[asterisk-users] sendmail file
Hi: How can I configure sendmail file to asterisk send voicemails to my mail.sendmail file in /usr/sbin is a read only file. I'd appreciate any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CTI Intergration with the CRM
Hi All; I see that Asterisk has call center, but is it possible to have CRM Integration? If yes, then how the integration will be? Is it via CTI? From where I can get the CTI API's to include it in the CRM application and let it communicate with the CTI server to complete the Integration. Any advise? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendmail file
your mail is not clear at all. if you want to change the path of sendmail ,do this with mailcmd, in the voicemail.conf, if you want to send a voicemail to a class of emails, using dbase is more easier. let me to know more, about your problem. --- On Sun, 6/29/08, fateme fatah [EMAIL PROTECTED] wrote: From: fateme fatah [EMAIL PROTECTED] Subject: [asterisk-users] sendmail file To: asterisk-users@lists.digium.com Date: Sunday, June 29, 2008, 12:18 PM Hi: How can I configure sendmail file to asterisk send voicemails to my mail.sendmail file in /usr/sbin is a read only file. I'd appreciate any help.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Druid Open Source Events - Druid Miami Meetup (18 Jul), OSCON (21-25 Jul), Druid London Meetup (22 Jul) LinuxWorld (4-7 Aug)
Dear Asterisk users, Voiceroute will be at exhibiting and presenting at the below open source communications related conferences, Druid meetups speaking about Druid Asterisk. We would like to meet with other fellow Asterisk enthusiasts who may be at OSCON LinuxWorld. Mark Spencer will be speaking at OSCON 2008 http://en.oreilly.com/oscon2008/public/schedule/speaker/6807 1) Druid Meetup Miami Florida Date: 18 Jul 2008, 6pm-8pm EST, Redfone Communications Miami Florida For more details and sign up http://druidmiami.eventbrite.com 2) OSCON 2008 (Portland Oregon) Date: 21-25 Jul 2008, Oregon Convention Center, Booth 221 Navin Kumar, will be giving a talk on Druid Building an Open Source Unified Communications Solution - The Druid Project 5:20pm - 6:05pm Thursday, 24 jul 2008 http://en.oreilly.com/oscon2008/public/schedule/speaker/27379 For more details and sign up, http://druidoscon.eventbrite.com 3) Druid Meetup (West End London, UK) Date: 22 Jul 2008, 6pm-8pm BET Location: Thames Valley University - Room TC43 For more details and sign up http://druidlondon.eventbrite.com 4) LinuxWorld 2008 (Moscone Center, San Francisco CA, USA) Date: 4-7 Aug 2008, 10am-4pm PST Location: Moscone Center, San Francisco, Booth 1626 For more details and sign up http://druidlinuxworld.eventbrite.com Some of the hot new stuff we will be demoing on Open Source at these events - Druid Communicator on Blackberry v1.5 launched! - Druid Communicator Adobe Air Application: SugarCRM/SalesForce Integration desktop application - Cool stuff like Blackberry Desktop Integration using Druid SOAP API - Druid SOAP API: Your own Druid Asterisk integration application in 10 mins for Druid Regards, Ming -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Palyback and CDR records
exten = _078.,3,Playback(platna|noanswer) Thank you everything work perfect now :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk seg fault
Hi, Am Mittwoch, den 25.06.2008, 08:42 -0400 schrieb Jerry Geis: I am running asterisk from svn check out from yesterday Jun 24. I started with 1.4.20, then 1.4.21 then svn. I am getting: pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed segment fault. I am running debian i386, on a 486 sx machine. I am connecting to the Console/DSP and then I get the seg fault. Only thing in asterisk I changed from the default was turning off codec_lpc10. Which I am not using anyway. What should I do with this error? If You are realy using a 486sx, please remember, that this CPU does not have a math copro. Maybe that's the cause for the failure. Do You have the Math-Emulation in kernel options activated? Other reasons can be some optimisations this processor doesn't support. As Tilghman wrote, the error does not occur in asterisk. According to Your second posting, I would suspect the alsa-stuff. HTH, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [FreeBSD 6.3] Why not use safe_asterisk?
Hello I'm running Asterisk 1.4.20.1 on a FreeBSD 6.3 host, and unless I'm mistaken, it seems like /usr/local/etc/rc.d/asterisk script doesn't make use of /usr/local/sbin/safe_asterisk to restart Asterisk in case it crashes. Is this correct, and if yes, why not use it? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hint() extension in AEL
Hi,
I've been trying to setup hinting recently on 1.4.20.1, and was
wondering if there is a more elegant way to do the following
piece of dialplan without repeating the hints for every existing
extension/user?
context Main {
hint(SIP/10301) 10301 = call(${EXTEN});
hint(SIP/10301) 301 = call(10${EXTEN});
// [snip]
hint(SIP/10327) 10327 = call(${EXTEN});
hint(SIP/10327) 327 = call(10${EXTEN});
_3XX= call(10${EXTEN});
_103XX =call(${EXTEN});
}
macro call( ext ) {
Dial(SIP/${ext},20,otL(360:61000:3));
switch(${DIALSTATUS}) {
case BUSY:
Voicemail([EMAIL PROTECTED],b);
break;
default:
Voicemail([EMAIL PROTECTED],u);
};
catch a {
VoiceMailMain([EMAIL PROTECTED]);
return;
};
Hangup;
};
So far I tried having the sip extension to ' = jump _103XX' or simply
'= Noop()'. Neither worked,
latter just overwriting the _103XX extension and causing just noop and
hangup executed when you call that extensions.
Any ideas?
___
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Re: [asterisk-users] [FreeBSD 6.3] Why not use safe_asterisk?
Vincent wrote: Hello I'm running Asterisk 1.4.20.1 on a FreeBSD 6.3 host, and unless I'm mistaken, it seems like /usr/local/etc/rc.d/asterisk script doesn't make use of /usr/local/sbin/safe_asterisk to restart Asterisk in case it crashes. Is this correct, and if yes, why not use it? Thank you. I've been wondering about that myself for a while too :) MySQL is known to be using that method under FreeBSD for quite some time similarly, by running the the /usr/local/bin/mysqld_safe from /usr/local/etc/rc.d/mysql-server. Any active contributors to the net/asterisk wanna shed some light on this mystery? Vahan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] indicating call on d channel when no b chan available
Hello all! I have HFC ISDN Card running in NT mode connected to another PBX what I want to do is to indicate new incomming call when both B channels are congested on D channel with CallerID. The other PBX system used to do it with my phone provider, so that is not a problem. I am using bristuff 0.4.0 RC2. I tried almost everything but nothing seems to work, anyone can give me some clues where should I look for solution? All the best. Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?
Sherwood McGowan wrote: Gentlemen, I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime. This system works fine with 1.2.28, and everything loads fine with no errors, but when I log an agent in I see the extra message (not in use) by their listing and they are not rang by asterisk when their queue is called. Any ideas? Nobody else? -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] indicating call on d channel when no b chan available
On Sun, Jun 29, 2008 at 04:32:40PM +0200, michael_t Gazeta.pl wrote: Hello all! I have HFC ISDN Card running in NT mode connected to another PBX what I want to do is to indicate new incomming call when both B channels are congested on D channel with CallerID. The other PBX system used to do it with my phone provider, so that is not a problem. I am using bristuff 0.4.0 RC2. I tried almost everything but nothing seems to work, anyone can give me some clues where should I look for solution? Do you have a bri debug trace of this attempt? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] indicating call on d channel when no b chan available
Debug show completely nothing. I think there is something missing in my config files. Should it work just with callwaiting and callwaitingcallerid in zapata.conf or something more? my zapata.conf looks like: switchtype = euroisdn signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel = yes echotraining = 100 echocancelwhenbridged = yes overlapdial = yes cancallforward = yes callwaiting = yes callwaitingcallerid = yes threewaycalling = yes transfer = yes immediate = no hidecallerid = no usecallerid = yes group = 1 context = isdn facilityenable = yes channel = 2-3 2008/6/29, Tzafrir Cohen [EMAIL PROTECTED]: On Sun, Jun 29, 2008 at 04:32:40PM +0200, michael_t Gazeta.pl wrote: Hello all! I have HFC ISDN Card running in NT mode connected to another PBX what I want to do is to indicate new incomming call when both B channels are congested on D channel with CallerID. The other PBX system used to do it with my phone provider, so that is not a problem. I am using bristuff 0.4.0 RC2. I tried almost everything but nothing seems to work, anyone can give me some clues where should I look for solution? Do you have a bri debug trace of this attempt? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [VOIP-Users-Conference] Re: A Flood Of Asterisk Appliances
We'd be happy to put our appliance up against the rest - I've cc'd in the asterisk list as it doesn't have to be a commercial magazine or something like that but if anyone wants to organize an 'asterisk appliance shootout' just sit down and post a methodology or what exactly the want to test for and get 5 or 6 vendors to submit their appliances I'd love to submit I'd be happy to provide a vdex-40 appliance. http://www.taa.com/products-vdex-40.html Cheers, Dean [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Sunday, 29 June 2008 1:14 PM To: [EMAIL PROTECTED] Subject: [VOIP-Users-Conference] Re: A Flood Of Asterisk Appliances On Sun, 29 Jun 2008 13:04:05 -0400, Dean Collins wrote: Cool - either way the more development in this space the better. Cheers, Dean I'm hoping that someone big enough to have a meaningful test facility makes a point of evaluating the bunch. It's difficult to know the comparative strengths/weeknesses of each at this point. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Sunday, 29 June 2008 12:32 PM To: [EMAIL PROTECTED] Subject: [VOIP-Users-Conference] Re: A Flood Of Asterisk Appliances Dean If you listen to Alec Saunder's podcast they actually say that it's Asterisk based now, but has been tested with Freeswitch as well. Perhaps their marketing collateral is a little out of sync with their deliverable device? Michael On Sun, 29 Jun 2008 12:19:02 -0400, Dean Collins wrote: Hey Michael, according to this page Jazinga uses Freeswitch http://www.jazinga.com/technology.html#3 So it's not asterisk based. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Sunday, 29 June 2008 12:06 PM To: [EMAIL PROTECTED] Subject: [VOIP-Users-Conference] A Flood Of Asterisk Appliances The trend in new offerings of Asterisk appliances continues unabated. On a VUC call a couple of weeks ago Dean Collins Co. detailed the VDEX-40 from Technoco, with The Amanda Company providing US distribution. That conference call was great, with lots of detail about the hardware, software (a Druid variant) and target market. To the extend that VUC includes some very technical people, including resellers, and met the apparent demands of the group, it appears that the device has a bright future. It features a dual processor hardware design setting it apart from others, as well as an attractive price point. Dean certainly knows what he's doing so I expect we'll be seeing more from them as time goes by. Read the rest of the po here: http://food4wine.ning.com/profiles/blog/show?id=1348225%3ABlogPost%3A6 0 2 1 Or http://blog.mgraves.org/2008/06/29/asterisk-appliances-continue-to-exp l o de-on-scene/ Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] --~--~-~--~~~---~--~~ Your participation in the conference is always appreciated! Please try to be there live when it happens. You received this message because you are subscribed to the Google Groups Asterisk Users Conference group. To post to this group, send email to [EMAIL PROTECTED] To unsubscribe from this group, send email to [EMAIL PROTECTED] For more options, visit this group at http://groups.google.com/group/VOIP-Users-Conference?hl=en -~--~~~~--~~--~--~--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?
On Sun, Jun 29, 2008 at 7:02 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Sherwood McGowan wrote: Gentlemen, I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime. This system works fine with 1.2.28, and everything loads fine with no errors, but when I log an agent in I see the extra message (not in use) by their listing and they are not rang by asterisk when their queue is called. Any ideas? Nobody else? Have you checked call-limit and state information for SIP peers? That was changed between 1.2 and 1.4, and could affect queue state. See the UPGRADE notes. Otherwise You'll have to set core set debug 2 and core set verbose 3, and post full log (debug+verbose) where agents got logged in (if you have also realtime members, just execute queue show on CLI. Then you'll have to give one call to agent, talk for little and disconnect. Then just post that log here. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Timeout between digits for fxs station
Hi All; How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs? Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timeout between digits for fxs station
On Jun 29, 2008, at 6:35 PM, bilal ghayyad wrote: Hi All; How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs? Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number? Usually best to use WaitExten() or TIMEOUT(digit). For most stuff I do, I just use waitexten. Fred Posner [EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup?
I've got a unique situation and think it may be the lack of the Hangup command in the dialplan that is creating the issue.Can anyone elaborate on why it is, or is not, important to use hangup in the dialplan. Presently I don't have the first instance of it in my dialplan, however, I see some things in the debugging that might be cleaner if I did implement hangup I have approximately 140 extensions provisioned off this asterisk server and about 8 IVRs...So as you might expect, it is quite busy... Thanks, -Joe ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk to Broadvoice SIP peer fails in 1.6.9-beta9
In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now, with the same SIP configuration, I cannot establish the peer. I've enclosed a SIP log in the hope that someone can help me analyze this failure. I'd guess the issue is NAT related and wondering if someone can spot a problem in the logs, below. Some details to help read this log (I've changed these numbers for privacy purposes): . My Asterisk server is behind a firewall. It's internal address is 192.168.71.1. . My public IP address is 123.123.123.123 . I am calling 2125551212 . My Broadvoice phone number is 9145551212 Here is the log: == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Macro(SIP/siegeld-00e08e00, dial-sip,[EMAIL PROTECTED] broadvoice.com) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/siegeld-00e08e00, SIP/[EMAIL PROTECTED]) i\ n new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.71.7 port 18596 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (NAT) to 123.123.123.123:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport Max-Forwards: 70 From: David Siegel sip:[EMAIL PROTECTED];tag=as57923ac4 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-beta9 Date: Mon, 30 Jun 2008 05:13:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 256 v=0 o=root 593017814 593017814 IN IP4 192.168.71.7 s=Asterisk PBX 1.6.0-beta9 c=IN IP4 192.168.71.7 t=0 0 m=audio 18596 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called [EMAIL PROTECTED] stsca1*CLI --- SIP read from UDP://123.123.123.123:5060 --- SIP/2.0 100 Trying Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE From: David Siegel sip:[EMAIL PROTECTED];tag=as57923ac4 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd Content-Length: 0 - --- (7 headers 0 lines) --- stsca1*CLI --- SIP read from UDP://123.123.123.123:5060 --- SIP/2.0 403 Forbidden Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE From: David Siegel sip:[EMAIL PROTECTED];tag=as57923ac4 To: sip:[EMAIL PROTECTED];tag=lmno Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd User-Agent: Asterisk PBX 1.6.0-beta9 Content-Type: application/sdp Content-Length: 188 v=0 o=1213832004 593017814 593017814 IN IP4 192.168.71.7 s=- c=IN IP4 192.168.71.7 t=0 0 m=audio 18596 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 - --- (9 headers 9 lines) --- Transmitting (NAT) to 123.123.123.123:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport Max-Forwards: 70 From: David Siegel sip:[EMAIL PROTECTED];tag=as57923ac4 To: sip:[EMAIL PROTECTED];tag=lmno Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0-beta9 Content-Length: 0 --- [Jun 30 01:13:51] WARNING[3023]: chan_sip.c:14738 handle_response_invite: Received response: Forbidden f\ rom 'David Siegel sip:[EMAIL PROTECTED];tag=as57923ac4' -- SIP/sip.broadvoice.com-00e0ddb0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/siegeld-00e08e00, s-CONGESTION,1) in new stack -- Goto (macro-dial-sip,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] PlayTones(SIP/siegeld-00e08e00, congestion) in new st\ ack -- Auto fallthrough, channel 'SIP/siegeld-00e08e00' status is 'CONGESTION' Really destroying SIP dialog '[EMAIL PROTECTED]' Method: INVI ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
