[Asterisk-Users] Need to interface to BRIs
Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 new update, v0.5.9
No, this is really a bug. Actually, a nasty, untraceable bug. Check bugnotes for details. Michael. Brian West wrote: Was this bug fixed or was it really a bug. I'm reading the bug notes and it doesn't appear to be a bug in asterisk from what Mark said on the notes. bkw On Wed, 11 Feb 2004, Michael Manousos wrote: This new version contains a workaround to an Asterisk bug (see http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001029). This bug caused random segfaults in H.323/SIP calls. Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Need to interface to BRIs
Look at this: http://www.junghanns.net/asterisk/page17.html Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jim Archer Gesendet: Montag, 16. Februar 2004 10:11 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] Need to interface to BRIs Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to interface to BRIs
Hi Jim, we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. You can find more information about it at: http://www.junghanns.net/asterisk/page17.html best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-02-16 um 10.10 schrieb Jim Archer: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to interface to BRIs
I forgot to mention, I am in North America. --On Monday, February 16, 2004 4:10 AM -0500 Jim Archer [EMAIL PROTECTED] wrote: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - oh323 - Cisco CallManager
Did you try the latest version (0.5.9)? Michael. Tomica Crnek wrote: Hi everyone, Does anyone know the answer for this situation? I have Asterisk with E1 PRI links, with SIP phones registered to Asterisk and with h.323 connection to Cisco CallManager. I am using oh323. I think I have a problem with codecs but I do not know exactly what is wrong. this is working ok: -- Call from CallManager (7960) to SIP phone on Asterisk (X-Lite or 7960 with SIP image) - working OK Call from CallManager (7960) to E1 PRI trunk to PSTN through Asterisk - working OK Call from E1 PRI trunk from PSTN through Asterisk to CallManager (7960) - working OK here is the problem --- Call from SIP phone to CallManager - rings the phone, in the moment when called party picks the receiver Asterisk crashes with core dump Interesting is that if you establish a call in opposite direction (from CallManager to SIP phone) prior to that one, Asterisk wouldn't crash sometimes I will appreciate if anyone can help Tomica ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: SIP 481 subscription does not exist with SJPhone
Does anyone have any ideas on how to stop these messages from the SJPhone? everything i've seen says they're harmless, but they're filling my console and if anyone has any ides on how to make them go away i would be appreciative. thanks, yair _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to interface to BRIs
Jim Archer ([EMAIL PROTECTED]) wrote: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks Hi Jim, use Zaptel BRI from www.junghanns.net We are using 2 cards of them (one internally, one at a customer) without any problems. --jan -- Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc. Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need to interface to BRIs
Hi, -Original Message- I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? www.junghanns.net/asterisk Checkout ZapBRI. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Need to interface to BRIs
Hi, On Mon, 16 Feb 2004 at 04:10, Jim Archer wrote: I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? http://www.junghanns.net/asterisk/page17.html cu Reinhard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LDAP authentication
Hi, I'm using SIP channel, and i would like to authenticate users with a LDAP server. Is this feature implemented in Asterik? I have read some posts about it, but i don't know if it's currently available. Thank you very much. .G ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to interface to BRIs
Hi Jim, i forgot to mention that the drivers do not yet support NI-1, but will support it in the near future. Until then the only solution for you will be the Eicon Diva Server 4BRI-8M and chan_capi. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-02-16 um 10.53 schrieb Jim Archer: I forgot to mention, I am in North America. --On Monday, February 16, 2004 4:10 AM -0500 Jim Archer [EMAIL PROTECTED] wrote: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to interface to BRIs
so you need North America ISDN, not EuroIsdn. The only way is a diva server card with capi driver, Klaus zapBri doesn't support NI (as far as I know) Matteo. Il lun, 2004-02-16 alle 10:53, Jim Archer ha scritto: I forgot to mention, I am in North America. --On Monday, February 16, 2004 4:10 AM -0500 Jim Archer [EMAIL PROTECTED] wrote: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: SIP 481 subscription does not exist with SJPhone
This has been talked about many times. They are harmless as far as anyone know. To get rid of them, get a different client. Or tell SJPhone to change there code. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yair hakak Sent: Monday, February 16, 2004 5:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] re: SIP 481 subscription does not exist with SJPhone Does anyone have any ideas on how to stop these messages from the SJPhone? everything i've seen says they're harmless, but they're filling my console and if anyone has any ides on how to make them go away i would be appreciative. thanks, yair _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zhone + call transfer
After finding a spot to put the Zhone Zplex so that the fan noise doesn't annoy anyone, I've got everything working to an acceptable level except for call transfers. No matter what I do the Zhone doesn't seem to be passing 'flash' key presses on to asterisk, ie whenever I try to transfer a call, nothing happens. The DTMF tones pressed after the 'flash' key are simply heard over the conversation. Running asterisk with -vvvc doesn't show anything when trying to transfer a call which leads me to believe that it has something to do with the Zhone. Can anyone confirm that call transfers do in fact work with the Zhone Zplex? Is there anything obvious that I may have missed? ...and yes, I've added the following to Zapata.conf for the appropriate channels: threewaycalling = yes transfer = yes cancallforward = yes Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Need to interface to BRIs
Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... (OP: there's also a 4BRI from Eicon, IIRC. It'll work with * through CAPI, but I'm quite sure that it's a bit more expensive than KP's card ;-)) -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - oh323 - Cisco CallManager
I have just installed 0.5.9. Up to this moment it didn't crash :P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Monday, February 16, 2004 11:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk - oh323 - Cisco CallManager Did you try the latest version (0.5.9)? Michael. Tomica Crnek wrote: Hi everyone, Does anyone know the answer for this situation? I have Asterisk with E1 PRI links, with SIP phones registered to Asterisk and with h.323 connection to Cisco CallManager. I am using oh323. I think I have a problem with codecs but I do not know exactly what is wrong. this is working ok: -- Call from CallManager (7960) to SIP phone on Asterisk (X-Lite or 7960 with SIP image) - working OK Call from CallManager (7960) to E1 PRI trunk to PSTN through Asterisk - working OK Call from E1 PRI trunk from PSTN through Asterisk to CallManager (7960) - working OK here is the problem --- Call from SIP phone to CallManager - rings the phone, in the moment when called party picks the receiver Asterisk crashes with core dump Interesting is that if you establish a call in opposite direction (from CallManager to SIP phone) prior to that one, Asterisk wouldn't crash sometimes I will appreciate if anyone can help Tomica ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapRAS + RADIUS authentication
Does it work? I can't find if there is a config option in pppd config to make pppd authenticate user against radius server. Tomica
Re: [Asterisk-Users] Zhone + call transfer
ah ah what is the trasnfer hook time for you phone? Usa phones seems to have a very long time... like 1secs, and that's the default for zaptel... here in italy we have flash time about 120 msec, instead so, try that: edit zaptel.h in your zaptel src dir, search for ZT_DEFAULT_RXFLASHTIME (line 802 in current cvs), and lower the value from 1250 to 200, for example AND lower ZT_MAXPULSETIME (line 805) from (150 * 8) to something like (20 * 8)... but not lower than ZT_MINPULSETIME (15 * 8). compile, install, reload modules, restart asterisk, and let us know. Matteo. Il lun, 2004-02-16 alle 12:50, Kent Williams ha scritto: After finding a spot to put the Zhone Zplex so that the fan noise doesn't annoy anyone, I've got everything working to an acceptable level except for call transfers. No matter what I do the Zhone doesn't seem to be passing 'flash' key presses on to asterisk, ie whenever I try to transfer a call, nothing happens. The DTMF tones pressed after the 'flash' key are simply heard over the conversation. Running asterisk with -vvvc doesn't show anything when trying to transfer a call which leads me to believe that it has something to do with the Zhone. Can anyone confirm that call transfers do in fact work with the Zhone Zplex? Is there anything obvious that I may have missed? ...and yes, I've added the following to Zapata.conf for the appropriate channels: threewaycalling = yes transfer = yes cancallforward = yes Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Need to interface to BRIs
Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zhone + call transfer
Kent Williams wrote: After finding a spot to put the Zhone Zplex so that the fan noise doesn't annoy anyone, I've got everything working to an acceptable level except for call transfers. No matter what I do the Zhone doesn't seem to be passing 'flash' key presses on to asterisk, ie whenever I try to transfer a call, nothing happens. The DTMF tones pressed after the 'flash' key are simply heard over the conversation. Running asterisk with -vvvc doesn't show anything when trying to transfer a call which leads me to believe that it has something to do with the Zhone. Can anyone confirm that call transfers do in fact work with the Zhone Zplex? Is there anything obvious that I may have missed? ...and yes, I've added the following to Zapata.conf for the appropriate channels: threewaycalling = yes transfer = yes cancallforward = yes Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If your phone has a setting for flash button timing try increasing it. I'm not sure how to tweek the timing of flash within *. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi Phones
Miguel, IPC5000 doesn't support G729 (8 kbps) (it only support G711 64kbps) Be carefull with what you buy. Hector. - Original Message - From: Miguel Cavazos [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Sunday, February 15, 2004 9:38 AM Subject: [Asterisk-Users] Wifi Phones Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based. I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or seen it working, sorry about the unnesesary traffic to the list, my question is simple would this work against asterisk if anyone knows any other Wifi phones besides Wisip and Ciscos expensive toy please tell me. Miguel Cavazos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overhead Paging
Well, I've made a little progress with this. In zaptel.conf I set em=11 (11 is the channel). In zapata.conf I set signalling=em_w and usecallerid=no In extensions.conf I set 311,1,Dial(Zap/11,15) In the Adtran, I set the port to FXS DPO. At the beginning of each page, I hear two DTMF 1 tones (the channel number?), but then I am able to page. Does anyone know how to get rid of the DTMF tones? Thanks, Michael Welter Michael Welter wrote: I've a Valcon V-2001A paging controller connected to an Adtran 750 FXS port. The V-2001A looks like an FXS loop start extension. When I call the extension, I can hear ringing tones and CallerID through the speaker, but the paging controller doesn't answer--it continues to ring. I also hear a relay clicking with each ring in the paging controller. Does anyone have experience with configuring these devices for paging? Thank you, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail extension - hangup
Hello, I have configured 8500 to access voicemailmain. With whatpriority does the control exit when the user hangsup the phone without pressing #. I want to execute an app when the control exits from voicemailmain. Any inputs? Regards Deepak
Re: [Asterisk-Users] Re: Need to interface to BRIs
Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium Zaptel cards? Matteo Brancaleoni wrote: Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analogical FXO vs. BRI dialing speed
When dialing out, will a call be established significantly faster by an ISDN adapter such as an Eicon Diva server compared to an analogical FXO such as Digium's X100P ? signature.asc Description: This is a digitally signed message part
RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface
There seems to be some trouble with either the maillist or my client. I haven't received any of the posted replys on this topic, but found the replys through the asterisk.linkx.net search engine. But anyways here is the reply on the mail from: James H dot Cloos Jr. cloos at jhcloos.com If it is a pri I'd give SetCallerID() a try in the dialplan. It is a PRI and I've tried the SetCallerId() which displays my PRI main number, like the other experiments I've tried. I talked to my Telco provider which said that it isn't possible to hide the number via shortcodes but that it should be done via my PABX. -Original Message- From: Mickey Binder [mailto:[EMAIL PROTECTED] Sent: 13. februar 2004 12:14 To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 license
Hello all, I wanted to know if is there a way to see which of my 4 g729b license is registered in one specific Asterisk box. Is that possible? I could not find any registration record on my box to compare with the license... best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analogical FXO vs. BRI dialing speed
Am Mo, 2004-02-16 um 14.39 schrieb Jean-Marc V. Liotier: When dialing out, will a call be established significantly faster by an ISDN adapter such as an Eicon Diva server compared to an analogical FXO such as Digium's X100P ? Yes, ISDN uses digital signalling so call setup times on the last mile (from your NT1 to the telco switch) are close to 0. Also the callerID on incoming calls is available immediately with ISDN (with analog lines you usually get it after the first ring). best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 30VIP Phones
Hi Has anyone go the 30VIP phone to work with asterisk? If so how good us the usability of the Cisco 30VIP phone with asterisk either using chan_sccp or Chan_skinny? Thanks for your Help Robb -- Robert Boardman Tel:01617737929 FWD:86263 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP Carrier recommendations?
I need to get PSTN connectivity for my asterisk server. Either IAX2 or SIP. Does anyone have recommendations of carriers that provide US termination and will work (doesn't have to be a supported platform) with Asterisk? OH: And you need not recommend Galaxyvoice. I'm not waiting 72hrs for them to set up an account. What a joke. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mailing list lag again
Hi, today's BRI thread showed, that the mailing list has a delay of about an hour again. Is this still due to the Digium relocation, or is something else going on with the list server? cu Reinhard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Messages (SIMPLE)
Hi How can i configure Asterisk for proxing SIP/SIMPLE Messages when the target is registered? How can the user retrieve the waiting-messages? Thank you very much. .G ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g729 license
When you start * from console use -vvvc and the number of detected licenses will be shown when the g729 translator is loaded. Only why that I know of to check this. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Osvaldo Mundim Sent: Monday, February 16, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] g729 license Hello all, I wanted to know if is there a way to see which of my 4 g729b license is registered in one specific Asterisk box. Is that possible? I could not find any registration record on my box to compare with the license... best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IaxTel: Using IaxTel Numbers As Asterisk DIDs
I was wondering if there was any easy way to direct incoming calls from IaxTel to specific extensions without having to create a separate context for each? It seems to be pretty strait forward with FWD, SIPPhone and other SIP based services -- you just add /Extension to the end of the registration. If there isn't any direct way to do that from within the registration statement, how hard would it be for somebody to work with Mark and create an edit screen for your IaxTel account that allowed you to set the context/extension for the call to reach. That would make the number useful ONLY as an Asterisk DID, but that's how many of us use IAXtel. Thoughts? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
On Sat, Feb 14, 2004 at 02:47:11AM -0500, Ryan Finnesey wrote: Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Carrier recommendations?
John, I've recently used nufone and voicepulse, both with great results. Thanks, Chris Clifton - Original Message - From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 16, 2004 9:13 AM Subject: [Asterisk-Users] VOIP Carrier recommendations? I need to get PSTN connectivity for my asterisk server. Either IAX2 or SIP. Does anyone have recommendations of carriers that provide US termination and will work (doesn't have to be a supported platform) with Asterisk? OH: And you need not recommend Galaxyvoice. I'm not waiting 72hrs for them to set up an account. What a joke. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 license
The problem is that I have 2 licenses of 8 channels. One is being used in one of my boxes and the other one is not. What I want is to be sure that the one which I will use in a new Asterisk box is not the one which is being used... Any suggestion? regards Osvaldo On Feb 16, 2004, at 11:57 AM, Wes Marderness wrote: When you start * from console use -vvvc and the number of detected licenses will be shown when the g729 translator is loaded. Only why that I know of to check this. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Osvaldo Mundim Sent: Monday, February 16, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] g729 license Hello all, I wanted to know if is there a way to see which of my 4 g729b license is registered in one specific Asterisk box. Is that possible? I could not find any registration record on my box to compare with the license... best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent / Queue help
Hi, First let me apologize if I sent this to the list twice. Is it possible to kick a caller out of a queue after 5 minutes and goto the next priority in the context where they were assigned to the queue ? My desired result is that even though one agent is dynamically logged into the queue and is on a call, I would like the 2nd caller to stay in the queue for 5 minutes and then timeout to the next priority if the agent is still busy and can't get to the call. Some observations: I have tried the n option with queue (if I don't the 2nd caller will stay in the queue infefinately) eg: exten = 401,1,Queue(support1|n) The problem with using n is that with one agent logged into the queue and he is busy on a call, when the 2nd call is placed in the queue it immediately timesout and goes to the next priority in the context even if timeout=300 is set in queue.conf. Any help appreciated. -bh Here are the configs: extensions.conf [supportq] exten = 401,1, Queue(support1|t) agents.conf [agents] autologoff=15 ackcall=no ;wrapuptime=5000 musiconhold = default queues.conf general] [support1] music = default strategy = leastrecent ;context = leavemessage timeout = 300 retry = 2 maxlen = 0 -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re-Invites and Studder.
Billy Huddleston [EMAIL PROTECTED] wrote: I've been using Asterisk with a Cisco GW and ATA's.. I have it setup with re-invites. When a call is first answered, the 1st second or so of the conversation is stuttered, garbled, whatever you want to call it.. I believe this is due to Asterisk shifting the media stream directly to the Gateway or ATA. Is their a way to eliminate this stutter without disabling re-invites? This is very discontenting to our customers and employees... Why don't you turn off re-invites as a test and see if the problem goes away. Then you will know for sure if this is causing the problem. Or to put it another way, the first step in solving a problem is identifying the problem. :) Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Just to clarify this from a different direction, Oftel/Ofcom approach these things by say that they are 'technology neutral', i.e. as standard they don't care how the service is delivered, it is the service that is regulated and not the delivery mechanism. This means in theory the rules for VoIP are the same for copper, wireless, mobile etc. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pingtel Phones?
[EMAIL PROTECTED] wrote: Hello All, Does anyone here have any experience with pingtel Xpressa hard phones? I am considering buying a couple. Already have Snom200s, but want something with better CTI and full duplex speakerphone. Michael Michael, I used some Pingtel phones with * and ended up having to scrap them. Pingtel assured that they would work with * but when they didn't, they put it all off on Asterisk. They also didn't seem to be interested in getting them to work. In my opinion they were dishonest with through the whole deal and I wouldn't recommend doing business with them or using that phone. One persistent issue is that the phone would lock up if you had more that 2 calls even though they said it could handle 11 simultaneous calls. I talked to at least one other user who had the same issues. Everything was fine in the lab, but upon roll out the phones didn't work right. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
Steve Kennedy wrote: On Sat, Feb 14, 2004 at 02:47:11AM -0500, Ryan Finnesey wrote: Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Steve I am going to an Oftel meeting to discuss VoB regulation next week.. Hopefully this will help to see where it is heading.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface
The correct way to hide your callerid on a PRI interface is to set the presentation indicator. Some CO switches do a basic sanity check on the callerid they receive. If you set the number string to empty but the presentation indicator to allow the number they will replace the number string by your main number. I do not know how or if possible to change the presentation indicator on * but a look in libpri should give some clues. - Alfred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mickey Binder Sent: Monday, February 16, 2004 6:13 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface There seems to be some trouble with either the maillist or my client. I haven't received any of the posted replys on this topic, but found the replys through the asterisk.linkx.net search engine. But anyways here is the reply on the mail from: James H dot Cloos Jr. cloos at jhcloos.com If it is a pri I'd give SetCallerID() a try in the dialplan. It is a PRI and I've tried the SetCallerId() which displays my PRI main number, like the other experiments I've tried. I talked to my Telco provider which said that it isn't possible to hide the number via shortcodes but that it should be done via my PABX. -Original Message- From: Mickey Binder [mailto:[EMAIL PROTECTED] Sent: 13. februar 2004 12:14 To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g729 license
I know that during the Registration that the file /var/lib/va-certificate is created. Maybe this will help, file is encrypted so it don't offer much information. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Osvaldo Mundim Sent: Monday, February 16, 2004 11:03 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] g729 license The problem is that I have 2 licenses of 8 channels. One is being used in one of my boxes and the other one is not. What I want is to be sure that the one which I will use in a new Asterisk box is not the one which is being used... Any suggestion? regards Osvaldo On Feb 16, 2004, at 11:57 AM, Wes Marderness wrote: When you start * from console use -vvvc and the number of detected licenses will be shown when the g729 translator is loaded. Only why that I know of to check this. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Osvaldo Mundim Sent: Monday, February 16, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] g729 license Hello all, I wanted to know if is there a way to see which of my 4 g729b license is registered in one specific Asterisk box. Is that possible? I could not find any registration record on my box to compare with the license... best regards Osvaldo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: GS BT-100 echo
Tim Sailer wrote: I picked up a GS 100 phone based on the overall good response I've heard of these phones. One thing I'm fighting with, which I can't find any info on, is a *real* bad local echo on the GS. The remote end doesn't hear it, and all the docs I see about echocancel deal with hardwired phones/ports (fxs/fso). Phone software is: Software Version: Program--1.0.4.45Bootloader--1.0.0.13HTML--1.0.0.20 if that matters. sip.conf for the phone is: [gs1] type=friend username=gs1 secret= host=dynamic canreinvite=no nat=yes qualify=1000 disallow=all allow=alaw allow=ulaw Tim Tim, Look harder in the mailing list and at the WIKI. There are literaly hundreds, if not thousands, of posts on this exact issue. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
Linus Surguy wrote: Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Just to clarify this from a different direction, Oftel/Ofcom approach these things by say that they are 'technology neutral', i.e. as standard they don't care how the service is delivered, it is the service that is regulated and not the delivery mechanism. This means in theory the rules for VoIP are the same for copper, wireless, mobile etc. Linus As I understand it that is what the Ofcom VoB discussion next week is all about.. The standard line telco's have to be required to provide a service in an emegency eg during a power failure, but this is impossible for a VoIP provider sine the provider does not have control over the full path or the electricity supply.. That is only one example where VoIP cannot be regulated in the same way as standard telephone services.. In my mind there will have to be seperate regulations, there may well be some common clauses but they will still be seperate regulations.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN LCR USING ASTERISK
hi everybody,
here is what I've done to make my asterisk* act as a LCR.
first, you'll have to install isdnrate (part of isdnutils) and get a recent
rate-??.dat (check rates4linux.sourceforge.net for that.)
to test isdnrate just try the following command:
lcr -o -b3 -l60 *any_number_you_want_to_test*
the -o tells isdnrate to only use provides activated in /etc/isdn/rate.conf
(e.g. if you have some preselection providers or tisdn-xxl)
-b3 is for the best 3 providers
-l60 says call duration 60 seconds (the default-value I also used in my AGI
is 153 secs. so if you want to use annother duration please change the
commandline in the agi)
then I wrote a very little (and simple) AGI.
-
/lcr.agi ---
---
#!/usr/bin/perl
$|=1;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
$number = $ARGV[0];
$length = 60;
$raw = `/usr/bin/isdnrate -o -b1 -L -l$length $number`;
$raw =~ /([0-9]*)_.;(.*?);/;
$prefix = $1;
$provider = $2;
print VERBOSE \Using LCR Provider $provider - $prefix!\\n;
$result = STDIN;
print SET VARIABLE LCR $prefix\n;
$result = STDIN;
as you can see, my AGI just sets a variable called LCR.
here is how I use it in my dialplan:
exten = _0.,1,Answer
exten = _0.,2,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = _0.,3,ResponseTimeout,10 ; Set Response Timeout to 10
seconds
exten = _0.,4,agi,/lcr.agi|${EXTEN:${TRUNKMSD}}
exten = _0.,5,Dial,CAPI/@6294096:b${LCR}${EXTEN:${TRUNKMSD}}|60|T
I know that this is all very simple - and maybe there are some errors in my
setup but I just wanted to share my expirence with you.
bye
... thorsten
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Re: [Asterisk-Users] Hide outgoing CallerId on Zap interface
Mayby you missed my reply as well. Here it is again ... When I need to hide callerid ( sip phones ), I will configure this in sip.conf. You need to include restrictcid=yes for each user that needs to be hidden. -- Pertti Mickey Binder wrote: There seems to be some trouble with either the maillist or my client. I haven't received any of the posted replys on this topic, but found the replys through the asterisk.linkx.net search engine. But anyways here is the reply on the mail from: James H dot Cloos Jr. cloos at jhcloos.com If it is a pri I'd give SetCallerID() a try in the dialplan. It is a PRI and I've tried the SetCallerId() which displays my PRI main number, like the other experiments I've tried. I talked to my Telco provider which said that it isn't possible to hide the number via shortcodes but that it should be done via my PABX. -Original Message- From: Mickey Binder [mailto:[EMAIL PROTECTED] Sent: 13. februar 2004 12:14 To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - Carrier Access Bank Ring through
Title: Asterisk - Carrier Access Bank Ring through I'm currently having a very odd problem with Asterisk. I have a 24-Port FXO Access Bank I, and everything appears ok on the T1. No alarms, no issues at all. The channel bank seems to be answering the line immediately, and I'm getting some strange background noises. Idea's anyone? Derek Samford Net Phone Blue, Inc. (zapata.conf) [channels] context=default signalling=fxs_ks group=1 signalling=fxs_ks channel = 1-24 (zaptel.conf) span=1,1,0,esf,b8zs fxsks=1-24
Re: [Asterisk-Users] Pingtel Phones?
Andy, Thanks for the feedback! I ended up buying one phone on Ebay for $202.50, which seems like a good price. It would be used in a very small office under light load. Perhaps I can get it nto work suffient for my needs. Otherwise, back to Ebay to sell the beast ;-) Michael On Mon, 16 Feb 2004 10:42:22 -0600, Andy Hester wrote: [EMAIL PROTECTED] wrote: Hello All, Does anyone here have any experience with pingtel Xpressa hard phones? I am considering buying a couple. Already have Snom200s, but want something with better CTI and full duplex speakerphone. Michael Michael, I used some Pingtel phones with * and ended up having to scrap them. Pingtel assured that they would work with * but when they didn't, they put it all off on Asterisk. They also didn't seem to be interested in getting them to work. In my opinion they were dishonest with through the whole deal and I wouldn't recommend doing business with them or using that phone. One persistent issue is that the phone would lock up if you had more that 2 calls even though they said it could handle 11 simultaneous calls. I talked to at least one other user who had the same issues. Everything was fine in the lab, but upon roll out the phones didn't work right. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] Behind every great man is a great woman rolling her eyes. ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZapRAS + RADIUS authentication
On Mon, 2004-02-16 at 06:15, Tomica Crnek wrote: Does it work? I can't find if there is a config option in pppd config to make pppd authenticate user against radius server. It should, but that is really a question for a ppp list or FAQ. First link from google for me was this... http://www.xs4all.nl/~evbergen/radius-pppd.html Please do not make us start the UTFG or UTFW all over again with a vengeance. I only used the terms pppd linux radius. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analogical FXO vs. BRI dialing speed
On Mon, 2004-02-16 at 07:39, Jean-Marc V. Liotier wrote: When dialing out, will a call be established significantly faster by an ISDN adapter such as an Eicon Diva server compared to an analogical FXO such as Digium's X100P ? Analog, nothing logical there. ISDN will be faster dialing out as you will communicate with asterisk via the dialpad where you want to be connected too, and if you are on a analog line, asterisk will repeat the digits to the telco switch in analog just like you did but at a specific cadence. Since a DTMF digit is around 450 to 800 msec, and in that time frame you can transfer all the call setup information digitally, the call could be setup in the equivalent of a single digits time, let alone the next 6-10 digits. Incoming, the calls are again signaled digitally and acknowledged with the switch in less time than it takes to make the first half of a ring. On analog you will want to wait till the second or third ring to get the CallerID, but it was there to start with on the ISDN call. On my PRI line, calls are answered and prompts played without a single ring event being heard by the caller. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 30VIP Phones
On Mon, 2004-02-16 at 08:10, Robert Boardman wrote: Hi Has anyone go the 30VIP phone to work with asterisk? If so how good us the usability of the Cisco 30VIP phone with asterisk either using chan_sccp or Chan_skinny? Thanks for your Help I have had mine working with the chan_skinny. There is new work being done with chan_sccp to support more features. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
The problem with the Ofcom consultation as I see it is that it seems to be regressive wrt to the position now being taken by the FCC. There are probably not many more than 250,000 VoB users worldwide so now is not the time to impose significant market constraints. The new EU regulatory framework actually imposes very few constraints on new service providers in emerging markets such as VoIP being based as it is on the concept of significant market power (SMP). I don't think any carrier has SMP in VoB so the real issue is the extent to which Ofcom tinkers in the interpretation of the rules. Unfortunately they seem to be focusing on the red herrings of emergency service support and lawful intercept - neither of which are of much interest to users. Fixed and mobile services already provide acceptable emergency access. The real issue is the umbrella topic of Universal Service Provision and what the impact of VoIP will be on that. The tone of the Ofcom invitation to the VoB briefing focused on issues that could limit the market rather than promote it. Let's hope that the VoB briefing is followed up by some balanced and broad based consultation. Iain --On Monday, February 16, 2004 5:55 pm + WipeOut [EMAIL PROTECTED] wrote: Linus Surguy wrote: Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Just to clarify this from a different direction, Oftel/Ofcom approach these things by say that they are 'technology neutral', i.e. as standard they don't care how the service is delivered, it is the service that is regulated and not the delivery mechanism. This means in theory the rules for VoIP are the same for copper, wireless, mobile etc. Linus As I understand it that is what the Ofcom VoB discussion next week is all about.. The standard line telco's have to be required to provide a service in an emegency eg during a power failure, but this is impossible for a VoIP provider sine the provider does not have control over the full path or the electricity supply.. That is only one example where VoIP cannot be regulated in the same way as standard telephone services.. In my mind there will have to be separate regulations, there may well be some common clauses but they will still be separate regulations.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
Just to clarify this from a different direction, Oftel/Ofcom approach these things by say that they are 'technology neutral', i.e. as standard they don't care how the service is delivered, it is the service that is regulated and not the delivery mechanism. This means in theory the rules for VoIP are the same for copper, wireless, mobile etc. As I understand it that is what the Ofcom VoB discussion next week is all about.. The standard line telco's have to be required to provide a service in an emegency eg during a power failure, but this is impossible for a VoIP provider sine the provider does not have control over the full path or the electricity supply.. That is only one example where VoIP cannot be regulated in the same way as standard telephone services.. Thats not completely true - UK regulations say that a standard POTS analogue phone line must work in the event of power failure, and the same is true for a single ISDN line installation, but nothing else is actually covered - if you have a PRI ISDN30 install it is actually your responsibily to make it work in a power failure condition by providing UPS etc - if you want to. Equally VoIP tends not to fall under this requirement. I think we can expect that the meeting next week is going to primarily concentrate on a) 999 emergency calling requirements and b) numbering issues. Whilst there may be some other coverage of PATS/non-PATS issues* I'm sure these will be the main focus. Linus * Other PATS issues are things like directory enquiries/operator assistance/providing directories/itemised billing/service for the blind etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC-S cards?
hi everybody, Acer ISDN-Surf, Billion Bipac ISDN, Trust PCI ISDN Modem, D-LINK DMI-128+ to name a few ;-) anyone knows where to get one of theses cards (or any other based on the HFC-S chipset) in germany? my computer-trader maybe can get d-link's card but he don't know how long it could take. does anyone (hello kapejod ;)) ) know wich one should be my first choise, just in case I find more than one of these babys. thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Need to interface to BRIs
The FritzCard has CAPI drivers and does NOT provide zaptel timing. The quadBRI PCI has zaptel drivers and does provide zaptel timing. Am Mo, 2004-02-16 um 14.41 schrieb Master Abi: Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium Zaptel cards? Matteo Brancaleoni wrote: Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Vonage -- no make that VoicePulse Connect :)
VoicePulse--- If I get 2 simultaneous calls? For one local number from voicepulse, is it possible to get simultaneous incoming calls? Jeff Chen www.mutualphone.com Yahoo messenger ID: jeffcheny2k -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Knox Sent: February 16, 2004 2:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Vonage -- no make that VoicePulse Connect :) Thanks Michael, VoicePulse does have local number, so I just provisioned one :) Now I am setting it up, no problem so far, the next question is. If I get 2 simultaneous calls on my inbound will one ring busy or will asterisk handle this for me? I would like to be able to receive multiple simultaneous calls if possible for an application I am developing. Also, I assume to go through my router (NAT) I just need to open up the one port, 5036, right? Whopee! I can give Vonage it's device and high prices back soon :) Too bad they couldn't play nice and allow other devices on their service. I used Vonage for a year, up until last month in fact. However, there doesn't seem to be a way to avoind using their ATA/MTA if you use their service. You won't be able to connect directly to their servers with your * box. SI switched to VoicePulse Connect. I actually prefer their 2.9 cent/minute rates as opposed to a flat $35/month. I rarely use 1000 minutes/month so I'm saving money over Vonage. There was a down side in that VP doesn't offer DIDs in my area. Therefore I still have 2 POTS lines for incomming calls, but that was likely going to stay that way anyhow, as abackup to when the ISP or ITSP have problems. Also, in order to pass SIP through your router you're going to have open up potentially a lot of ports. I prefer to connect to VPC using IAX2, which requires that I open only one port. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] I am easily satisfied with the very best - Winston Churchill The questions arisen, is this a prison? Some say it is, but I say it isn't. - Ian Hunter --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.577 / Virus Database: 366 - Release Date: 2/4/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FS: Adtran TotalAccess 850 Channel Bank,Router,4x4FXS
For Sale: Adtran 850 Channel Bank with Router, running latest firmware A.04.04.26 Four FXS cards (that's 16 FXS ports for 16 separate asterisk extensions). This is a great channel bank for Asterisk, when paired with a T100P card. Auction ends in about six hours, currently at $370. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3077414453 Thanks, rm - Roderick Montgomery [EMAIL PROTECTED] URL:http://thecomplex.com/ the fool stands only to fall, but the wise trip on grace... [Sarah Masen] - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Need to interface to BRIs
Klaus-Peter Junghanns [EMAIL PROTECTED] said: Yes, like any zaptel device it supports echo cancelation (in software). Good. You can get 2 quadBRI PCI for the price of 1 Eicon 4BRI-8M. Only? ;-) -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ZapRAS + RADIUS authentication
thanks :) From: [EMAIL PROTECTED] on behalf of Steven Critchfield Sent: pon 16.2.2004 20:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ZapRAS + RADIUS authentication On Mon, 2004-02-16 at 06:15, Tomica Crnek wrote: Does it work? I can't find if there is a config option in pppd config to make pppd authenticate user against radius server. It should, but that is really a question for a ppp list or FAQ. First link from google for me was this... http://www.xs4all.nl/~evbergen/radius-pppd.html Please do not make us start the UTFG or UTFW all over again with a vengeance. I only used the terms pppd linux radius. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
[Asterisk-Users] X100P analogue cards and impedance matching
Hi - I have looked but can't find details of the impedance that the analogue PSTN interface card X100P presents. Am I right in assuming it is a resistive 600 ohm match? If so, is there anything I can do (in software or hardware?!) to 'tweak' its impedence to the complex impedence normally presented by a UK phone line? The reason I ask is that old chestnut - echo...and I wonder whether resolving the impedence match might overcome some of the echo issues. Rgds Tim Robinson Basingstoke, UK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Need to interface to BRIs
What is the advantage of having zaptel timing? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter Junghanns Sent: Monday, February 16, 2004 11:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Need to interface to BRIs The FritzCard has CAPI drivers and does NOT provide zaptel timing. The quadBRI PCI has zaptel drivers and does provide zaptel timing. Am Mo, 2004-02-16 um 14.41 schrieb Master Abi: Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium Zaptel cards? Matteo Brancaleoni wrote: Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrading asterisk yields broken pipe
Hello list, I am attempting to upgrade asterisk on a production box. I have opted to set INSTALL_PREFIX to /usr/local/asterisk-0.7.2 which is ugly (since it makes /usr, /var, /etc directories in there), but I didn't want the new install to overwrite my existing installation. The new asterisk runs, but when a call tries to go through, I get six of Ouch ... error while writing audio data: : Broken pipe errors, and then a segfault. Reading elsewhere, I have discovered that the error is coming from mpg123. The process table yields 6 mpg123 processes which appear to be playing the on hold music (onhold_low.mp3, onhold_high.mp3, etc.) Presently, I don't know where to go. 1. I am trying to test a new compile of asterisk while retaining the ability to revert to the old compilation. Am I going about it the right way (by setting the INSTALL_PREFIX to a different dir)? 2. If the answer to #1 is yes, then what might be the problem with mpg123? - Web Design and Web Hosting provided by http://www.ezITsolutions.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Got my DID, getting an error.
Hi there, Got my DID from VoicePulse. Very fast and quite cheap :) I configured the iax.conf with the info they provided, I am getting a good connect to their server, but when I try to dial my number I am seeing the following on the console. Anyone got an idea? Oh I replaced the number below with a generic # :) Thanks! Connected to Asterisk CVS-02/15/04-11:38:41 currently running on Asterisk-Test (pid = 11062) Feb 16 17:00:01 NOTICE[1150528304]: chan_iax2.c:4689 socket_read: Rejected connect attempt from 66.234.228.132, request '[EMAIL PROTECTED]' does not exist --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.577 / Virus Database: 366 - Release Date: 2/5/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ZapRAS + RADIUS authentication
I really should have toned down my response before. For whatever reason, google gives different results to different people. One may only experience this when they use someone else's computer to do well known searches. I apparently have good google juice when it comes to linux related searches. Sometimes other searches take a bit more time to dig through. I have noticed that some things I search for, the links I want come right to the top of the list, or at least on the first page. Yesterday I was on someone else's computer and dug 5 pages deep without seeing the link I wanted on known exact search terms. Anyways, happy googling. On Mon, 2004-02-16 at 15:26, Tomica Crnek wrote: thanks :) From: [EMAIL PROTECTED] on behalf of Steven Critchfield Sent: pon 16.2.2004 20:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ZapRAS + RADIUS authentication On Mon, 2004-02-16 at 06:15, Tomica Crnek wrote: Does it work? I can't find if there is a config option in pppd config to make pppd authenticate user against radius server. It should, but that is really a question for a ppp list or FAQ. First link from google for me was this... http://www.xs4all.nl/~evbergen/radius-pppd.html Please do not make us start the UTFG or UTFW all over again with a vengeance. I only used the terms pppd linux radius. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can you savage a failed call transfer
I have a couple of cummsy user who always lose a call when the transfer is not done properly ie due to dialing a wrong digit, etc. My question is that is it possible to savage a failed call transfer? -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
Re: [Asterisk-Users] Asterisk and Vonage -- no make that VoicePulse Connect :)
On Sun, 15 Feb 2004 21:00:53 -0500, Tom Knox wrote: Thanks Michael, VoicePulse does have local number, so I just provisioned one :) Now I am setting it up, no problem so far, the next question is. If I get 2 simultaneous calls on my inbound will one ring busy or will asterisk handle this for me? I would like to be able to receive multiple simultaneous calls if possible for an application I am developing. I think this is possible, but since I don't have a DID from them I can't say for sure. I can say that I can make up to 6 calls at the same time with one account. They love that since it eats up minutes. I usew to to conference in co-workers. Also, I assume to go through my router (NAT) I just need to open up the one port, 5036, right? Not 5036, that's for SIP. Use 4569 for IAX2. Whopee! I can give Vonage it's device and high prices back soon :) Too bad they couldn't play nice and allow other devices on their service. Yeah, I felt the same way. If you've had the service for a while they leave you with the ATA. One day I'll unlock mine and puit it to use. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] With us or against us isn't a policy worthy of a democratic superpower. -- Zbigniew Brzezinski, Former US National Security Advisor ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
On Mon, Feb 16, 2004 at 07:38:15PM +, Iain Stevenson wrote: The problem with the Ofcom consultation as I see it is that it seems to be regressive wrt to the position now being taken by the FCC. There are probably not many more than 250,000 VoB users worldwide so now is not the time to impose significant market constraints. The new EU regulatory framework actually imposes very few constraints on new service providers in emerging markets such as VoIP being based as it is on the concept of significant market power (SMP). I don't think any carrier has SMP in VoB so the real issue is the extent to which Ofcom tinkers in the interpretation of the rules. Not on the end-users, but as a Communications Service Provider what differentiates you from a regular POTS Communications Service Provider ? In the UK it's all covered by the Communications Act and a telco (as they were) has to meet 21 obligations under the Act. SMP refers to companies such as BT in the UK (who dominate the market), and potentially Mercury in the past, as these are specifically regulated. Unfortunately they seem to be focusing on the red herrings of emergency service support and lawful intercept - neither of which are of much interest to users. Fixed and mobile services already provide acceptable emergency access. The real issue is the umbrella topic of Universal Service Provision and what the impact of VoIP will be on that. Emegency service support isn't a red herring, it's an obligation for fixed line operators (and definately to residential users). Currently VoB is mainly geared at people with DSL or cable modem access i.e. they already have a phone line (this is definately true for BT's service they recently introduced, you MUST be a residential customer with an existing phone service, no QoS guarantees etc). Lawful intercept is also a must both as part of the Comms Act and RIP Act (fixed and mobile operators have facilities to do this, a VoIP provider shouldn't be treated differently). The tone of the Ofcom invitation to the VoB briefing focused on issues that could limit the market rather than promote it. Let's hope that the VoB briefing is followed up by some balanced and broad based consultation. Ofcom is ensuring the Comms Act is adhered to ... Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM card loses Dial tone
I have a cordless phone that causes this same thing to happen every time I plug it into a digium fxs port. I have an old style tdm card and a new one, same results with both. Don't know what it is about the phone that makes this happen; It works fine plugged into a pots line. The phone is a uniden 900mhz dss. -Jeff On Sat, 14 Feb 2004 14:49:49 -0500 Ulexus [EMAIL PROTECTED] wrote: Same here. I, too have received replacement cards from Digium, and I have even tried replacing the proSLICs, all to no avail. Also to note: the same port on each (of three) cards always goes out first. On Thursday, 12 February, 2004 19:22, John Vozza wrote: Same here... Usually after several of these show up in my system log: Power alarm on module 1, resetting! Need to unload/reload module wcfxs in order to get the dial tone back. Happens several times a week, sometimes more frequently. John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Thu, 12 Feb 2004, Youness El Andaloussi wrote: I experienced similar problems too with a 4 chan tdm400. This seems to especially happen when you make configuration changes. It has nothing to do with runing X or no, it does not even have to do with redhat... I experienced the same problem on mandrake. One thing you have to be extra careful is when restarting, make sure that all the modules have entirely reloaded before expecting a dialtone with an asterisk debug console asterisk -r... many of the times I thought there was no dialtone and the asterisk process had gone cukoo, I noticed that configuration was not entirely reload. Yet, reloading many times seems to get some of the TDM400 channels hung. On the other hand, this problem does not seem to happen as extensively when no reloads are made ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What can cause a Red alarm?
I know having it unplugged from the line will cause this, but it's not. It's an X101P single port FXO card. Most of the time it works fine but occasionally wigs out. In this case zttool shows a red alarm. Other times I call into it and it answers but I just hear a buzzing sound. In a day or two I'll try it again and it'll be back to normal I have an el cheapo POTS phone right there to make sure the line is good. There is a second, identical FXO card in the machine as well, which we haven't used in a long time. Thoughts? (Besides swap the two cards and try the other one) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Need to interface to BRIs
On Monday 16 February 2004 15:52, Tim Petlock wrote: What is the advantage of having zaptel timing? There are a host of features, such as conferencing and music-on-hold which require a hardware device for timing. Not just any device will work (it has to generate exactly 1000 cycles per second). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Got my DID, getting an error.
On Mon, 2004-02-16 at 16:16, Tom Knox wrote: Hi there, Got my DID from VoicePulse. Very fast and quite cheap :) I configured the iax.conf with the info they provided, I am getting a good connect to their server, but when I try to dial my number I am seeing the following on the console. Anyone got an idea? Oh I replaced the number below with a generic # :) Thanks! Connected to Asterisk CVS-02/15/04-11:38:41 currently running on Asterisk-Test (pid = 11062) Feb 16 17:00:01 NOTICE[1150528304]: chan_iax2.c:4689 socket_read: Rejected connect attempt from 66.234.228.132, request '[EMAIL PROTECTED]' does not exist Could it be that to dial out you need a 1 in front of the 10 digit number. The 1 should probably be important, and without it you fail to match an outbound extension pattern matching on it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can you savage a failed call transfer
On Tue, 2004-02-17 at 02:42, dkwok wrote: I have a couple of cummsy user who always lose a call when the transfer is not done properly ie due to dialing a wrong digit, etc. My question is that is it possible to savage a failed call transfer? What kind of transfer, what kind of telephony device? On Zap if you use 3 way calling, if you miss dial the extension, you just flash hook back and try it again. On Zap, you could always park the call and then dial the otherside and tell them how to connect to the call. This one benefits from the potential of music on hold and the ability to get the call back, or even taking it somewhere else. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Need to interface to BRIs
On Mon, 2004-02-16 at 15:52, Tim Petlock wrote: What is the advantage of having zaptel timing? Music on hold, and meetme require some form of timing. Also it makes your sip phones better able to deal with VAD(silence suppression) by providing a throttle to the audio. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter Junghanns Sent: Monday, February 16, 2004 11:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Need to interface to BRIs The FritzCard has CAPI drivers and does NOT provide zaptel timing. The quadBRI PCI has zaptel drivers and does provide zaptel timing. Am Mo, 2004-02-16 um 14.41 schrieb Master Abi: Does the Fritz!Card PCI and Quad BRI also provide timing like the Digium Zaptel cards? Matteo Brancaleoni wrote: Il lun, 2004-02-16 alle 12:49, Cees de Groot ha scritto: Klaus-Peter Junghanns [EMAIL PROTECTED] said: we have a 4 BRI solution for Asterisk, the quadBRI PCI ISDN. One thing I'd like to know about this card: Echo Cancellation? I've replaced by Fritz!Card PCI by a Diva Server 2M, and the difference is remarkable... since is zaptel based, it shares same zaptel routines for EC, as far as I know. Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analogical FXO vs. BRI dialing speed
Steven Critchfield wrote: On Mon, 2004-02-16 at 07:39, Jean-Marc V. Liotier wrote: When dialing out, will a call be established significantly faster by an ISDN adapter such as an Eicon Diva server compared to an analogical FXO such as Digium's X100P ? Analog, nothing logical there. ISDN will be faster dialing out as you will communicate with asterisk via the dialpad where you want to be connected too, and if you are on a analog line, asterisk will repeat the digits to the telco switch in analog just like you did but at a specific cadence. Since a DTMF digit is around 450 to 800 msec, and in that time frame you can transfer all Eh? A DTMF digit is about 100ms - roughly 50ms on and 50ms off. Your overall conclusion is right though. Digital is much faster. On a PRI T1 some managers complain they only get 23 channels, while they would get 24 if the used robbed bit lines. However, for lines carrying lots of short calls the faster call setup on a PRI means it is usually a significant win overall. the call setup information digitally, the call could be setup in the equivalent of a single digits time, let alone the next 6-10 digits. Incoming, the calls are again signaled digitally and acknowledged with the switch in less time than it takes to make the first half of a ring. On analog you will want to wait till the second or third ring to get the CallerID, but it was there to start with on the ISDN call. On my PRI line, calls are answered and prompts played without a single ring event being heard by the caller. This can be a little confusing for the caller, but thankfully it also screws up a lot of telemarketer systems. Dialogic et al don't recognise the phone as properly answered if they never hear the ringback tone :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk for a call center?
Hi All... I am using Asterisk successfully in my small office as a fairly ordinary PBX. I am quite happy with it. I have a friend who needs to build a call center. The call center will be used to take orders. I have two big questions. First, can Asterisk be configured accept calls on a bunch of incoming lines, answering with a greeting and telling the person that they will be transferred to the next available operator. Then, can it watch all the extensions, and route the calls to these extensions on a first in, first out basis? Can operators somehow tell Asterisk they are ready for another call or are on break? Second, could I use a VoIP service instead of BRI lines? I experimented with iConnect quite a while ago, and the biggest problem I had with it was that I could only have one line of iConnect. I expect the software has improved since then. Thanks! Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
Iain Stevenson wrote: The problem with the Ofcom consultation as I see it is that it seems to be regressive wrt to the position now being taken by the FCC. There are probably not many more than 250,000 VoB users worldwide so now is not the time to impose significant market constraints. Why do you quote VoB, when the use of broadband versus other internet connections is totally arbitrary? The figure you quote seems far too low for voice over internet (rather than VoIP, since a lot of the IP is on private nets). I think you will find each of the major producers of VoIPs phone has produced rather more than that. Business users alone, dumping their PBXs, must accounts for millions of lines by now. Some of that traffic goes branch to branch over private nets, but they do a lot of interconnecting with the PSTN too. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi Phones
yes it does and g723.1 also, im about to buy it, i will be sending feedback to the list as soon as i get my unit. I really like alot how it looks, hopefully i will also love how it works:) Miguel On Mon, 2004-02-16 at 12:57, HQ wrote: Miguel, IPC5000 doesn't support G729 (8 kbps) (it only support G711 64kbps) Be carefull with what you buy. Hector. - Original Message - From: Miguel Cavazos [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Sunday, February 15, 2004 9:38 AM Subject: [Asterisk-Users] Wifi Phones Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based. I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or seen it working, sorry about the unnesesary traffic to the list, my question is simple would this work against asterisk if anyone knows any other Wifi phones besides Wisip and Ciscos expensive toy please tell me. Miguel Cavazos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2827 - 16 msgs
if my question is stupid just ignore please. using SIP is the communication between the IP phones and the asterisk server secure encrypted ? Kemal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pingtel Phones?
I have 3 Pingtel phones and have tested them since they were prototypes. I have had no lockups or weird problems with them on Asterisk. I will says this about them though: These phones are BIG on features and extensibility through Java at the cost of quality. It doesn't take a lot of work to find that the handsets are noisy and that non-typical use can easily hose up he phone. Also, these phones appear to leak resources which cause them to need a reboot after being up for several weeks. The Cisco 7960 or similar phones are the most solid SIP phones that I have worked with. I plan on selling 2 of my Pingtel's and getting a couple of the Ciscos. Andy Hester wrote: [EMAIL PROTECTED] wrote: Hello All, Does anyone here have any experience with pingtel Xpressa hard phones? I am considering buying a couple. Already have Snom200s, but want something with better CTI and full duplex speakerphone. Michael Michael, I used some Pingtel phones with * and ended up having to scrap them. Pingtel assured that they would work with * but when they didn't, they put it all off on Asterisk. They also didn't seem to be interested in getting them to work. In my opinion they were dishonest with through the whole deal and I wouldn't recommend doing business with them or using that phone. One persistent issue is that the phone would lock up if you had more that 2 calls even though they said it could handle 11 simultaneous calls. I talked to at least one other user who had the same issues. Everything was fine in the lab, but upon roll out the phones didn't work right. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cannot find -lXext when building * ?
Howdy.. I am building * on a barebones server, running just the minimum config (no X, etc..) when I build, I get this error, and I'm trying to track it down. Has anyone ran into this before or have a general idea? gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread` /usr/i386-slackware-linux/bin/ld: cannot find -lXext collect2: ld returned 1 exit status make[1]: *** [pbx_gtkconsole.so] Error 1 Thx, cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Room Monitor
Do any of you know of a cost effect device that could be connected to an Asterisk station port to provide room monitoring? I'm looking to replace the wireless baby monitor we currently have, since there is too much interference between our daughter's room and our room for it to work effectively. I've found a few items[1-4] that seem to provide the feature I'm looking for, but they seem much more expensive than necessary. Essentially, I'm looking for something that I can assign an extension on asterisk to and then call from another station to activate monitoring. Any ideas are welcome. [1] - http://www.spyandsecuritystore.com/informer.html [2] - http://shop.store.yahoo.com/spytechagency/11435.html [3] - http://www.talkingelectronics.com/security/room_devices.html [4] - http://www.surveillance-spy-cameras.com/room-monitor.htm -- Jamin W. Collins To be nobody but yourself when the whole world is trying it's best night and day to make you everybody else is to fight the hardest battle any human being will fight. -- E.E. Cummings ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speech between Grandstream phones sounds like talking under water
When I make a simple phone call from one Budgetone 101 to another, the speech sounds slurred and slow, sort of like the person is talking under water. Both phones and the Asterisk server are on the same subnet. Both phones are configured to use the PCMU (ulaw) codec as first choice, and the Voice Frames per TX parameter is set to 2. Incidentally, if I directly IP dial from one phone to the other (bypassing Asterisk) the speech sounds excellent. I'm running a CVS build from Feb. 1, 2004, and there is a Digium X100P card with one incoming CO line in my machine. The first part of my sip.conf looks like this: [general] port=5060 binaddr=0.0.0.0 disallow=all allow=ulaw [200] type=friend username=200 host=dynamic context=home reinvite=no canreinvite=no [201] type=friend username=201 host=dynamic context=home reinvite=no canreinvite=no I turned on sip debug, and noticed the following in the output: v=0 s=SIP Call c= IN IP4 192.168.2.29 m= audio 5004 RTP/AVP 0 a=rptmap:0 PCMU/8000 a=ptime:20 Found audio format UNKN Found description format PCMU Capabilities: us - 4, them 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined 0 Does anyone know why this could be happening? Thanks, Ron attachment: winmail.dat
[Asterisk-Users] New to the list - some (unsolved) questions
Dear all, I'm new to the list and new to Asterisk, so please bear with me ;) I've been googling the web but couldn't find my answers... my apologies if these have been already discussed before. Nowdays I'm interested in setting up some VoIP-based solution on our offices and I think Asterisk is the right choice. I've been browsing here and there but couldn't find any of those success stories from customers using Asterisk for their everyday needs. So, any hints on this issue will be much appreciated, as I need some support materials to sell Asterisk to my managers :-) Furthermore, I read the documentation (http://digium.com/index.php?menu=documentation) site and couldn't find something like Asterisk Setup Crash Course for Dummies or the like ;) I'd like to know the minimal requirements for Asterisk to work. Is there any suggested card for ISDN lines? Besides the ones on Asterisk's website, has anyone any experience with NMS boards (http://www.nmscommunications.com/)? Thanks in advance! Best regards, Martin -- Martin Mielke [EMAIL PROTECTED] THALES Information Systems http://www.thales-is.com/ UNIX is user-friendly... It´s just selective about who its friends are. [ echo \$0\$0_;chmod +x _;./_ ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Good source for moh files
You'll probably want to re-quant them to 8kHz, but there are quite a few classical tracks available at: http://hebb.mit.edu/FreeMusic/ -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
On Mon, Feb 16, 2004 at 04:51:08PM +, WipeOut wrote: I am going to an Oftel meeting to discuss VoB regulation next week.. Hopefully this will help to see where it is heading.. Of course Oftel doesn't exist anymore, it's all Ofcom now ... Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip in the EU
Well, since they restricted attendance to service providers and representatives of consumer organisations I wouldn't be too optimistic for a balanced outcome ;-) Iain --On Monday, February 16, 2004 4:51 pm + WipeOut [EMAIL PROTECTED] wrote: Steve Kennedy wrote: On Sat, Feb 14, 2004 at 02:47:11AM -0500, Ryan Finnesey wrote: Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Steve I am going to an Oftel meeting to discuss VoB regulation next week.. Hopefully this will help to see where it is heading.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hide outgoing CallerId on Zap interface
This is excactly what restrictcid=yes does in sip.conf. Eg. when it is used you'll see this in pri debug: Calling Number (len=12) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation prohibited, user number passed network screening (33) 12345 12345 is your number but it will not be passed to the other party that is being called. -- Pertti Alfred R. Nurnberger wrote: The correct way to hide your callerid on a PRI interface is to set the presentation indicator. Some CO switches do a basic sanity check on the callerid they receive. If you set the number string to empty but the presentation indicator to allow the number they will replace the number string by your main number. I do not know how or if possible to change the presentation indicator on * but a look in libpri should give some clues. - Alfred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mickey Binder Sent: Monday, February 16, 2004 6:13 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Hide outgoing CallerId on Zap interface There seems to be some trouble with either the maillist or my client. I haven't received any of the posted replys on this topic, but found the replys through the asterisk.linkx.net search engine. But anyways here is the reply on the mail from: James H dot Cloos Jr. cloos at jhcloos.com If it is a pri I'd give SetCallerID() a try in the dialplan. It is a PRI and I've tried the SetCallerId() which displays my PRI main number, like the other experiments I've tried. I talked to my Telco provider which said that it isn't possible to hide the number via shortcodes but that it should be done via my PABX. -Original Message- From: Mickey Binder [mailto:[EMAIL PROTECTED] Sent: 13. februar 2004 12:14 To: Asterisk maillist Subject: [Asterisk-Users] Hide outgoing CallerId on Zap interface Hi there I know I have asked a somehow similar question earlier but since then I've tried some different things which isn't working. I want to completely hide my outgoing CallerId when dialing out on my Zap interface. I've tried a lot of different settings in sip.conf and hoped that zap would hide the CallerId if sip was told to do so, but that wasn't the case. Then I've tried to set hidecallerid=yes in zapata.conf (and restarted *) but this only results in my main number CallerId being displayed. Is it somehow possible to completely hide the CallerId, like when someone from a secret number is calling and the display on my mobile says Secret number ? And if that is possible, is it then possible to do it on a per-user basis configured via sip.conf? regards, Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Eicon Diva Server card, where to purchase ?
Where can we get the Eicon Dive server card with BRI ? I mean, what is the lowest cost supplier ? I have heard that we need to get the version after 2.x since the echo cancellation is only supported after 2.x. Regards DL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Got my DID, getting an error.
VP seems to be working fine for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of matt Sent: Monday, February 16, 2004 9:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Got my DID, getting an error. I'm getting the same but they say they are working on it...may pay to find out when you signed up DID and request credit for percentage of month it wasn't working by the time they come back online. Matt Tom Knox wrote: I am getting my message on an inbound call, haven't tried outbound yet ;-) Anyone else have any ideas? Patiently waiting Trying to learn a wonderful new system :) On Mon, 2004-02-16 at 16:16, Tom Knox wrote: Hi there, Got my DID from VoicePulse. Very fast and quite cheap :) I configured the iax.conf with the info they provided, I am getting a good connect to their server, but when I try to dial my number I am seeing the following on the console. Anyone got an idea? Oh I replaced the number below with a generic # :) Thanks! Connected to Asterisk CVS-02/15/04-11:38:41 currently running on Asterisk-Test (pid = 11062) Feb 16 17:00:01 NOTICE[1150528304]: chan_iax2.c:4689 socket_read: Rejected connect attempt from 66.234.228.132, request '[EMAIL PROTECTED]' does not exist Could it be that to dial out you need a 1 in front of the 10 digit number. The 1 should probably be important, and without it you fail to match an outbound extension pattern matching on it. -- Steven Critchfield [EMAIL PROTECTED] --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.577 / Virus Database: 366 - Release Date: 2/5/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wifi Phones
I am also going to share this with the list since it may be helpful for other Wisip/* users out there. There are two tricks that I discovered in getting the Wisip to work with *. The first thing is to go into the Wisip's web config and select the SW/Update section. There will be an entry for a tftp server. This is tricky because it looks like an entry to get software updates but it is actually a tftp config file loader. As with many other entries in the wisip it will not allow you to put a blank entry. Just put in an fake entry and it works fine. If you don't do this you will find that no matter how many times you reenter your settings you will lose all the sips settings on each bootup :-). Note that this will be an issue even if you explicitly ordered the phone without a FWD config. Even though they offer the option on the order form they call come configured this way. The second trick is to change the outgoing proxy port from the fwd/wisip default to 5060. Otherwise it is a pretty standard sip setup. Here is the entry from my sip.conf file. If you don't have a G.729 licence then you need to switch the settings to uLaw or aLaw in the codec. Also make sure you match up the dtmf settings. [wisip] type=friend secret=blah host=dynamic context=local dtmfmode=rfc2833 callerid=Joe Schmoe nat=yes qualify=1000 When you get it going please compare notes on functionality. So far I am finding the Wisip likes my brand new Cisco AP1100 a lot more than either an airport or linksys ap. This could however be wep as my new Ap doesn't have that setup yet. I also noticed that the volume level varies considerably depending on how you place the phone next to your ear. It really needs to be lined up just right. Once I got connected with the cisco and everything running, and the phone placed correctly against my ear I had one really clear 20 minute call with the unit while I roamed around the building. The range was really pretty good. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Gene Brown [EMAIL PROTECTED]: Could you send me a copy of your sip.conf for the wisip from Pulver. And the config for the phone? I bought one and it worked with FWD but I haven't gotten it to work with *. Thanks Gene Brown [EMAIL PROTECTED] I don't know if anyone else has worked with Spectralink, but I tried to get some demo units to test with a while back and I was really disapointed. At first they claimed they were SIP complient. Then they sent me a contract for the demo. They wouldn't send a demo unless I agreed to have them do an onsite install to the tune of nearly $5000. I would only have been obligated for the install fees if I decided to buy, but not being happy about the install fees and wanting to know why I learned more about how the technology works. My sales rep shared that the phones aren't actually SIP compliant and work through the SVP server to provide SIP compliance to the PBX connection. He also shared that they want provide warranty on their products unless they do the install. Really turned me off, smelled very proprietary, although SVP QOS is pretty cool. I just received a Wisip last week and it looks pretty promising, although I think my unit may be damaged. Pulver support was also very up front with me that they technically only support the Wisip with FWD. They have been good to work with me so far, even rushed me a unit when I explained I was researching for a large purchase. It definetly connects to asterisk, but I think my unit has a bad antena or transmiter. The audio drops in and out and the signal strength indicator shows only one bar even when only about 25' from the access point. Anyone else actually gotten their hands one of these to try with Asterik? I would like to buy a couple hundred of them, but they need work reliably. I would love to compare notes with someone to see if my experiences are a typical. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Craig Waddington [EMAIL PROTECTED]: Those phones look good, but, only have 10 milliwatt output. Have you looked at these: http://www.spectralink.com/products/nl-wts.html 100mw output. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miguel Cavazos Sent: 15 February 2004 12:39 To: Asterisk Users Subject: [Asterisk-Users] Wifi Phones Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here
Re: [Asterisk-Users] Asterisk monitor with Daemontools
Jacky wrote: Hi All, I which to use Daemontools to watch asterisk process. EVIL! Asterisk fork's a new process shortly after starting (unless you run with a console) Find safe_asterisk in the contrib directory. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oh323 question
I have the following config: Asterisk compiled with oh323 on a public IP Grandstream behind a NAT Aseterisk sending calls to a Nextone MSW H.323 My Grandstream phone registers to my * server via SIP fine. When I place a call that goes from my * server to my Nextone via H.323, I seem to loose my IP address and the Nextone blocks the call. Seems to work fine when I go from SIP phone to SIP provider, but fails when I go from SIP Phone to H.323 provider. All outbound. When I look at the CDR's on the Nextone, I see 0.0.0.0 as my IP address. Any ideas?? Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x100p dropping incoming calls
I have been experiencing hung up when answering incoming calls through x100p. NOTICE[1242768320]: chan_zap.c:4584 ss_thread: Got event 2 (ring/Answered).. -- Executing Wait(Zap/1-1,1) in new stack -- Executing Answer(Zap/1-1,) in new stack -- Executing DigitTimeout(Zap/1-1.5) in new stack -- Set digit timeout to 5 -- Executing ResponseTimeout(Zap/1-1,5) in new stack -- Set Response Timeout to 5 -- Executing Playback(Zap/1-1,cpswelcom) in new stack -- Playing 'cpswelcome' (language 'en') -- Executing BackGround(Zap/1-1,cpswelcome5) in new stack -- Playing 'cpswelcome5' (language 'en') . . . . -- Called 1001 -- Sip/1001-a924 is ringing -- SIP/1001-a924 answered Zap/1-1 == Spawn extension (operator,s,2) exited non-zero on 'Zap/1-1' -- Executing Hangup(Zap/1-1,) in new stack == Spawn extension (operator, h,1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' As soon I pick up the phone when it rings. The call was hungup. It seems to be occasional and not happens all the time. Anyone has any comment on what possible cause. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
RE: [Asterisk-Users] HFC-S cards?
Hi, -Original Message- Acer ISDN-Surf, Billion Bipac ISDN, Trust PCI ISDN Modem, D-LINK DMI-128+ to name a few ;-) anyone knows where to get one of theses cards (or any other based on the HFC-S chipset) in germany? my computer-trader maybe can get d-link's card but he don't know how long it could take. does anyone (hello kapejod ;)) ) know wich one should be my first choise, just in case I find more than one of these babys. Just take a look at any cheap ISDN card you can find. The HFC-S are identified by a little chip that has the towers of Cologne on them, and the marking 'HFC-S' (You'd never guess ;-). .Creatix ISDN-S0/PCI .Trust PCI-Modem .Acer ISDN 128 Surf PCI .D-Link DMI-128I+ .Billion/Asuscom (Asuscom/Askey) .HFC cards from Conrad Elektronik (Bestellnummer 95 50 78) .Neolec FREEWAY ISDNPCI See also http://www.pbx4linux.com/download/doc/PBX4Linux.htm, under chapter 2.1 (ISDN cards) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hi all. If I want to use the * only as a GW to PSTN and allow only one external proxy to place calls. how is the smartest way to do this ? I dont want the world to be able to do invites only a specific IP, in this case my proxy that handles all the users. /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
