Re: [asterisk-users] Configuring Sangoma A101D with Asterisk 1.2.18 zaptel-1.2.17.1
It would help to know exactly what Dell Poweredge you were considering. They do vary. I have Dell Power Edge 850 Also how do I enable DTMF hardware detection. There are no drivers which support it. I have the lastest Beta drivers installed, they seem to show yes in the logs, but the hardware DTMF didnt work, so I wrote a mail, to the developer of the drivers he said they are still working in the lab probably have one within a week. Stephen Bosch [EMAIL PROTECTED] wrote: Deepak Naidu wrote: Hi, I have a Dell Power Edge server planning yo buy Sangoma A101D card. To configure with my Asterisk 1.2.18 zaptel-1.2.17.1 Free-PBX setup. It would help to know exactly what Dell Poweredge you were considering. They do vary. If you compile your kernel with SMP and IO-APIC support, you shouldn't have any problems. The Sangoma cards are very tolerant. So I wanted to know the steps any issue which I may come accross if any. I have googled have some docs handy wrt Trixbox-2.2. Just wanted to get some notes from user with custom install setup when used with Asterisk+freepbx+Sangoma. On the hardware side, the experience shouldn't be any different. Also how do I enable DTMF hardware detection. As far as I know, that is the default. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CONSOLE=Phone/phone0 and CONSOLE=Console/dsp and Zap/g2
Hi List; I know that we can use SIP/john and IAX2/jack/613 but I do not know what are these: Phone/phone0 Console/dsp Any advise? Regards Bilal Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CONSOLE=Phone/phone0 and CONSOLE=Console/dsp and Zap/g2
On Sun, Jul 29, 2007 at 02:18:15AM -0700, bilal ghayyad wrote: Hi List; I know that we can use SIP/john and IAX2/jack/613 but I do not know what are these: Phone/phone0 This is from chan_phone . Look into its documentations. Console/dsp This is from chan_oss or chan_alsa . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/Dont know how to handle Accepted
Hi Victor, Victor Toofic wrote: Hi, I've finally got running Asterisk 1.2.14 with UniCall MFC/R2 patches, I can generate calls and all seems OK but I cannot receive any call, this is what I get: Unicall/3 event Offered CRN 32769 - Offered on channel 0 (ANI: 814777, DNIS: 83329276, Cat: 1) MFC/R2 UniCall/3 Call control(5) MFC/R2 UniCall/3 Accept call MFC/R2 UniCall/3 1 on - [2/OFFERED /Group B /Accepted Paid] MFC/R2 UniCall/3 - 2 off [2/OFFERED /Group B /Accepted Paid] MFC/R2 UniCall/3 1 off - [2/OFFERED /Group B /Accepted Paid] MFC/R2 UniCall/3 Answer guard expired Unicall/3 event Accepted Unicall/3 Don't know how to handle signalling event Accepted What versions of software did you use to get a screwed up result like that? The message Don't know how to handle signalling event Accepted is printed at the end of a case statement which does handle that event. I the publicly available versions of unicall, and can't see how that could go wrong, even if you mix components from different versions. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Brazilian.
Some brazilian here on list? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connections broken
Michael Munger wrote: I agree it is the NAT in the router. Does anyone know what the ip tables command would be to pass IAX to an Asterisk box on the LAN? It depends a lot on what your current setup is, but something akin to: iptables -A PREROUTING -t nat -p tcp -i eth0 --dport 4569 -j DNAT --to ip-of-asterisk-box:4569 should work, assuming you have the relevant parts compiled in. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial from Phonebook of Evolution or Thunderbird
Hi, does anyone know about a plugin that allows dialling a contact from the phonebook of evolution or T-bird? -- Alexander Topolanek http://www.topolanek.at ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon 2007
Saw the website but can't find a schedule or even an email address to contact someone. Anyone know more about Astricon? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reposting (was: better subject needed [was: Re: Query1])
Note that some of us newbies have posted the same question two or three times because we didn't see our own post (let alone a reply) in a timely manner. After a while on the list, we learn that many of us experience delays of a few hours to several days and we become accustomed to following threads in which the posts arrive out of sequence. (I recognize that the delays may be a result of my own email configuration, but I have no problem with other lists or email correspondents.) --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Saturday, July 28, 2007 2:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] better subject needed [was: Re: Query1] On Sat, Jul 28, 2007 at 12:03:33PM +0530, [EMAIL PROTECTED] wrote: Hi, I am facing problem in configuring D-channel for TE120P card.I did the following things /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf [ snip ] /etc/asterisk/zaptel.conf . Hmm.. sounds familiar. Haven't I answered it already. I also recall someone replying to it just today... You have already posted that question. Two of us have already posteated follow-ups on it. Please reply to (at least one) of them rather than re-posting your question. Furthermore, your posts have no meaningful subject. This post could use a subject such as: problem in configuring D-channel for TE120P card Or even just: configuring D-channel for TE120P card -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Wiki
On 7/27/07, Jared Smith wrote: Yes, the second edition of the book will be out very soon now. I'm glad to hear you enjoyed the book. Hopefully you'll like the second edition even better. :-) I am sure I will, I am also glad to see that there is someone like O'Reilly out there facilitating such a publication for us geeks. fyi, I have been plugging your first book in a few places ( see ack section :-) http://nooss.org/wiki/Installing_Asterisk_From_Source thnx again. -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial from Phonebook of Evolution or Thunderbird
Alexander, Check out Snap. It plugs into Thunderbird on Windows. On 7/29/07, Alexander Topolanek [EMAIL PROTECTED] wrote: Hi, does anyone know about a plugin that allows dialling a contact from the phonebook of evolution or T-bird? -- Alexander Topolanek http://www.topolanek.at ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reposting (was: better subject needed [was: Re: Query1])
On 7/29/07, Don Kelly [EMAIL PROTECTED] wrote: Note that some of us newbies have posted the same question two or three times because we didn't see our own post (let alone a reply) in a timely manner. True. I could swear that when I post to biz, I get a post confirmation message immediately but not on users. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID from POTS to SIP
Thanks for the reply. Yep, the telco is providing the callerid since it's getting into the asterisk call database just fine, in all cases. It appears that the telco has changed the cadence for the distinctive ring again. I modified the dring contexts to handle a 0,0,0 dring and now callerid appears to make it through to the phones, sometimes. I'm therefore led to believe that there is weird behavior around distinctive ring and the passing of CID. I can only assume that if an exact matching dring context isn't found and it goes to [default] then callerid isn't passed out but still is logged, which is just wonky to me. On Sat, Jul 28, 2007 at 01:49:13PM -0400, Stephen Bosch wrote: What happens when you connect a regular caller ID device to the line and call the distinctive ring number? Is the telco even supplying CID info? This sounds like a programming problem on the telco side. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
Yep. From Brasilia-DF ! :) 2007/7/29, Jozeph Brasil [EMAIL PROTECTED]: Some brazilian here on list? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Wiki
On 7/27/07, Jared Smith [EMAIL PROTECTED] wrote: On Fri, 2007-07-27 at 06:26 -0700, bilal ghayyad wrote: For example: what is the best (shortest) way to search for information related to the command playbak()? I find that the fastest and most up-to-date information regarding the dialplan applications is the online help in the Asterisk CLI. For example, type core show application Playback at the Asterisk CLI, and I haven't looked at 1.4 but in 1.2 the quality is wildly variable. Some of the apps have descriptions that read like assembly langauge comments: ADDA,B; add A to B Dial is an exception, it has everything you need to know right there. Someone should probably take a look at every app and make sure the description is enlightening. Another idea would be to include a unique ID in that CLI output that could be used to build a URL that gave a man page like listing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
São Paulo, SP here ;) On 7/29/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote: Yep. From Brasilia-DF ! :) 2007/7/29, Jozeph Brasil [EMAIL PROTECTED]: Some brazilian here on list? -- Hugo da Costa Rebelo Tel: +55 (11) 7165-4630 www.hugorebelo.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID from POTS to SIP
[EMAIL PROTECTED] wrote: The other context, the one with distinctive ring that's not passing caller id, actually does a little more: [main-open] exten = s,1,Answer exten = s,n,Wait(3) exten = s,n,Background(opengreeting) exten = s,n,Dial(SIP/ht1SIP/gxp3,20) But even if I remove those extra bits from that context and make it look like the other one, it still doesn't work. Any more suggestions? Yes - you're answering before you have a chance to receive the Caller ID -- on an analog line you cannot receive Caller ID once you have answered. Try this: [main-open] exten = s,1,Wait(1) exten = s,n,Answer exten = s,n,Background(opengreeting) exten = s,n,Dial(SIP/ht1SIP/gxp3,20) Regards, Trevor Peirce -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
One more, at least... Sds, Gustavo From: Jozeph Brasil [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: [asterisk-users] Brazilian. Date: Sun, 29 Jul 2007 09:02:09 -0300 Some brazilian here on list? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Verifique já a segurança do seu PC com o Verificador de Segurança do Windows Live OneCare! http://onecare.live.com/site/pt-br/default.htm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling to users in other asterisk servers
I configured only the sip.conf: ... [usera] type=friend host=dynamic secret=pass disallow=all allow=ulaw context=sip [userb] type=friend host=dynamic secret=pass disallow=all allow=ulaw context=sip ... And the extensions.conf: ... exten = 1000,1,Dial,SIP/usera exten = 2000,1,Dial,SIP/userb ... I found it in a asterisk tutorial on google... with it I get the asterisk working on my LAN for test my softphone project... I think that is a firewall problem, right or wrong? I will make tests for check it... Thanks very much!!! On 7/28/07, dave cantera [EMAIL PROTECTED] wrote: aryjunior, is your dialplan and registration configured to connect to another * server?...include your config so we can analyze it... daveC Carlos Rojas wrote: Hello, Do you have porf forwardin for SIP protocol in your firewall? SIP: 5060 udp rtp 1 - 2 udp (default) and IAX2 4569 udp Best Regards Carlos Rojas On 7/28/07, *Ary Junior* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Im a asterisk newbie and I've configured an asterisk server here in my house... in my LAN two users can login and call to each other, but when I try to call an user in another asterisk server outside my LAN ( sip:[EMAIL PROTECTED] mailto:sip:[EMAIL PROTECTED] ) it dont work... if the person outside is conected on my server it works fine... My asterisk server is behind a firewall and portfowarding... it is possible? Thanks very much!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.22/921 - Release Date: 07/26/2007 11:16 PM -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial from Phonebook of Evolution or Thunderbird
On Sun, Jul 29, 2007 at 04:05:02PM +0200, Alexander Topolanek wrote: Hi, does anyone know about a plugin that allows dialling a contact from the phonebook of evolution or T-bird? On Kontact you could use an arbitrary script. Not sure about Evolution or Thunderbird. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Autoreply: Re: SunRocket / ALLO / etc special offer
I'll take either Actually now that I have had a chance to think about what I did (sorry bad week here). Yes, I will admit I did patrionize the users list... sorry if I offended anyone. I just figured I'd try to help any SunRocket users out that may not be on the biz list.If you review my history, you'll see I only post business stuff to the biz list. This is an exception. On 7/26/07, Baji Panchumarti [EMAIL PROTECTED] wrote: On 7/26/07, Matt Hoppes wrote: I would agree... intended to send that to biz, sorry. I see that you also sent it to the biz-list. And if you fail the lie detector test how about agreeing to a full boycott of your service or at least a M.L.D.P. (mailing list death penalty :-) ? -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
On 7/29/07, Jozeph Brasil [EMAIL PROTECTED] wrote: Some brazilian here on list? There are many more here: http://www.asteriskbrasil.org/ -- Kristian Kielhofner ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-1.2.22 DeadAGI Hangup
Arun Kumar wrote: I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI scripts are not working properly. Like after hangup I used to do some more work now its not working. Try, at your own risk, this: http://svn.digium.com/view/asterisk/branches/1.2/res/res_agi.c?r1=71065r2=71656 Original message from lamer on Fri Jul 27, 2007 5:36 am: This happens due to change in res_agi behaviour. Thus, EXEC DIAL will hangup straight away even SIGHUP is ignored as EXEC DIAL works as an underlying app since 1.2.20 (and probably 1.4.8). Dial with 'g' seems to solve half of the problem but there are some side effects. It's currently reported here http://bugs.digium.com/view.php?id=10315 Solution is to revert the change in: http://svn.digium.com/view/asterisk/branches/1.2/res/res_agi.c?r1=71065r2=71656 Source: http://forum.asterisk2billing.org/viewtopic.php?p=8118#8118 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
As it turns out the telco was not routing the calls to us, a little misktake they said after 3 days of being with no service. The line was not CAS, it was CCS, no need to compile unicall. Whatever they meant with your card has to be configured with DSS1 will remain in mystery. Maybe someone here can tell me what they mean. The configuration I previously listed is valid for lines in Panama City, Panama. With the telco being Cable Wireless Panama and the asterisk with a sangoma A102. If there's any Cable wireless tech reading this. Guys, your support s*cks big time. Thanks to all for your kind and prompt help. On 7/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: If you do not have any alarms and PRI debug span 1 still gives you nothing then you need to call your telco and say I'm not getting any Q.931 messages on the D-Channel. Stephen Bosch wrote: Erick Perez wrote: Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. Hmn, well, that's telling. Are you using the correct cable? Is the cable plugged into the correct port on the card? The 102 is a two-port. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling to users in other asterisk servers
Ok, my firewall port forward rules: TCP5004 - 5082192.168.254. 2UDP5004 - 5082192.168.254. 2TCP4569192.168.254. 2UDP4569192.168.254. 2UDP1 - 2192.168.254. 2 And it dont works... Any configuration in special for make call the to users in another asterisk servers? Thanks very much!!! On 7/28/07, Carlos Rojas [EMAIL PROTECTED] wrote: Hello, Do you have porf forwardin for SIP protocol in your firewall? SIP: 5060 udp rtp 1 - 2 udp (default) and IAX2 4569 udp Best Regards Carlos Rojas On 7/28/07, Ary Junior [EMAIL PROTECTED] wrote: Hi, Im a asterisk newbie and I've configured an asterisk server here in my house... in my LAN two users can login and call to each other, but when I try to call an user in another asterisk server outside my LAN ( sip:[EMAIL PROTECTED] ) it dont work... if the person outside is conected on my server it works fine... My asterisk server is behind a firewall and portfowarding... it is possible? Thanks very much!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reposting
randulo wrote: On 7/29/07, Don Kelly [EMAIL PROTECTED] wrote: Note that some of us newbies have posted the same question two or three times because we didn't see our own post (let alone a reply) in a timely manner. True. I could swear that when I post to biz, I get a post confirmation message immediately but not on users. And also with the Dev list. It's also a known problem and I'm sure they are working on it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Query
On 26 Jul 2007 17:25:30 +0530, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Jul 26 17:09:42 WARNING[11925]: chan_zap.c:9148 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. == Primary D-Channel on span 1 down Jul 26 17:09:43 WARNING[11925]: chan_zap.c:2438 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channelanyway! Can anybody tell me how to overcome this error. Sanchal: If you will refer to my message of two days ago it explains exactly how to fix the issue. Best regards, Andrew ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN: Problems starting off [solved]
Hi,
Am Samstag, 28. Jul 2007, 17:56:43 +0200 schrieb Armin Schindler:
On Fri, 27 Jul 2007, Bertram Scharpf wrote:
My `capi.conf' is like show in many tutorial on the web. In
`extensions.conf' I just added the following lines:
please provide your capi.conf.
Which chan-capi version do you use?
Sorry, I thought the whole was too much information to post
at once.
Seems that the MSN or even `capi-in' cannot be found at all.
Yes, chan-capi seems to wait because of no match.
This was the hint I needed and I found an option 'immediate'
which I had to set to 'yes'.
I still do not know what it means to renounce to wait for
SETUP/SENDING-COMPLETE. Anyway, commands like this one give
the right MSN:
exten = _Z.,n,Verbose(===${DNID}===)
Cool. Thanks!
Bertram
--
Bertram Scharpf
Stuttgart, Deutschland/Germany
http://www.bertram-scharpf.de
___
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Re: [asterisk-users] What is the best softphone work with Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 bilal ghayyad wrote: Hi Vicky; Thanks a lot for your reply. Where to download Idefisk/zoiper? http://www.zoiper.com/ - -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGrSazDQNt8rg0Kp4RAuCXAJ0dFlW5b0nuDceydBEur7Uug+2GgQCgpNZH P3RfOuatupqXelriG2bg09I= =ffkM -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] :THIS IS A SPAM: Re: Sangoma on Fedora 7 x86_64
Hi john, Thank you for your reply, i finally stumbled on google what the problem is. The driver does not compile on kernel newer than 2.6.19. Regards, Nhadie John Novack wrote: Sangoma gives EXCELLENT technical support. I would suggest you try there first. The few problems I have had with installation were addressed promptly and when driver fixes proved necessary, corrected in short order. Also the cards have a 5 year warranty! John Novack ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom custom ring tones (slightly OT)
I even got a Polycom here saying I'll be back which was funny for about an hour, then not funny at all. PaulH On Fri, 2007-07-27 at 12:36 +0800, James Andrewartha wrote: Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: PAGE_BEEP se.pat.ringer.13.name=Page Beep se.pat.ringer.13.inst.1.type=chord se.pat.ringer.13.inst.1.value=12 se.pat.ringer.13.inst.1.param=200 se.pat.ringer.13.inst.2.type=chord se.pat.ringer.13.inst.2.value=15 se.pat.ringer.13.inst.2.param=600 se.pat.ringer.13.inst.3.type=branch se.pat.ringer.13.inst.3.value=-2/ Alternatively, since you can use .wav files for ring tones, do people have any recommendations for where to find some distinctive rings? Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Brazilian.
bruno RS Kristian Kielhofner wrote: On 7/29/07, Jozeph Brasil [EMAIL PROTECTED] wrote: Some brazilian here on list? There are many more here: http://www.asteriskbrasil.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues with logged in agents that are not reachable
Hello, I am using 1.4.8 and have a question about Queues. I noticed that if I have an agent logged in using AgentCallBackLogin and that agent is unreachable for some reason (SIP phone unplugged) calls to him/her will completely yack. For example: 1-Agent 500 is the only one logged into queue number 1. 2-A call comes into queue number 1 3-The call is pushed to agent 500 at extension 21 which is unreachable because the ethernet cable is unplugged to extension 21's handset. 4-The caller gets hungup on entirely instead of the call going to another agent or leaving the caller in the queue I don't expect this to happen but I want to be sure all bases are covered on light days during shift changes etc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Programming with libiax2
Hi everyone, I am considering writing some code using libiax2. Are there any good resources to get started with this? Books? Sites? Thanks -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Programming with libiax2
Hello, Le lundi 30 juillet 2007 à 14:19 +1000, Devraj Mukherjee a écrit : Hi everyone, I am considering writing some code using libiax2. Are there any good resources to get started with this? Books? Sites? There is a small example here : http://proformatique.org/spip.php?article101 Sorry it's in french because it was writing for a french magasine. Sylvain ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4.8 and google talk - no audio
Hi all, Iam using asterik 1.4.8 and connected to google talk. When iam calling from my google talk account to sip phone i can hear the voice (2 way). (this happens only within the LAN). when my friend tries to call my asterisk server (connects to the public ip) using his googletalk client it comes to my sip phone but either party cant hear a voice. I have fully allowd both tcp,udp on my router. and i have a public IP. (no nat). i have disabled my firewall on my asterisk box (just for testing). this is my configuration gtalk.conf [general] context=default allowguest=yes bindaddr=147.120.203.190 externip=203.xx.xx.xx [guest] disallow=all ;allow=alaw allow=ulaw context=guest [google] disallow=all allow=ulaw ;allow=alaw context=default connection=asterisk jabber.conf [general] debug=yes autoprune=yes autoregister=yes [asterisk] type=client serverhost=talk.google.com [EMAIL PROTECTED] secret=xx port=5222 usetls=yes usesasl=yes timeout=1000 rtp.conf iam using lower ports... rtpstart=1650 rtpend=4560 sip.conf [general] context=default bindport=5060 bindaddr=0.0.0.0 ;srvlookup=yes dtmfmode=rfc2833 relaxdtmf=no disallow=all allow=ulaw ;allow=alaw ;allow=gsm maxexpirey=30 defaultexpirey=180 canreinvite=yes ;nat=no UserAgent=Asterisk [312] type=friend context=default regexten=312 username=312 secret=312 fromuser=1 callerid=tharanga host=dynamic ;nat=no canreinvite=no dtmfmode=RFC2833 incominglimit=3 mailbox=1 is this a bug ?? or something missing in my configuration ?. thxs in advance Tharanga - Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Programming with libiax2
Thanks. Got the source, will explore further :) On 7/30/07, Sylvain Boily [EMAIL PROTECTED] wrote: Hello, Le lundi 30 juillet 2007 à 14:19 +1000, Devraj Mukherjee a écrit : Hi everyone, I am considering writing some code using libiax2. Are there any good resources to get started with this? Books? Sites? There is a small example here : http://proformatique.org/spip.php?article101 Sorry it's in french because it was writing for a french magasine. Sylvain ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
