Re: [asterisk-users] WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to 'unknown sip:[EMAIL PROTECTED]
Hi Olle, that was a phone misconfigurationa parameter had a wrong value. The message has disappeared and now the phone seems to work! Thank you! Giorgio Johansson Olle E wrote: 10 jan 2008 kl. 16.48 skrev gincantalupo: Hi, I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom always rings but sometimes (it happens randomly!) no voice is passing thru (2 ways). Asterisk CLI shows this warning: Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to 'unknown sip:[EMAIL PROTECTED] I have already set localnet and externip parameters inside the general section of my sip.conf: localnet = 192.168.4.0/24 externip = xx.xx.xx.xxx Is there anybody who knows how to solve this problem? The error message has nothing to do with no voice is passing thru. The error message clearly indicates that you have bad credentials for an INVITE. In order for anyone to help you, you need to reveal more about the setup and the involved parties in the communication. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to open master device '/dev/zap/ctl'
there is no /proc/zap folder .. can you tell how can I create /dev nodes. I have tested the same configurations on FC5 and these device links were created ... drwxr-xr-x 2 root root 160 Jan 17 10:59 . drwxr-xr-x 13 root root 3640 Jan 17 11:00 .. crw--- 1 root root 196, 1 Jan 17 10:59 1 crw--- 1 root root 196, 254 Jan 17 10:59 channel crw--- 1 root root 196, 0 Jan 17 10:59 ctl crw--- 1 root root 196, 255 Jan 17 10:59 pseudo crw--- 1 root root 196, 253 Jan 17 10:59 timer crw-rw 1 root root 196, 250 Jan 17 10:59 transcode -ag create nodes and links /proc/zap On Jan 16, 2008 3:39 PM, Chris Bagnall lists at minotaur.cc wrote: Make sure asterisk is in the dialout group in /etc/passwd The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout, and if you're using the gentoo ebuild of asterisk, it'll run as asterisk:asterisk, so you need to make sure asterisk is a member of the dialout goup otherwise it'll never be able to access /dev/zap/* FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well worth updating to 2007.0 if you can spare the time - it'll save you a lot of messing around with gcc versions etc. later down the line. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Single T1 with DIDs
Can anyone share their experience with me? I am looking for a provider that delivers Dialtone over T1 to terminate to my asterisk box and also provide DIDs. Does the DIDs come with the T1 services or those are purchased/charged seperately. Any help greatly appreciated. My target markets are Philadelphia and Washington DC Metro areas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to open master device '/dev/zap/ctl'
*Walter Willis, *Thanks a lot, got the commands from zap Makefile and it worked, now can create conference room, my question still stands why it didn't create itself. Will go through make file to get an answer to that. Anyone else facing the issue can resolve by running following commands mknod /dev/zap/ctl c 196 0 mknod /dev/zap/transcode c 196 250 mknod /dev/zap/timer c 196 253 mknod /dev/zap/channel c 196 254 mknod /dev/zap/pseudo c 196 255 -ag any version of asterisk not create nodes into /proc/zap create to command, view into make file how to create nodes On Jan 16, 2008 8:48 PM, Walter Willis walterwn at gmail.com wrote: create nodes and links /proc/zap On Jan 16, 2008 3:39 PM, Chris Bagnall lists at minotaur.cc wrote: Make sure asterisk is in the dialout group in /etc/passwd The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout, and if you're using the gentoo ebuild of asterisk, it'll run as asterisk:asterisk, so you need to make sure asterisk is a member of the dialout goup otherwise it'll never be able to access /dev/zap/* FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well worth updating to 2007.0 if you can spare the time - it'll save you a lot of messing around with gcc versions etc. later down the line. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] callerid on atxfer
Hello. I have a little problem with the callerid shown to the callee if he recieves an atxfer (*2) call. The display of the calees phone is showing (s) and thats not what i want. I wanna see the callerid from the user who is transfering the call. Example: 12345 calls 123, 123 transfers (atxfer) the call to 124. 124 should see 123 in his display not (s). Is there a way to accomplish that? thanks an regards t. -- knowledgeTools® ... managing complexity. -- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz -- This eMail communication (and any attachment/s) may contain confidential or privileged information and is intended only for the individual(s) or entity named above and to others who have been specifically authorized to receive it. If you are not the intended recipient, please do not read, copy, use or disclose the contents of this communication to others. Please notify the sender that you have received this e-mail in error by reply e-mail, and delete the e-mail subsequently. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson However, you'll need to do similar things to your asterisk box router if it's behind NAT for IAX as you do for SIP. (You will need a static IP address on the NAT router and port-forward 4569 to the asterisk box, just as you'd port-forward 5060 and 1-2 for SIP) Please correct me if I'm wrong, for Iax clients you don't need to do static port-forwarding as they will create upon registration one entry in NAT table with UDP port for both signalling and media. On the other hand, sip clients (without Stun) are difficult to manage behind Nat because of RTP/RTCP ports. I don't want to start a flame Iax vs Sip, just to clarify respective advantages. Best Regards, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 42, Issue 51
On zapata.conf use the parameter callerid. On Jan 17, 2008 3:33 AM, sandeep [EMAIL PROTECTED] wrote: hi all, how to set the caller id facility for the TDM400p card. Please help me thanks, sandeep.s -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk desktop tools for OS X
Hi everyone, I have been long working on a project (http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open master device '/dev/zap/ctl'
On Wed, Jan 16, 2008 at 10:09:54PM -0500, Walter Willis wrote: any version of asterisk not create nodes into /proc/zap create to command, view into make file how to create nodes Do you suggest to use mknod manually? This will work. Unless you use udev. And almost everybody use it. What is the output of: mount -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Meetme MeetMeAdmin cmd info-use
Hi All I need to set my Asterisk conference such way that , during confernce Admin Can kick 1 or all user , Same for mute fuction.As well as Admin can increase or decrease conf user volume. for that i used MeetMeAdmin like this exten = 600,1,MeetMeAdmin(,ekKLmMNS,7010)where is conf number 7010 is Admin user ityself Also for all other user i use like this with same conf number exten = 601,1,MeetMe(|Mps) when 2 user in conf wait for Admin. then Admin Dail 600 but i found no response when admin dail 600 its blank no response at console. All Dail Plan config are ok. But for 600 its balnk. what i need to do so can Admin Handle user who dail 601 enter in same conf room .Means i want to controld all the users? Also What is the use of user parameter in MeetMeAdmin cmd. is that normal user or Admin Itself Plz give me the solution so i can handle such conf condition as Admin. Regards Amit ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk desktop tools for OS X
Hi, Im interested, Please send me copy Thanks On Jan 17, 2008 7:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project (http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Lito Manansala Network Operations (VoIP) VoiceValley Group of Companies Phone: +61-7-30188461 Fax: +61-7-30188499 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk desktop tools for OS X
Looks interesting. I couldn't get it working because a few of the preference fields were not responding (current svn, build on Leopard). Looks like a nice elegant solution though. Let me know if there's anything you want help on and I'll dust off my cocoa! Simon Simon Elliston Ball [EMAIL PROTECTED] On 17 Jan 2008, at 13:06, Lito Manansala wrote: Hi, Im interested, Please send me copy Thanks On Jan 17, 2008 7:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project ( http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Lito Manansala Network Operations (VoIP) VoiceValley Group of Companies Phone: +61-7-30188461 Fax: +61-7-30188499 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP client in asterisk not trying to contact IMAP server
On Thu, 17 Jan 2008 6:34 +0200, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: And now in make menuselect you have to go to voicemail options and set IMAP support to on. Thanks, if that was in any of the docs I just completely glossed over it. I'll give it a shot. Thanks again, --J(K) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timing on TE405P
On 1/17/08, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jan 17, 2008 at 03:09:59PM +0200, Atis Lezdins wrote: Hi, I'm wondering why zttest shows Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469 Shouldn't it be 100% as timing is hardware and comes from PRI? Am I missing some kernel config? It may be slightly different. Your system clock may be slightly off. But more importantly, zttest doesn't start and stop messuring time at exactly the right spot. Anything i can improve? I think - zttest should do it correctly, as manpage says - definite pass is 100% or 99.99% I'm just having some issues with faxing, so i thought this could be a problem. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel timing on TE405P
Hi, I'm wondering why zttest shows Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469 Shouldn't it be 100% as timing is hardware and comes from PRI? Am I missing some kernel config? Regards, Atis My /etc/zaptel.conf is span=1,4,0,esf,b8zs span=2,3,0,esf,b8zs span=3,2,0,esf,b8zs span=4,1,0,esf,b8zs #lspci 07:03.0 Communication controller: Digium, Inc. Wildcard TE405P Quad-Span togglable E1/T1/J1 card 5.0v (rev 02) #lsmod | grep -P (zap|zt|wc) zttranscode17808 0 wcusb 25088 0 wctdm 46272 0 wcfxo 21536 0 wctdm24xxp121024 0 wcte11xp 34848 0 wct1xxp23456 0 wct4xxp 323904 96 zaptel202984 215 zttranscode,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 10817 1 zaptel -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Single T1 with DIDs
On Jan 17, 2008 5:23 AM, broadband Voice [EMAIL PROTECTED] wrote: Can anyone share their experience with me? I am looking for a provider that delivers Dialtone over T1 to terminate to my asterisk box and also provide DIDs. Does the DIDs come with the T1 services or those are purchased/charged seperately. Any help greatly appreciated. My target markets are Philadelphia and Washington DC Metro areas. I would be glad to help you out with this as I have T1s in both PA and MD and have been through all the paces with all of the big players in the area from T1s to T3s. I pay $.65 per DID per month on top of the loop and minute charges. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL includes?
voip*CLI ael reload Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown root token '#include' Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box, and I don't want to upgrade our only production computer. Jay Rodrigo R Passos wrote: Jay, What error? Jay Moore wrote: How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL includes?
On 1/17/08, Jay Moore [EMAIL PROTECTED] wrote: How do I include a file (not a context) in AEL? #include filename returns an error. What's the error? For me this works: #include extensions_db.ael; #include extensions_utils.ael; #include extensions_ivr.ael; #include extensions_globals.ael; However i'm using #aelparse -d -n -w -q extensions.ael to dump ael into .conf format Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL includes?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore Sent: Thursday, January 17, 2008 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AEL includes? How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay That is exactly the syntax that you should be (and I am) using. I don't know why that wouldn't work, unless you're using an older version of Asterisk and are using fully-qualified paths. - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL includes?
How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel timing on TE405P
On Thu, Jan 17, 2008 at 03:09:59PM +0200, Atis Lezdins wrote: Hi, I'm wondering why zttest shows Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469 Shouldn't it be 100% as timing is hardware and comes from PRI? Am I missing some kernel config? It may be slightly different. Your system clock may be slightly off. But more importantly, zttest doesn't start and stop messuring time at exactly the right spot. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Single T1 with DIDs
Steve, That is very helpful, How much are we talking about in terms of the loop and minute charges. If you want it offline I can send you a private my with my phone number. On 1/17/08, Steve Totaro [EMAIL PROTECTED] wrote: On Jan 17, 2008 5:23 AM, broadband Voice [EMAIL PROTECTED] wrote: Can anyone share their experience with me? I am looking for a provider that delivers Dialtone over T1 to terminate to my asterisk box and also provide DIDs. Does the DIDs come with the T1 services or those are purchased/charged seperately. Any help greatly appreciated. My target markets are Philadelphia and Washington DC Metro areas. I would be glad to help you out with this as I have T1s in both PA and MD and have been through all the paces with all of the big players in the area from T1s to T3s. I pay $.65 per DID per month on top of the loop and minute charges. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip channel - redirection - which context is used
Hi, When asterisk receives 302 Moved Temporary sip response what is the logic for selecting the domain and context to use? Thanks for any help Tomasz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL includes?
Jay, What error? Jay Moore wrote: How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL includes?
On 1/17/08, Jay Moore [EMAIL PROTECTED] wrote: voip*CLI ael reload Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown root token '#include' Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box, and I don't want to upgrade our only production computer. I suppose, that it doesn't support AEL2. You can dump ael to conf file with command i posted before. Oh, and you will need to grab 1.4, and compile aelparse from it. Regards, Atis Jay Rodrigo R Passos wrote: Jay, What error? Jay Moore wrote: How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Encryption
Cavalera Claudio Luigi wrote: Is this the libiax used currently on asterisk http://ftp.digium.com/pub/libiax/ ? No. Asterisk has its own IAX2 implementation. I would like to understand if someone is using this in production. I have no idea if anyone is using it. It's easy to use, so I assume that some people are ... Moreover which Iax client do you use to test this? I'm actually not aware of any IAX clients that have implemented encryption. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] modem through Zaptel/Digium?
This is just a low priority curiosity question because I have a usable workaround. I have Digium card that uses the Zaptel driver (can't get to my home machine right now to get the exact model, but it probably doesn't matter). It's a card with one POTS line and three extension hookups. I'm using Asterisk 1.4 and Zaptel 1.4.7 . One of the extension ports is connected to a modem on another computer. This is a FAX modem that works well; I have * programmed to detect incoming faxes and route them to this modem, and it works seamlessly. I can also send outbound faxes with no problem. The curiosity is that this modem does not work for dialup unless I bypass the * server and connect it directly to the wallplate, then it works fine. I don't see why it would be able to detect carrier and negotiate with a fax machine through * and Zaptel, but not with a dialup server. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SVN mirror back up to date
The public Asterisk SVN mirror is back up to date. I apologize for the inconvenient downtime. Re-syncing with a repository that has almost 100,000 revisions took a while. :) -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iax Encryption
Hello, from what I've understood Iax2 should support aes128 encryption. I've found this old info: http://www.voip-info.org/wiki/view/IAX+encryption and this (unanswered?) post http://lists.digium.com/pipermail/asterisk-security/2005-August/60.h tml Is this the libiax used currently on asterisk http://ftp.digium.com/pub/libiax/ ? I would like to understand if someone is using this in production. Moreover which Iax client do you use to test this? Best Regards, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL includes?
AEL was an experimental feature in Asterisk 1.2.x and you may not implement all funcionts. Jay Moore wrote: voip*CLI ael reload Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown root token '#include' Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box, and I don't want to upgrade our only production computer. Jay Rodrigo R Passos wrote: Jay, What error? Jay Moore wrote: How do I include a file (not a context) in AEL? #include filename returns an error. Thanks, Jay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Device state of SIP doesn't change
Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Encryption
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant I would like to understand if someone is using this in production. I have no idea if anyone is using it. It's easy to use, so I assume that some people are ... I guess what you are meaning here is it's easy to configure on asterisk side. So this encryption is now considered robust enough to be used in production? I'm asking this because of comments I've found here: http://www.voip-info.org/wiki/index.php?page=IAX%20encryption about beta stage encryption. Thanks, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?
Cavalera Claudio Luigi wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson However, you'll need to do similar things to your asterisk box router if it's behind NAT for IAX as you do for SIP. (You will need a static IP address on the NAT router and port-forward 4569 to the asterisk box, just as you'd port-forward 5060 and 1-2 for SIP) Please correct me if I'm wrong, for Iax clients you don't need to do static port-forwarding as they will create upon registration one entry in NAT table with UDP port for both signalling and media. On the other hand, sip clients (without Stun) are difficult to manage behind Nat because of RTP/RTCP ports. I don't want to start a flame Iax vs Sip, just to clarify respective advantages. Best Regards, Claudio I believe you are correct, as long as the client sends *something* to the server at frequent enough intervals that the router keeps the connection in it's active list. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] More voicemail cards needed...
Thank you all for the voicemail cards you sent. If you have the following in PDF or laying around (scan): * ATT/Cingular flow voicemail card * Verizon flow voicemail card * Sprint flow voicemail card * TMobile flow voicemail card * Alltel flow voicemail card * Avaya Nortel Octel flow voicemail card * Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one I will work on getting these integrated with EVM. Users will be able to select via user prefs and admin on a per user setting of their preferred VM flow. Final prompts are coming this week; need the cards for any additions. I have the following: Audix, 3COM, IC IMail, NEC EliteMail LX, Nortel Call Pilot, Olle's, and a customized Octel. Feel free to send others that may be of interest. Send all cards to: nt_jnewman at yahoo.com. Justin Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modem through Zaptel/Digium?
Greg Woods wrote: This is just a low priority curiosity question because I have a usable workaround. I have Digium card that uses the Zaptel driver (can't get to my home machine right now to get the exact model, but it probably doesn't matter). It's a card with one POTS line and three extension hookups. I'm using Asterisk 1.4 and Zaptel 1.4.7 . One of the extension ports is connected to a modem on another computer. This is a FAX modem that works well; I have * programmed to detect incoming faxes and route them to this modem, and it works seamlessly. I can also send outbound faxes with no problem. The curiosity is that this modem does not work for dialup unless I bypass the * server and connect it directly to the wallplate, then it works fine. I don't see why it would be able to detect carrier and negotiate with a fax machine through * and Zaptel, but not with a dialup server. --Greg I think asterisk has the ability to detect fax tones and disable echo cancellation for those calls. I don't know if that is the case with a regular modem call. I'd check to make sure that echo cancellation is disabled on the extension the modem is plugged into. The only other idea is to try connecting at a lower speed (I would think this would happen automatically though). -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More voicemail cards needed...
TMOB http://support.t-mobile.com/knowbase/root/public/tm22131.htm Thanks, Steve Totaro On Jan 17, 2008 1:54 PM, Justin Newman [EMAIL PROTECTED] wrote: Thank you all for the voicemail cards you sent. If you have the following in PDF or laying around (scan): * ATT/Cingular flow voicemail card * Verizon flow voicemail card * Sprint flow voicemail card * TMobile flow voicemail card * Alltel flow voicemail card * Avaya Nortel Octel flow voicemail card * Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one I will work on getting these integrated with EVM. Users will be able to select via user prefs and admin on a per user setting of their preferred VM flow. Final prompts are coming this week; need the cards for any additions. I have the following: Audix, 3COM, IC IMail, NEC EliteMail LX, Nortel Call Pilot, Olle's, and a customized Octel. Feel free to send others that may be of interest. Send all cards to: nt_jnewman at yahoo.com. Justin Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk desktop tools for OS X
Yaah!!! Mac! I am a big user of OS X. Can't help it. Macs eye candy draws me in like my wofe. :) And.. I've never had a single issue with it. I also host virtual Ubuntu, Red Hat and XP :( on the same box using VMware. Sorry about the Mac rant. Just glad to see some Mac / Asterisk attention... I have multiple Asterisk servers in place and would REALLY be interested in your tool set. I can test it on Leopard or Tiger as I have both in available. Thanks, Jim - Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project (http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modem through Zaptel/Digium?
Is it bridging the Zap channels? We have asterisk doing FXO-FXS modem calls working fine, the key is making sure the channels are bridging and EC is NOT turning on. If you have anything preventing that the modem calls won't work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Thursday, January 17, 2008 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] modem through Zaptel/Digium? Greg Woods wrote: This is just a low priority curiosity question because I have a usable workaround. I have Digium card that uses the Zaptel driver (can't get to my home machine right now to get the exact model, but it probably doesn't matter). It's a card with one POTS line and three extension hookups. I'm using Asterisk 1.4 and Zaptel 1.4.7 . One of the extension ports is connected to a modem on another computer. This is a FAX modem that works well; I have * programmed to detect incoming faxes and route them to this modem, and it works seamlessly. I can also send outbound faxes with no problem. The curiosity is that this modem does not work for dialup unless I bypass the * server and connect it directly to the wallplate, then it works fine. I don't see why it would be able to detect carrier and negotiate with a fax machine through * and Zaptel, but not with a dialup server. --Greg I think asterisk has the ability to detect fax tones and disable echo cancellation for those calls. I don't know if that is the case with a regular modem call. I'd check to make sure that echo cancellation is disabled on the extension the modem is plugged into. The only other idea is to try connecting at a lower speed (I would think this would happen automatically though). -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Proxy Issues
On Jan 17, 2008 2:28 PM, Nicholas Blasgen [EMAIL PROTECTED] wrote: I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using X-Lite I have no issue with settings as follows: Display Name: Any Name User name: 0057510 Password: 0057510 Authorization user name: blank Domain: directnationalloan.com Checked Register with domain and Send outbound via: Proxy Address: las-obproxy.voipzone.us X-Lite has no issues with registration or placing calls. Now the fun part, Asterisk I've been able to get to register. register = [EMAIL PROTECTED]: 0057510:[EMAIL PROTECTED] It's the placing of calls that I'm getting an error. I've tried so many different configurations that it's somewhat pointless to show you my settings. The one I've been playing around with most recently is: [voipexten] auth=0057510:[EMAIL PROTECTED] username=0057510 secret=0057510 fromdomain= directnationalloan.com type=peer qualify=yes insecure=port,invite outboundproxy=las-obproxy.voipzone.us But of corse that doesn't work. Maybe someone here has an idea. -- /Nick Try dropping the auth line and changing the outboundproxy to host= ? Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PostgreSQL query results truncated 255 characters
I am querying an postgresql database from my 1.4.13 system and the results seem to be truncating each column at 255 characters. The columns are typed as character varying 1000. Any suggestion on how to remove this limit? TIA Vic ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Callback
Hi all Someone has make a voicemail callback on * ? Thanks -- Gilberto Nunes Itajaí - SC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Proxy Issues
I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using X-Lite I have no issue with settings as follows: Display Name: Any Name User name: 0057510 Password: 0057510 Authorization user name: blank Domain: directnationalloan.com Checked Register with domain and Send outbound via: Proxy Address: las-obproxy.voipzone.us X-Lite has no issues with registration or placing calls. Now the fun part, Asterisk I've been able to get to register. register = [EMAIL PROTECTED]: 0057510:[EMAIL PROTECTED] It's the placing of calls that I'm getting an error. I've tried so many different configurations that it's somewhat pointless to show you my settings. The one I've been playing around with most recently is: [voipexten] auth=0057510:[EMAIL PROTECTED] username=0057510 secret=0057510 fromdomain=directnationalloan.com type=peer qualify=yes insecure=port,invite outboundproxy=las-obproxy.voipzone.us But of corse that doesn't work. Maybe someone here has an idea. -- /Nick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modem through Zaptel/Digium?
On Jan 17, 2008 1:28 PM, Jeremy Mann [EMAIL PROTECTED] wrote: Is it bridging the Zap channels? We have asterisk doing FXO-FXS modem calls working fine, the key is making sure the channels are bridging and EC is NOT turning on. If you have anything preventing that the modem calls won't work. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Thursday, January 17, 2008 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] modem through Zaptel/Digium? Greg Woods wrote: This is just a low priority curiosity question because I have a usable workaround. I have Digium card that uses the Zaptel driver (can't get to my home machine right now to get the exact model, but it probably doesn't matter). It's a card with one POTS line and three extension hookups. I'm using Asterisk 1.4 and Zaptel 1.4.7 . One of the extension ports is connected to a modem on another computer. This is a FAX modem that works well; I have * programmed to detect incoming faxes and route them to this modem, and it works seamlessly. I can also send outbound faxes with no problem. The curiosity is that this modem does not work for dialup unless I bypass the * server and connect it directly to the wallplate, then it works fine. I don't see why it would be able to detect carrier and negotiate with a fax machine through * and Zaptel, but not with a dialup server. --Greg I think asterisk has the ability to detect fax tones and disable echo cancellation for those calls. I don't know if that is the case with a regular modem call. I'd check to make sure that echo cancellation is disabled on the extension the modem is plugged into. The only other idea is to try connecting at a lower speed (I would think this would happen automatically though). -Dave Try setting the modem to 9600 baud. It will probably work. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] buffer-issue when piping live-streams into musiconhold
Hi Folks, I'm currently trying to configure musiconhold (on a asterisk-1.4.17) for replaying a live mp3-stream (Icecast2). after reading the related material on voip-info and several other pages, I've successfully tried out mpg132, madplay and mplayer to pipe a stream into moh. however, there is one major problem involving some kind of buffer-issue. let me try to explain this problem using a timeline: 10:00 I'm calling the pbx, musiconhold starts correctly to play the live-stream (almost live, with very small delay) - that's OK. 10:01 I hangup. -- than I pause for 20 min -- 10:20 I'm calling a second time. However moh now doesn't stream live, but starts to continue playing the stream from 10:01. This goes on for about 30secs, then the replay stops for a second and continues at the correct position (once again, rather live). along I get this message at the console: [Jan 17 20:37:15] NOTICE[6220]: res_musiconhold.c:544 monmp3thread: Request to schedule in the past?!?! [Jan 17 20:37:15] NOTICE[6220]: res_musiconhold.c:544 monmp3thread: Request to schedule in the past?!?! I've installed the ztdummy-module as I've read that the message Request to schedule in the past?!?! might have something to do with that, however this didn't help. It looks like there's some kind of buffering going on... Thanks a lot for any suggestions, at this point I'm rather clueless ;) regards, michael musiconhold.conf: [default] mode=custom application=/etc/asterisk/mohstream.sh mohstream.sh #!/bin/bash /usr/bin/wget -q -O - http://my.stream.com:8000 | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 -a -12 - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Device state of SIP doesn't change
Atis Lezdins wrote: Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. Of course, i checked UPGRADE.txt, and lot of other resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in order for SIP devices to report proper device state. I see in your sip.conf file that you have set call-limit in the general section. This setting, however, may only be set per peer (or user). Unfortunately, there's no warning message output if an unrecognized option is set in the general section. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk desktop tools for OS X
I'm interested too Devraj, please send a copy of if possible to try it. Thanks. On Jan 17, 2008 12:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project (http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk desktop tools for OS X
Thanks for your response guys. There are still some issues with the code (Svn on SourceForge). I am working on getting these fixed up and will post a message when its ready for download. I will yell out if I need some Asterisk/Cocoa help. Thanks a lot. On Jan 18, 2008 7:19 AM, Adrià Vidal [EMAIL PROTECTED] wrote: I'm interested too Devraj, please send a copy of if possible to try it. Thanks. On Jan 17, 2008 12:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project ( http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] buffer-issue when piping live-streams into musiconhold
thx a lot russel...your hack actually works!! :) Meanwhile I've found something about the musiconhold-conf-option cachertclasses, which might help in starting a separate instance for every caller. however, that didn't really work for me... probably this option only works for mode=files?! http://www.asterisk.org/doxygen/trunk/Config_moh.html http://lists.digium.com/pipermail/asterisk-commits/2007-November/017911.html anyway, thx a lot for your suggestions :) regards, michael On Jan 17, 2008 9:52 PM, Russell Bryant [EMAIL PROTECTED] wrote: Michael Kamleitner wrote: 10:00 I'm calling the pbx, musiconhold starts correctly to play the live-stream (almost live, with very small delay) - that's OK. 10:01 I hangup. -- than I pause for 20 min -- 10:20 I'm calling a second time. However moh now doesn't stream live, but starts to continue playing the stream from 10:01. This goes on for about 30secs, then the replay stops for a second and continues at the correct position (once again, rather live). along I get this message at the console: snip musiconhold.conf: [default] mode=custom application=/etc/asterisk/mohstream.sh mohstream.sh #!/bin/bash /usr/bin/wget -q -O - http://my.stream.com:8000 | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 -a -12 - Most players don't work quite correctly with Asterisk MOH. For it to work the way you expect, the player you are using must throw away the audio when Asterisk isn't currently reading from the stream. There was a magic version of mpg123 (0.59r IIRC) that did that, and that is why it was the recommended version. If you're reading from a raw TCP stream, then you can use the small streamplayer utility included with Asterisk. Otherwise, I don't really have a good suggestion for you right now. I suppose that you could use some sort of hack to ensure that music on hold is always playing so that the stream is being serviced. extensions.conf: [moh_hack] exten = hack,1,Answer exten = hack,n,StartMusicOnHold(default) exten = hack,n,While(1) exten = hack,n,Wait(300) exten = hack,n,EndWhile() *CLI originate Local/[EMAIL PROTECTED] application Echo -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mag. Michael Kamleitner - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - E-Mail: [EMAIL PROTECTED] Xing: https://www.xing.com/profile/Michael_Kamleitner - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Phone: +43 699 116 07 923 - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Web: http://www.kamleitner.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk desktop tools for OS X
Hi Tzafrir, Yes it does use the Manager Interface. It account does require call level access. That may then result in umlimited access to Asterisk (well to originate calls anyway). However I have made real conscious efforts to filter messages that are being transmitted over the socket so the application doesn't listen or talk on behalf of a single extension. If this is a concern, is every desktop application that integrates using the Manager Interface a problem for Asterisk administrators? Also, what is a way around it then? I see desktop tools for Asterisk being one of the biggest advantages over traditional PBXes. On Jan 18, 2008 7:19 AM, Adrià Vidal [EMAIL PROTECTED] wrote: I'm interested too Devraj, please send a copy of if possible to try it. Thanks. On Jan 17, 2008 12:25 PM, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have been long working on a project ( http://asterisktools.org, to be released under GPL) that aims to provide desktop tools for Macs. I am finally getting to the release stages of this application and hope to have an early BETA available next weekend. If there is anybody who is interested in this tool, please send me an email as I am looking for people who can test the application for me before we make a final release. The code is already available via SVN and there are some really cool and thoughtful features. Thanks a lot. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] not understanding Cisco call manager connection for incoming calls
I am connected to CCM and have a sip.conf entry like: [CCMHEART] type=friend host=X.y.X.A allow=ulaw allow=alaw allow=all canreinvite=yes qualify=yes context=CCMHEART In extensions.conf I have a context of: [CCMHEART] exten = s,1,Goto(default,s,1) exten = 45801,1,Goto(default,s,1) exten = 4545801,1,Goto(default,s,1) I do have a default context. However calling the above 4545801 number asterisk does not answer as it says it cannot find the 45801 in the current context. Once I put the 3 context lines above in the default context asterisk answers just fine. Why do I need to put the 3 lines in the default context? The sip.conf entry has the context being CCMHEART shouldnt it look there? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging Recording File
I am looking to see if anyone has seen this problem before. I am
setting the MEETME_RECORDINGFILE variable in a macro, then using the r
option with the Page application to record the page. But the page is
only recorded to the file specified in MEETME_RECORDINGFILE
sometimes... Sometimes it works and sometimes it doesn't. When it
doesn't work it places the recorded file in the sounds dir with a
meetme-conf-. name. Here is my Macro.
Basically it is getting my phones that begin with a certain number
from the realtime database to create a variable with a value that ='s
SIP/6001SIP/6002SIP/6003 this is passed to the macro as ARG1
I added a System command to log the variables to a text file so I know
when the page is made, the variables are correct.
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to be paged
; ARG2 = Device for the PA system. If the user selected to
; page the PA system. That will be included.
;
exten = s,1,Set(MEETME_RECORDINGFORMAT=wav)
exten = s,2,Set(MEETME_RECORDINGFILE=custom/paging/${EPOCH})
exten = s,3,System(/bin/echo ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} $
{MEETME_RECORDINGFORMAT} ${MEETME_RECORDINGFILE} /var/log/asterisk/
pagemacro_var.log)
exten = s,4,MYSQL(Connect connid ${realdb_host} ${realdb_user} $
{realdb_pass} ${realdb_db})
exten = s,5,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\
WHERE\ name\ LIKE\ '${ARG1}%')
exten = s,6,MYSQL(Fetch fetchid ${resultid} number)
exten = s,7,GoToIf($[${fetchid} = 1]?8:10)
exten = s,8,Set(pagedevice=${pagedevice}SIP/${number})
exten = s,9,GoToIf($[${fetchid} = 1]?6:10)
exten = s,10,Set(pagedevice=${pagedevice:1})
exten = s,11,MYSQL(Clear ${resultid})
exten = s,12,MYSQL(Disconnect ${connid})
exten = s,13,GoToIf($[${ARG2} != ]?14:15)
exten = s,14,Set(pagedevice=${pagedevice}${ARG2})
exten = s,15,SIPAddHeader(Call-Info:answer-after=0)
exten = s,16,SIPAddHeader(Alert-Info: Ring Answer)
exten = s,17,NoOp(Page Recording ${MEETME_RECORDINGFILE})
exten = s,18,Set(CALLERID(all)=System Page 1010)
exten = s,19,Page(${pagedevice},r)
;On hangup, run script that will email the recording to shared
conference.
exten = h,1,System(/var/lib/asterisk/scripts/mail_lastpage ${ARG1} $
{MEETME_RECORDINGFILE})
exten = h,2,Hangup()
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk between two Asterisks
This is my configuration in the extensions.conf, iax.conf at Site A and Site B, so anyone can help why the call refused? Site A: [IPLink] type=friend context=IPLinkIncoming host=192.168.2.3 usename=IPLink secret=password canreinvite=no nat=no [SiteBInternal] exten = _2XX,1,Dial(IAX2/[EMAIL PROTECTED]) exten = _2XX,2,Playback(vm-nobodyavail) exten = _2XX,3,Hangup() exten = _2XX,102,Playback(tt-allbusy) exten = _2XX,103,Hangup() [IPLinkIncoming] include = SiteBInternal include = SiteBExternal And at Site B: [IPLink] type=friend context=IPLinkIncoming host=192.168.2.2 usename=IPLink secret=password canreinvite=no nat=no [SiteAInternal] exten = _2XX,1,Dial(IAX2/[EMAIL PROTECTED]) exten = _2XX,2,Playback(vm-nobodyavail) exten = _2XX,3,Hangup() exten = _2XX,102,Playback(tt-allbusy) exten = _2XX,103,Hangup() [IPLinkIncoming] include = SiteAInternal include = SiteAExternal Regards Bilal -- Hi All; I did an IP Trunk using IAX between two Asterisk boxes, now Asterisk A can send a call for B but B refuse it. The IAX type was configured to be friend in the iax.con for Asterisk A and B, is there any thing else need to be done to let B accept the call from A? Also, I used an static IP address for the host when I configured the iax client in the iax.conf file. Any help? Regards Bilal I used to see this problem when I used to use IAX2. Sometimes it would just go away. I seem to remember using insecure=very to get it working but I may be wrong. Anyways, post the relevant parts of your IAX2 confs from both boxes and someone might be able to spot something right off the bat. Thanks, Steve Totaro Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Remotely Cancel Call Forward
Kevin Kiely wrote: I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn’t work and removing the mac-phone.cfg caused problems Remove the XML element tag from within mac-phone.cfg that it updated with the forwarding information and then reboot it again. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Remotely Cancel Call Forward
I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn't work and removing the mac-phone.cfg caused problems ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] buffer-issue when piping live-streams into musiconhold
Michael Kamleitner wrote: 10:00 I'm calling the pbx, musiconhold starts correctly to play the live-stream (almost live, with very small delay) - that's OK. 10:01 I hangup. -- than I pause for 20 min -- 10:20 I'm calling a second time. However moh now doesn't stream live, but starts to continue playing the stream from 10:01. This goes on for about 30secs, then the replay stops for a second and continues at the correct position (once again, rather live). along I get this message at the console: snip musiconhold.conf: [default] mode=custom application=/etc/asterisk/mohstream.sh mohstream.sh #!/bin/bash /usr/bin/wget -q -O - http://my.stream.com:8000 | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 -a -12 - Most players don't work quite correctly with Asterisk MOH. For it to work the way you expect, the player you are using must throw away the audio when Asterisk isn't currently reading from the stream. There was a magic version of mpg123 (0.59r IIRC) that did that, and that is why it was the recommended version. If you're reading from a raw TCP stream, then you can use the small streamplayer utility included with Asterisk. Otherwise, I don't really have a good suggestion for you right now. I suppose that you could use some sort of hack to ensure that music on hold is always playing so that the stream is being serviced. extensions.conf: [moh_hack] exten = hack,1,Answer exten = hack,n,StartMusicOnHold(default) exten = hack,n,While(1) exten = hack,n,Wait(300) exten = hack,n,EndWhile() *CLI originate Local/[EMAIL PROTECTED] application Echo -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP client in asterisk not trying to contact IMAP server
On Jan 17, 2008 7:55 AM, KodaK [EMAIL PROTECTED] wrote: Thanks, if that was in any of the docs I just completely glossed over it. I'll give it a shot. Yes, I skipped over that in the docs. I'm good at that. Thanks for the help. I've also written up a quickie how-to on how to enable this on a trixbox system. Don't know how helpful it is, but it's there. http://www.trixbox.org/wiki/trixbox-imap --J(K) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] buffer-issue when piping live-streams into musiconhold
Michael Kamleitner wrote: thx a lot russel...your hack actually works!! :) Awesome. :) Meanwhile I've found something about the musiconhold-conf-option cachertclasses, which might help in starting a separate instance for every caller. however, that didn't really work for me... probably this option only works for mode=files?! http://www.asterisk.org/doxygen/trunk/Config_moh.html http://lists.digium.com/pipermail/asterisk-commits/2007-November/017911.html Well, that option only exists in Asterisk trunk, and is only relevant when using realtime for music on hold. I assume you're probably using one of the released versions of Asterisk, so this wouldn't be available. anyway, thx a lot for your suggestions :) You're quite welcome. I'm glad I could help out. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC errors
On Wed, 2008-01-16 at 15:52 -0800, Steven wrote: I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri 1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a TE220B. The system load is below 0.10. I moved the server into production, with one PRI, on Friday. On that day we handled a couple thousand calls and I only saw one HDLC abort message. On Saturday half the calls and two abort messages an hour apart. On Sunday, after 1500 when there was only a couple calls, the HDLC messages went crazy. We're getting non-stop Abort messages, with Bad FCS thrown in about every tenth message. They come in bunches, with short 10-30 second breaks. Then every once and awhile there is an 30 minute break, sometimes a 3 hour break. The messages seems completely separate from system load. The system will be idle and get the messages and have no messages when I load up dozens of calls on it (using call files to complete calls) I'm not sure what to try next, other than calling the telco and asking them to check their equipment. Does any one have a suggestion before I do that? Hi Steven Some quick remarks. Generally, you would get this kind of messages if either the signal level is to low, or distorted. The clock-signal for the RX is regenerated from the RX-signal. Now-a-days, these lowerlevel protocols are (should be) completely handeled by the hw of the board (not sure, don't have one here), not by the system. So system load should be not an issue. Those dl360 are not capable of hosting much power greedy pci-boards, so i would not suspect your psu. You wrote you received before thousand calls without much of a problem. So although ESD-damage of the board can not entirely ruled out, its unlikely. You mention went into production, Did this imply moving of the system from a testing room into a server-location? Other (longer) cables? Perhaps you can check with your telco wether they receive bad frames coming from you hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Remotely Cancel Call Forward
Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I re-touched the file and followed your suggestion. Any way of removing the call forwarding feature via the xml configs? Kevin Kiely wrote: I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn't work and removing the mac-phone.cfg caused problems Remove the XML element tag from within mac-phone.cfg that it updated with the forwarding information and then reboot it again. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.5/1228 - Release Date: 1/16/2008 9:01 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC errors
I have a suggestion. Have you contacted Digium technical support for assistance with resolving this issue? Excellent suggestion. Make sure you can give them SSH access and screen so you can see what they are doing. Before that, check (remake) your T1 cables and if it is punched down on a block, re-punch it. I'm used to vendors that aren't responsive, so I never even thought of it. They've told me to try running patlooptest (which I will tonight), to see if the problem is in the card. Thanks for your suggestions. Hopefully I'll learn something tonight. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDLC errors
Andrew Joakimsen wrote: I'll assume you chose trixbox to make your life easier when it comes to dealing with others regarding the PBX. Pretty much, yes. What is between the smartjack and your T1 card? What sort and length of cable? Any splices? Punchdown or patch panels? About 100 feet of RJ48 (yes, STP) which tests fine. Though I was thinking of moving the server to be in the same room as the telco box to see ensure its not the cables. Also I'm not sure if Trixbox has this but ssh in and see if there is an application called zttool. What are the statistics it is providing? I don't have any IRQ misses: IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 24/ 24/ 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Remotely Cancel Call Forward
When setting a forward on the phone, the phone will upload to your ftp server a modified macaddr-phone.cfg XML file that (amongst other locally made changes) contains an OVERRIDE statement similar to this: OVERRIDE reg.1.fwdContact= reg.1.fwdStatus=1 ... / Change the .fwdStatus attribute to 0, then reboot the phone (sip notify polycom-check-cfg peername). That will removed the forward just fine, at least in my setup here. Works the other way as well: modify the XML file to list a valid .fwdContact and set .fwdStatus to 1, then reboot the phone. That phone won't ring again until the forward is disabled :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk
Hi there this is an interesting topic that I see here and a problem that I am trying to solve too. But I was wondering if the forwarding solution will work for my case. So I have two Asterisk boxes A and B. A is behind a corporate NAT such that A can SSH to B, but not vice versa( One-way SSH ) . The UDP port 5060 of the corporate NAT is blocked off and I will not be able to have it unblocked for security reasons. Hence, is my only choice using an SSH tunnel between A and B for the IAX connection to work? Will it work though with that One-way SSH factor mentioned before? Thanks John From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 2 Jan 2008 16:29:45 + Subject: Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk Sure, but if (as is often the case) you only have control over the firewall at one end of the link, you set the forwarding at the end you control and have the far end to register to you every 30 seconds. Tim. On 2 Jan 2008, at 15:13, Rob Hillis wrote: Perhaps. I've never been one to trust that firewalls operate as they should - I've been bitten far too many times by a firewall that doesn't quite behave as you expect. Also, when diagnosing network connectivity problems, I find that it helps to have the rules in place rather than having to infer the rule. Tim Panton wrote: If you are careful, you only need to setup a port forward at one end of the IAX trunk. Have one Asterisk register (regularly) with the other. The second asterisk (server) will need to have port 4569 forwarded through it's router. The first asterisk (client) wont need any port forwarding. Tim. On 2 Jan 2008, at 10:18, Rob Hillis wrote:The reason that IAX2 is considered good for NAT issues is that it uses only one port for both control messages and voice traffic as opposed to SIP that uses a predictable port for control messages and an unpredictable one for voice/video traffic. If both servers are behind NAT servers, you will need to ensure that the appropriate UDP port (by default 4569) are forwarded to your Asterisk servers. Only this port is required - RTP isn't used by IAX2. bilal ghayyad wrote: Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like TCP/UDP port mapping or IP forwarding)? How that will be it if possible? Or I have to do a kind of port mapping? If I will need to use port mapping, then I have to map the TCP and UDP ports that are determined in iax.conf and rtp.conf files at site A for asterisk ip address at site A? Or I have to map the TCP and UDP ports that are in iax.conf and rtp.conf at site B for asterisk ip address at site A? In other words, if I am at site B then I have to go for router B and do mapping for TCP/UDP ports of the asterisk at site B or the asterisk at site A? Any help. Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Shed those extra pounds with MSN and The Biggest Loser!!
Re: [asterisk-users] HDLC errors
You mention went into production, Did this imply moving of the system from a testing room into a server-location? Other (longer) cables? Unplugged the current system and hooked up a new, longer, cable to the asterisk system. The cable is RJ48 STP, about 100 feet. However we ran several cables and swapping them around doesn't make a difference; they all test good too. We could be having bad luck with them :-) I was thinking of moving the server to be beside the telco box, but that is a large undertaking. Perhaps you can check with your telco wether they receive bad frames coming from you I gave them a call and they'll run a report and get back to me. Hopefully the patlooptest tonight will point to the problem. Thank you everyone for all your suggestions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7910 Handsets: Skinny protocol?
Dear all. I have about 30 Cisco 7910 handsets, and my basic research has told me that they are not SIP based handsets. Not to worry for now, I just need them to connect to my asterisk server. They are giving me a bit of a hard time. Has anyone here had any experience on how to do this? Documentation on the internet is seriously lacking. Best regards, Mr Gabriel Ogunleye IT Administrator Fusis Group Fusis House, 4 Maple Grove Business Centre Lawrence Road Hounslow, TW4 6DR, UK T: +44(0)845 9000 375 F: +44(0)845 9000 376 M: +44(0)7956 540 134 E: [EMAIL PROTECTED] W: www.fusis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys PAP2 NA
Dear all, I have managed to connect this device to my asterisk box, but it is giving me a bit of a hard time. I can call other extensions from this box, but I am not able to call this one. It seems to permanently remain engaged. When I dial it, this is the message I get. Is there a know issue in this regard? What can I do to test the actual current status of the line? Best regards, Mr Gabriel Ogunleye IT Administrator Fusis Group Fusis House, 4 Maple Grove Business Centre Lawrence Road Hounslow, TW4 6DR, UK T: +44(0)845 9000 375 F: +44(0)845 9000 376 M: +44(0)7956 540 134 E: [EMAIL PROTECTED] W: www.fusis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk
On Thu, 2008-01-17 at 17:09 -0800, John Constalgie wrote: Hence, is my only choice using an SSH tunnel between A and B for the IAX connection to work? Will it work though with that One-way SSH factor mentioned before? It's my understanding that SSH tunneling will only work with TCP traffic. IAX2 uses UDP packets, so I don't think that'll work. You might try setting up a VPN or something along those lines. (Also, IAX2 defaults to port 4569, not port 5060.) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 NA
On 1/18/08, Mr Gabriel Ogunleye [EMAIL PROTECTED] wrote: Dear all, I have managed to connect this device to my asterisk box, but it is giving me a bit of a hard time. I can call other extensions from this box, but I am not able to call this one. It seems to permanently remain engaged. When I dial it, this is the message I get. Is there a know issue in this regard? What can I do to test the actual current status of the line? What do you get? Enable sip set debug in CLI. Is it behind NAT? We have a lot of them successfully working. Sometimes they crash and needs reboot, but generally they are ok. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?
Diax is probably the smallest Windows softphone. Add to that list Mozphone (http://mozphone.mozdev.org/) that can be installed in Firefox Kiax : http://sourceforge.net/projects/kiax shameless plugMy MediaX softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php/shameless plug iaxcomm : http://iaxclient.sourceforge.net/iaxcomm/ The one from Sokol associates : http://www.sokol-associates.com/?q=node/29 There is other ones also, Google is your friend As for a hardware IAX phone, I can't recommend one as I never tried one. hth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Remotely Cancel Call Forward
I guess I was interested in Disabling the forwarding feature completely via the config. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Thursday, January 17, 2008 7:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Remotely Cancel Call Forward When setting a forward on the phone, the phone will upload to your ftp server a modified macaddr-phone.cfg XML file that (amongst other locally made changes) contains an OVERRIDE statement similar to this: OVERRIDE reg.1.fwdContact= reg.1.fwdStatus=1 ... / Change the .fwdStatus attribute to 0, then reboot the phone (sip notify polycom-check-cfg peername). That will removed the forward just fine, at least in my setup here. Works the other way as well: modify the XML file to list a valid .fwdContact and set .fwdStatus to 1, then reboot the phone. That phone won't ring again until the forward is disabled :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.5/1228 - Release Date: 1/16/2008 9:01 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Remotely Cancel Call Forward
I misread then. Even though your original message said you wanted to un-forward a phone. That can be done with the recipe BJ and I outlined. I am not aware of any way to disable the forward function, i.e. prevent a user from forwarding in the first place. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk
Jared Smith wrote: On Thu, 2008-01-17 at 17:09 -0800, John Constalgie wrote: Hence, is my only choice using an SSH tunnel between A and B for the IAX connection to work? Will it work though with that One-way SSH factor mentioned before? It's my understanding that SSH tunneling will only work with TCP traffic. IAX2 uses UDP packets, so I don't think that'll work. You might try setting up a VPN or something along those lines. (Also, IAX2 defaults to port 4569, not port 5060.) OpenVPN works great for this. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Remotely Cancel Call Forward
In your per-phone configuration: phone1 reg ... divert divert.fwd.1.enabled = 0 divert.fwd.2.enabled = 0 divert.fwd.3.enabled = 0 divert.fwd.4.enabled = 0 divert.fwd.5.enabled = 0 divert.fwd.6.enabled = 0 / This removes the soft-key and disallows the option from the menu. I can't stand that feature as the soft-key is terribly misplaced, everytime you go hit 'end call', if the other user hangs up first, half our users ended up forwarding their phone to an invalid extension on accident. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Kiely Sent: Thursday, January 17, 2008 17:48 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Remotely Cancel Call Forward I guess I was interested in Disabling the forwarding feature completely via the config. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Thursday, January 17, 2008 7:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Remotely Cancel Call Forward When setting a forward on the phone, the phone will upload to your ftp server a modified macaddr-phone.cfg XML file that (amongst other locally made changes) contains an OVERRIDE statement similar to this: OVERRIDE reg.1.fwdContact= reg.1.fwdStatus=1 ... / Change the .fwdStatus attribute to 0, then reboot the phone (sip notify polycom-check-cfg peername). That will removed the forward just fine, at least in my setup here. Works the other way as well: modify the XML file to list a valid .fwdContact and set .fwdStatus to 1, then reboot the phone. That phone won't ring again until the forward is disabled :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.5/1228 - Release Date: 1/16/2008 9:01 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk
Good question. I have never tried tunneling IAX over SSH but it seems like it should work just like anything else. How about a port opened up for OpenVPN. You know you can run IAX on any port you wish, port 80 may work for you if you have some extra external IPs not being used for HTTP. The same is true for OpenVPN. Thanks, Steve Totaro On Jan 17, 2008 8:09 PM, John Constalgie [EMAIL PROTECTED] wrote: Hi there this is an interesting topic that I see here and a problem that I am trying to solve too. But I was wondering if the forwarding solution will work for my case. So I have two Asterisk boxes A and B. A is behind a corporate NAT such that A can SSH to B, but not vice versa( One-way SSH ) . The UDP port 5060 of the corporate NAT is blocked off and I will not be able to have it unblocked for security reasons. Hence, is my only choice using an SSH tunnel between A and B for the IAX connection to work? Will it work though with that One-way SSH factor mentioned before? Thanks John -- From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 2 Jan 2008 16:29:45 + Subject: Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk Sure, but if (as is often the case) you only have control over the firewall at one end of the link, you set the forwarding at the end you control and have the far end to register to you every 30 seconds. Tim. On 2 Jan 2008, at 15:13, Rob Hillis wrote: Perhaps. I've never been one to trust that firewalls operate as they should - I've been bitten far too many times by a firewall that doesn't quite behave as you expect. Also, when diagnosing network connectivity problems, I find that it helps to have the rules in place rather than having to infer the rule. Tim Panton wrote: If you are careful, you only need to setup a port forward at one end of the IAX trunk. Have one Asterisk register (regularly) with the other. The second asterisk (server) will need to have port 4569 forwarded through it's router. The first asterisk (client) wont need any port forwarding. Tim. On 2 Jan 2008, at 10:18, Rob Hillis wrote: The reason that IAX2 is considered good for NAT issues is that it uses only one port for both control messages and voice traffic as opposed to SIP that uses a predictable port for control messages and an unpredictable one for voice/video traffic. If both servers are behind NAT servers, you will need to ensure that the appropriate UDP port (by default 4569) are forwarded to your Asterisk servers. Only this port is required - RTP isn't used by IAX2. bilal ghayyad wrote: Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like TCP/UDP port mapping or IP forwarding)? How that will be it if possible? Or I have to do a kind of port mapping? If I will need to use port mapping, then I have to map the TCP and UDP ports that are determined in iax.conf and rtp.conf files at site A for asterisk ip address at site A? Or I have to map the TCP and UDP ports that are in iax.conf and rtp.conf at site B for asterisk ip address at site A? In other words, if I am at site B then I have to go for router B and do mapping for TCP/UDP ports of the asterisk at site B or the asterisk at site A? Any help. Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users
Re: [asterisk-users] Linksys PAP2 NA
What message? NAT? On Jan 17, 2008 8:18 PM, Mr Gabriel Ogunleye [EMAIL PROTECTED] wrote: Dear all, I have managed to connect this device to my asterisk box, but it is giving me a bit of a hard time. I can call other extensions from this box, but I am not able to call this one. It seems to permanently remain engaged. When I dial it, this is the message I get. Is there a know issue in this regard? What can I do to test the actual current status of the line? Best regards, Mr Gabriel Ogunleye IT Administrator Fusis Group Fusis House, 4 Maple Grove Business Centre Lawrence Road Hounslow, TW4 6DR, UK T: +44(0)845 9000 375 F: +44(0)845 9000 376 M: +44(0)7956 540 134 E: [EMAIL PROTECTED] W: www.fusis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 NA
There is no NAT involved, just a straight connection Mr Gabriel Ogunleye IT Administrator -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: 18 January 2008 05:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys PAP2 NA What message? NAT? On Jan 17, 2008 8:18 PM, Mr Gabriel Ogunleye [EMAIL PROTECTED] wrote: Dear all, I have managed to connect this device to my asterisk box, but it is giving me a bit of a hard time. I can call other extensions from this box, but I am not able to call this one. It seems to permanently remain engaged. When I dial it, this is the message I get. Is there a know issue in this regard? What can I do to test the actual current status of the line? Best regards, Mr Gabriel Ogunleye IT Administrator Fusis Group Fusis House, 4 Maple Grove Business Centre Lawrence Road Hounslow, TW4 6DR, UK T: +44(0)845 9000 375 F: +44(0)845 9000 376 M: +44(0)7956 540 134 E: [EMAIL PROTECTED] W: www.fusis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 NA
There is no NAT involved. I think I will try to sip set debug. What exactly should I be looking for? How did you configure these devices - maybe something I missed in the config? Mr Gabriel Ogunleye IT Administrator -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: 18 January 2008 01:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys PAP2 NA On 1/18/08, Mr Gabriel Ogunleye [EMAIL PROTECTED] wrote: Dear all, I have managed to connect this device to my asterisk box, but it is giving me a bit of a hard time. I can call other extensions from this box, but I am not able to call this one. It seems to permanently remain engaged. When I dial it, this is the message I get. Is there a know issue in this regard? What can I do to test the actual current status of the line? What do you get? Enable sip set debug in CLI. Is it behind NAT? We have a lot of them successfully working. Sometimes they crash and needs reboot, but generally they are ok. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
