Re: [Asterisk-Users] Re: Re: quadbri NT_mode S-Bus Problem
I'm currently looking for a good solid solution that works here in the US with BRI-U NI-1 off a DMS100 or 5ESS. bkw - Original Message - From: Ben Bosshardt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:18 PM Subject: [Asterisk-Users] Re: Re: quadbri NT_mode S-Bus Problem Are you using Santis Siemens ISDN NT1 ? If yes , we have the same from Siemens Switzerland , What I've done is to get one cable from ISDN NT -- ISDN MODEM in * Machine ( HFC - S Modem Euro 30 - 40 ) and then used bristuff ( google for it ) , and used that , it just works! . I can send you my configs if you need som ehlp I gladly look at your config files to see what I have done wrong. At the moment the setup is hooked up that I can make inbound and outbound calls (from ISDN and SIP clients), just with the limitations as below : 1. On outbound calls, I get the normal rining call progress tone althought the the other party has not even been reached. This then changes from normal ringing suddenly to busy when the other party is sending a busy signal. I'd rather have the call progress send a busy signal right away. 2. Internal calls between two ISDN client phones on the S-bus is not possible. The phone rings but the call is dropped as soon as it is answered. Kind Regards, Ben ___ Asterisk-Users mailing list [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] Re: Updated Grandstream configurator
I was even considering going further and writing a crossplatform or a webapp for configuring. However I was thinking if someone has written some notes on the config file specification that could save a lot of time. I have no intention of competing with gsconfigure since I think it's an excellent app although I have to boot into windows to use it. Hehe, make it a web app *) and a special TFTP server that understands the Grandstream Options, so that the phones get only handed *.bin files if there is something to upgrade. Or is there any open source TFTP server available that can do this? *) Using DeStar as a web app for this should be quite simple. The web frame for adding configure objects (Phones, Lines, Permissions, or in this case Phone configs), editing them, deleting them, using the info in them to write config files is fully there. http://www.holgerschurig.de/destar.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Gui client
Beware: I might be biased! http://67.109.153.236/*web/ It edits extensions.conf after some customization.However unable to update sip.conf. I guess you made yourself lot's of work. However, there is already a PHP based web editor that works on this low level of editing *.conf directly. I personally don't see the benefit of this. To edit Asterisk on the *.conf level, you have to know how that works, e.g. all the subleties of the extension logic, of the options for the Dial command and so on. So you need training (or self-training by reading the Wiki) to be able to edit at this level. And if you know all of this, then you probably also know vi, emacs or joe. And then a web interface is suddenly too clumsy. Your mileage may, of course, vary. :-) http://67.109.153.236/asterisk-stat/cdr.php Link to the CDR Tool. Seems to be the one that I know already :-) http://67.109.153.236/cgi-bin/am/am-main.pl The perl based Asterisk GUI Management system. Help is available online in same panel. This code is a bit cumbersome and I am not going to attempt developing this. PHP is much more preferrable. This GUI has a non-describing structure. When I clicked Edit profile and did not enter a name, but pressed enter, I was presented with a web form. But I was not sure what I was editing. I felt like staying in limbo. Also the web interface looks a little boring. Once the code reaches some useful level, I am going to post the source code back here, through a download link. Which one is yours? Would you mind looking at http://www.holgerschurig.de/destar.html and install it? Sorry, I can't give you an link for an online presentation because I don't have access to some server where I can install it. Please look critically at the program and give me back any feedback. The program starts being usable, but generally I find every day some bugs and add every second day some features. My goal is to use this program in production in about one month's time. It's not PHP, it's Python. But when you look at some cfg_*.py file, you see that it is actually easy to add or modify stuff, even for PHP, Perl etc programmers. Im using very Pythonic techniques only in other files :-) This program is aimed to be a hand-holding program. Currently, you cannot edit *.conf files directly. Instead, you edit high level objects. E.g. a SIP phone, a CAPI line, test applications like echo etc. When you save, it will re-create all needed *.conf files for you. The program can display a description of what you edit and even hints for the lines (althought I didn't used this). There's infrastructore to translated all displayed text into any language. Later I'll add user stuff (like personal phone book, last N received calls, last N dialled calls), admin stuff (office phone book, CDR display analysis). Also I need more infrastructure, e.g. User Identities with logon. But most of the infrastructure is now there. Greetings, Holger -- MN Solutions GmbH http://www.mn-solutions.de Holger Schurig Dieselstr. 18 61191 Rosbach v.d.Höhe Tel: (+49) 6003 9141 0 Fax: (+49) 6003 9141 49 ___ Asterisk-Users mailing list [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] chan_vpb
Hi, Has anyone usingchan_vpb noticed that only one splash of ringback is provided to the PSTN? I have tried several different permutations in extensions.conf and interworking to mgcp sip and iax. I am using the Voicetronix supplied chan_vpb and asterisk from the 1.0 cvs source tree. thanks darren
Re: [Asterisk-Users] SIP to H323 call timeout
My SIP UA is an ATA186 and my H323 gateway is a Cisco 5300 and a Nextone. From: administrator tootai [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP to H323 call timeout Date: Tue, 20 Jul 2004 02:34:31 +0200 Fred Lee a écrit : Hi all, I have the following setup: UAs SER -- ASTERISK --GNUGK - GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out immediately, so call setup is never completed. On GNUGK the call request comes in followed by a normal call drop. Any ideas on what could be the problem ?? Do you use the h323 - Nufone? Is it a recent installation? If so, could be the problem that GW need FastStart and the * h323 don't send it. My asterisk configuration, debug and console output are as follow : SIP.CONF == [general] port = 5080 bindaddr = 10.10.1.170 context = to_GNUGK disallow=all allow=g729 H323.CONF === [general] port = 1720 allow = g729 gatekeeper = 64.80.103.12 allowgkrouted = yes context = to_SER EXTENSIONS.CONF [general] static = yes writeprotect = yes [to_GNUGK]] exten = _.,1,Dial(h323/[EMAIL PROTECTED]:1720,60,C) [to_SER] exten = _.,1,Dial(SIP/[EMAIL PROTECTED]:5060,60) DEBUG File == Jul 15 16:14:10 DEBUG[65541]: Check for res for Jul 15 16:14:10 DEBUG[65541]: is not a local user Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop: sip:[EMAIL PROTECTED];ftag=661806388;lr=on Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username: 15613021234 Jul 15 16:14:10 DEBUG[311316]: [EMAIL PROTECTED]:1720, timeout=0. Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t Jul 15 16:14:23 DEBUG[311316]: [EMAIL PROTECTED]:1720, timeout=0. Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter Jul 15 16:14:31 DEBUG[311316]: is not a local user Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found CONSOLE Output == *CLI -- Executing Dial(SIP/-08121388, h323/[EMAIL PROTECTED]:1720|60|C) in new stack -- Called [EMAIL PROTECTED]:1720 == No one is available to answer at this time -- Timeout on SIP/-08121388 == CDR updated on SIP/-08121388 _ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [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 _ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [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] Echo on a PRI
On Mon, 2004-07-19 at 19:52, TC wrote: Might as well come join the * SIG [EMAIL PROTECTED] bare your sole there ... This fragmentation helps us how? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls from H323 to SIP unsuccessful
Hello, I have managed to make calls from sip to h323 through Asterisk and Gnugk, but I cannot make calls from h323 to sip through Gatekeeper and Asterisk, the gatekeeper says "called party not registered"... does someone have a successful configuration fors this ? Thank you very much... Soumaya Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !
[Asterisk-Users] Grandstream transfer button
I am not having much luck searching on why the transfer button on my grandstream bt102 stopped working. anyone have any ideas where to look?
Re: [Asterisk-Users] Echo on a PRI
On Mon, 2004-07-19 at 19:12, David Goldfein wrote: Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Your PRI and the T1 itself cannot introduce echo on their own. What you may see though is that you are introducing a delay as you traverse the asterisk link. Asterisk will buffer 8 bits per channel from the PRI before it send it down the T1 line to the PBX. This is a new delay that is now added on to the latency your PBX introduces. A guess is that you also get the 2 machines fighting against each other on the echo. I doubt you can turn off echo cancel in the PBX so you should try turning it off in asterisk. It should help reduce some latency in asterisk and let the PBX handle the rest of the echo cancel on it's own. -- 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] Cisco 7960 SIP V6 and distinctive ring.
Scott
I managed to get the line working.but I can't hear a difference in cadence.
I read in the wiki there is a bug logged with cisco to make distinctive ring more
distinctive so i'm gonna wait till then before pursuing it further.
I'm going to focus on xml services in the short termgod these phones are powerful.
Thanks for your help.
P
-Original Message-
From: Scott Laird [mailto:[EMAIL PROTECTED]
Sent: Monday, July 19, 2004, 11:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP V6 and distinctive ring.
On Jul 19, 2004, at 9:29 AM, [EMAIL PROTECTED] wrote:
Hi
Can anyone with distinctive ring on their 7960's possibly post how
they've got it to work?
I understand that the ALERT_INFO variable is involved but using the
examples for the variable value from the WiKi I'm just getting an
error message from the Asterisk concole.
I'm setting it to 'Bellcore-dr1' through 'Bellcore-dr4'. I'm grabbing
the value out of Asterisk's database and sticking it into ALERT_INFO
like this:
[macro-setalertinfo]
exten = s,1,DBGet(ALERT_INFO=distinctivering/${CALLERIDNUM})
Works fine for me. You should also be able to do
'SetVar(ALERT_INFO=Bellcore-dr1)' without problems. Can you show us
the line that's generating errors?
Scott
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Re: [Asterisk-Users] Cheap PoE switches/injectors?
Jonathon Ebay items:- 5710513834 5710609468 P -Original Message- From: Jonathan Moore [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004, 12:09 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cheap PoE switches/injectors? Has anyone tried the new dlink powered switches? I remember seeing an online voip store selling these as a good option for providing power in a voip application. They were price at 1100 for a 24 port model. The lowest cost solution I have seen are the individual 3com power injectors which can be had for between $16-$25. I have done some minimal testing with one for use with wireless access points and it seems workable, although not a good solution for a high density environment. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Scott Laird [EMAIL PROTECTED]: On Jul 19, 2004, at 9:03 AM, [EMAIL PROTECTED] wrote: Look out for 3c17205 switches from 3com and read the QOS thread posting here at the moment. So $1600 for 24 ports. That's not *too* bad. HP seems to have a similar model (2626-PWR) for a similar price. 3com also seems to have a 24-port injector for $800. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users 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] SIP to H323 call timeout
Fred Lee a écrit : My SIP UA is an ATA186 and my H323 gateway is a Cisco 5300 and a Nextone. My question was which * h323 channel you're using? (h323 Nufone or oh323) Don't know about Cisco and Nextone but I also use an ATA186 as SIP UA with GnuGK and have this problem. If you install an earlier that 20/05/04 CVS asterisk version with H323 Nufone channel it works. Don't know how it works with the stable branch. Daniel From: administrator tootai [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP to H323 call timeout Date: Tue, 20 Jul 2004 02:34:31 +0200 Fred Lee a écrit : Hi all, I have the following setup: UAs SER -- ASTERISK --GNUGK - GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out immediately, so call setup is never completed. On GNUGK the call request comes in followed by a normal call drop. Any ideas on what could be the problem ?? Do you use the h323 - Nufone? Is it a recent installation? If so, could be the problem that GW need FastStart and the * h323 don't send it. My asterisk configuration, debug and console output are as follow : SIP.CONF == [general] port = 5080 bindaddr = 10.10.1.170 context = to_GNUGK disallow=all allow=g729 H323.CONF === [general] port = 1720 allow = g729 gatekeeper = 64.80.103.12 allowgkrouted = yes context = to_SER EXTENSIONS.CONF [general] static = yes writeprotect = yes [to_GNUGK]] exten = _.,1,Dial(h323/[EMAIL PROTECTED]:1720,60,C) [to_SER] exten = _.,1,Dial(SIP/[EMAIL PROTECTED]:5060,60) DEBUG File == Jul 15 16:14:10 DEBUG[65541]: Check for res for Jul 15 16:14:10 DEBUG[65541]: is not a local user Jul 15 16:14:10 DEBUG[65541]: build_route: Record-Route hop: sip:[EMAIL PROTECTED];ftag=661806388;lr=on Jul 15 16:14:10 DEBUG[65541]: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Jul 15 16:14:10 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:10 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:10 DEBUG[311316]: Host: 10.10.1.12:1720 Username: 15613021234 Jul 15 16:14:10 DEBUG[311316]: [EMAIL PROTECTED]:1720, timeout=0. Jul 15 16:14:13 DEBUG[213006]: Cleaning up our mess Jul 15 16:14:23 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:23 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:23 DEBUG[311316]: Host: 10.10.1.12:1720 Username: t Jul 15 16:14:23 DEBUG[311316]: [EMAIL PROTECTED]:1720, timeout=0. Jul 15 16:14:24 DEBUG[213006]: Cleaning up our mess Jul 15 16:14:31 DEBUG[311316]: SIMPLE DIAL (NO URL) Jul 15 16:14:31 DEBUG[311316]: type=h323, format=256, [EMAIL PROTECTED]:1720. Jul 15 16:14:31 DEBUG[311316]: Host: 10.10.1.12:1720 Username: h Jul 15 16:14:31 DEBUG[311316]: find_user() - decrement inUse counter Jul 15 16:14:31 DEBUG[311316]: is not a local user Jul 15 16:14:31 DEBUG[65541]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found CONSOLE Output == *CLI -- Executing Dial(SIP/-08121388, h323/[EMAIL PROTECTED]:1720|60|C) in new stack -- Called [EMAIL PROTECTED]:1720 == No one is available to answer at this time -- Timeout on SIP/-08121388 == CDR updated on SIP/-08121388 _ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [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 _ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [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] Calls from H323 to SIP unsuccessful
Hello Fathallah, Tuesday, July 20, 2004, 2:44:25 PM, you wrote: FS Hello, FS I have managed to make calls from sip to h323 through Asterisk and Gnugk, FS but I cannot make calls from h323 to sip through Gatekeeper and Asterisk, FS the gatekeeper says called party not registered... does someone have a FS successful configuration fors this ? Have you tried AcceptUnregisteredCalls=1 in your GnuGK conffile? -- Best regards, Olegmailto:[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] CID, international style?
On Mon, 2004-07-19 at 02:13, Steve Murphy wrote: I'm thinking of doing an app to work with the CID that's gotten from the Zap channel. All the CID's I see from within the US are 10 digit numbers. I'm out in the rural areas of the US, and no-one ever calls me from overseas. If they did, what would the CID look like? What does the CallerID look like overseas? How many countries provide it? CID is esentially part of any digital subscriber line anywhere in the world, so everybody uses it. On PSTN there are 3 (maybe more) standards: the way it's done in the US, UK and Sweden (which also is used in Denmark and the Netherlands). Asia might use something different again, or not. International calls CID look depends on the Telco you are connected to. If you are on PSTN, usually they would add the 00 or in the U.S. the 011 in front of the international dialcode and then let the number follow. On digital subscriber lines (at least the DSS1 specs), you will get the number without prefix and the information, what dialplan it belongs to (international, national, local, unscreened, etc.) The number itself depends on the dialplan used in that country (Ireland: variable, but national prefix + subscriber no. allways 6 digits, not counted the 0 in front, Denmark: fixed 8 digits, Norway: fixed 8 digits, Sweden: variable, Germany: variable) So all in all, your CID can have any length, depending on the countries dialplan, where somebody would be calling you from. Kind regards, Martin List-Petersen ___ Asterisk-Users mailing list [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] Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723
Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723 Thanks, From: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 19 Jul 2004 19:48:02 -0500 To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4610 - 12 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: Re: Cisco 7960 SIP V6 and distinctive ring. (Sam Tilders) 2. Re: Asterisk + NEC Electra Elite IPK Integration (Jason Kawakami) 3. RE: Polycom IP 500 Voicemail (Wiley E. Siler) 4. Re: uip200 clips audio prompts (Ryan Courtnage) 5. MWI - Config Stupidity or Notify Issues? (Robert Jackson) 6. RE: RE:RE: [Asterisk-Users] Codecs - Advantages (Wiley E. Siler) 7. RE: Polycom IP 500 Voicemail (Wiley E. Siler) 8. RE: Polycom IP 500 Voicemail (Chris A. Icide) 9. Echo on a PRI (David Goldfein) 10. Suscription (Carlos Clemares) 11. RE: Echo on a PRI (Wiley E. Siler) 12. Re: SIP to H323 call timeout (administrator tootai) --__--__-- Message: 1 Date: Tue, 20 Jul 2004 09:25:11 +1000 From: Sam Tilders [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Cisco 7960 SIP V6 and distinctive ring. Reply-To: [EMAIL PROTECTED] On Mon, Jul 19, 2004 at 02:09:34PM -0700, asteriskstuff @ ziplip. com wrote: Thanks..it's a numeric value!! in the wiki it refers to a text field!! The wiki is also correct... I have: exten = 101,1,SetVar(ALERT_INFO=Bellcore-dr1) And that works fine. What was the error message you were getting? -- -- Sam Tilders [EMAIL PROTECTED] (Move to Jupiter) --__--__-- Message: 2 From: Jason Kawakami [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Date: Mon, 19 Jul 2004 17:28:09 -0600 Reply-To: [EMAIL PROTECTED] Date: Mon, 19 Jul 2004 14:54:44 -0500 From: Christopher L. Wade [EMAIL PROTECTED] Organization: Unistar-Sparco Computers, Inc. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Reply-To: [EMAIL PROTECTED] Would the TLI(2)-U10 ETU work as well? That is a 2 port analog tie line card, I don't think that Digium has a card that can be set up as an analog 4W EM trunk. bad idea anyway, the t-1 will be a much better interface and if you ever press the eject on the IPK you could use the t-1 as a PSTN interface. --__--__-- Message: 3 Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail Date: Mon, 19 Jul 2004 16:28:25 -0700 From: Wiley E. Siler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Mine does the same. Once in Message center I can choose selection 1.Message Center and then soft key Select.Then I select the registered line that I want to check voice mail on. That is no less than 4 key strokes just to get into your voice mail. Not many to me but tons to an unskilled user. However, in the documentation regarding the bypassInstantMessage value, supposedly, setting bypassInstantMessage to 1 is supposed to allow you to go right into voice mail without navigating the Message Center. That is the big question on my mind at this point. I have yet to get this to work and I also don't think I am receiving any SIMPLE messages ti show me that I have messages waiting. Do you get a message waiting indicator? W -Original Message- From: Chris A. Icide [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 3:03 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail On 12:40 PM 7/19/2004, Wiley E. Siler wrote: My Polycom is on loan as a demo and I assume it is one of the first revision models. In fact it shows as Rev A on the back of the phone. I have all the same buttons you listed save for the Messages button. The 3rd from the bottom on the right column of buttons sayd Voice Mail on my version. That corresponds to the location of your button that says Messages. I assume this was changed by Polycom since their phone has other messaging capability (isntant message for instance) and it was easier to use Messages and unify the meaning instead of Voice Mail and lock it into one type of messaging. Does your Messages button dump you right into voice mail
Re: [Asterisk-Users] chan_capi: sending incoming calls to different contexts
On Mon, 2004-07-19 at 08:31, Holger Schurig wrote: Not sure if it works for you, but the simplest way is: [capi-in] exten = DIDNUM1,1,DoSomething exten = DIDNUM2,1,DoSomething exten = DIDNUM3,1,DoSomething where DIDNUMX is the direct indial number. Much nicer than goto statements with complicated splits. AFAIK you have only a DIDNUM if you have DID, that is with ISDN P2P, but not with P2MP. Or am I wrong? Are the multiple MSNs handled like DIDs? DID=Durchwahlnummern DID=Direct Inward Dialing, yes, but there is not much difference between the configuration of P2MP and P2P. The snippet shown is correct. It depends though a bit on the telco, what number they are sending. Most telco's send the full number, when talking P2MP, and only the last digits when talking P2P. Eircom only sends the last 4 digits, no matter what (don't ask me why) on P2MP. Haven't tried on P2P yet, since they are nasty slow in processing their orders. Anyhow the example up there fits for anything, you just need to figure out what your Telco is sending to you :o) Kind regards, Martin List-Petersen ___ Asterisk-Users mailing list [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] Calls from H323 to SIP unsuccessful
yes I tried it and doesnt work :( -- Oleg A. Arkhangelsky [EMAIL PROTECTED] a écrit : Hello Fathallah, Tuesday, July 20, 2004, 2:44:25 PM, you wrote: FS Hello, FS I have managed to make calls from sip to h323 through Asterisk and Gnugk, FS but I cannot make calls from h323 to sip through Gatekeeper and Asterisk, FS the gatekeeper says called party not registered... does someone have a FS successful configuration fors this ? Have you tried AcceptUnregisteredCalls=1 in your GnuGK conffile? -- Best regards, Oleg mailto:[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 Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.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] quadbri NT_mode S-Bus Problem
The HFC-S ISDN card is the same as the quadbri (basically).
Difference is just, that the quadbri has 4 ports and possibility for
termination and power on the ports, if necessary.
Kind regards,
Martin List-Petersen
On Tue, 2004-07-20 at 03:41, Junaid Uppal wrote:
Are you using Santis Siemens ISDN NT1 ? If yes , we have the same from
Siemens Switzerland , What I've done is to get one cable from ISDN NT
-- ISDN MODEM in * Machine ( HFC - S Modem Euro 30 - 40 ) and then
used bristuff ( google for it ) , and used that , it just works! . I
can send you my configs if you need som ehlp
regards
~uppal
On Sun, 18 Jul 2004 23:47:47 +0200, Ben Bosshardt
[EMAIL PROTECTED] wrote:
What type is your ISDN house telephone system?
Without more specific information all we can do is guess...
Our system is a just the basic subscription to SWISSCOM, which is the main
phone company in Switzerland. We have BRI with 2 Channels which can be used
simulaniously and a Siemens NT that has only the function of feeding our
S-bus with 4 phones connected.
For a sollution to 1 ... drop the r option of dial...
exten = _X.,1,Dial(Zap/g1/${EXTEN})
I will give it a try.
You might need pridialplan/prilocaldialplan set to local for local
calls... or both to unknown... just experiment with those values.
I am still looking for any documentation regarding the use of
pridialplan/prilocaldialplan. I don't know how to find out what SWISSCOM
requires.
Thanks for your help.
Ben
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Re: [Asterisk-Users] Re: Re: quadbri NT_mode S-Bus Problem
I think you are still stuck with CAPI cards then, that support DMS100 or 5ESS. the quadbri/octobri/hfc-s drivers so far only support DSS1 if i'm not mistaken. Kind regards, Martin List-Petersen On Tue, 2004-07-20 at 07:17, Brian K. West wrote: I'm currently looking for a good solid solution that works here in the US with BRI-U NI-1 off a DMS100 or 5ESS. bkw - Original Message - From: Ben Bosshardt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:18 PM Subject: [Asterisk-Users] Re: Re: quadbri NT_mode S-Bus Problem Are you using Santis Siemens ISDN NT1 ? If yes , we have the same from Siemens Switzerland , What I've done is to get one cable from ISDN NT -- ISDN MODEM in * Machine ( HFC - S Modem Euro 30 - 40 ) and then used bristuff ( google for it ) , and used that , it just works! . I can send you my configs if you need som ehlp I gladly look at your config files to see what I have done wrong. At the moment the setup is hooked up that I can make inbound and outbound calls (from ISDN and SIP clients), just with the limitations as below : 1. On outbound calls, I get the normal rining call progress tone althought the the other party has not even been reached. This then changes from normal ringing suddenly to busy when the other party is sending a busy signal. I'd rather have the call progress send a busy signal right away. 2. Internal calls between two ISDN client phones on the S-bus is not possible. The phone rings but the call is dropped as soon as it is answered. Kind Regards, Ben ___ Asterisk-Users mailing list [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] what is :
what is your current flashware? didi you spill anything on the phone? Have you tried going back 1 flash and then testing the button and the flash your phoneagain and test? ___ Asterisk-Users mailing list [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] Calls from H323 to SIP unsuccessful
Fathallah Soumaya wrote: yes I tried it and doesnt work :( you didn't forget setting the prefix in oh323.conf? -- Oleg A. Arkhangelsky [EMAIL PROTECTED] a écrit : Hello Fathallah, Tuesday, July 20, 2004, 2:44:25 PM, you wrote: FS Hello, FS I have managed to make calls from sip to h323 through Asterisk and Gnugk, FS but I cannot make calls from h323 to sip through Gatekeeper and Asterisk, FS the gatekeeper says called party not registered... does someone have a FS successful configuration fors this ? Have you tried AcceptUnregisteredCalls=1 in your GnuGK conffile? -- Best regards, Oleg mailto:[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 Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
On Mon, 2004-07-19 at 13:36, Christopher L. Wade wrote:
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't think I'll need the U30, but I'm not entirely sure.
Thanks,
Chris
I used the DTI-U10 (DTI-24-U10). Got it from GTS Telephone
Inc.(732-323-8620) for $300.00 (reconditioned).
My voice t1 comes into asterisk via the first T100P, and attaches to the
nec t1 via the second T100P using em_wink (as a trunk).Then with LCR I
make it add a 9 to the outgoing trunk so asterisk will route it to the
T1.
I grouped the channels that sales calls come into, and I grouped the
channels that go to the nec, so I could use a dial string like:
[sales]
exten = s,1,Playback,transfer|skip ; Please hold while...
exten = s,2,Dial,zap/g7/210 ; Ring, Nec sales group
exten = s,3,Hangup
and to ring extensions on the nec I did this:
; nec bridge
exten = _1XX,1,Dial(zap/g7/${EXTEN})
g7 (group7 is the T1 trunk); Extension 210 is a virtual extension set to
ring 5 other nec extensions; and the 1XX will match extensions in the
100 range that are not on the asterisk.
I should get started on the doc's didn't realize how far I'd come
till now.
Problems I still have:
1. If someone dials slowly from an nec extension - the nec sends the
first group, asterisk then tells them the number is not in service.
2. IAX2 connection to remote office is still choppy occasionally.
Don't know if the pix 501 is getting overwhelmed by the encryption of
voice packets or what?
t o n y
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[Asterisk-Users] callparking vs calltransfer
HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad
Re: [Asterisk-Users] Calls from H323 to SIP unsuccessful
Hello again, Do I need really a prefix? can you tell me exactly where shall I put a prefix in the oh323.conf? and also what are the sections concerned in the gnugk.ini?? I would be very grateful Thanks a lot Soumaya you didn't forget setting the prefix in oh323.conf? -- Oleg A. Arkhangelsky [EMAIL PROTECTED] a écrit : Hello Fathallah, Tuesday, July 20, 2004, 2:44:25 PM, you wrote: FS Hello, FS I have managed to make calls from sip to h323 through Asterisk and Gnugk, FS but I cannot make calls from h323 to sip through Gatekeeper and Asterisk, FS the gatekeeper says called party not registered... does someone have a FS successful configuration fors this ? Have you tried AcceptUnregisteredCalls=1 in your GnuGK conffile? -- Best regards, Oleg mailto:[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 Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo! Messenger sur http://fr.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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Yahoo! Messenger: dialoguez instantanément avec vos amis. Téléchargez GRATUITEMENT sur http://fr.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] Echo on a PRI
On the subject of echo on a PRI, I too get this, but only when calling people in certain rate centers. Calls within my LATA (primarily VZ) are completely free of echo. Calls to a neighboring LATA (all Sprint) have echo on almost every rate center. I wish I knew more about this so I could rip Sprint a new one and tell them to fix their trunking, but... -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, July 20, 2004 7:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Echo on a PRI On Mon, 2004-07-19 at 19:12, David Goldfein wrote: Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Your PRI and the T1 itself cannot introduce echo on their own. What you may see though is that you are introducing a delay as you traverse the asterisk link. Asterisk will buffer 8 bits per channel from the PRI before it send it down the T1 line to the PBX. This is a new delay that is now added on to the latency your PBX introduces. A guess is that you also get the 2 machines fighting against each other on the echo. I doubt you can turn off echo cancel in the PBX so you should try turning it off in asterisk. It should help reduce some latency in asterisk and let the PBX handle the rest of the echo cancel on it's own. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] Voicetronix
Hi. With voicetronix Openswitch12, I have installed the latest drivers, Asterisk 1.0-RC1 and so far so good. I want the simplest of all cases to begin, all 12 channels FXS. But, it doesn't work. When I pick up the phone, and dial 1, which in my extensions.conf should make my sip phone ring, asterisk doesn't register that I've pushed the 1 on the analoge phone. But Asterisk has registered that I've picked up, and is sending the dialtone. If I try to ring to it from my sip phone, the analouge phone rings for an instance then a hangup is done. If I stop Asterisk, rmmod vpbhp, and insmod vpbhp, and try again, sometimes it works, and I can ring, but if I make the call from analouge to sip, then I can hear nothing in the sip phone, in the analouge perfect. If I call from the sip phone, bouth parties hear perfect. Even worse if I make a call from analouge to analouge, I hear perfect in bouth phones. But when I hang up, it is not registered, and there is a bridged call left in Asterisk and only way to get rid of it is to close Asterisk. My conf files are below: extensions.conf [vpb-fxs] exten = s,1,Wait,4 exten = s,2,Answer exten = s,3,Hangup ; call to sip, dial 1 exten = 1,1,Wait,2 exten = 1,2,Dial(SIP/116,30,t) ; to make call analouge to analouge (line 3) dial 2 exten = 2,1,Wait,2 exten = 2,2,Dial(vpb/1-3/,30,t) [from-sip] exten = _41,1,Dial(vpb/1-1/,30,t) exten = _42,1,Dial(vpb/1-2/,30,t) exten = _43,1,Dial(vpb/1-3/,30,t) vpb.conf [general] cards = 1 type = v12pci [interfaces] board = 1 context = vpb-fxs mode = dialtone channel = 1 channel = 2 channel = 3 channel = 4 channel = 5 channel = 6 channel = 7 channel = 8 channel = 9 channel = 10 channel = 11 channel = 12 ___ Asterisk-Users mailing list [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] Echo on a PRI
I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Welcome to the club. ;) You have the same exact problem I've got. The only difference is I'm using dual Xeon 2.4s and a Supermicro SuperWorkstation 7033A-T (X5DAL-TG2 motherboard http://supermicro.com/products/motherboard/Xeon/E7505/X5DAL-TG2.cfm ). Echo training=800 on a recent CVS helped, but did not totally resolve the issue. Best regards, Ryan Thrash ___ Asterisk-Users mailing list [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] Echo on a PRI
On Tuesday 20 July 2004 06:39, Steven Critchfield wrote: Might as well come join the * SIG [EMAIL PROTECTED] bare your sole there ... This fragmentation helps us how? You know, I was wondering the same thing -- I got subscribed to it and I do have to say that the technical discussion there is better than anything I've seen here, although the conclusions are the same... :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on a PRI
On Tuesday 20 July 2004 09:04, Troy Settle wrote: On the subject of echo on a PRI, I too get this, but only when calling people in certain rate centers. Calls within my LATA (primarily VZ) are completely free of echo. Calls to a neighboring LATA (all Sprint) have echo on almost every rate center. I wish I knew more about this so I could rip Sprint a new one and tell them to fix their trunking, but... Are you sure it's Sprint's fault? I mean perhaps calling within your own LATA has less delay than calling neighbour LATAs and, combined with the delay that the T100P/TE405P introduces, presents enough delay to perceive echo... IME It's not a telco problem. I also have echo when calling certain numbers, both within and outside of my own LATA (at least I think it is) ... My conclusion is that *s echo cancellation is very hardware-specific, but with echotraining=800 it's good enough for what we use it for. The only problem with echotraining=yes is the 8/10s delay before audio is heard -- sometimes the start of conversations is cut off since our receptionist doesn't have to life a receiver from the cradle to her ear. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on a PRI
I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Oops. I need to correct my last post: I don't have the PBX in the mix. My config is dual Xeon 2.4s, 1GB RAM, HW SATA RAID, SuperMicro X5DAL-TG2 motherboard connected to: Telco - PRI (T100P) - Asterisk - SIP Phones (Budgetones 102s/Snom 200) The premise is still the same though: echo present despite our digital PRI that *should* make this impossible. It's usually only echo on our side when calling out as has been discussed here previously ad nauseum with no one being able to really figure out its source. I wish I knew where to really start poking around to try to help get to the bottom of this. Best regards, Ryan Thrash ___ Asterisk-Users mailing list [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] Modem chipset Intel
Hi, I've a question. I need a fxo card to use with *, but i saw on site that also modem with chipset intel works. My question is: are there other chipset (beyond to written ones), intel or not, that works? If not, has anyone from Italy bought a modem with chipset intel 537 or md3200? because i don't succeed to find it!! Thanx, Bob __ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27 luglio. Per te gratis il modem in comodato e l'attivazione. In piu' navighi a soli 1,5 euro l'ora per i primi tre mesi. Cosa aspetti? Attivala subito! http://abbonati.tiscali.it/adsl/prodotti/640Kbps/ ___ Asterisk-Users mailing list [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] callparking vs calltransfer
A call transfer in * is a blind transfer where there is an attempt to bridge the other end of the call with another extension in your dial plan. Call parking is placing the other end of the call into a holding area where they can listen to MOH until somebody picks up the parked call. Transferring is used in parking because you transfer the call to your parking extension. -Seth On Mon, 2004-07-19 at 21:45, mohammad mirzaee wrote: HI ALL; Anybody can explain the difference between call parking vs call transfer Regards mohammad -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callparking vs calltransfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 19 July 2004 09:45 pm, mohammad mirzaee wrote: HI ALL; Anybody can explain the difference between call parking vs call transfer Regards mohammad Sure, Call Parking is when you place a call, on hold, but released from your phone, at a public location, where everyone can get to it. Call Transfer is when you actually transfer a call to a different location. But then you have Supervised Transfer where you can talk to the other party before dropping the call on them, and Unsupervised Transfer which means it's gone the moment you punch the extension, or in the case ofVoIP phones, when you press the send/ok button. - -- 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) iD8DBQFA/R0wljK16xgETzkRAqMGAKCnwxUvkvaEPmMMSAYDVRaSt4/H3ACgzLdr sj/FEBM9omQhI5GzT3J01DA= =fR+6 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wireless SIP Phones
Hello, I found serveral discussions about the Zyxel ePhone Prestige P2000W and the WiSip from Pulver Innovations on this mailings list but still have some questions: 1) are there other affordable wireless SIP Phones on the market? I haven't seen or found anything else till now ... 2) is p2000w and wisip the same hardware?? so could I use firmware from both companies regardless of what phone I buy?? 3) does any of these phones have major bugs or will it be usable in a productive environment without getting mad or sleepless ?? 4) any security issues with these phones? Last but not least does anyone who knows both phones recommend anyone of these?? Or should I just buy the cheaper one? And one offtopic question ... does anyone know or can recommend a gsm to pbx adapter which costs less than 300 Euro?? Best regards g23 ___ Asterisk-Users mailing list [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] NAT problems with ZIP 4x4
I'm trying to get a ZIP 4x4 working behind a NAT server, talking to * on a public address. When I use the same sip.conf configuration (and same NAT server) that works for Grandstream and Sipura phones, the 4x4 can register and make calls, calls *to* the 4x4 do not make it to the phone. I can see from the sip trace that the sip packets to the phone are being retried by *, but I don't understand why. I can only assume, since it works for other phones, the problem is in the phone config and not *. Would anyone who has experience getting this to work, be willing to share their wisdom? 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
Re: [Asterisk-Users] Asterisk Gui client
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 20 July 2004 03:33 am, Holger Schurig wrote: Would you mind looking at http://www.holgerschurig.de/destar.html and install it? Sorry, I can't give you an link for an online presentation because I don't have access to some server where I can install it. Please look critically at the program and give me back any feedback. The program starts being usable, but generally I find every day some bugs and add every second day some features. My goal is to use this program in production in about one month's time. A program that can hold you hand as you go along is very nice. What you could also do is add help screens that gives more in depth descriptions. But I'm looking forward to see your product as it grows! - -- 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) iD8DBQFA/R+sljK16xgETzkRAsG9AKCAsl3fEWSmzjom9Ick+yJboGPdugCgs4WV 41FmmNZeYv9imQjbPqkiP/A= =x1l7 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codec translate
HI ALL; Is astersik enable to translate between different codecs. I have couple ofSIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad
Re: [Asterisk-Users] chan_vpb
VoiceTronix and I debugged some issues with the 1.0 driver. The one that's in HEAD is incompatible but has several changes. As much as it pains me to say it, the HEAD source is much more stable that the STABLE source. Chris On Tue, 2004-07-20 at 18:39 +1000, Darren McIntosh wrote: Hi, Has anyone using chan_vpb noticed that only one splash of ringback is provided to the PSTN? I have tried several different permutations in extensions.conf and interworking to mgcp sip and iax. I am using the Voicetronix supplied chan_vpb and asterisk from the 1.0 cvs source tree. thanks darren -- Chris Tooley / Network and Development Services Networking Technologies Resource Center, LLC (NTRC) 8650 Spicewood Springs Road, Suite 105 Austin TX 78759 512-250-8985 / Fax 512-250-5909 www.ntrc.net / www.ntrcstore.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] Random Dropped Called
I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs which may indicate a problem Jul 15 15:32:41 WARNING[11276]: PRI: !! Got reject for frame 30, retransmitting frame 30 now, updating n_r! Jul 15 17:03:20 WARNING[11276]: PRI: !! Got reject for frame 95, but we only have others! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 124 now, updating n_r! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 125 now, updating n_r! Jul 15 17:11:56 WARNING[11276]: PRI: Read on 66 failed: Unknown error 500 Jul 15 23:08:37 WARNING[5126]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 30406 (Response) Jul 16 05:39:08 NOTICE[11276]: PRI got event: 8 on span 1 Jul 16 06:25:04 NOTICE[5126]: Request to schedule in the past?!?! Jul 17 14:43:43 WARNING[11276]: Ring requested on channel 1 already in use on span 1. Hanging up owner. This issue has had me baning my head on my desk for weeks, any information that you may have that could clear this up will be much appreciated. --Paul M. Oster [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] Echo on a PRI
On Tuesday 20 July 2004 09:22, Ryan Thrash wrote: Oops. I need to correct my last post: I don't have the PBX in the mix. My config is dual Xeon 2.4s, 1GB RAM, HW SATA RAID, SuperMicro X5DAL-TG2 motherboard connected to: Telco - PRI (T100P) - Asterisk - SIP Phones (Budgetones 102s/Snom 200) The premise is still the same though: echo present despite our digital PRI that *should* make this impossible. It's usually only echo on our side when calling out as has been discussed here previously ad nauseum with no one being able to really figure out its source. I wish I knew where to really start poking around to try to help get to the bottom of this. No, the PRI does NOT make echo impossible. It makes it highly unlikely that YOU will generate echo. You never hear echo YOU generate; you hear the echo being generated on the other side. Has anyone you've called complained of echo? My * servers are SuperMicro 7043P-8R; single Xeon 2.8 HT processor in a dual-capable X5DP8-G2. More than enough balls to get the job done, but perhaps some PCI issues? Norstart Meridian MICS (12 PSTN trunk lines) - Adit600 FXS - T100P - IAX2 - TE405P - Bell Canada PRI The IAX2 link is only one hop (* server connected to each side over a PairGain Megabit Modem 300S on a dedicated ethernet port). Long distance calls are through Nufone and my internet link to Nufone is 8 hops. I have never heard compliant of echo when calling long distance, only through our local PRI. 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
[Asterisk-Users] Asterisk CVS compile error YDL 3.0.1
Hi, i'm trying to compile Asterisk under YDL 3.0.1, libpri, zaptel compile ok, but at make install in asterisk give me this error, have an idea because it can be? Thanks in advance. k\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -Wall -Werror -fPIC -O3 -march=ppc -funroll-l oops -fomit-frame-pointer -c -o anaFilter.o anaFilter.c cc1: invalid option `arch=ppc' make[2]: *** [anaFilter.o] Error 1 make[2]: Leaving directory `/usr/src/asterisk/codecs/ilbc' make[1]: *** [ilbc/libilbc.a] Error 2 make[1]: Leaving directory `/usr/src/asterisk/codecs' make: *** [subdirs] Error 1 Adrià Vidal [EMAIL PROTECTED]
[Asterisk-Users] New CVS version
I yesterday brought up to date the version of * the CVS and now I have a problem. I cannot effect the RELOAD that * it breaks. Somebody can help or say as to load new users without stopping * ? Thank´s Excuse my English Joao Carlos Moura ___ Asterisk-Users mailing list [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] Zap channels not working in morning
I set up an Asterisk system here, running 4 FXO channels. 2 are currently configured for use. It seems that overnight the Zap channels stop responding to calls, and will not run outgoing calls. The Asterisk screen does not show them as active, and no response occurs on the screen when they stop responding. I checked the log files but nothing seems to be happening out of the ordinary. I'm using a SPA-2000 to connect the extension analog phones to it. It works perfect all day then it stops responding for no apparent reason. It stayed up yesterday morning fine, but it was reset the day before. This morning it stopped working again. Any ideas? A reboot solves it every time but I'd rather not do that if I didn't have to. 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] Echo on a PRI
On the subject of echo on a PRI, I too get this, but only when calling people in certain rate centers. Calls within my LATA (primarily VZ) are completely free of echo. Calls to a neighboring LATA (all Sprint) have echo on almost every rate center. I wish I knew more about this so I could rip Sprint a new one and tell them to fix their trunking, but... Troy, Given the reseach that is currently going on, etc, I would not bet any more then a cup of coffee that Sprint (or any other carrier) has an echo problem right now. There _appears_ to be an issue with the echo cancellation routines in asterisk that is impacting more then a couple of implementations. The research to date suggests the current echo canceler works well in some cases, and not so well in other cases. In other words, there are a certain set of undocumented/unknown conditions (and/or PC systems) that fall within the thresholds of the current canceler that work, and other conditions (and PC systems) that fall outside the limits of the canceler that are less then acceptable. The limits and thresholds are _not_ black white and may end up being one of the more difficult problems to solve within asterisk. (E.g, it's entirely possible that your calls via Sprint fall outside the limits of *'s canceler.) As you've probably seen earlier on this list, there is a high probability that internal system issues (eg, interrupt servicing latency, possibly PCI bus issues, etc) that are impacting this in _some_ specific cases. In some cases, swapping the motherboard did in fact impact the cancellation quality. However, be very carefull not to read anything more into that at this time. There is no one at this time that knows factually what those limits, thresholds, etc, happen to be (not even Mark). Rich ___ Asterisk-Users mailing list [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] 7960 Dynamic DNS?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 3:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7960 Dynamic DNS? I can't think of any router that supports this You could put it in as a request to www.sveasoft.com for their firmware for the wrt54g (great box...runs linux and lots of features and functionality). Not only does the Sveasoft firmware already support dynamic DNS, the original Linksys firmware does as well. It was very common junk router feature (and by junk I mean anything you can buy at Staples that claims to be a router). Daryl ___ Asterisk-Users mailing list [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] codec translate
HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Yes. Also, Google works pretty good too. A simple Google Search for: Asterisk Translate Codec, would have returned a lot of useful searches. I included the link below in case you didn't know where/what google is. http://www.google.com/search?hl=enlr=ie=UTF-8q=Asterisk+Translate+codecbtnG=Search The sixth result looks like a winner. Additionally, Read the WiKi. http://www.voip-info.org - 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] Random Dropped Called
Run zttool and see if you the T1 card is missing interrupts. If so, put the following statement in your rc.local : # unmask interrupts /sbin/hdparm -u1 /dev/hda This will tell the ide driver not to mask interrupts while servicing disk i/o and the missing interrupts on your T1 card will likely go away. If this isn't the problem, zttool might still give you a hint if there are problems on the PRI itself. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 20 Jul 2004, Paul Oster wrote: I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs which may indicate a problem Jul 15 15:32:41 WARNING[11276]: PRI: !! Got reject for frame 30, retransmitting frame 30 now, updating n_r! Jul 15 17:03:20 WARNING[11276]: PRI: !! Got reject for frame 95, but we only have others! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 124 now, updating n_r! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 125 now, updating n_r! Jul 15 17:11:56 WARNING[11276]: PRI: Read on 66 failed: Unknown error 500 Jul 15 23:08:37 WARNING[5126]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 30406 (Response) Jul 16 05:39:08 NOTICE[11276]: PRI got event: 8 on span 1 Jul 16 06:25:04 NOTICE[5126]: Request to schedule in the past?!?! Jul 17 14:43:43 WARNING[11276]: Ring requested on channel 1 already in use on span 1. Hanging up owner. This issue has had me baning my head on my desk for weeks, any information that you may have that could clear this up will be much appreciated. --Paul M. Oster [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
Re: [Asterisk-Users] Wireless SIP Phones
Bodo Hahnke wrote: Hello, I found serveral discussions about the Zyxel ePhone Prestige P2000W and the WiSip from Pulver Innovations on this mailings list but still have some questions: 1) are there other affordable wireless SIP Phones on the market? I haven't seen or found anything else till now ... AFAIK not in large quantities - seen some other phones on paper, don't know if they exist though... 2) is p2000w and wisip the same hardware?? so could I use firmware from both companies regardless of what phone I buy?? The firmwares of the p2000W and the WiSIP are interchangeable. So yes. 3) does any of these phones have major bugs or will it be usable in a productive environment without getting mad or sleepless ?? The P200W and the WiSIP have quite a lot of bugs and usability issues. I would not use them in a productive environment unless it is with people with a technical background. The firmware and the phone are not there yet. The only Wireless SIP phone I would use in a productive environment would be the Cisco 7920. 4) any security issues with these phones? WEP is NOT secure - if you need security use wired and encrypted communication. Last but not least does anyone who knows both phones recommend anyone of these?? Or should I just buy the cheaper one? Buy the one you like better and use the WiSIP firmware on Monday's and Wednesday's then change it to ZyXEL for the rest of the week. :-) hope this helps Dominique ___ Asterisk-Users mailing list [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] Sound files - uncompressed versions available?
Hi, When listening to GSM-compressed voice prompts from either G.729 or iLBC codec, the sound quality is distinctly sub-optimal due to the use of multiple transcoding. Are the standard Asterisk sound files available in uncompressed format? - I have no problems with disk-space... PS Am aware that John Todd makes his extras available in uncompressed format: http://www.loligo.com/asterisk/sounds/AIF/ Thanks a lot, Fran. ___ Asterisk-Users mailing list [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] collect calls
Is it possible to set in Asterisk? Not to accept collect calls? Oz From: Osvaldo Mundim Junior [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] collect calls Date: Mon, 19 Jul 2004 16:33:19 -0300 Hi, Does anybody knows where can I change timing for collect calls? tks Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP 2 ISDN
Hello List, I'm from Germany and I want to use a Asterisk System. I have a few Accounts at my SIP-Provider www.sipgate.de and now I want to use my ISDN-Phone on the Sip-System. My idea was i set up a Asterisk-System and i will put in an ISDN Card where I can plug a ISDN Phone, I will have to use an ISDN card with the NT-Mode. The Asterisk has to register is at the SIP Provider and if a Call comes to me the Asterisk has to gibe the call to the ISDN card where the Telephone will ring. If the SIP Account 1 rings the telephone should get the MSN 1 and if Account 2 rings, the telephone should get the MSN 2. I will use Asterisk behind a NAT Router. If the Internetconnection interrupts the Asterisk has to wait 20 seconds, then has to register at the SIP-Provider. How can I do this, can somebody please help me? How is it possible to get the SIP Calls to the ISDN card? Would be very nice if you could help me. Thanks Moritz Beierlein
Re: [Asterisk-Users] codec translate
Asterisk can certainly do transcoding but the g729 codec requires a license unless you are using it in pass-thru mode. http://www.voip-info.org/wiki-Asterisk+G.729+licensing -Seth On Mon, 2004-07-19 at 22:36, mohammad mirzaee wrote: HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Gui client
A program that can hold you hand as you go along is very nice. What you could also do is add help screens that gives more in depth descriptions. But I'm looking forward to see your product as it grows! Yeah, help screens could be good. Or, when you enter the config, it can say * You should add at least one phone. * Having a telco line could be useful. ... and so on. So that a newbie knows what to do. For help: in the configlegs, where the VarType(...) definitions are, you can add a hint, e.g. you change VarType(ext, title=_(Extension), optional=True) into VarType(ext, title=_(Extension), hint=_(If you define an extension, then you can call the phone with this number. A phone without an extension can still be used as a target for direct dialin or calling groups.), optional=True) This text will then show up to the right of the web form. Later, I'd like to make this a popup window. ___ Asterisk-Users mailing list [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] codec translate
To translate with g729 you need licenses... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Brent Franks Enviado el: Martes, 20 de Julio de 2004 10:01 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] codec translate HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Yes. Also, Google works pretty good too. A simple Google Search for: Asterisk Translate Codec, would have returned a lot of useful searches. I included the link below in case you didn't know where/what google is. http://www.google.com/search?hl=enlr=ie=UTF-8q=Asterisk+Translate+codecb tnG=Search The sixth result looks like a winner. Additionally, Read the WiKi. http://www.voip-info.org - 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 ___ Asterisk-Users mailing list [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] New CVS version
I yesterday brought up to date the version of * the CVS and now I have a problem. Did you remove /usr/lib/asterisk/modules/res_parking.so before you installed the new asterisk modules? If not, you end up with res_parking.so and res_features.so fighting each other ... ___ Asterisk-Users mailing list [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] received a call waiting CONNECT_IND
My * just crashed and the nearest events to the time of the core dump
were two received a call waiting CONNECT_IND within 23 seconds of each
other according to the code in chan_capi.c this is because of the
following:-
if ((CONNECT_IND_BCHANNELINFORMATION(CMSG)[1] == 0x02) (!
capi_controllers[controller]-isdnmode)) {
// this is a call waiting CONNECT_IND with BChannelinformation[1]
== 0x02
// meaning no B or D channel for this call, since we can't do
anything with call waiting now
// just reject it with user busy
// however...if we are a p2p BRI then the telco switch will allow
us to choose the b channel
// so it will look like a callwaiting connect_ind to us
ast_log(LOG_ERROR,received a call waiting
CONNECT_IND\n);
What can I do to handle this correctly?
--
Dave Cotton [EMAIL PROTECTED]
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Re: [Asterisk-Users] New CVS version
You are probably having a problem with parking being renamed to features. Try a make clean then a make install. If that doesn't work then delete the res_parking.so module from /usr/lib/asterisk/modules/. -Seth On Tue, 2004-07-20 at 09:58, AsteriskList wrote: I yesterday brought up to date the version of * the CVS and now I have a problem. I cannot effect the RELOAD that * it breaks. Somebody can help or say as to load new users without stopping * ? Thanks Excuse my English Joao Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: quadbri NT_mode S-Bus Problem
I'm currently looking for a good solid solution that works here in the US with BRI-U NI-1 off a DMS100 or 5ESS. I've had luck with the Diva Server card and chan_capi. It worked great until we tried EKTS so I could get BellSouth to provision multiple DNs. Last I heard kapejod just needed the NI-1 specs and some time/money to implement NI-1 for the HFC cards. Duno what the status there is. --Justin ___ Asterisk-Users mailing list [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] Wireless SIP Phones
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 20 July 2004 09:26 am, Bodo Hahnke wrote: 1) are there other affordable wireless SIP Phones on the market? I haven't seen or found anything else till now ... Last but not least does anyone who knows both phones recommend anyone of these?? Or should I just buy the cheaper one? Why not just get a converter where you plugin a normal wireless phone to asterisk. Then you can use any phone you want/like. Digium has one and so does Grandstream. - -- 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) iD8DBQFA/TKTljK16xgETzkRAvEvAKCzV2Jg9cBQDVgXU9S41Cz9jhZ5HwCeKXRA DsdQRpU4OcBA7RbaHErpBYQ= =OeSI -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Numbering Plan and Siemens EWSD
Trace from their analyzer attached. switchtype was already set to euroisdn, so that shouldn't be the problem. I first configured pridialplan=unknown, but the telecom partner asked me to change the TON (type of number) to unknown, and the NPI to ISDN/Telephony Numbering Plan (E.164/E.163). Setting the pridialplan to local was not allowed (a TON of 'subscriber number' wouldn't work on their switch). So I changed some code and added a PRI_PROVIDER constant 0x01 to libpri.h (TON: unknown, NPI: ISDN/Telephony NP), like they wanted. Bruno, so you're just using pridialplan=local / national ? Do you also use the prilocaldialplan ? The guys from Siemens told me that it was highly uncommon to connect a softswitch directly to the Siemens EWSD. Our telecom partner in Belgium is TTG / Ventelo, and we are the only ones who connect a softswitch to their Siemens. If anyone has some info, please let me know. Thanks in advance. LU4 TLink4B SLink4 2004 July 20 12:20:31.725 === LAPD === --- ADDRESS --- SAPI : 0 = call control procedures CR : ..1. EA0: ...0 TEI: 0 = non-automatic TEI assignment user equipment EA1: ...1 --- CONTROL --- --- S FRAME --- S FORMAT : ..01 SFB: 00.. = RR (receive ready) SPARE : P/F: ...1 N(R) : 33 LU4 TLink4A SLink4 2004 July 20 12:20:31.728 === LAPD === --- ADDRESS --- SAPI : 0 = call control procedures CR : ..1. EA0: ...0 TEI: 0 = non-automatic TEI assignment user equipment EA1: ...1 --- CONTROL --- --- S FRAME --- S FORMAT : ..01 SFB: 00.. = RR (receive ready) SPARE : P/F: ...1 N(R) : 34 LU4 TLink4B SLink4 2004 July 20 12:20:41.724 === LAPD === --- ADDRESS --- SAPI : 0 = call control procedures CR : ..1. EA0: ...0 TEI: 0 = non-automatic TEI assignment user equipment EA1: ...1 --- CONTROL --- --- S FRAME --- S FORMAT : ..01 SFB: 00.. = RR (receive ready) SPARE : P/F: ...1 N(R) : 33 LU4 TLink4A SLink4 2004 July 20 12:20:41.728 === LAPD === --- ADDRESS --- SAPI : 0 = call control procedures CR : ..1. EA0: ...0 TEI: 0 = non-automatic TEI assignment user equipment EA1: ...1 --- CONTROL --- --- S FRAME --- S FORMAT : ..01 SFB: 00.. = RR (receive ready) SPARE : P/F: ...1 N(R) : 34 LU4 TLink4B SLink4 2004 July 20 12:20:51.724 === LAPD === --- ADDRESS --- SAPI : 0 = call control procedures CR : ..1. EA0: ...0 TEI: 0 = non-automatic TEI assignment user equipment EA1: ...1 --- CONTROL --- --- S FRAME --- S FORMAT : ..01 SFB: 00.. = RR (receive ready) SPARE : P/F: ...1 N(R) : 33 LU4 TLink4A SLink4 2004 July 20 12:20:51.728 === LAPD === --- ADDRESS --- SAPI : 0 = call control procedures CR : ..1. EA0: ...0 TEI: 0 = non-automatic TEI assignment user equipment EA1: ...1 --- CONTROL --- --- S FRAME --- S FORMAT : ..01 SFB: 00.. = RR (receive ready) SPARE : P/F: ...1 N(R) : 34 LU4 TLink4A SLink4 2004 July 20 12:20:55.140 === LAPD === --- ADDRESS --- SAPI : 0 = call control procedures CR : ..0. EA0: ...0 TEI: 0 = non-automatic TEI assignment user equipment EA1: ...1 --- CONTROL --- --- I FRAME --- I FORMAT : ...0 N(S) : 33 P : ...0 N(R) : 34 === ETSI ISDN === PROT DISC : 08h = Q.931 user-network call control message LEN CALL R : 2 SPARE : 0 FLAG : 0... = the message is sent from the side that originates the call reference CALL REF : 3 MESS TYPE : 05h = Setup --- SETUP --- IE ID : 4 LEN: 3 --- BEARER CAP --- EXT: 1... CODING STD : .00. = CCITT standardized coding INFO TC: ...0 = speech EXT: 1... TRANS MODE : .00. = circuit mode INFO TR: ...1 = 64 kbit/s EXT: 1... LAYER ID : .01. USRINFO L1 : ...00011 = recommendation G.711 A-law IE ID : 24 LEN: 3 --- CHANNEL ID --- EXT: 1... INT ID PRS : .0.. = interface implicitly identified INT TYPE : ..1. = other interface SPARE : ...0 PREF/EXCL : 1... = exclusive: only the indicated channel is acceptable D-CHANNEL : .0.. = the
Re: [Asterisk-Users] Sound files - uncompressed versions available?
Hi, When listening to GSM-compressed voice prompts from either G.729 or iLBC codec, the sound quality is distinctly sub-optimal due to the use of multiple transcoding. Would sox sound.gsm sound.au help a little bit? ___ Asterisk-Users mailing list [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] Echo on a PRI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andrew Kohlsmith wrote: ~ The only problem with echotraining=yes is the 8/10s delay before audio is | heard -- sometimes the start of conversations is cut off since our | receptionist doesn't have to life a receiver from the cradle to her ear. I normally wait about a second after I pick up the phone until I hear a very small click. I think that might be the end of the training period. ~ Then I proceed with my introduction. It seems to work quite well. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFA/TR0uYsUrHkpYtARAndWAJ9bWApl75x0YLKKYYN29pQ3SkpzRgCggVAi ko1u/UIYgX7UFwjbneheZ9A= =OBXB -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [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] Affordable SIP Phone - Stiil a Myth?
D'Link has thrown one more spanner into my works. Here is the clarification I have got about the STUN functionality. We need to buy their SIP server, if we have to make the D'Link Phones work transparently or Asterisk Server should have a Stun Server running on the same IP. Seshu Kanuri From: G Rao [EMAIL PROTECTED] Add to Address Book To: Seshu Kanuri [EMAIL PROTECTED] CC: [EMAIL PROTECTED] Subject: Re: STUN server settings Date: Tue, 20 Jul 2004 20:30:26 +0530 Dear Mr. Seshu Kanuri, Thanks for your mail. 1. D-Link IP Phones do not have any settings for the STUN server. 2. D-Link (SIP + H.323) Server has in-built STUN functionality support. 3. If one uses this D-Link Server the D-Link IP Phones (DPH-80) can work behind a NAT also. 4. In general if the Stun functionality, if in-built into the SIP server, then the end-devices do need to have any STUN support. 5. If D-Link end devices (IP Phones) if used with any other SIP server which do not have in-built STUN support, then they may not work behind NAT. I hope I am clear. Thanks / Regards, Rao KVSSS GUNNESWARA RAO D-Link (India) Limited. Phone: +91 22 2650 6271 Mobile: +91 98212 18057 - Original Message - From: Seshu Kanuri [EMAIL PROTECTED] To: G Rao [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 7:09 PM Subject: Re: STUN server settings Dr Rao Garu, I need a lttle more clarification on this. 1) Do we or dont we need a Stun Server running on our SIP server IP? 2) Does the D'Link Phones use D'Link's STUN server by default? /* 1. D-Link India has one SIP + H.323 server which has the in-built STUN server support. 2. So the end-devices (IP Phones) do not have any separate stun suport. */ 3) Point 2 - Does this mean that D'Link Phones by default dont have Stun Support and will not work from NAT? Please clarify. Thank You Seshu Kanuri --- G Rao [EMAIL PROTECTED] wrote: Dear Mandar, You are right. 1. D-Link India has one SIP + H.323 server which has the in-built STUN server support. 2. So the end-devices (IP Phones) do not have any separate stun suport. Thanks / Regards, Rao KVSSS GUNNESWARA RAO D-Link (India) Limited. Phone: +91 22 2650 6271 Mobile: +91 98212 18057 - Original Message - From: Mandar Pise To: [EMAIL PROTECTED] Cc: Seshu Kanuri Sent: Tuesday, July 20, 2004 4:08 PM Subject: STUN server settings Dear Mr. Rao, This is in reference to our telecon few minutes ago; I am listing the setting that I understood. The STUN server must be run on the SIP server IP address to resolve NAT issue of IP phone. There is no necessity of separate field for STUN server address in IP phone. Kindly correct me if I misunderstood something so I can convey the same to our US technicians. Thanks Regards, Mandar Pise ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Gui client
The source code found heere http://www.holgerschurig.de/destar.html is in an unsupported TAR format. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Sent: Tuesday, July 20, 2004 9:36 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Gui client -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 20 July 2004 03:33 am, Holger Schurig wrote: Would you mind looking at http://www.holgerschurig.de/destar.html and install it? Sorry, I can't give you an link for an online presentation because I don't have access to some server where I can install it. Please look critically at the program and give me back any feedback. The program starts being usable, but generally I find every day some bugs and add every second day some features. My goal is to use this program in production in about one month's time. A program that can hold you hand as you go along is very nice. What you could also do is add help screens that gives more in depth descriptions. But I'm looking forward to see your product as it grows! - -- 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) iD8DBQFA/R+sljK16xgETzkRAsG9AKCAsl3fEWSmzjom9Ick+yJboGPdugCgs4WV 41FmmNZeYv9imQjbPqkiP/A= =x1l7 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] Echo on a PRI
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, July 20, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Echo on a PRI On Tuesday 20 July 2004 09:04, Troy Settle wrote: On the subject of echo on a PRI, I too get this, but only when calling people in certain rate centers. Calls within my LATA (primarily VZ) are completely free of echo. Calls to a neighboring LATA (all Sprint) have echo on almost every rate center. I wish I knew more about this so I could rip Sprint a new one and tell them to fix their trunking, but... Are you sure it's Sprint's fault? I mean perhaps calling within your own LATA has less delay than calling neighbour LATAs and, combined with the delay that the T100P/TE405P introduces, presents enough delay to perceive echo... Pretty sure. Severe echo problems are only apparent when calling destinations within certain rate centers in this particular Sprint LATA (956) from my LATA (244). What's weird, is that inbound calls /from/ these same rate centers seem to have much less echo problem. It's possible that there's a something wrong with the trunking between my telco (KMC Telecom), the tandem (Verizon), and my LD carrier (MCI), then going to the destination (Sprint). The reverse call path is Sprint = Sprint = KMC = me. Fortunately, most of our calls are inbound, so it's not a huge issue at this time. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI dead in USA?
Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were well, BRI is getting quite antiquated, and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built-in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.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] New CVS version
On Tue, 20 Jul 2004 10:49:51 -0400, Seth Remington [EMAIL PROTECTED] wrote: You are probably having a problem with parking being renamed to features. Try a make clean then a make install. If that doesn't work then delete the res_parking.so module from /usr/lib/asterisk/modules/. You may need to change modules.conf to load res_features rather than res_parking ___ Asterisk-Users mailing list [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: gnophone and asterisk
On Tue, Jul 13, 2004 at 09:11:46AM +0700, Isianto Istiadi wrote: On Mon, 12 Jul 2004 14:51:55 +0200 Stefan Tichy [EMAIL PROTECTED] wrote: On Mon, Jul 12, 2004 at 03:30:24PM +0700, Isianto Istiadi wrote: and then I do nmap -sU ip (I don't see port 4569 or 5036 available). I can't register gnophone with *, when I do ethereal, I can see that gnophone tried to connect to port 5036, but the * replied destination unreachable. Is there something wrong with my config? gnophone 0.2.4 uses iax only not iax2. -- Stefan Tichy [EMAIL PROTECTED] Is that means I can't use gnophone with cvs *? will version 0.2.5 works? The version 0.2.4 is more than 2 years old and it looks as if there is no further development for GnoPhone. November 13th, 2001 - 11:58am CST - GnoPhone 0.2.4 released http://www.gnophone.com/ If you really need a iax2 capable softphone, you may check this: http://www.holgerschurig.de/files/linux/qtiax-0.1.tar.bz2 -- Stefan Tichy [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] New CVS version
Thanks. It decided my problem. Joao Carlos Moura - Original Message - From: Seth Remington [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 11:49 Subject: Re: [Asterisk-Users] New CVS version You are probably having a problem with parking being renamed to features. Try a make clean then a make install. If that doesn't work then delete the res_parking.so module from /usr/lib/asterisk/modules/. -Seth On Tue, 2004-07-20 at 09:58, AsteriskList wrote: I yesterday brought up to date the version of * the CVS and now I have a problem. I cannot effect the RELOAD that * it breaks. Somebody can help or say as to load new users without stopping * ? Thanks Excuse my English Joao Carlos Moura ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Updated Grandstream configurator
Maron Kristófersson wrote: I was even considering going further and writing a crossplatform or a webapp for configuring. However I was thinking if someone has written some notes on the config file specification See: http://www.mail-archive.com/[EMAIL PROTECTED]/msg43052.html Also, refer to the sources of GSConfigure. that could save a lot of time. I have no intention of competing with gsconfigure since I think it's an excellent What? Competition is good! Stephen R. Besch ___ Asterisk-Users mailing list [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] New CVS version
Thank´s. It decided my problem. Joao Carlos Moura - Original Message - From: Holger Schurig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 11:42 Subject: Re: [Asterisk-Users] New CVS version I yesterday brought up to date the version of * the CVS and now I have a problem. Did you remove /usr/lib/asterisk/modules/res_parking.so before you installed the new asterisk modules? If not, you end up with res_parking.so and res_features.so fighting each other ... ___ Asterisk-Users mailing list [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] Re: Numbering Plan and Siemens EWSD
Trace from their analyzer attached. Can they send an EWSD trace??? switchtype was already set to euroisdn, so that shouldn't be the problem. I first configured pridialplan=unknown, but the telecom partner asked me to change the TON (type of number) to unknown, and the NPI to ISDN/Telephony Numbering Plan (E.164/E.163). A smart technician must avoid to use TON=Unknown. Correct, E164 must be used. Setting the pridialplan to local was not allowed (a TON of 'subscriber number' wouldn't work on their switch). Bad data in tables, I presume... or you are sending crap. So I changed some code and added a PRI_PROVIDER constant 0x01 to libpri.h (TON: unknown, NPI: ISDN/Telephony NP), like they wanted. Bruno, so you're just using pridialplan=local / national ? Do you also use the prilocaldialplan ? The guys from Siemens told me that it was highly uncommon to connect a softswitch directly to the Siemens EWSD. Softwitch? What softswitch? For EWSD, asterisk it's just a PBX, because is connected thru PRI! Our telecom partner in Belgium is TTG / Ventelo, and we are the only ones who connect a softswitch to their Siemens. If anyone has some info, please let me know. Thanks in advance. ___ Asterisk-Users mailing list [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] quadBRI
Hello Asterisk users.
We have a quadBRI card installed and have the following problem. When
starting Asterisk, the card is up and works perfect. But if no one uses
it for 2-3 hours, the card seems to change status. If I try making a
call from my sip phone to an extarnal telephone then asterisk registers
that I'm trying to call, i.e. it's not a bridged but a show channels
gives this output:
Channel (ContextExtensionPri ) State Appl. Data
Zap/1-1 (isdn-ingresso s1 ) Dialing AppDial
(Outgoing Line)
SIP/116-c24d (from-sip 03926969736 1 )Ring
Dial ZAP/g1/392736|30|t
2 active channel(s)
But in the sip phone there is only silence exept for som weak clicking
sound.
If instead I make the call from outside, then the sip phone rings,
when I answer, in the sip phone there is absolute silence and the
calling phone keeps ringing. Asterisk says that the call is up, i.e.
bridged.
Does anyone know anything about this syndrome? Below are the conf
files involved.
[from-sip]
exten = _0.,1,Dial(ZAP/g1/${EXTEN:1},30,t)
[isdn-ingresso]
exten =27380773,1,Dial(SIP/116,20,t)
exten = s,1,Answer
exten = s,2,Playback(pbx-invalid)
exten = s,3,Hangup
zapata.conf:
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12
zaptel.conf:
switchtype = euroisdn
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=8.2
txgain=-1.0
signalling = bri_cpe_ptmp
pridialplan = local
prilocaldialplan = local
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = local
context=isdn-ingresso
group = 1
channel = 1-2
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = local
context=isdn-ingresso
group = 2
channel = 4-5
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = local
context=isdn-ingresso
group = 3
channel = 7-8
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan = local
context=isdn-ingresso
group = 4
channel = 10-11
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Re: [Asterisk-Users] Echo on a PRI
I installed a server in Australia with a Wildcard X100P in it. From my server in the U.S, I pushed a call via IAX to the server in Australia which then pushed it out that card. Severe echo, only I could hear it though. The remote side heard nothing. Definately been reading up on this echoing issue. I thought the main reason was latency, and my ping to that server in Australia reveals 200ms response times. However, they have a HT286 Converter there in Australia on the same connection, and it connects to my Asterisk server here in the US via SIP and it places calls all day long with no problems. I'm going to try more testing, like connecting a HT286 here in the U.S straight to their Asterisk server there and trying to make local calls. Will then try Asterisk to Asterisk communication via SIP instead of IAX. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 10:17 AM Subject: RE: [Asterisk-Users] Echo on a PRI On the subject of echo on a PRI, I too get this, but only when calling people in certain rate centers. Calls within my LATA (primarily VZ) are completely free of echo. Calls to a neighboring LATA (all Sprint) have echo on almost every rate center. I wish I knew more about this so I could rip Sprint a new one and tell them to fix their trunking, but... Troy, Given the reseach that is currently going on, etc, I would not bet any more then a cup of coffee that Sprint (or any other carrier) has an echo problem right now. There _appears_ to be an issue with the echo cancellation routines in asterisk that is impacting more then a couple of implementations. The research to date suggests the current echo canceler works well in some cases, and not so well in other cases. In other words, there are a certain set of undocumented/unknown conditions (and/or PC systems) that fall within the thresholds of the current canceler that work, and other conditions (and PC systems) that fall outside the limits of the canceler that are less then acceptable. The limits and thresholds are _not_ black white and may end up being one of the more difficult problems to solve within asterisk. (E.g, it's entirely possible that your calls via Sprint fall outside the limits of *'s canceler.) As you've probably seen earlier on this list, there is a high probability that internal system issues (eg, interrupt servicing latency, possibly PCI bus issues, etc) that are impacting this in _some_ specific cases. In some cases, swapping the motherboard did in fact impact the cancellation quality. However, be very carefull not to read anything more into that at this time. There is no one at this time that knows factually what those limits, thresholds, etc, happen to be (not even Mark). Rich ___ Asterisk-Users mailing list [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] Wireless SIP Phones
yet. The only Wireless SIP phone I would use in a productive environment would be the Cisco 7920. I don't see a SIP load for the 7920. Are you sure it is SIP enabled? Ray. - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on a PRI
On Tuesday 20 July 2004 09:22, Ryan Thrash wrote: Oops. I need to correct my last post: I don't have the PBX in the mix. My config is dual Xeon 2.4s, 1GB RAM, HW SATA RAID, SuperMicro X5DAL-TG2 motherboard connected to: Telco - PRI (T100P) - Asterisk - SIP Phones (Budgetones 102s/Snom 200) The premise is still the same though: echo present despite our digital PRI that *should* make this impossible. It's usually only echo on our side when calling out as has been discussed here previously ad nauseum with no one being able to really figure out its source. I wish I knew where to really start poking around to try to help get to the bottom of this. No, the PRI does NOT make echo impossible. It makes it highly unlikely that YOU will generate echo. You never hear echo YOU generate; you hear the echo being generated on the other side. Unless I've totally misunderstood the double negatives in that statement, I don't believe your statement is accurate at all. The echo problem that most of us have (or had) does result from audio initiated by sip phones (etc) passing out through any number of zap oriented cards/adapters to the pstn (regardless of who the pstn provider happens to be). The technical issue seems to be oriented around... a. rtp packet arrives at asterisk via the LAN (as an example only), b. asterisk queues the rtp packet/bytes for transmission via a zap channel, c. the system sends pkts/bytes to zap card, and for _lots_ of different reasons, some of the audio (pkts/bytes) are reflected back towards the inbound side of the card (to asterisk code) via the PCI and interrupt structure, d. the current echo canceler removes that reflection **if** the pkts/bytes arrive (in * code) within a certain amount of time, e. if the reflection falls outside the current canceler's limits, or if some other audio interference is involved, or if an interrupt or two is missed, the reflected audio is not removed by the current canceler (as it falls outside it's limits) and we hear echo. The echotraining=800 enhancement represents one step towards reducing critical timing part of pulsing the zap channel and pre-loading the canceler with something reasonable. That, in effect, removed the 5-to-20 second training period for the canceler. It had nothing to do with addressing the limits or thresholds of the canceler itself. Regardless of whether one uses a zap driven PRI, T1, or analog line, there is reflected energy (eg, audio) that needs to be removed by the echo canceler. The amount of reflected energy varys by type of facility (eg, PRI vs analog line), by call destination, the efficiency of any hybrid involved (if any), etc, but its still there in all cases, period. What remains for the current echo issues seem to boil down to two somewhat unrelated issues (there might be more): a. internal system delays possibly resulting from interrupt service latency, internal PCI structure, etc. (Those systems with this issue seem to have some degree of echo on all calls. Swapping motherboards is known to impact this one to some degree.) b. echo on certain zap calls where it appears the reflected energy falls outside the limits of the current canceler. (Those are likely to relate to significant time delays in the reflected energy and _might_ be related to the type of facilities used within the carrier's network. Likely in the Sprint case noted.) Whether one has echo with NuFone calls or not is totally irrelevant as those calls are not sent through zap channels, and are not subjected to the same echo canceler issues noted above. Trying to identify _factually_ what the various limits happen to be with interrupt latency (etc), reflected energy from both local and outside sources is not an trevial task. Changing the echo canceler to support whatever those limits happen to be is likely to be far more difficult. As Steve Underwood noted earlier, one of the only ways to identify the issues in (a) is to write a test application that sends data out through the zap card, loop that data back into the receive side, and measure the delays (and variation in delay) assoicated with that path. There could be multiple issues uncovered, and some are likely to be system dependent. We'd all like to hope that changes to address (a) would be sufficient to also address the issues in (b). We'll have to wait and see. Rich ___ Asterisk-Users mailing list [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] Wireless SIP Phones
yet. The only Wireless SIP phone I would use in a productive environment would be the Cisco 7920. Does it work in SCCP mode with good results in Asterisk? - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
I am looking for a provider that will provide an equivalent of DID/DOD trunks via IAX, IAX2 or SIP using numbers in Metro Chicago (prefer Skokie), LA (prefer West Hollywood or Venice), and/or Orlando (prefer Winter Garden). If I can migrate some of my existing numbers using LNP, that would be even better, but it is not a requirement. While I know that there are several companies that will terminate VoIP number using these protocols, none offers a functional equivalent of ILEC DID service. From my ILEC, I can purchase one or more DID trunks and a block of phone numbers (usually for between $0.01 and $0.10 a number). I can receive as many calls simultaneously as I have trunks, after that callers receive a busy signal. All VoIP trunk providers that I have found, want to charge me several dollars per phone number, but will allow me unlimited incoming calls per number. I want many phone numbers so that each phone in my facility has its own phone number, but I really do not need that many simultaneous calls and it would be cost prohibitive to pay several dollars for each phone number. Thanks in advance. /carmi ___ Asterisk-Users mailing list [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] Up to date?
Hi, before you start throwing stones to me let me tell you that I am a bit new to Linux. I downloaded Asterisk from the cvs server on Wednesday 15 July 2004, as described in Andy Powell's Getting Started with Asterisk (http://www.automated.it/guidetoasterisk.htm). Thanks Andy! I read about the Asterisk 1.0 RC1, and I would like to download it and install it. Could someone tell me what is the best way to proceed, considering that I already have a configuration that I would not like to loose, and that I would like to have the option to roll-back to the version I already have, if all goes pear-shaped? TIA Yiannis. ___ Asterisk-Users mailing list [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] Suscription
Carlos Clemares [EMAIL PROTECTED] wrote: Name: Carlos Clemares Of course you are. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ 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] how to configure my cisco 7960?!
hi everybody, just tried to install my cisco 7960 - but without much success :-( I want to set it up as a sip phone - but I can not setup the phone's IP address... after plugging it in it says Configuring IP - I unlocked it and entered the Network Configuration. I can see the edit-buttons but when I trie to press then it says That key is not active here so how can I tell the phone the IP of my tftp-server when the phone doesn't let me adjust it's settings?! 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] BRI dead in USA?
Scott Stingel wrote: Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. So why is the subject of your message PRI dead in USA? G What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built-in feature. I have a client who had 8 BRI lines (just recently turned off) used as trunk service for a Nortel MICS system. They worked fine, delivered calling number _and_ name (in QWest territory), service was excellent and voice quality was too (duh, it's digital :-)). I suspect that BRI is going to go away in the next couple of years, since it has never really taken off and now there are other alternatives for high-speed data-only usage. ___ Asterisk-Users mailing list [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] what is :
Richard Neese [EMAIL PROTECTED] wrote: what is your current flashware? Cisco 7960G firmware: 7.1. Sipura SPA-2000 firmware: 2.0.9(d). didi you spill anything on the phone? I try not to do silly things like that. Have you tried going back 1 flash and then testing the button and the flash your phoneagain and test? No. Why would I want to do that? Is this a general survey, or did you just forget to include some context in your article? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ 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
RE: [Asterisk-Users] PRI dead in USA?
Well they fail to realize that ISDN is used for more than data. I just wanna scream at them and say IT DOES VOICE TO YOU NINNY!.. Rates are far from reasonable. 167/mth here is what I would have to pay for ISDN-BRI. SBC is lame. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Tuesday, July 20, 2004 10:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI dead in USA? Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were well, BRI is getting quite antiquated, and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built- in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.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
Re: [Asterisk-Users] PRI dead in USA?
Hi Scott, Local ISDN BRI service is definitely on it's way out. We recently have canceled several ISDN BRI accounts and replaced them with ADSL lines. More bandwidth and less cost. If you intend on using the lines for voice only, then FXO is the better option. If you looking to use voicedata the I would suggest 1 FXO line with ADSL over it. We believe the Digium cards with Asterisk in a small Linux box will provide a best combination of flexibility and services. Doug, Voippages.com Scott Stingel wrote: Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were well, BRI is getting quite antiquated, and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built-in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.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
RE: [Asterisk-Users] BRI dead in USA?
BTW - the title of this was supposed to be BRI dead in USA?! (too early!) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Tuesday, July 20, 2004 8:37 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI dead in USA? Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were well, BRI is getting quite antiquated, and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built-in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.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
RE: [Asterisk-Users] Echo on a PRI
You can have echo even w/ PRI's, it's just that the echo isn't INTRODUCED at the PRI demarcation point, it is INTRODUCED somewhere along the call path, usually where the 4 wire digital signal to 2 wire analogue signal conversion point exist. We found that moving to a Compaq DL380 or 6400R and compiling in a few extra options (see one of my previous post) totally abolished our echo problem. W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Thrash Sent: Tuesday, July 20, 2004 8:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Echo on a PRI I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Welcome to the club. ;) You have the same exact problem I've got. The only difference is I'm using dual Xeon 2.4s and a Supermicro SuperWorkstation 7033A-T (X5DAL-TG2 motherboard http://supermicro.com/products/motherboard/Xeon/E7505/X5DAL-TG2.cfm ). Echo training=800 on a recent CVS helped, but did not totally resolve the issue. Best regards, Ryan Thrash ___ Asterisk-Users mailing list [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] PRI dead in USA?
i work for a local telco and BRI is avoided due to the amount of hardware it can take to get to an end user. DSL is simply easier and cheaper to provide. Not sure why you can't get caller id, i know when can add that feature to BRI - hcir On Jul 20, 2004, at 7:36 AM, Scott Stingel wrote: Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were well, BRI is getting quite antiquated, and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built-in feature. ___ Asterisk-Users mailing list [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] Polycom IP 500 Voicemail
I tried this configuration and it still does not work for me. In fact, now I cannot dial in using the menu system of the message center. Here is how I have now mine configured and what I get... msg msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe=8 msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=8 msg.mwi.2.subscribe= msg.mwi.2.callBackMode=registration msg.mwi.2.callBack= msg.mwi.3.subscribe= msg.mwi.3.callBackMode=registration msg.mwi.3.callBack= msg.mwi.4.subscribe= msg.mwi.4.callBackMode=registration msg.mwi.4.callBack= msg.mwi.5.subscribe= msg.mwi.5.callBackMode=registration msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6.callBackMode=registration msg.mwi.6.callBack=/ /msg nat nat.ip= nat.signalPort= nat.mediaPortStart=/ user_preferences up.headsetMode=0 up.useDirectoryNames=0 up.oneTouchVoiceMail=1/ The relevent fields being the msg. fields and up.oneTouchVoicemail This allows me voicemail via the Messages button but it is not direct. I have to navigate still through allt he menus. W -Original Message- From: John Baker [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 10:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail My Polycom Message button goes straight to voicemail. Here's how: 1) Use the latest firmware, available on the Wiki 2) In your phone.cfg file (for each phone) set msg msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=76 3) In your extensions.conf, have something like: exten = 76,1,VoiceMailMain2([EMAIL PROTECTED]) (Let's assume your voice mailbox is the same as your extension) Then when you push the message button, asterisk will ask for your password! You're in! John Chris A. Icide wrote: On 04:28 PM 7/19/2004, Wiley E. Siler wrote: Mine does the same. Once in Message center I can choose selection 1.Message Center and then soft key Select.Then I select the registered line that I want to check voice mail on. That is no less than 4 key strokes just to get into your voice mail. Not many to me but tons to an unskilled user. However, in the documentation regarding the bypassInstantMessage value, supposedly, setting bypassInstantMessage to 1 is supposed to allow you to go right into voice mail without navigating the Message Center. That is the big question on my mind at this point. I have yet to get this to work and I also don't think I am receiving any SIMPLE messages ti show me that I have messages waiting. Do you get a message waiting indicator? I do get MWI, there are a few things you need to set, and I forget what off the top of my head, soon as I can look and post it here. I haven't tried the bypassInstantMessage value, but I'll take a look and see if I can get it to work. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] Errors and Warnings with Galaxyvoice
Hello, I am receiving the following repeated Errors and Warnings with Galaxyvoice. I have placed the sip context below, perhaps someone can offer suggestions how I could troubleshoot this. Thanks Kevin Jul 20 12:35:48 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 181 (Critical Request) Jul 20 12:36:02 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again [galaxyvoice] port=5060 fromuser=2035551212 fromdomain=216.229.127.40 username=V00X type=friend secret=X auth=md5 host=216.229.127.40 ;defaultip=216.229.127.40 reinvite=no canreinvite=no dtmfmode=inband context=inbound-galaxy qualify=yes disallow=all allow=gsm allow=ulaw callerid=2035551212 defaultexpirey=3600 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI dead in USA?
Never rely on a telco for correct information, they will very often be wrong... unless you luck out and actually talk to someone who knows something... Both PRI and BRI are capable of ANI (Caller ID) by using their D-Channel to send/receive this information digitally... A regular T1 (read non-ISDN) can also receive Caller ID if it is done in-band (I.E. Between the first and second rings like an analog line does...) This is the old-school way of doing it, but you get the benefit of not losing that last channel... - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 8:36 AM Subject: [Asterisk-Users] PRI dead in USA? Hi- Because a majority of my customers are in Europe, I've gotten quite used to working with ISDN (PRI) and BRI on a regular basis. Recently one of my customers asked me if I could terminate a few lines locally here in the USA (California), so I called up SBC to enquire as to how much it would cost to install a BRI here. Although the rates were reasonable (except the installation), I got the distinct impression that they really didn't want to install BRI's. Their comments were well, BRI is getting quite antiquated, and the like. They said with the advent of ADSL, there's not much of a market anymore, as most of past usage was modem related. I'm a little worried about the pricing going up, and availability going down in the near future. I don't have the volume yet to justify PRI. What are other's experience in the US with BRI? Also, they mentioned that I couldn't get caller ID with the BRI service, which I thought was a built-in feature. Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.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
Re: [Asterisk-Users] Echo on a PRI
On Tuesday 20 July 2004 11:04, Jason A. Pattie wrote: I normally wait about a second after I pick up the phone until I hear a very small click. I think that might be the end of the training period. ~ Then I proceed with my introduction. It seems to work quite well. I agree and do that myself; it's just a matter of training the staff :-) -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] question regarding Asterisk. X-Lite, and firewall
Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite (phone A) is on Win2K, with IP 192.168.1.100. They are both behind a router with dynamic IP address. Assume its public IP is aaa.bbb.ccc.ddd. I have another X_Lite SIP phone (phone B) that is NOT in the LAN I mentioned above. Rather, it has its own public IP address, say eee.fff.ggg.hhh. I have configured the router to forward all traffic to its port 5161 to Asterisk server's 5060 port, and configured SIP phone A to use 192.168.1.102:5060 and phone B aaa.bbb.ccc.ddd:5161 as proxy server respectively. Both phones registered successfully. Now, I used phone B to call phone A. The entire SIP hand-shake went through successfully. However, I can only get voice from phone A to phone B, not the other direction. I found that RTP traffic went from phone A - Asterisk - phone B. However, on the other direction, phone B tried to use 192.168.1.102 as destination of Asterisk to send voice too. Obviously, the IP is a private IP, hence, is not reachable. How do I change configuration of Asterisk so that phone B can use aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address? By the way, both directions use UDP protocol. Thanks! Michael Wang [EMAIL PROTECTED] 2004-07-20
Re: [Asterisk-Users] Wireless SIP Phones
You are right, there is no SIP firmware for the 7920 - SCCP is currently the only choice for *. Ray Burkholder wrote: yet. The only Wireless SIP phone I would use in a productive environment would be the Cisco 7920. I don't see a SIP load for the 7920. Are you sure it is SIP enabled? Ray. - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [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] 7960 Dynamic DNS?
Eats humble pie!! I'd never seen it in the settings and sure enough it's there. Sorry for misguiding. P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004, 7:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 7960 Dynamic DNS? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 3:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7960 Dynamic DNS? I can't think of any router that supports this You could put it in as a request to www.sveasoft.com for their firmware for the wrt54g (great box...runs linux and lots of features and functionality). Not only does the Sveasoft firmware already support dynamic DNS, the original Linksys firmware does as well. It was very common junk router feature (and by junk I mean anything you can buy at Staples that claims to be a router). Daryl ___ Asterisk-Users mailing list [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] FREE (305) and (786) termination. Anyone interested?
I have an Asterisk box with free local termination to area codes (305) and (786) [Miami area, US]. I want to configure it to accept incomming VoIP traffic (cant use IAX) and terminate calls over the PSTN network. I need help with the configuration and also some incoming traffic for testing purposes. Please contact me if you can help. Regards, Alejandro.
[Asterisk-Users] Installing X100P
I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it installed as is? I really do not want to change my version of Asterisk since it is running well at this point. Is it possible to just update and add the card? Thanks, Wiley
Re: *****SPAM FOUND***** [Asterisk-Users] how to configure my cisco 7960?!
Turn off dhcp first. Option 25 in network configuration. - Original Message - From: xfastjackx [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 12:34 PM Subject: *SPAM FOUND* [Asterisk-Users] how to configure my cisco 7960?! hi everybody, just tried to install my cisco 7960 - but without much success :-( I want to set it up as a sip phone - but I can not setup the phone's IP address... after plugging it in it says Configuring IP - I unlocked it and entered the Network Configuration. I can see the edit-buttons but when I trie to press then it says That key is not active here so how can I tell the phone the IP of my tftp-server when the phone doesn't let me adjust it's settings?! 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 ___ Asterisk-Users mailing list [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] Echo on a PRI
Thanks Everyone! I appreciate all the feed back. Right now I am using a Digium T400P card and my system, although it is fast, has a slight load, about 15% due to some mysql activity. I know that Digium as a new card the TE410P. Does anyone have any experience in the new card and is the speed difference likely to help with the echo? Also, if I put in a second processor, is that likely to help with the echo? Thanks Again, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Goldfein Sent: Monday, July 19, 2004 5:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo on a PRI Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Thanks, 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
[Asterisk-Users] Call Queue: strategies and penalties
All, For the last 10 months, I've been using strategy=ringall. This has worked fine and did what I wanted, but at this point, I'm needing to implement a 'penalty' or delay for some members of the call queue. 1: remote users(remote flunkies) 2: level-1 support (flunkies) 3: level-2 support (glorified flunkies) 4: level-3 support (super flunkies) When a call comes in, I want it to ring the first group for 30 seconds, and if there's no answer, ring groups 1-2 for 30 seconds. If no answer, ring groups 1-3 for 30 seconds, and if still no answer, ring all 4 groups until the call is answered. What do I need to do to get this behavior? If the answer involves $$, tell me about it, I'm not afraid to spend some cash to help streamline my business. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
