the UK, Police, fire, etc, potentially even ISPs for abuse purposes - but they are not meant to abuse the privilige). ___ 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] tdm400p static - out of ideas
Ryan Courtnage wrote: Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. Attempts to send outbound calls on any Zap channel will result in hearing a loud 'static' noise on the line. Let's look at some possibilities of line problems: What time does it stop answering? Is it ever during ALIT times (usually very early morning)? Have you tried calling the telco to see if it could be their problem? How far away from the CO/mux are you? Have you tried a new/different card? If you didn't want to fork out the cash for a new one, you could try a X100P/knockoff* on one of the lines to see if that fixes the problem... if so you can deduce a bad card. Nick *I usually don't recommend the knockoffs, but for a day of testing $10 sure beats $100... everybody else should support Digium! :-) ___ 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] Parking call problem
On Wednesday 07 July 2004 18:06, James Jones wrote: I been having a issue with call parking. I can park calls from internal extensions. But call from the outside can not be parked. When I recieve call from the outside I press the # key and nothing happens. Does any one have any thoughts? P.S. I am allowing the to be transferable. Show us your incoming call context, especially the Dial() line that rings the internal extensions. My guess is that you're using T instead of t. 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] VoIP hackers gut Caller ID
rich allen wrote: this is really simple, companies like Nortel, Lucent need to change their code for caller id, if the number should be blocked then dont transmit it to the far end switch That's a really bad idea. Even worse than top-posting. My local PSAP should know what number I'm calling from, because I'd like police/fire/EMS units to show up at my house if I can't tell them where I'm calling from. My phone company would also enjoy knowing where the call came from for the sake of preventing toll fraud from any Tom, Dick, and Harry with a SS7 connection. If CLID is blocked (or presentation restricted in SS7 ISUP parlance) only networks should see the Caller*ID, never users. This is a situation where network operators must not abrogate their responsibly to make and enforce policy; software solutions to policy problems are never panacean, just as policy can't fix an unencrypted password file. Nick ___ 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 hackers gut Caller ID
Chad Whitten wrote: this is true, but Bellsouth (our local RBOC) only allows numbers in our DID range to pass. I can set the outbound caller id to anything, but if its not in our DID range, then the lead number of the DID range is sent out. Are other telco's not doing this? No, not as a rule. And if you complain, the ones that do can make it go away, Nick ___ 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] Uniden consult transfer
This is how they implemented it, you cannot get the caller back.
Consultation transfer allows you--the transfering party to talk to
the transfered-to party before hangup up then the call be transfered.
(Xfer+number+dialthen hangup)
Or blind transfer where after pressing Xfer and number, transfering
party can just hang up to have the call transfered.
I was told that future firmware release will have additional
capability where the
transfering party can cancel a transfer and switch back to the
transfered party. This would be useful in case when transfered-to
party doesn't want to talk to transfred party, and allows the
transferring party to resume conversation with caller (transfered
party)
Ryan Courtnage [EMAIL PROTECTED] wrote:
On 6-Jul-04, at 4:36 PM, brian wrote:
Well if you xfer the call why is it asterisk job to know to bring the
call
back.. the transfer happened.
It's not ... this question is specific to Uniden UIP200 (ie: I'm not
referring to the *'s # transfer)
I'd like to know if it's possible to get the caller back after pressing
the button labeled 'XFER' on the phone - which doubles a a 'Flash'
button. The SIP call itself is not actually transfered by * until the
UIP200 user hangs up the phone. Uniden just kind of implemented their
flash transfer functions in a weird way.
Now you could get kinky with ${DNIS} or
atleast I think you can.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ryan Courtnage
Sent: Tuesday, July 06, 2004 10:53 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Uniden consult transfer
Hi all,
I curious to know if other UIP200 users have this same issue:
You flash (XFER button) to consult-transfer a caller to another
extension.
If
the transfer target party is unavailable (ie: voicemail), there
appears to
be
no way to get the original caller back.
If it's a known limitation, has anyone come up with a functional work
around?
Thank
--
..
Ryan Courtnage
Coalescent Systems Inc
403.244.8089
www.voxbox.ca
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Re: [Asterisk-Users] tdm400p static - out of ideas
On July 8, 2004 03:22 am, Nicholas Bachmann wrote: Ryan Courtnage wrote: Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. Attempts to send outbound calls on any Zap channel will result in hearing a loud 'static' noise on the line. Let's look at some possibilities of line problems: What time does it stop answering? Is it ever during ALIT times (usually very early morning)? It's totally random - morning/evening/afternoon. Once it stops answering, that's it, a reboot or module-reload is needed. If ALIT for some reason prevents the card from answering, it should be able to recover and begin answering after the ALIT is complete. Have you tried calling the telco to see if it could be their problem? When the card goes into the non-functional state, I can plug a regular phone into any of the lines and make calls just fine. After verifying working lines and plugging them back into the tdm400p card, I still can't dial out (the Zap channel will answer, but I will hear only static, and the call to the pstn is never placed). As well, incoming calls will not be answered (* console will not even show the 'started simple switch on zap/x' message). How far away from the CO/mux are you? Not too sure - it's in downtown Calgary - so probably not far. There is the possibility that _something_ with the phone line is triggering the problem. Maybe it's some noise, an unexpected signal, some crosstalk ... things that will cause unexpected behavior ... but also things that shouldn't put the entire card into a non-functioning state. Have you tried a new/different card? If you didn't want to fork out the cash for a new one, you could try a X100P/knockoff* on one of the lines to see if that fixes the problem... if so you can deduce a bad card. I may have to push for a replacement tdm400p card from Digium. Nick *I usually don't recommend the knockoffs, but for a day of testing $10 sure beats $100... everybody else should support Digium! :-) An acquaintance who is having the same problem has reluctantly replaced his card with an openline4. I would like nothing more than to stick with Digium hardware - this thread and obtaining a replacement card is my last kick at the cat. ___ 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 hackers gut Caller ID
If he is routing tandem traffic he would be running IMTs and be SS-7 interconnected. Hopefully his switching/prepaid equipment would have authentication capabilities to allow the registered caller id be generated. Note this peeve is against end-users manipulating it, not service providers. This comment is aimed at ISDN BRIs, PRIs, and PBX (trunk-side) DS1s where the end-user currently is able to spoof anything desired to the service provider's switch. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent: Wednesday, July 07, 2004 17:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Timothy R. McKee Sent: Wednesday, July 07, 2004 11:58 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID This has always been one of my pet peeves, even as I worked in the industry. A telco switch operating a DS1 on trunk side should enforce caller-id numbers to be within the range of DID numbers assigned to that trunk. There should be a default DID number that is used to replace any *invalid* numbers sent on that trunk. Note that blocked caller ids would still be blocked, but the rest of the data should be corrected. Blocking ID is ok, lying about it is not. Blind trust of a non-SS7 link is a _bad_ thing. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Wednesday, July 07, 2004 10:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 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 How then should a service provider who is routing tandem traffic place a call through any other network? This would preclude the ability for pre-paid or post paid providers to send out traffic at the originating customers request with correct callerid! Dave ___ 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] Uniden consult transfer
On July 8, 2004 03:53 am, sean mai wrote: This is how they implemented it, you cannot get the caller back. Consultation transfer allows you--the transfering party to talk to the transfered-to party before hangup up then the call be transfered. (Xfer+number+dialthen hangup) Or blind transfer where after pressing Xfer and number, transfering party can just hang up to have the call transfered. I was told that future firmware release will have additional capability where the transfering party can cancel a transfer and switch back to the transfered party. This would be useful in case when transfered-to party doesn't want to talk to transfred party, and allows the transferring party to resume conversation with caller (transfered party) I was told the same thing. They refered to it as Transfer with Shuttle which is under investigation for Phase 3 (whenever that is). ___ 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 hackers gut Caller ID
Insofar as I know it wasn't a feature in our DMS500 software load, if it was the translations/provisioning folks didn't seem to be aware of if. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Whitten Sent: Wednesday, July 07, 2004 17:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID this is true, but Bellsouth (our local RBOC) only allows numbers in our DID range to pass. I can set the outbound caller id to anything, but if its not in our DID range, then the lead number of the DID range is sent out. Are other telco's not doing this? On Wednesday 07 July 2004 11:04, brian wrote: Anyone with a PRI/ISDN line can set callerid to anything... Not just voip, not just asterisk. Come on guys. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Wednesday, July 07, 2004 9:01 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 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 -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ 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] Mandrake 10, Request for comments.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 07 July 2004 07:32 pm, [EMAIL PROTECTED] wrote: My * is presently running fine on Mandrake 9.2, but Ive been entertaining moving to Mandrake 10.0 to enjoy the obvious improvement in kernal speed Im seeing on other 10.0 boxes Ive recently built for other applications. (10.0 is the first implementation of the 2.6 kernal) Any comments from anyone who's running on 10.0? IS anyone running * on Mandrake 10.0? If so, any issues stand out? I'm hesitant because of the dot zero release of anything is always broken, and so far this has not been an exception, but not insurmountable. Thanks in advance. Marc To have solve the latter problem I have multiple boots. This gives me a stable and dev version. I create a couple of * partitions that can survive multiple Linux versions. /etc/asterisk /var/lib/asterisk I allocate around 5G total for each install. Now I can test different distro's with a minimum of fuss. To answer the MDK 10 question. I have it on two different boxes. One worked fine and the other I have not debugged but it could not find zapata. MDK have really nice security options which will tie down the box pretty tight with ongoing tests. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA7M3SljK16xgETzkRAtkSAJ9p0KWpPn+VtwrBpRDeYtLIZSk+AgCdHDN5 bYu9i3qMhUJ7TMKDm2UqGbM= =7aqg -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] Re: Re: iax or sip
Consider it backwards compatibility, sure, use [EMAIL PROTECTED] where you can, but I surely know if I told my parents to call me at ... Right now my grandstream bt-100 and asterisk team up to deliver 6001 as the number that I can be reached at to any remote caller. Somehow I don't think that my non-FQTN (Fully Qualified Telephone Number) is going to deliver much joy to folks hoping their return call button is going to do something useful. Would programming wolfgang at wsrcc dot com (damn spam-bots!) as the sip phone number allow a significant percentage of the folks to dial me back? (Assuming I have my _sip._udp SRV crap set right.) Do any commercial SIP providers lookup SRV? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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 hackers gut Caller ID
hi... here in Italy is almost impossible to set an invalid cid, if is out of your allowed space. ie. if you have X numbers on your PRI, you can only set one of these. nothing more. on bri you simply cannot do nothing. just my 2 cents. Indeed I've noticed here in Australia on BRI-ISDN (2x B channels) with DID I can't spoof numbers to the exchange... it's been a while since I toyed with the system, but from memory I could attempt to set any 9 digit number I wanted for the CallerID string, however the exchange would not allow that to go through and instead passed through the correct group directory number (primary number) for the service. However if I set the CallerID digits to anywhere within our 100-number block DID range, the exchange will happily pass on the specific number... guess it might be a combination of Euro ISDN standards and how the local telco's configure the exchanges. ___ 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] Parking call problem
Hi James, If you are finding that the # key does nothing, then the transfer feature of the Dial app is not working for you. This feature is not related to call parking. Check that your dial statement (when coming in from outside) uses the correct case of t :- t: Allow the called user to transfer the call T: Allow the calling user to transfer the call A lower case t is appropriate for your incoming calls, and a tT would probably be what you want for internal calls. Regardless of having call parking set up, pressing the hash key with these options should always result in the transfer message being played. Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 08/07/2004, at 8:06 AM, James Jones wrote: ___ 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] Mandrake 10, Request for comments.
On Wed, 2004-07-07 at 19:32 -0400, [EMAIL PROTECTED] wrote: My * is presently running fine on Mandrake 9.2, but Ive been entertaining moving to Mandrake 10.0 to enjoy the obvious improvement in kernal speed Im seeing on other 10.0 boxes Ive recently built for other applications. (10.0 is the first implementation of the 2.6 kernal) Any comments from anyone who's running on 10.0? IS anyone running * on Mandrake 10.0? If so, any issues stand out? I'm running one system on 10.0 and another on 10.1 Cooker with no issues. But I do use plain vanilla kernels from kernel.org. Just remember make linux26 for zaptel compilation. -- Dave Cotton [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] Intermittent cidname lookups
I'm having a problem with intermittent lookup of Caller ID Name info using LookupCIDName. The same problem occurs when doing: asterisk -rx database show cidname No data is returned on every fourth or fifth query. No errors are being logged. I'm currently running CVS-HEAD-07/07/04-17:04:31 and first noticed the problem a few weeks ago. Is anyone seeing a similar problem? ___ 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] X100P bad sound after period of time
Hi folks. I am using a X100P card and after some random amount of time of correct operation, say 8-20 hours, the card starts acting up and producng horrid sound quality which is all broken up. All other channels appear to work fine. One thing I noticed, is that zap show channel 1 always shows the Actual Hookstate: Offhook as soon as the telco line is plugged in. Is this normal? Maybe a bug in the status program or might this be indicative of my problem somewhere? The card claims to be sharing an interrupt with the SCSI controller and I don't see any way to change that. I put a second card in a different machine and it too, shared the interrupt but with the usb-uhci instead. It too shows zap show channel 1 as offhook as soon as the line is plugged in. Is there something real basic I am missing here? I'm on CVS-HEAD-06/27/04-23:21:33 /proc/interrupts CPU0 0: 541808 XT-PIC timer 1: 1203 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-ohci 8: 1 XT-PIC rtc 10:5281916 XT-PIC aic7xxx, wcfxo 11: 22266 XT-PIC aic7xxx, eth0, Cyclom-Y 12: 32 XT-PIC PS/2 Mouse 14: 0 XT-PIC ide0 NMI: 0 ERR: 0 My modules.conf looks like: alias eth0 e100 alias scsi_hostadapter aic7xxx alias usb-controller usb-ohci options torisa base=0xd alias char-major-196 torisa options wcfxo debug=1 options torisa debug=1 options wcfxs debug=1 options zaptel debug=1 zaptel.conf fxsks=1 loadzone = us defaultzone=us zapata.conf [trunkgroups] [channels] context=demo signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no context = PSTN-in channel = 1 Thanks, dbc. -- David Cook ___ 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 hackers gut Caller ID
Here is what you can possibly do: - Steal calling cards if they are useing caller id authentication scheme - Get access to personal banking information (Citibank uses callerid as part of authentication process.) - Purchase goods and services backed up by calling verification. I can go on and on for hours. Main point of story that [EMAIL PROTECTED] will hit the fan and VOIP will be regulated badly. Especially if some known terrorist will confess about using Vonage in Afaganistan.or some of drug dealers/weapon traders will be cought . Bug generraly author of that article is an idiot. He does not understand the difference beteween VOIP and ISDN PRI. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listas iPfone Sent: Wednesday, July 07, 2004 6:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID This is very interesting... Regulations..USA... But... what can i do faking a caller id? stolen what? what is the point? miklos - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 12:56 PM Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID why regulate? nobody regulates the return address on a letter sent via USPS. - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:00 AM Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 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] E100P
wct1xxp is meant for E100P, you will need to do a modprobe or ztcfg for wct1xxp instead of wce1xxp (which is for the T100P if I am not wrong). Hope this helps On Wed, 07 Jul 2004 17:45:48 -0700, Ing. Angel Gomez [EMAIL PROTECTED] wrote: Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? The antistatic bag had a small label that has E100P written on it, and the card is a bit different than the T100P I already have, Does Digium use the same boards for both cards ? I don't have an E1 link here, can I test the card just by loading the driver and run zttool to see how many channels it shows ? I don't find a wce1xxp driver only wct1xxp... ___ 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] E100P
wct1xxp is meant for E100P, you will need to do a modprobe or ztcfg for wct1xxp instead of wce1xxp (which is for the T100P if I am not wrong). Hope this helps On Thu, 8 Jul 2004 13:57:41 +0800, Caleb Kow [EMAIL PROTECTED] wrote: wct1xxp is meant for E100P, you will need to do a modprobe or ztcfg for wct1xxp instead of wce1xxp (which is for the T100P if I am not wrong). Hope this helps On Wed, 07 Jul 2004 17:45:48 -0700, Ing. Angel Gomez [EMAIL PROTECTED] wrote: Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? The antistatic bag had a small label that has E100P written on it, and the card is a bit different than the T100P I already have, Does Digium use the same boards for both cards ? I don't have an E1 link here, can I test the card just by loading the driver and run zttool to see how many channels it shows ? I don't find a wce1xxp driver only wct1xxp... ___ 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] E100P
Ing. Angel Gomez wrote: Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? I think this was asked just a few days ago...the answer is YES. -- Andres Network Admin http://www.telesip.net ___ 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] E100P
Hello- Yes, they have the same artwork. You can tell an E100P by looking at the clock generator chip at U7. An E1 should be marked with 2.*** MHz (can't read all the digits on mine) Cheers Scott 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 Ing. Angel Gomez Sent: Wednesday, July 07, 2004 5:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E100P Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? The antistatic bag had a small label that has E100P written on it, and the card is a bit different than the T100P I already have, Does Digium use the same boards for both cards ? I don't have an E1 link here, can I test the card just by loading the driver and run zttool to see how many channels it shows ? I don't find a wce1xxp driver only wct1xxp... ___ 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] Small Linux Distro
checkout .. http://www.automated.it/asterisk/ and http://knopsterisk.com/ Feedback back to the forum once you make progress would be useful. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 08 July 2004 03:50 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Small Linux Distro Does anyone have a current, stripped linux distro which has only asterisk and net drivers? If so do you have it available somewhere? I guess also, my question could be, does anyone know of a small distro, which will run asterisk. When I say small I mean 700Mb Also, anyone got any sites on hand which would point to ways to make linux start up faster? (BTW this is all in aid of making Asterisk boxes, with LCDs and buttons as opposed to keyboard and screen - i will also write an interface for Asterisk to LCDproc, so that it can be controlled from buttons mounted next to the screen, and make it GPL). Any help, pointers greatly appreciated. Matt ___ 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] Perl library to manipulate 'ini files'
Can anyone tell where can I find the perl library for manipulating 'ini files'? Thanks, Asterisk config files are not really INI files. E.e. one line can come up several times: disallow=any allow=ABCD allow=BCDE and so on. If you want python and not perl and only generate (not read) config files, you might look at the source code of DESTAR, http://www.holgerschurig.de/destar.html. Starting point is asterisk.py Other Python classes are at http://sourceforge.net/projects/pyst/, see astconfig.py in their CVS. There might be perl code out there, but I'm not very much anymore in Perl :-) ___ 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] Small Linux Distro
Does anyone have a current, stripped linux distro which (Bhas only (Basterisk and net drivers? (B (BIf so do you have it available somewhere? (B (BI guess also, my question could be, does anyone know of a (Bsmall (Bdistro, which will run asterisk. (B (BWhen I say small I mean 700Mb (B (BWe mostly use SuSE for building Asterisk servers and the (Bsmallest you can do with the standard SuSE installer just (Babout fits onto 500MB. With a little bit of effort (Bremoving more stuff you could easily fit everything into a (B512MB compact flash card, including Asterisk and some free (Bspace for logs and voicemail. (B (BYou might want to use an IDE/CF adapter that can hold two (BCF cards. This way you could stick in a smaller second (Bcard -say- 32 or 64MB and mount stuff like /etc/asterisk, (B/var/log, /var/run and voicemail etc on that one, possibly (Bleaving the main card mounted read-only for most of the (Btime. (B (BAlso, keep in mind that you don't necessarily have to keep (Bkernel sources and development tools around, which can be (Ba siginficant reduction in required disk space. (B (BIf you have the time to put some more effort into this, (BI'd recommend you look into Coyote Linux ... (B (Bhttp://www.coyotelinux.com (B (BThis is an embedded Linux distro put together by Joshua (BJackson for the purpose of creating very small footprint (Bfirewall routers and VPN servers. (B (BHe's got three applications based on this so far, one is a (Bfirewall that fits entirely onto a floppy disk, another is (Ba VPN server and firewall router that fits into a 32MB (Bcompact flash card and the third one is an intrusion (Bdetection system with similar properties. (B (BWith a bit of effort you could possibly build a very small (Bfoot print (=64MB ?) Asterisk server on top of Coyote (BLinux. (B (BOther places to check out are (B (Bhttp://www.siliconpenguin.com (B (Bhttp://www.embedded-linux.org (B (Bhttp://www.linuxdevices.com (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.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
R: [Asterisk-Users] VoIP hackers gut Caller ID
hi... here in Italy is almost impossible to set an invalid cid, if is out of your allowed space. ie. if you have X numbers on your PRI, you can only set one of these. nothing more. on bri you simply cannot do nothing. just my 2 cents. In Switzerland CLI is also impossible to spoof - by default. If you ask the BRI/PRI provider, and you have an ISDN connection with DDI, they enable CLIP Special Arrangement, which allows to add a presentation number to the real CLI. So you can't really abuse of it, because your real number is always transmitted together with your pretend-to-be CLI. The advantage of this is that anyone can change his CLI, for example to make outgoing calls and show a 0800 number in the customer's cell phone. We use this feature in our company, because our customers know us by our 0800 number, not the real number hiding behind it. The disadvantage is that not all networks accept presentation numbers, for example Orange Mobile. In this case, the caller's real CLI will be displayed instead of the presentation number. If you get yourself an SS7 link that's a different story, but in this case you're supposed to be a trusted entity, and you shall not spoof and play with numbers that you're not allowed to use. IMHO, trusted entities with SS7 links that abuse of that power should simply be disconnected from the public network. Not every kid with a couple $1000 spare should be allowed to play with this. -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.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] VoIP hackers gut Caller ID
Alex wrote: Bug generraly author of that article is an idiot. He does not understand the difference beteween VOIP and ISDN PRI. Right on! I agree completely. 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] VoIP hackers gut Caller ID
On Wed, Jul 07, 2004 at 07:19:44PM -0800, rich allen wrote: what do you mean not quite right??? if the clid is supposed to be blocked then don't send it. if the far end is a law enforcement or emergency agency then the clid is NOT supposed to be blocked!! if the originating switch had the ability to send or not send, problem solved for voip providers from getting a blocked clid CLID is NEVER blocked at the SS7 level (well almost), it flagged as withheld. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ 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 hackers gut Caller ID
However if I set the CallerID digits to anywhere within our 100-number block DID range, the exchange will happily pass on the specific number... guess it might be a combination of Euro ISDN standards and how the local telco's configure the exchanges. Interesting. Our incumbent Deutsche Telekom sells a disabled screening on a BRI port for 2,x Euros per month to anybody who asks. To be fair they will set the screening indicator to 'user provided, not screened', so in theory a called party could tell (one can on another BRI line). Unfortunately, the screening indicator does not appear on analog lines or mobiles. This feature really comes in handy, if you forward calls coming from 3rd parties to your mobile as you can preserve the original callerid and can return any calls missed. Thilo P.S.: Don't worry about fake callerid coming from Germany. Any numbering plan indicators will be set to national. ___ 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] E100P
Andres wrote: Ing. Angel Gomez wrote: Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? I think this was asked just a few days ago...the answer is YES. If people would read the included documentation from Digium they would have known this little fact. 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] Unreliable dtmf digit generation from tdm400p
Al,
I spoke with a tech support person at Digium today. He suggested adding a
'w' to the dial string. I did this and thus far it appears to have solved
the problem.
; we must add a pause (w) in the dial string for some reason otherwise the
dtmf digits sent won't be complete.
; appears only to be a problem with tdm400 cards, x100p cards seem to dial
ok without the w
exten = _9X.,1,Dial(ZAP/10/w${EXTEN:1})
exten = _9X.,2,Congestion
As a side item, while doing all the testing to figure this out, I noticed
that there is a delay in the dialing of the last digit. So if you dial
5551212 the card actually dials 555121 pause 2. Does anyone know why it
does this? The Digium support guy seemed to think it was a bug and
suggested I report it on bugs.digium.com, which I'll do unless someone
here can enlighten me as to the a meanful purpose to the delay. It
already takes longer than is really comfortable to dial a number so I'd
like to speed it up as much as possible.
Mark
I have the *exact* same problem. Please let me know if you have found
any solution. Thanks!
In my setup I have 2 of the TDM400P cards, with four FXO modules each.
Al
[EMAIL PROTECTED] wrote:
I have a tdm400p 4 port fxo card which is not reliably creating the dtmf
dialed digits when making a call. I have placed a linemans handset in
monitor mode on the line and can hear that what the system reports it is
dialing is not what the card is actually dialing. This happens about
25-50% of the time. The remaining time the digits dialed are correct and
the call goes through properly.
For example, I dial 5551212
== Spawn extension (default, 95551212, 1) exited non-zero on
'SIP/102-8da7'
-- Executing Dial(SIP/102-07cb, Zap/2/5551212) in new stack
-- Called 2/5551212
-- Zap/2-1 answered SIP/102-07cb
The system logs that it's dialing 5551212 to channel zap/2.. great.
Now when I actually listen to what the card is dialing, it doesn't dial
5551212 but something like 555212. I don't know what exactly it's
dialing
since I can't decode dtmf in my head, but it's clearly missing a digit or
two. As a result, the telco comes back with a your call can't be
completed because the full phone number wasn't dialed.
I have a X100P which is also in the system which works just fine.. it
never has this problem.
This is a brand new card, and I only have this one, so I can't test with
any others. Maybe it's defective?I've spent all day trying to
troubleshoot this - I've tried different phone lines, even put the card
into another box I built to try and troubleshoot. Always get the same
intermittant problem.
Also I've noticed in this testing that there is a slight pause before the
last digit is dialed. This always occurs and I'm curious why it does
this.
Thanks for anyones help!
Mark
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--
Alok K. Dhir [EMAIL PROTECTED]
Symplicity Corporation
http://solutions.symplicity.com
703 351 6987 (w) | 703 351-6357 (f)
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Re: [Asterisk-Users] Hangup's not detected correctly
Martin Pycko wrote: Well first of all if you're outside of US or callprogress-supported zones then you can use only busydetect. And that will only work if after the remote hangup your telco gives the fast-busy or any type of busy. You can tweak the duration of tone/pause and increase the count and it *will* work properly. regards Martin One thing to watch for here is RX gain if busydetect does not seem to be working after trying all the combinations. I had a 2 x X101P setup which busydetected perfectly - TX and RX gains were at the default levels. The X101Ps were replaced by a TDM card with 4 x FXO modules and with no config changes, busydetect stopped working. After incrementing in 1dB steps, an RX gain of 3.0 brought back reliable busydetect. I look forward to Rich Adamsons forthcoming writeup on setting up the gain distribution in an Asterisk system, to get everything working optimally. Regards, Richard ___ 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] Digium FXO Interfaces don't support groundstart???
Hi All, I was surprised to be told by a Digium support person today that Digium's FXO interfaces (X100P, TDM400P FXO modules) don't support groundstart signalling. This surprises me because as far as I know in a typical PBX configuration with analog trunk lines, groundstart signalling is the only way to prevent Glare. I just purchased two TDM400P's for a system I'm building to replace our office PBX (Altigen). Since there are no statements anywhere on Digium's website about lack of groundstart support (Actually, to the contrary they boast about all the signalling support in their sales slick), I now need to decide if I want to return the products and switch to a T1 / channel bank configuration. I remember when we setup our current Altigen PBX, we had problems with glare and disconnect detection and so I went through the process of figuring out what was going on and learning about groundstart. After we switched to groundstart everything worked great. In a high use system, it's highly likely that a trunk will experience glare, which is annoying for incoming callers and system users. I'm just a bit baffled as to why Digium wouldn't support groundstart on cards designed to be PBX trunk lines. Someone please tell me I'm missing something. Mark ___ 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] ISDN, AVM C4, HFC-cards and echo
the hfc-pci cards use the same echo cancelation (in software) that any zaptel device uses. Am Do, 2004-07-08 um 09.47 schrieb Peer Oliver Schmidt: [interfaces] msn=123456 echosquelsh=1 make that echosquelch=1 incomingmsn=* controller=1 softdtmf=0 context=default ;echocancel=yes ;echotail=64 ;deflect=12345678 devices=2 callgroup=1 best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/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
RE: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo
I have 2 HFC cards. There is no echo. If you use the bri-stuff drivers they use the native zaptel echo cancellers. And I have no echo. None. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver Schmidt Sent: 08 July 2004 08:47 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo Hello, I've been using * for quite a while using AVM C4 card and a few Grandstream Budgetone 102 hard phones plus couple of HandyTone 286. Echo is a big problem. I am getting used to it, but some users complain. Anyone has experience w/ regards to echo comparing the AVM C4 with two HFC-cards? Before shelling out time and money, maybe someone else already has done so, and could tell me if it is wortwhile. I only need four B-channels, so two HFC cards should be all I need. Any help, pointers and tips are greatly appreciated. Thanks for your time. rgds pos PS: Current CAPI.CONF settings: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=123456 echosquelsh=1 incomingmsn=* controller=1 softdtmf=0 context=default ;echocancel=yes ;echotail=64 ;deflect=12345678 devices=2 callgroup=1 ___ 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: [spam] Re: [Asterisk-Users] Fax Detection
Bkw thanks for the advcicel no joy though: I tried: Exten = 08700686XXX,1,Goto(textextension,7000,1) [testextension] Exten = 7000,1,Answer Exten = 7000,2,Ringing Exten = 7000,3,Wait(5) Exten = 7000,4,Dial(SIP/104) Exten = fax,1,rxfax(/var/spool/asterisk/incoming/testfax.tif) Exten = fax,2,congestion Exten = fax,102,congestion Any other tips? Cheers Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian K. West Sent: 08 July 2004 02:22 To: [EMAIL PROTECTED] Subject: [spam] Re: [Asterisk-Users] Fax Detection Try Answer Then Ringing and wait about 2-3 seconds. Then Dial bkw - Original Message - From: Matt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 2:44 PM Subject: [Asterisk-Users] Fax Detection Hi all I've tried Google, wiki and mailing list and IRC but still haven't gotten to the bottom of this. Hopefully someone might be able to help. I'm using telappliant to provide my inbound and outbound calls. * plays host to 30 cisco's and they are all working great using G711 A-law. I've managed to get SpanDSP to compile and install and I can send a receive a fax on a dedicated extension. What I'm trying to do now and can't seem to nail is getting an inbound fax to be detected and then handled. I've tried the examples from the wiki and the sites linked on the wiki; messed about trying my own weird and wonderful methods but still no joy. All the calls are using G711 A-law. Here is the test context I'm using XXX = hiden Exten = 08700686XXX,1,Goto(textextension,7000,1) [testextension] Exten = 7000,1,Answer Exten = 7000,2,Dial(SIP/104) Exten = fax,1,rxfax(/var/spool/asterisk/incoming/testfax.tif) Exten = fax,2,congestion Exten = fax,102,congestion Calls hit the testextension contect but don't get detected as a fax. Cheers Matt ___ 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] Can X100P/X101P detect reversal of line polarity?
On Thu, Jul 08, 2004 at 08:30:59PM +1200, Richard Scobie wrote: One thing to watch for here is RX gain if busydetect does not seem to be working after trying all the combinations. I had a 2 x X101P setup which busydetected perfectly - TX and RX gains were at the default levels. The X101Ps were replaced by a TDM card with 4 x FXO modules and with no config changes, busydetect stopped working. After incrementing in 1dB steps, an RX gain of 3.0 brought back reliable busydetect. I look forward to Rich Adamsons forthcoming writeup on setting up the gain distribution in an Asterisk system, to get everything working optimally. In Sweden we don't get a busy signal when the remote part hangs up. Instead remote hang up is signaled by reversing the polarity of the line. Can X100P/X101P detect polarity reversal when off hook, on hook, or both? Regards, Mikael Magnusson ___ 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 placed towards a trange called number 'h'
Sometime I observe that my Asterisk is resending a ISDN call with a strange called number equals to 'h'. Is there a possibility to avoid that ? /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] x100p and two hfc isdn cards
hello, i have a problem starting asterisk with one x100p digium and two hfc chipset isdn cards with bri-stuff.0.0.2. ztcfg -vv shows me a this info: Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: D-channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: D-channel (Default) (Slaves: 07) 7 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) - cat /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1 loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=2-3,5-6 dchan=4,7 and # cat /etc/asterisk/zapata.conf [channels] switchtype = euroisdn ; p2p TE mode signalling = bri_cpe ; prilocaldialplan=national pridialplan = unknown ; echocancel=yes group = 1 context=isdn channel = 2-3,5-6 group = 2 context=gsm signalling=fxs_ks channel = 1 - but when i start asterisk i got this errors: Parsing '/etc/asterisk/zapata.conf': Found Jul 8 13:53:58 WARNING[16384]: chan_zap.c:682 zt_open: Unable to specify channel 2: No such device or address Jul 8 13:53:58 ERROR[16384]: chan_zap.c:5397 mkintf: Unable to open channel 2: No such device or address here = 0, tmp-channel = 2, channel = 2 Jul 8 13:53:58 ERROR[16384]: chan_zap.c:7668 setup_zap: Unable to register channel '2-3' Jul 8 13:53:58 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Jul 8 13:53:58 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Segmentation fault what to do? i have latest CVS asterisk .. thank you, Tomaz ___ 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] Rollover oddity
Hello, I've got 2 analogue lines (from SBC) coming into a TDM22B. SBC have put rollover from the first to the second line. The rollover works fine when handsets are connected directly to the lines (ie when Asterisk is not involved), but when the lines are connected to Asterisk, the rollover fails: the caller just hears the line ringing, and the person on the first (busy) line hears call waiting interrupts. I have proved that it's the rollover that's not taking place (ie it is not that the rollover happens, but the second line isn't answered) So how (and why) does Asterisk affect the rollover in this way, and how can a busy line going through Asterisk 'look' different to the telco from a normal handset? And, of course... how do I solve this? Thanks Bob ___ 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] E100P
JM If people would read the included documentation from Digium JM they would have known this little fact. What documents? What do the documents say? Can we get one scanned and posted in the wiki? (Please). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: 08 July 2004 09:07 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E100P Andres wrote: Ing. Angel Gomez wrote: Hi, i just received an E100P, this is the first one I have ever seen, and notice that the board reads T100P. Is this right ? I think this was asked just a few days ago...the answer is YES. If people would read the included documentation from Digium they would have known this little fact. 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] strange problem with oh323 loaded!
What exactly is the problem with v0.6.3(a)? Michael. Anthony Law wrote: I too tried 0.6.3 and it is behaving the same. I have now downloaded oh323 to 0.6.2a and it seems fine. Regards, Anthony ___ 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 -- *** * * *G R E E C E * * * * EUROPEAN CHAMPION EURO 2004 * * * *** ___ 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 hackers gut Caller ID
Because if p2p voip means i get the same volume of junk phonecalls as i currently do spam emails i am not even going to _think_ about adopting it. We _need_ authentification. Steve Totaro [EMAIL PROTECTED] wrote: __ why regulate? nobody regulates the return address on a letter sent via USPS. - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:00 AM Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 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] Cisco 7960 NAT question
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 firewall-ip D N 255.255.255.255 60665 Unmonitored tp2/tp2 firewall-ip D N 255.255.255.255 60646 Unmonitored tp1/tp1 firewall-ip D N 255.255.255.255 60649 Unmonitored Now, the Cisco phones are set to use nat (nat = 1) and in the SIP configuration, the phones are also configured for SIP. [tp1] type=friend secret=tp1 host=dynamic nat=yes callerid=Test Phone 1 I can make calls out over the phones, but can't get anything back in. If I call voicemail say, then that's fine. But if I try and call another phone behind the firewall, it just sits there :/ IS there a specific port range I need to open? Should I be using a different sip config? Cheers for any help, Ben www.griffin.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] Small Linux Distro
Thanks to everybody for your links and support. I think I have convinced all of the shareholders (tonight) that GPL is the way for anything we develop with regards to Asterisk, and they are going to put some more money into building a couple of prototypes boxes...(fun!!! hacking time!!!) So, once I have the hardware all setup and working properly, I'll post again here and find out what kind of features people would like to see in the LCD/button interface and what kind of statistics and options would be beneficial. Kind regards, Matt Riddell ___ 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 7960 NAT question
I had the same problem. What I found is I needed to set register with proxy to yes in the sip config. Hope this helps -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills Sent: Thursday, July 08, 2004 7:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 NAT question I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 firewall-ip D N 255.255.255.255 60665 Unmonitored tp2/tp2 firewall-ip D N 255.255.255.255 60646 Unmonitored tp1/tp1 firewall-ip D N 255.255.255.255 60649 Unmonitored Now, the Cisco phones are set to use nat (nat = 1) and in the SIP configuration, the phones are also configured for SIP. [tp1] type=friend secret=tp1 host=dynamic nat=yes callerid=Test Phone 1 I can make calls out over the phones, but can't get anything back in. If I call voicemail say, then that's fine. But if I try and call another phone behind the firewall, it just sits there :/ IS there a specific port range I need to open? Should I be using a different sip config? Cheers for any help, Ben www.griffin.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 ___ 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] Minimum install required for Asterisk + voicemail SIP friends from mysql
I have been trying to install asterisk with MySQL for voicemail and SIP friends. Using redhat 9 I am installing all the base components required for asterisk, mysql, mysql server and mysql-devel. If I do a make clean, make install without enabling the mysql options in the /apps/Makefile and /channels/Makefile all goes well and make completes. If I enable the options Make fails, when it gets to -L/usr/lib/mysql -lmysqlclient -lz with an error /usr/bin/ld: cannot find -lz Now If I do the same after installing Redhat 9 doing a complete install make completes successfully. Ideally I would (I am sure others would too) like to install the minimum required. Help/advice will be greatly appreciated. Umar. ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.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] ISDN, AVM C4, HFC-cards and echo
I have a HFC-S Card with European Signaling , Can you tell me what distro/kernel are you using and how did you get the card detected with linux ? a small procedure ? regards ~uppal On Thu, 8 Jul 2004 10:12:04 +0100, Robinson Tim-W10277 [EMAIL PROTECTED] wrote: I have 2 HFC cards. There is no echo. If you use the bri-stuff drivers they use the native zaptel echo cancellers. And I have no echo. None. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver Schmidt Sent: 08 July 2004 08:47 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo Hello, I've been using * for quite a while using AVM C4 card and a few Grandstream Budgetone 102 hard phones plus couple of HandyTone 286. Echo is a big problem. I am getting used to it, but some users complain. Anyone has experience w/ regards to echo comparing the AVM C4 with two HFC-cards? Before shelling out time and money, maybe someone else already has done so, and could tell me if it is wortwhile. I only need four B-channels, so two HFC cards should be all I need. Any help, pointers and tips are greatly appreciated. Thanks for your time. rgds pos PS: Current CAPI.CONF settings: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=123456 echosquelsh=1 incomingmsn=* controller=1 softdtmf=0 context=default ;echocancel=yes ;echotail=64 ;deflect=12345678 devices=2 callgroup=1 ___ 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] ISDN, AVM C4, HFC-cards and echo
RH9 and the drivers from http://capi4linux.thepenguin.de/download/asterisk/ You don't use CAPI or isdn4linux. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Junaid Saeed Uppal Sent: 08 July 2004 13:08 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo I have a HFC-S Card with European Signaling , Can you tell me what distro/kernel are you using and how did you get the card detected with linux ? a small procedure ? regards ~uppal On Thu, 8 Jul 2004 10:12:04 +0100, Robinson Tim-W10277 [EMAIL PROTECTED] wrote: I have 2 HFC cards. There is no echo. If you use the bri-stuff drivers they use the native zaptel echo cancellers. And I have no echo. None. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver Schmidt Sent: 08 July 2004 08:47 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo Hello, I've been using * for quite a while using AVM C4 card and a few Grandstream Budgetone 102 hard phones plus couple of HandyTone 286. Echo is a big problem. I am getting used to it, but some users complain. Anyone has experience w/ regards to echo comparing the AVM C4 with two HFC-cards? Before shelling out time and money, maybe someone else already has done so, and could tell me if it is wortwhile. I only need four B-channels, so two HFC cards should be all I need. Any help, pointers and tips are greatly appreciated. Thanks for your time. rgds pos PS: Current CAPI.CONF settings: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=123456 echosquelsh=1 incomingmsn=* controller=1 softdtmf=0 context=default ;echocancel=yes ;echotail=64 ;deflect=12345678 devices=2 callgroup=1 ___ 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] Question about Cisco IP Phone 7960
I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic ___ 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 hackers gut Caller ID
The real problem here is that people shouldn't be using callerid as an authentication scheme. Lots of people have had the ability to set arbitrary clid for years and yet banks and other institutions have stupidly used it to authenticate callers. Complaints should be directed to them and not the VoIP industry. -brian Alex wrote: Here is what you can possibly do: - Steal calling cards if they are useing caller id authentication scheme - Get access to personal banking information (Citibank uses callerid as part of authentication process.) - Purchase goods and services backed up by calling verification. I can go on and on for hours. Main point of story that [EMAIL PROTECTED] will hit the fan and VOIP will be regulated badly. Especially if some known terrorist will confess about using Vonage in Afaganistan.or some of drug dealers/weapon traders will be cought . Bug generraly author of that article is an idiot. He does not understand the difference beteween VOIP and ISDN PRI. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listas iPfone Sent: Wednesday, July 07, 2004 6:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID This is very interesting... Regulations..USA... But... what can i do faking a caller id? stolen what? what is the point? miklos - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 12:56 PM Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID why regulate? nobody regulates the return address on a letter sent via USPS. - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:00 AM Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP hackers gut Caller ID
It is imperative that the ability to set caller ID's is kept as we need this in everyday business. stuart - Original Message - From: Brian Cuthie [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 1:28 PM Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID The real problem here is that people shouldn't be using callerid as an authentication scheme. Lots of people have had the ability to set arbitrary clid for years and yet banks and other institutions have stupidly used it to authenticate callers. Complaints should be directed to them and not the VoIP industry. -brian Alex wrote: Here is what you can possibly do: - Steal calling cards if they are useing caller id authentication scheme - Get access to personal banking information (Citibank uses callerid as part of authentication process.) - Purchase goods and services backed up by calling verification. I can go on and on for hours. Main point of story that [EMAIL PROTECTED] will hit the fan and VOIP will be regulated badly. Especially if some known terrorist will confess about using Vonage in Afaganistan.or some of drug dealers/weapon traders will be cought . Bug generraly author of that article is an idiot. He does not understand the difference beteween VOIP and ISDN PRI. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listas iPfone Sent: Wednesday, July 07, 2004 6:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID This is very interesting... Regulations..USA... But... what can i do faking a caller id? stolen what? what is the point? miklos - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 12:56 PM Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID why regulate? nobody regulates the return address on a letter sent via USPS. - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:00 AM Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN, AVM C4, HFC-cards and echo
I have a HFC-S Card with European Signaling , Can you tell me what distro/kernel are you using and how did you get the card detected with linux ? a small procedure ? Isn't this already decently written in the wiki and in the mailing list archive? If not, can you please refer us to the wiki pages where you miss information? ___ 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 hackers gut Caller ID
See bottom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Timothy R. McKee Sent: Thursday, July 08, 2004 12:05 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID If he is routing tandem traffic he would be running IMTs and be SS-7 interconnected. Hopefully his switching/prepaid equipment would have authentication capabilities to allow the registered caller id be generated. Note this peeve is against end-users manipulating it, not service providers. This comment is aimed at ISDN BRIs, PRIs, and PBX (trunk-side) DS1s where the end-user currently is able to spoof anything desired to the service provider's switch. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent: Wednesday, July 07, 2004 17:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Timothy R. McKee Sent: Wednesday, July 07, 2004 11:58 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID This has always been one of my pet peeves, even as I worked in the industry. A telco switch operating a DS1 on trunk side should enforce caller-id numbers to be within the range of DID numbers assigned to that trunk. There should be a default DID number that is used to replace any *invalid* numbers sent on that trunk. Note that blocked caller ids would still be blocked, but the rest of the data should be corrected. Blocking ID is ok, lying about it is not. Blind trust of a non-SS7 link is a _bad_ thing. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Wednesday, July 07, 2004 10:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 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 How then should a service provider who is routing tandem traffic place a call through any other network? This would preclude the ability for pre-paid or post paid providers to send out traffic at the originating customers request with correct callerid! Dave ___ 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 No , you don't have to be using SS7 signaling on your IMT's, 4Wire EM configured for DTMF or MF digits will provide the capability to send out ANI/Callerid to the PSTN. When 800 inbound traffic is delivered over FGD circuits the typical pattern received when set for (DTMF) is
Re: [Asterisk-Users] Cisco 7960 NAT question
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 firewall-ip D N 255.255.255.255 60665 Unmonitored tp2/tp2 firewall-ip D N 255.255.255.255 60646 Unmonitored tp1/tp1 firewall-ip D N 255.255.255.255 60649 Unmonitored Now, the Cisco phones are set to use nat (nat = 1) and in the SIP configuration, the phones are also configured for SIP. [tp1] type=friend secret=tp1 host=dynamic nat=yes callerid=Test Phone 1 I can make calls out over the phones, but can't get anything back in. If I call voicemail say, then that's fine. But if I try and call another phone behind the firewall, it just sits there :/ IS there a specific port range I need to open? Should I be using a different sip config? The sonicwall has a user selectable option to support sip. Have you tried to enable it? (Don't know how well that actually works, never tested 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] VoIP hackers gut Caller ID
Institutions using caller ID could just impliment a callback feature to verify identity, but even then a phone guy could be sitting outside your house or business with a butt set. In all reality, there is no way to ID someone without knowing them AND conducting a face to face transaction (and even then, how can you really be sure that you know them?) Username and password are a joke, voice is easily recorded and manipulated, biometrics can be fooled with scotch tape or other means. Someone can swipe your RSA FOB etc... I am sure terrorist are using VoIP, they arent stupid (when it comes to technology). They have been merging messages into images and posting them on the internet for years. That takes more know how than placing a voip call. Thanks, Steve Totaro - Original Message - From: Brian Cuthie [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 8:28 AM Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID The real problem here is that people shouldn't be using callerid as an authentication scheme. Lots of people have had the ability to set arbitrary clid for years and yet banks and other institutions have stupidly used it to authenticate callers. Complaints should be directed to them and not the VoIP industry. -brian Alex wrote: Here is what you can possibly do: - Steal calling cards if they are useing caller id authentication scheme - Get access to personal banking information (Citibank uses callerid as part of authentication process.) - Purchase goods and services backed up by calling verification. I can go on and on for hours. Main point of story that [EMAIL PROTECTED] will hit the fan and VOIP will be regulated badly. Especially if some known terrorist will confess about using Vonage in Afaganistan.or some of drug dealers/weapon traders will be cought . Bug generraly author of that article is an idiot. He does not understand the difference beteween VOIP and ISDN PRI. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listas iPfone Sent: Wednesday, July 07, 2004 6:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID This is very interesting... Regulations..USA... But... what can i do faking a caller id? stolen what? what is the point? miklos - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 12:56 PM Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID why regulate? nobody regulates the return address on a letter sent via USPS. - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:00 AM Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 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]
RE: [Asterisk-Users] VoIP hackers gut Caller ID
Correct, I was trying to not muddy the waters with lots of detail. Basically I was saying that inter-provider trunk links should be trusted and trunk links directly to end-users (where DIDs are assigned) should not be. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent: Thursday, July 08, 2004 08:51 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID See bottom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Timothy R. McKee Sent: Thursday, July 08, 2004 12:05 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID If he is routing tandem traffic he would be running IMTs and be SS-7 interconnected. Hopefully his switching/prepaid equipment would have authentication capabilities to allow the registered caller id be generated. Note this peeve is against end-users manipulating it, not service providers. This comment is aimed at ISDN BRIs, PRIs, and PBX (trunk-side) DS1s where the end-user currently is able to spoof anything desired to the service provider's switch. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent: Wednesday, July 07, 2004 17:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Timothy R. McKee Sent: Wednesday, July 07, 2004 11:58 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID This has always been one of my pet peeves, even as I worked in the industry. A telco switch operating a DS1 on trunk side should enforce caller-id numbers to be within the range of DID numbers assigned to that trunk. There should be a default DID number that is used to replace any *invalid* numbers sent on that trunk. Note that blocked caller ids would still be blocked, but the rest of the data should be corrected. Blocking ID is ok, lying about it is not. Blind trust of a non-SS7 link is a _bad_ thing. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Wednesday, July 07, 2004 10:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 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 How then should a service provider who is routing tandem traffic place a call through any other network? This would preclude the ability for pre-paid or post paid providers to send out traffic at the originating customers request with correct callerid! Dave ___ 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] Meetme and IAX
Hello everybody! I have been using MeetMe application with SIP and Zap devices and everything worked fine. Now I want to use an IAX client but when trying to connect to a conference I get a message telling that the conf number is not valid. Through documentation it seems to me that conference should be enabled for IAX clients but I don't know how. Please can someone tell me? Thanks in advance Lamine Copyrights © 2003 Groupe Chaka - http://www.chaka.sn ___ 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] zaphfc and ASUSCOM working in the US
Junaid Saeed Uppal wrote: I have a cologne chip isdn pci card too , can you tell me what distro are you using and what procedure did you follow to get it working with isdn4linux? regards ~uppal I'm using zaphfc (not i4l)with RH 9.0 and bristuffbri-stuff-0.0.2a-pp. I had to modify zaphfc.c and change to PCI_VENDOR_ID_ASUSTEK and PCI_DEVICE_ID_ASUSTEK_0675 (found these in pci_ids.h) zaptel.conf: span=1,1,3,ccs,ami bchan=1-2 dchan=3 ... zapata.conf signalling=bri_cpe_ptmp pridialplan=local prilocaldialplan=local echocancel=yes immediate=yes group=5 context=mainmenu channel=1-2 ... When I change the ccs/ami parameters to esf/b8zs (or anything else) there is no obvious effect (I had to modify zaphfc.c to allow other span parameters). However, I'm getting a segfault when I try pri intense debug span 1 (pri debug span 1 works ok). I'm trying to locate my gdb book in order to analyze the core file. Jonathan Sadler says IE 0x2a (IE 42) is for display text such as CID name info. Good luck with your card. -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.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] VoIP hackers gut Caller ID
On Jul 8, 2004, at 9:51 AM, Steve Kennedy wrote: On Wed, Jul 07, 2004 at 07:19:44PM -0800, rich allen wrote: what do you mean not quite right??? i[..]blocked clid CLID is NEVER blocked at the SS7 level (well almost), it flagged as withheld. Bingo, if you have a SS7 switch at the net then you can send whatever you want. Michael ___ 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] sample config file for GS BT101?
If you have an example of a config file for a Grandstream BT101/102, I would appreciate if you would share it with me. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] OH323-COMPILE
HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but Icome up with dead end. In factit makes mecrazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07, 2004-07-02, .) 4- oh323 v.5-10 / oh323 v.5.9 5- my linux box is redhat 8.0 the error looks like the following: make[1]: *** [chan_oh323.o] Eroor 1 make[1]: Leaving directory '/root/asterisk/asterisk-oh323-0.5.9/asterisk-driver' make: *** [subdirs_all] Error 1 I think there is a mismatch between my oh323 and asterisk. But I donot know the excat asterisk CVS I will be waiting for your help warmest regards mohammad
Re: [Asterisk-Users] Question about Cisco IP Phone 7960
On 08/07/2004 at 08:21 Hall, Eric M. wrote: I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic Make sure you changed ALL the configs to point to the CORRECT image file.. OS79XX.TXT should contain P0S3-07-1-00 and your SIPmac or SIPDefault.cnf should contain image_version: P0S3-07-1-00 iirc the default in OS79XX.TXT is the unsigned image... HTH Andy ___ 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] GR303
The GR303 support is network side only right now and documentation is minimal. Client side operation is in development right now and should be available around the end of the month. The network model will be * connecting to a 5E on a GR303 trunk acting as a sip gateway for GR303 subscribers in the switch. After we get this up and working I would be glad to post configurations for anyone who might be interested. Jody N. Rudolph Heartland Internet Services [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Wednesday, July 07, 2004 9:07 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GR303 where can i find documentation on Asterisk's support for GR303??? Multiple posters have asked the same question, however no one seems to know for sure. General opinion seems to be the code that does exist was probably intended to communicate with an access box of some sort, and not with telco central offices. ___ 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] Kerry/Edwards campaign and VOIP
They could spoof their caller ID to the Bush/Cheney campain and call people at 3AM to ask for their support!!! - Original Message - From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 7:56 PM Subject: Re: [Asterisk-Users] Kerry/Edwards campaign and VOIP Bill Merriam wrote: I am trying find a way to help the local Kerry campaign and it occurs to me that VOIP and Asterisk could be a big help. I have never worked on a Bill, You'll find that the FEC has VERY strict guidelines regarding things like this. John ___ 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] x100p and two hfc isdn cards
hi I had the saem trouble, so I just took my x100p card out and the problem went away:) I know its not the ultimate solution, but I decided to use an ATA with my analgue phone instead. I would suggest trying to put the analogue lines as channel 7 and the isdn lines as channels 1-6 Good luck regards Clive On Thu, 08 Jul 2004 11:52:23 +0200 Tomaz [EMAIL PROTECTED] wrote: hello, i have a problem starting asterisk with one x100p digium and two hfc chipset isdn cards with bri-stuff.0.0.2. ztcfg -vv shows me a this info: Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: D-channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: D-channel (Default) (Slaves: 07) 7 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) - cat /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1 loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=2-3,5-6 dchan=4,7 and # cat /etc/asterisk/zapata.conf [channels] switchtype = euroisdn ; p2p TE mode signalling = bri_cpe ; prilocaldialplan=national pridialplan = unknown ; echocancel=yes group = 1 context=isdn channel = 2-3,5-6 group = 2 context=gsm signalling=fxs_ks channel = 1 - but when i start asterisk i got this errors: Parsing '/etc/asterisk/zapata.conf': Found Jul 8 13:53:58 WARNING[16384]: chan_zap.c:682 zt_open: Unable to specify channel 2: No such device or address Jul 8 13:53:58 ERROR[16384]: chan_zap.c:5397 mkintf: Unable to open channel 2: No such device or address here = 0, tmp-channel = 2, channel = 2 Jul 8 13:53:58 ERROR[16384]: chan_zap.c:7668 setup_zap: Unable to register channel '2-3' Jul 8 13:53:58 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Jul 8 13:53:58 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Segmentation fault what to do? i have latest CVS asterisk .. thank you, Tomaz ___ 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 _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ 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] Problem SIP no audio just noise
I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk box. Sorry for the long logs. Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2 From: damian sip:[EMAIL PROTECTED];tag=2667644054 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 42510 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103a Content-Length: 291 v=0 o=damian 23894728 23894788 IN IP4 10.1.1.11 s=X-Lite c=IN IP4 10.1.1.11 t=0 0 m=audio 8000 RTP/AVP 0 8 3 97 110 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 13 lines Using latest request as basis request Sending to 10.1.1.11 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 110 Found RTP audio format 101 Peer RTP is at port 10.1.1.11:0 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'phone1010' Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:4851 check_user: Setting NAT on RTP to 0 Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:6424 handle_request: Check for res for damian Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:1386 update_user_counter: damian is not a local user Looking for 99826816 in default Jul 8 16:47:21 DEBUG[65541]: chan_sip.c:4115 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2 From: damian sip:[EMAIL PROTECTED];tag=2667644054 To: sip:[EMAIL PROTECTED];tag=as5b6158bb Call-ID: [EMAIL PROTECTED] CSeq: 42510 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED]:0 Content-Length: 0 to 10.1.1.11:5060 -- Executing Dial(SIP/damian-ff45, Zap/4/9826816) in new stack Jul 8 16:47:21 DEBUG[262159]: chan_zap.c:1576 zt_call: Dialing '9826816' Jul 8 16:47:21 DEBUG[262159]: chan_zap.c:1633 zt_call: Deferring dialing... -- Called 4/9826816 Jul 8 16:47:21 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception on 21, channel 4 Jul 8 16:47:21 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got event Hook Transition Complete(12) on channel 4 (index 0) Jul 8 16:47:23 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception on 21, channel 4 Jul 8 16:47:23 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got event Dial Complete(9) on channel 4 (index 0) Jul 8 16:47:23 DEBUG[262159]: chan_zap.c:1169 zt_enable_ec: No echocancellation requested Jul 8 16:47:23 DEBUG[262159]: chan_zap.c:1185 zt_train_ec: No echo training requested Jul 8 16:47:24 DEBUG[262159]: chan_zap.c:3596 __zt_exception: Exception on 21, channel 4 Jul 8 16:47:24 DEBUG[262159]: chan_zap.c:2944 zt_handle_event: Got event Dial Complete(9) on channel 4 (index 0) Jul 8 16:47:24 DEBUG[262159]: chan_zap.c:1169 zt_enable_ec: No echocancellation requested Jul 8 16:47:24 DEBUG[262159]: chan_zap.c:3007 zt_handle_event: Done dialing, but waiting for progress detection before doing more... We're at 10.1.1.2 port 10524 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.1.11:5060;rport;branch=z9hG4bK493E94339A0F46AAACD2EF6C557922C2 From: damian sip:[EMAIL PROTECTED];tag=2667644054 To: sip:[EMAIL PROTECTED];tag=as5b6158bb Call-ID: [EMAIL PROTECTED] CSeq: 42510 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED]:0 Content-Type: application/sdp Content-Length: 251 v=0 o=root 586 586 IN IP4 10.1.1.2 s=session c=IN IP4 10.1.1.2 t=0 0 m=audio 10524 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.1.11:5060 Jul 8 16:47:24 DEBUG[262159]: rtp.c:1123 ast_rtp_write: Ooh, format changed from UNKN to ULAW Jul 8 16:47:24 DEBUG[262159]: chan_sip.c:1976 sip_rtp_read: Oooh, format
[Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6
I'm installing the new Slackware 10.0 distribution - but not sure if i should go with the 2.4 kernal - which i think is the default install - or the new 2.6 kernal? anyone running * and slackware 10.0 with 2.6 kernal? thanks joe ___ 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] sample config file for GS BT101?
http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone the wiki seems to be VERY complete when it comes to GS - Original Message - From: Bruce Komito [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 9:31 AM Subject: [Asterisk-Users] sample config file for GS BT101? If you have an example of a config file for a Grandstream BT101/102, I would appreciate if you would share it with me. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING
Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler Honestly I don't know how much more clear this message can be. You need an exten = s,1,something in your [default] context. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of robert brown Sent: Thursday, July 08, 2004 2:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING Hello, Can anyone help with the output shown below? Its running on RH9, recent CVS of Asterisk and with one X100P card (2 channels), a budget tone 102 and Xlite softphone. CLI -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:32 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:40 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Due to various SIP errors, where I am unable to authenticate, I decided to blank out all configuration relating to any SIP phones. Robert Brown FWD: 290651 [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] FINALLY! a good book about Asterisk.
There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __ 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
[Asterisk-Users] Shady dial anyone??
wondering if anybody knows this..does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. Nauman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 5:48 PM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4448 - 10 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: VoIP hackers gut Caller ID ([EMAIL PROTECTED]) 2. Cisco 7960 NAT question (Ben Merrills) 3. Re: Small Linux Distro ([EMAIL PROTECTED]) 4. RE: Cisco 7960 NAT question (Hall, Eric M.) 5. Minimum install required for Asterisk + voicemail SIP friends from mysql (=?iso-8859-1?q?Umar=20Sear?=) 6. Re: ISDN, AVM C4, HFC-cards and echo (Junaid Saeed Uppal) 7. RE: ISDN, AVM C4, HFC-cards and echo (Robinson Tim-W10277) 8. Question about Cisco IP Phone 7960 (Hall, Eric M.) 9. Re: VoIP hackers gut Caller ID (Brian Cuthie) 10. Re: VoIP hackers gut Caller ID (Stuart Baggs) --__--__-- Message: 1 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID From: [EMAIL PROTECTED] Date: Thu, 8 Jul 2004 11:59:00 0100 Reply-To: [EMAIL PROTECTED] Because if p2p voip means i get the same volume of junk phonecalls as i currently do spam emails i am not even going to _think_ about adopting it. We _need_ authentification. Steve Totaro [EMAIL PROTECTED] wrote: __ why regulate? nobody regulates the return address on a letter sent via USPS. - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:00 AM Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 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 --__--__-- Message: 2 Date: Thu, 8 Jul 2004 12:00:55 +0100 From: Ben Merrills [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 NAT question Reply-To: [EMAIL PROTECTED] I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 firewall-ip D N 255.255.255.255 60665 Unmonitored tp2/tp2 firewall-ip D N 255.255.255.255 60646 Unmonitored tp1/tp1 firewall-ip D N 255.255.255.255 60649 Unmonitored Now, the Cisco phones are set to use nat (nat =3D 1) and in the SIP configuration, the phones are also configured for SIP. [tp1] type=3Dfriend secret=3Dtp1 host=3Ddynamic nat=3Dyes callerid=3DTest Phone 1 I can make calls out over the phones, but can't get anything back in. If I call voicemail say, then that's fine. But if I try and call another phone behind the firewall, it just sits there :/ IS there a specific port
Re: [Asterisk-Users] Kerry/Edwards campaign and VOIP
On Thursday 08 July 2004 09:55, Steve Totaro wrote: They could spoof their caller ID to the Bush/Cheney campain and call people at 3AM to ask for their support!!! HAHAHHAHAHAH +1 Funny -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audiocodes - Asterisk Implementation
Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 20587: Found Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:11 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:13 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:18 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:20 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:24 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:25 DEBUG[1133742896]: chan_sip.c:706 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' I am using CVS version Asterisk CVS-HEAD-06/18/04-11:53:43. I have tried changing just about every config option I can think of in both Asterisk and the Audiocodes box without any success. Any ideas? I have checked the web for documentation on this setup, and all I have found is that some people have it working, but that is about it, no details. Any help would be greatly appreciated. Thanks, Brian ___ 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: Some (lack of) answers regarding the wakeup call application...
Steve wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 06 July 2004 07:53 pm, Steve wrote: On Tuesday 06 July 2004 03:00 pm, Maron Kristófersson wrote: Also, I need a Linux tool to splice a series of gsm audio clips together in order to use one 'get_data' instead of multiple cat sound1.gsm target.gsm cat sound2.gsm target.gsm Maron cat sound1.gsm sound2.gsm sound3.gsm is easier. Haha, should only have had one A single means create a new file (over writing the old one if possible) and a means append to the file (creating a new one if it doesn't exist). That a short description of it means, I'm sure I've missed a few details but that close enough for government work. -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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] WellTech Wellgate 5250 E1 trunk gateway
Hi all, I would like to ask if anyone successfully connected a Welltech Wellgate 5250 E1 trunk gateway to Asterisk? Anyone can post a working config on the Asterisk or WG5250 would highly appreciated. Best regards. ___ 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] Can X100P/X101P detect reversal of line polarity?
On Thu, 2004-07-08 at 04:28, Mikael Magnusson wrote: On Thu, Jul 08, 2004 at 08:30:59PM +1200, Richard Scobie wrote: One thing to watch for here is RX gain if busydetect does not seem to be working after trying all the combinations. I had a 2 x X101P setup which busydetected perfectly - TX and RX gains were at the default levels. The X101Ps were replaced by a TDM card with 4 x FXO modules and with no config changes, busydetect stopped working. After incrementing in 1dB steps, an RX gain of 3.0 brought back reliable busydetect. I look forward to Rich Adamsons forthcoming writeup on setting up the gain distribution in an Asterisk system, to get everything working optimally. In Sweden we don't get a busy signal when the remote part hangs up. Instead remote hang up is signaled by reversing the polarity of the line. Can X100P/X101P detect polarity reversal when off hook, on hook, or both? I believe the chip is capable of that detection. It may need a driver tweak to get it there and into a meaningful message in zap. -- 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] Fax Detection
On July 7, 2004 09:19 pm, Matt wrote: Hi all I've tried Google, wiki and mailing list and IRC but still haven't gotten to the bottom of this. Hopefully someone might be able to help. I'm using telappliant to provide my inbound and outbound calls. I'm not familiar with teleppliant. Do you use a digium (zap) card? AFAIK, you need one and need faxdetect=yes in zapata.conf. When a fax comes in, does anything relevant get written to * console? * plays host to 30 cisco's and they are all working great using G711 A-law. I've managed to get SpanDSP to compile and install and I can send a receive a fax on a dedicated extension. What I'm trying to do now and can't seem to nail is getting an inbound fax to be detected and then handled. I've tried the examples from the wiki and the sites linked on the wiki; messed about trying my own weird and wonderful methods but still no joy. All the calls are using G711 A-law. Here is the test context I'm using XXX = hiden Exten = 08700686XXX,1,Goto(textextension,7000,1) [testextension] Exten = 7000,1,Answer Exten = 7000,2,Dial(SIP/104) Exten = fax,1,rxfax(/var/spool/asterisk/incoming/testfax.tif) Exten = fax,2,congestion Exten = fax,102,congestion Calls hit the testextension contect but don't get detected as a fax. Cheers Matt ___ 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 -- .. Ryan Courtnage Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ 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] Slackware 10.0 and asterisk and 2.4 vs 2.6
Hello, I am running Asterisk on Slackware 10.0 with the 2.4 kernel(default kernel) and it is very happy. Don't see too much difference from 9.1 except for the fact that most of the binutils have been updated and several of them run differently now(top, ps, ...) Haven't tried the 2.6 kernel yet, but may try it later. MATT--- -Original Message- From: Joe Baptista [mailto:[EMAIL PROTECTED] Sent: Thursday, July 08, 2004 10:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Slackware 10.0 and asterisk and 2.4 vs 2.6 I'm installing the new Slackware 10.0 distribution - but not sure if i should go with the 2.4 kernal - which i think is the default install - or the new 2.6 kernal? anyone running * and slackware 10.0 with 2.6 kernal? thanks joe ___ 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] Sip Peer Status
Hello, I am cruious what exactly status shows. If I do a sip show peers, I get this table: 2133/213310.10.60.9 D 255.255.255.255 5060 OK (95 ms) 2120/212010.10.60.2 D 255.255.255.255 5060 OK (112 ms) Now, if I exit asterisk, and ping from the same server, response times are never greater than 2ms. Interestingly enough, the one that shows up at a lower 95 ms is actually on a different switch with higher ping times than the 2120 peer. What gives, is this a bug? Thanks, - Brent ___ 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] VoicePulse Connect DID Problems
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I use IAX to VoicePulse Connect /w recent CVS HEAD with no problems... Except, of course, for voicepulse's service flaking out on a regular basis. - -jwb On Thursday 08 July 2004 01:51 am, Ken Wiesner wrote: I had similar issues. I ended up using SIP to connect to them. Everything was working fine until I did a recent CVS upgrade and now only the outbound calls work. When an inbound call comes in I get: Jul 8 00:14:22 NOTICE[1116941120]: chan_sip.c:6779 handle_request: Failed to authenticate user asterisk sip:[EMAIL PROTECTED];tag=as632396c0 Is this a related issue or am I doing something incorrect? Thanks, Ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of twisted Sent: Wednesday, July 07, 2004 4:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse Connect DID Problems On Wed, 2004-07-07 at 19:56, Andrew Joakimsen wrote: I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone. Yes. This is because they refuse to upgrade their servers to latest cvs, wether it be HEAD *OR* Stable. Voicepulse - if you are listening - this is a MAJOR issue that has been floating around for MONTHS now. We have tried to tell you, we have tried to contact you. If you happen to see this, UPGRADE YOUR SERVERS. Thank you, that is all. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA7WLWHyXYB+SEybkRAr4XAKCFgBMeF09a88j88tFAJrV5Mg3/OQCePD6i tgk99wdSI4h7jXkTkplIi8I= =wUDR -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] patlooptest output
On Wed, Jul 07, 2004 at 08:28:23PM -0600, Daniel Daley wrote: Does anyone know if patlooptest either doesn't work for fxo/fxs signaled channels or if you have to do it a different way? If I run You have to set your channels as clear channels for it to work (i.e., bchan == clear). It doesn't work with signaled channels. Matthew Fredrickson ___ 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] odd behavior - adtran ta 850 + t100p
[EMAIL PROTECTED] wrote: I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked to module 2 of the same fxs card then asterisk (correctly) only sees that module go off hook. When plugging a phone into any of the fxs pairs, I only get dial tone for a second or two and then I get silence. However, I can still dial extensions and get through. I'm not sure but maybe it is a config problem with the ta 850, as it takes a little more manual configuration than the ta 750 I worked with before. Anybody have any pointers? Here is the output on the console when I pick up a phone on module 1, and module 2, respectively: [EMAIL PROTECTED]:~# asterisk -r Asterisk CVS-HEAD-07/06/04-12:37:58, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-HEAD-07/06/04-12:37:58 currently running on slack1 (pid = 702) - Remote UNIX connection -- Starting simple switch on 'Zap/5-1' -- Starting simple switch on 'Zap/6-1' -- Starting simple switch on 'Zap/7-1' -- Starting simple switch on 'Zap/8-1' -- Hungup 'Zap/5-1' -- Hungup 'Zap/6-1' -- Hungup 'Zap/7-1' -- Hungup 'Zap/8-1' -- Starting simple switch on 'Zap/5-1' Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy -- Starting simple switch on 'Zap/6-1' -- Starting simple switch on 'Zap/7-1' Jul 6 22:30:03 WARNING[163850]: chan_zap.c:1288 zt_set_hook: zt hook failed: Device or resource busy -- Starting simple switch on 'Zap/8-1' -- Hungup 'Zap/5-1' -- Hungup 'Zap/6-1' -- Hungup 'Zap/7-1' -- Hungup 'Zap/8-1' -- Starting simple switch on 'Zap/6-1' -- Hungup 'Zap/6-1' Here is zaptel.conf: span=1,0,0,esf,b8zs loadzone = us defaultzone=us fxsks=1 fxoks=5-24 And here is zapata.conf: [channels] transfer=yes context=default language = en usecallerid = no hidecallerid = no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no signalling=fxs_ks echotraining=yes group = 0 channel = 1 context=trusted group = 1 signalling = fxo_ks rxwink = 300 channel = 5-24 Any help appreciated, -Jeff ___ 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 Well I tried setting up the the unused fxo ports, tried setting them to unused, and even moved the fxs cards around in the bank to see if it would make any difference. No joy though. Anybody know how to run some self tests on this bank to be sure its the problem? I'm pretty sure adtran will fix or replace the bank, but I'm sure they are going to want me to explain the problem but I'm not sure what info they'll need. ___ 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] Question about Cisco IP Phone 7960
You the MAN!!! I drop the P of the P0S3-07-1-00 Everything is golden now! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Thursday, July 08, 2004 9:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Question about Cisco IP Phone 7960 On 08/07/2004 at 08:21 Hall, Eric M. wrote: I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic Make sure you changed ALL the configs to point to the CORRECT image file.. OS79XX.TXT should contain P0S3-07-1-00 and your SIPmac or SIPDefault.cnf should contain image_version: P0S3-07-1-00 iirc the default in OS79XX.TXT is the unsigned image... HTH Andy ___ 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] i or s or whatever the invalid_exten is HELP !!!!!
Ok
So when i get into a context and dial a number that does not exist the 'i'
exten works fine.
I have a test exten that runs my menu system if i dial that then dial an
exten that does not exist ' i ' works.
BUT when i dial a number directly that does not exist the 'i' exten does
not get called i did try
exten = ${INVALID_EXTEN}, Do summit ( no good )
What i need is when a number that is not in a context is dialled to go to a
list of exten's and then redirect to the correct context . Mainly because we
have many different context's all starting with 2 different digits ie
65xx for one office
66xx for office 2
43xx for branch a
etc
Or can i test the first 2 digits of a number and send it to the right
context ? Seems a bit difficult when i can't even get * to let the number
come in.
I am using BT101's ( SIP ) and Asterisk CVS-HEAD-05/19/04-01:33:53
Best Regards
Simon
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[Asterisk-Users] Using Windows Messenger+Video in *
Has anybody used Windows Messenger with asterisk? All documents around (google - wiki - bugs.digium.com) say that asterisk supports windows messenger with video but i have no succes yet! I can establish connection with audio but no video yet. I've used a range of windows messengers from version 4.7 to 5.0.0482. - shabanip ___ 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] OH323-COMPILE
You are trying to compile an ancient asterisk-oh323 with fresh Asterisk code. It won't work. Download and install asterisk-oh323-0.6.3a. Also, download and compile the recommended versions of OpenH323/Pwlib (OpenH323/Pwlib 1.12.2/1.5.2 are too old). Michael. mohammad mirzaee wrote: HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07, 2004-07-02, .) 4- oh323 v.5-10 / oh323 v.5.9 5- my linux box is redhat 8.0 the error looks like the following: make[1]: *** [chan_oh323.o] Eroor 1 make[1]: Leaving directory '/root/asterisk/asterisk-oh323-0.5.9/asterisk-driver' make:*** [subdirs_all] Error 1 I think there is a mismatch between my oh323 and asterisk. But I donot know the excat asterisk CVS I will be waiting for your help warmest regards mohammad -- *** * * *G R E E C E * * * * EUROPEAN CHAMPION EURO 2004 * * * *** ___ 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] tdm400p static - out of ideas
We have not experience with Digium cards. However we had similar problems when installing legacy pbx. The problem: local ground. One easy way to test the local ground is with a voltmeter measure the voltage between the CO tip wire (in open loop state) and local ground. This must be less than 2 VDC and must be stable (not oscillations). Correcting the local ground all problems were fixed. When you plug a regular phone, the phone is floating and have not electrical contact with other devices, then there are not ground currents. My guess is that there are some hardware design that are more sensitive than others to ground currents. Hope this helps. Jorge Ryan Courtnage wrote: On July 8, 2004 03:22 am, Nicholas Bachmann wrote: Ryan Courtnage wrote: Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. Attempts to send outbound calls on any Zap channel will result in hearing a loud 'static' noise on the line. Let's look at some possibilities of line problems: What time does it stop answering? Is it ever during ALIT times (usually very early morning)? It's totally random - morning/evening/afternoon. Once it stops answering, that's it, a reboot or module-reload is needed. If ALIT for some reason prevents the card from answering, it should be able to recover and begin answering after the ALIT is complete. Have you tried calling the telco to see if it could be their problem? When the card goes into the non-functional state, I can plug a regular phone into any of the lines and make calls just fine. After verifying working lines and plugging them back into the tdm400p card, I still can't dial out (the Zap channel will answer, but I will hear only static, and the call to the pstn is never placed). As well, incoming calls will not be answered (* console will not even show the 'started simple switch on zap/x' message). How far away from the CO/mux are you? Not too sure - it's in downtown Calgary - so probably not far. There is the possibility that _something_ with the phone line is triggering the problem. Maybe it's some noise, an unexpected signal, some crosstalk ... things that will cause unexpected behavior ... but also things that shouldn't put the entire card into a non-functioning state. Have you tried a new/different card? If you didn't want to fork out the cash for a new one, you could try a X100P/knockoff* on one of the lines to see if that fixes the problem... if so you can deduce a bad card. I may have to push for a replacement tdm400p card from Digium. Nick *I usually don't recommend the knockoffs, but for a day of testing $10 sure beats $100... everybody else should support Digium! :-) An acquaintance who is having the same problem has reluctantly replaced his card with an openline4. I would like nothing more than to stick with Digium hardware - this thread and obtaining a replacement card is my last kick at the cat. ___ 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] Audiocodes - Asterisk Implementation
I have the same problem but with MP-124 FXS Gateway. Does anybody has it to work with *? - Original Message - From: Brian J. Rathman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 6:52 PM Subject: [Asterisk-Users] Audiocodes - Asterisk Implementation Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 20587: Found Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:11 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:13 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:18 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:20 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:24 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:25 DEBUG[1133742896]: chan_sip.c:706 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' I am using CVS version Asterisk CVS-HEAD-06/18/04-11:53:43. I have tried changing just about every config option I can think of in both Asterisk and the Audiocodes box without any success. Any ideas? I have checked the web for documentation on this setup, and all I have found is that some people have it working, but that is about it, no details. Any help would be greatly appreciated. Thanks, Brian ___ 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] Kerry/Edwards campaign and VOIP
Steve Totaro wrote: They could spoof their caller ID to the Bush/Cheney campain and call people at 3AM to ask for their support!!! - Original Message - From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 7:56 PM Subject: Re: [Asterisk-Users] Kerry/Edwards campaign and VOIP Bill Merriam wrote: I am trying find a way to help the local Kerry campaign and it occurs to me that VOIP and Asterisk could be a big help. I have never worked on a Bill, You'll find that the FEC has VERY strict guidelines regarding things like this. John That would be typical of the DNC. ___ 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] Rollover oddity
I had something similar happen -- or so I thought. Turns out my * wasn't configured right, and the call-waiting blip was generated by Asterisk as it was detecting ring on the second line. Without your extensions.conf and as much info as you can provide (hardware, extension phones, etc) nobody's going to be able to tell you more about your problem, though. -Original Message- From: Bob Bailey [mailto:[EMAIL PROTECTED] Sent: Thursday, July 08, 2004 4:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Rollover oddity Hello, I've got 2 analogue lines (from SBC) coming into a TDM22B. SBC have put rollover from the first to the second line. The rollover works fine when handsets are connected directly to the lines (ie when Asterisk is not involved), but when the lines are connected to Asterisk, the rollover fails: the caller just hears the line ringing, and the person on the first (busy) line hears call waiting interrupts. I have proved that it's the rollover that's not taking place (ie it is not that the rollover happens, but the second line isn't answered) So how (and why) does Asterisk affect the rollover in this way, and how can a busy line going through Asterisk 'look' different to the telco from a normal handset? And, of course... how do I solve this? Thanks Bob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-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] Audiocodes - Asterisk Implementation
Brian J. Rathman wrote: Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 20587: Found Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:11 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:13 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:18 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:20 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:24 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:25 DEBUG[1133742896]: chan_sip.c:706 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' I am using CVS version Asterisk CVS-HEAD-06/18/04-11:53:43. I have tried changing just about every config option I can think of in both Asterisk and the Audiocodes box without any success. Any ideas? I have checked the web for documentation on this setup, and all I have found is that some people have it working, but that is about it, no details. Any help would be greatly appreciated. Thanks, Brian Firmwire version? ___ 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] Small Linux Distro
FWIW, I'm using a crystalfontz 2x16 display and show a mini-log of ongoing calls by extension. It's not super-stable yet, but someday... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, July 08, 2004 6:21 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Small Linux Distro Thanks to everybody for your links and support. I think I have convinced all of the shareholders (tonight) that GPL is the way for anything we develop with regards to Asterisk, and they are going to put some more money into building a couple of prototypes boxes...(fun!!! hacking time!!!) So, once I have the hardware all setup and working properly, I'll post again here and find out what kind of features people would like to see in the LCD/button interface and what kind of statistics and options would be beneficial. Kind regards, Matt Riddell ___ 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] FINALLY! a good book about Asterisk.
Joe Babstock wrote: There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. And how do you know it's a good book? I wouldn't mind a review and I may purchase the book (I doubt I qualify as a reviewer as I haven't yet figured this VoIP stuff out yet). I'm not really sure a few pages qualifies for a review. BTW, please excuse me if Paul is a frequent contributor to the mail list. I just found the method of announcement a bit suspect (I'm not say Paul posted this either). -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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
